From 4b166da939012905f4c36fedada62067db31948e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Daniel=20Gl=C3=B6ckner?= Date: Sat, 28 Mar 2009 19:47:01 +0100 Subject: ASoC: Add driver for s6000 I2S interface MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds a driver for the I2S interface found on Stretch s6000 family processors. Signed-off-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/s6000/Kconfig | 10 + sound/soc/s6000/Makefile | 6 + sound/soc/s6000/s6000-i2s.c | 629 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/s6000/s6000-i2s.h | 25 ++ sound/soc/s6000/s6000-pcm.c | 497 ++++++++++++++++++++++++++++++++++ sound/soc/s6000/s6000-pcm.h | 35 +++ 8 files changed, 1204 insertions(+) create mode 100644 sound/soc/s6000/Kconfig create mode 100644 sound/soc/s6000/Makefile create mode 100644 sound/soc/s6000/s6000-i2s.c create mode 100644 sound/soc/s6000/s6000-i2s.h create mode 100644 sound/soc/s6000/s6000-pcm.c create mode 100644 sound/soc/s6000/s6000-pcm.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3d2bb6f..3304f9d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,6 +32,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..8943a14 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,4 +10,5 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig new file mode 100644 index 0000000..4bfc8bc --- /dev/null +++ b/sound/soc/s6000/Kconfig @@ -0,0 +1,10 @@ +config SND_S6000_SOC + tristate "SoC Audio for the Stretch s6000 family" + depends on XTENSA_VARIANT_S6000 + help + Say Y or M if you want to add support for codecs attached to + s6000 family chips. You will also need to select the platform + to support below. + +config SND_S6000_SOC_I2S + tristate diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile new file mode 100644 index 0000000..df15f87 --- /dev/null +++ b/sound/soc/s6000/Makefile @@ -0,0 +1,6 @@ +# s6000 Platform Support +snd-soc-s6000-objs := s6000-pcm.o +snd-soc-s6000-i2s-objs := s6000-i2s.o + +obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o +obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c new file mode 100644 index 0000000..dcc7904 --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.c @@ -0,0 +1,629 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch S6000 family + * + * Author: Daniel Gloeckner, + * Copyright: (C) 2009 emlix GmbH + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "s6000-i2s.h" +#include "s6000-pcm.h" + +struct s6000_i2s_dev { + dma_addr_t sifbase; + u8 __iomem *scbbase; + unsigned int wide; + unsigned int channel_in; + unsigned int channel_out; + unsigned int lines_in; + unsigned int lines_out; + struct s6000_pcm_dma_params dma_params; +}; + +#define S6_I2S_INTERRUPT_STATUS 0x00 +#define S6_I2S_INT_OVERRUN 1 +#define S6_I2S_INT_UNDERRUN 2 +#define S6_I2S_INT_ALIGNMENT 4 +#define S6_I2S_INTERRUPT_ENABLE 0x04 +#define S6_I2S_INTERRUPT_RAW 0x08 +#define S6_I2S_INTERRUPT_CLEAR 0x0C +#define S6_I2S_INTERRUPT_SET 0x10 +#define S6_I2S_MODE 0x20 +#define S6_I2S_DUAL 0 +#define S6_I2S_WIDE 1 +#define S6_I2S_TX_DEFAULT 0x24 +#define S6_I2S_DATA_CFG(c) (0x40 + 0x10 * (c)) +#define S6_I2S_IN 0 +#define S6_I2S_OUT 1 +#define S6_I2S_UNUSED 2 +#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c)) +#define S6_I2S_DIV_MASK 0x001fff +#define S6_I2S_16BIT 0x000000 +#define S6_I2S_20BIT 0x002000 +#define S6_I2S_24BIT 0x004000 +#define S6_I2S_32BIT 0x006000 +#define S6_I2S_BITS_MASK 0x006000 +#define S6_I2S_MEM_16BIT 0x000000 +#define S6_I2S_MEM_32BIT 0x008000 +#define S6_I2S_MEM_MASK 0x008000 +#define S6_I2S_CHANNELS_SHIFT 16 +#define S6_I2S_CHANNELS_MASK 0x030000 +#define S6_I2S_SCK_IN 0x000000 +#define S6_I2S_SCK_OUT 0x040000 +#define S6_I2S_SCK_DIR 0x040000 +#define S6_I2S_WS_IN 0x000000 +#define S6_I2S_WS_OUT 0x080000 +#define S6_I2S_WS_DIR 0x080000 +#define S6_I2S_LEFT_FIRST 0x000000 +#define S6_I2S_RIGHT_FIRST 0x100000 +#define S6_I2S_FIRST 0x100000 +#define S6_I2S_CUR_SCK 0x200000 +#define S6_I2S_CUR_WS 0x400000 +#define S6_I2S_ENABLE(c) (0x48 + 0x10 * (c)) +#define S6_I2S_DISABLE_IF 0x02 +#define S6_I2S_ENABLE_IF 0x03 +#define S6_I2S_IS_BUSY 0x04 +#define S6_I2S_DMA_ACTIVE 0x08 +#define S6_I2S_IS_ENABLED 0x10 + +#define S6_I2S_NUM_LINES 4 + +#define S6_I2S_SIF_PORT0 0x0000000 +#define S6_I2S_SIF_PORT1 0x0000080 /* docs say 0x0000010 */ + +static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val) +{ + writel(val, dev->scbbase + reg); +} + +static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg) +{ + return readl(dev->scbbase + reg); +} + +static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg, + u32 mask, u32 val) +{ + val ^= s6_i2s_read_reg(dev, reg) & ~mask; + s6_i2s_write_reg(dev, reg, val); +} + +static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel) +{ + int i, j, cur, prev; + + /* + * Wait for WCLK to toggle 5 times before enabling the channel + * s6000 Family Datasheet 3.6.4: + * "At least two cycles of WS must occur between commands + * to disable or enable the interface" + */ + j = 0; + prev = ~S6_I2S_CUR_WS; + for (i = 1000000; --i && j < 6; ) { + cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel)) + & S6_I2S_CUR_WS; + if (prev != cur) { + prev = cur; + j++; + } + } + if (j < 6) + printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n"); + + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF); +} + +static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel) +{ + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF); +} + +static void s6000_i2s_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_start_channel(dev, channel); +} + +static void s6000_i2s_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_stop_channel(dev, channel); +} + +static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + int after) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after) + s6000_i2s_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!after) + s6000_i2s_stop(substream); + } + return 0; +} + +static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev) +{ + unsigned int pending; + pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW); + pending &= S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN; + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending); + + return pending; +} + +static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + unsigned int errors; + unsigned int ret; + + errors = s6000_i2s_int_sources(dev); + if (likely(!errors)) + return 0; + + ret = 0; + if (errors & S6_I2S_INT_ALIGNMENT) + printk(KERN_ERR "s6000-i2s: WCLK misaligned\n"); + if (errors & S6_I2S_INT_UNDERRUN) + ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK; + if (errors & S6_I2S_INT_OVERRUN) + ret |= 1 << SNDRV_PCM_STREAM_CAPTURE; + return ret; +} + +static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev) +{ + int channel; + int n = 50; + for (channel = 0; channel < 2; channel++) { + while (--n >= 0) { + int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel)); + if ((v & S6_I2S_IS_ENABLED) + || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY))) + break; + udelay(20); + } + } + if (n < 0) + printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces"); +} + +static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + w = S6_I2S_SCK_IN | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBS_CFM: + w = S6_I2S_SCK_OUT | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBM_CFS: + w = S6_I2S_SCK_IN | S6_I2S_WS_OUT; + break; + case SND_SOC_DAIFMT_CBS_CFS: + w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + w |= S6_I2S_LEFT_FIRST; + break; + case SND_SOC_DAIFMT_IB_NF: + w |= S6_I2S_RIGHT_FIRST; + break; + default: + return -EINVAL; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + + return 0; +} + +static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct s6000_i2s_dev *dev = dai->private_data; + + if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2) + return -EINVAL; + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id), + S6_I2S_DIV_MASK, div / 2 - 1); + return 0; +} + +static int s6000_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + int interf; + u32 w = 0; + + if (dev->wide) + interf = 0; + else { + w |= (((params_channels(params) - 2) / 2) + << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK; + interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ? dev->channel_out : dev->channel_in; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT; + break; + default: + printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n", + params_format(params)); + return -EINVAL; + } + + if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf)) + & S6_I2S_IS_ENABLED) { + printk(KERN_ERR "s6000-i2s: interface already enabled\n"); + return -EBUSY; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf), + S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK, + w); + + return 0; +} + +static int s6000_i2s_dai_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + struct s6000_snd_platform_data *pdata = pdev->dev.platform_data; + + if (!pdata) + return -EINVAL; + + dev->wide = pdata->wide; + dev->channel_in = pdata->channel_in; + dev->channel_out = pdata->channel_out; + dev->lines_in = pdata->lines_in; + dev->lines_out = pdata->lines_out; + + s6_i2s_write_reg(dev, S6_I2S_MODE, + dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL); + + if (dev->wide) { + int i; + + if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES) + return -EINVAL; + + dev->channel_in = 0; + dev->channel_out = 1; + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = dai->capture.channels_min; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = dai->playback.channels_min; + + for (i = 0; i < dev->lines_out; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT); + + for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), + S6_I2S_UNUSED); + + for (; i < S6_I2S_NUM_LINES; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN); + } else { + unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED}; + + if (dev->lines_in > 1 || dev->lines_out > 1) + return -EINVAL; + + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = 8 * dev->lines_in; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = 8 * dev->lines_out; + + if (dev->lines_in) + cfg[dev->channel_in] = S6_I2S_IN; + if (dev->lines_out) + cfg[dev->channel_out] = S6_I2S_OUT; + + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]); + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]); + } + + if (dev->lines_out) { + if (dev->lines_in) { + if (!dev->dma_params.dma_out) + return -ENODEV; + } else { + dev->dma_params.dma_out = dev->dma_params.dma_in; + dev->dma_params.dma_in = 0; + } + } + dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.same_rate = pdata->same_rate | pdata->wide; + return 0; +} + +#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops s6000_i2s_dai_ops = { + .set_fmt = s6000_i2s_set_dai_fmt, + .set_clkdiv = s6000_i2s_set_clkdiv, + .hw_params = s6000_i2s_hw_params, +}; + +struct snd_soc_dai s6000_i2s_dai = { + .name = "s6000-i2s", + .id = 0, + .probe = s6000_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .capture = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .ops = &s6000_i2s_dai_ops, +} +EXPORT_SYMBOL_GPL(s6000_i2s_dai); + +static int __devinit s6000_i2s_probe(struct platform_device *pdev) +{ + struct s6000_i2s_dev *dev; + struct resource *scbmem, *sifmem, *region, *dma1, *dma2; + u8 __iomem *mmio; + int ret; + + scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!scbmem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_none; + } + + region = request_mem_region(scbmem->start, + scbmem->end - scbmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SCB region already claimed\n"); + ret = -EBUSY; + goto err_release_none; + } + + mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1); + if (!mmio) { + dev_err(&pdev->dev, "can't ioremap SCB region\n"); + ret = -ENOMEM; + goto err_release_scb; + } + + sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!sifmem) { + dev_err(&pdev->dev, "no second mem resource?\n"); + ret = -ENODEV; + goto err_release_map; + } + + region = request_mem_region(sifmem->start, + sifmem->end - sifmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SIF region already claimed\n"); + ret = -EBUSY; + goto err_release_map; + } + + dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dma1) { + dev_err(&pdev->dev, "no dma resource?\n"); + ret = -ENODEV; + goto err_release_sif; + } + + region = request_mem_region(dma1->start, dma1->end - dma1->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_sif; + } + + dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (dma2) { + region = request_mem_region(dma2->start, + dma2->end - dma2->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, + "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_dma1; + } + } + + dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_dma2; + } + + s6000_i2s_dai.dev = &pdev->dev; + s6000_i2s_dai.private_data = dev; + s6000_i2s_dai.dma_data = &dev->dma_params; + + dev->sifbase = sifmem->start; + dev->scbbase = mmio; + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + s6000_i2s_wait_disabled(dev); + + dev->dma_params.check_xrun = s6000_i2s_check_xrun; + dev->dma_params.trigger = s6000_i2s_trigger; + dev->dma_params.dma_in = dma1->start; + dev->dma_params.dma_out = dma2 ? dma2->start : 0; + dev->dma_params.irq = platform_get_irq(pdev, 0); + if (dev->dma_params.irq < 0) { + dev_err(&pdev->dev, "no irq resource?\n"); + ret = -ENODEV; + goto err_release_dev; + } + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + ret = snd_soc_register_dai(&s6000_i2s_dai); + if (ret) + goto err_release_dev; + + return 0; + +err_release_dev: + kfree(dev); +err_release_dma2: + if (dma2) + release_mem_region(dma2->start, dma2->end - dma2->start + 1); +err_release_dma1: + release_mem_region(dma1->start, dma1->end - dma1->start + 1); +err_release_sif: + release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1); +err_release_map: + iounmap(mmio); +err_release_scb: + release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1); +err_release_none: + return ret; +} + +static void __devexit s6000_i2s_remove(struct platform_device *pdev) +{ + struct s6000_i2s_dev *dev = s6000_i2s_dai.private_data; + struct resource *region; + void __iomem *mmio = dev->scbbase; + + snd_soc_unregister_dai(&s6000_i2s_dai); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6000_i2s_dai.private_data = 0; + kfree(dev); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 0); + release_mem_region(region->start, region->end - region->start + 1); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (region) + release_mem_region(region->start, + region->end - region->start + 1); + + region = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(region->start, (region->end - region->start) + 1); + + iounmap(mmio); + region = platform_get_resource(pdev, IORESOURCE_IO, 0); + release_mem_region(region->start, (region->end - region->start) + 1); +} + +static struct platform_driver s6000_i2s_driver = { + .probe = s6000_i2s_probe, + .remove = __devexit_p(s6000_i2s_remove), + .driver = { + .name = "s6000-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init s6000_i2s_init(void) +{ + return platform_driver_register(&s6000_i2s_driver); +} +module_init(s6000_i2s_init); + +static void __exit s6000_i2s_exit(void) +{ + platform_driver_unregister(&s6000_i2s_driver); +} +module_exit(s6000_i2s_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h new file mode 100644 index 0000000..2375fdf --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.h @@ -0,0 +1,25 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch s6000 family + * + * Author: Daniel Gloeckner, + * Copyright: (C) 2009 emlix GmbH + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _S6000_I2S_H +#define _S6000_I2S_H + +extern struct snd_soc_dai s6000_i2s_dai; + +struct s6000_snd_platform_data { + int lines_in; + int lines_out; + int channel_in; + int channel_out; + int wide; + int same_rate; +}; +#endif diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c new file mode 100644 index 0000000..83b8028 --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.c @@ -0,0 +1,497 @@ +/* + * ALSA PCM interface for the Stetch s6000 family + * + * Author: Daniel Gloeckner, + * Copyright: (C) 2009 emlix GmbH + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include + +#include "s6000-pcm.h" + +#define S6_PCM_PREALLOCATE_SIZE (96 * 1024) +#define S6_PCM_PREALLOCATE_MAX (2048 * 1024) + +static struct snd_pcm_hardware s6000_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000), + .rate_min = 0, + .rate_max = 1562500, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 0x7ffffff0, + .period_bytes_min = 16, + .period_bytes_max = 0xfffff0, + .periods_min = 2, + .periods_max = 1024, /* no limit */ + .fifo_size = 0, +}; + +struct s6000_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ +}; + +static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int channel; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = par->sif_out; + channel = par->dma_out; + } else { + src = par->sif_in; + dst = dma_pos; + channel = par->dma_in; + } + + if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel))) + return; + + if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) { + printk(KERN_ERR "s6000-pcm: fifo full\n"); + return; + } + + BUG_ON(period_size & 15); + s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel), + src, dst, period_size); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static irqreturn_t s6000_pcm_irq(int irq, void *data) +{ + struct snd_pcm *pcm = data; + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_runtime_data *prtd; + unsigned int has_xrun; + int i, ret = IRQ_NONE; + u32 channel[2] = { + [SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out, + [SNDRV_PCM_STREAM_CAPTURE] = params->dma_in + }; + + has_xrun = params->check_xrun(runtime->dai->cpu_dai); + + for (i = 0; i < ARRAY_SIZE(channel); ++i) { + struct snd_pcm_substream *substream = pcm->streams[i].substream; + unsigned int pending; + + if (!channel[i]) + continue; + + if (unlikely(has_xrun & (1 << i)) && + substream->runtime && + snd_pcm_running(substream)) { + dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + ret = IRQ_HANDLED; + } + + pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])); + + if (pending & 1) { + ret = IRQ_HANDLED; + if (likely(substream->runtime && + snd_pcm_running(substream))) { + snd_pcm_period_elapsed(substream); + dev_dbg(pcm->dev, "period elapsed %x %x\n", + s6dmac_cur_src(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])), + s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i]))); + prtd = substream->runtime->private_data; + spin_lock(&prtd->lock); + s6000_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } + } + + if (unlikely(pending & ~7)) { + if (pending & (1 << 3)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Underflow\n", + channel[i]); + if (pending & (1 << 4)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Overflow\n", + channel[i]); + if (pending & 0x1e0) + printk(KERN_WARNING + "s6000-pcm: DMA %x Master Error " + "(mask %x)\n", + channel[i], pending >> 5); + + } + } + + return ret; +} + +static int s6000_pcm_start(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + int srcinc; + u32 dma; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + srcinc = 1; + dma = par->dma_out; + } else { + srcinc = 0; + dma = par->dma_in; + } + s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma), + 1 /* priority 1 (0 is max) */, + 0 /* peripheral requests w/o xfer length mode */, + srcinc /* source address increment */, + srcinc^1 /* destination address increment */, + 0 /* chunksize 0 (skip impossible on this dma) */, + 0 /* source skip after chunk (impossible) */, + 0 /* destination skip after chunk (impossible) */, + 4 /* 16 byte burst size */, + -1 /* don't conserve bandwidth */, + 0 /* low watermark irq descriptor theshold */, + 0 /* disable hardware timestamps */, + 1 /* enable channel */); + + s6000_pcm_enqueue_dma(substream); + s6000_pcm_enqueue_dma(substream); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_stop(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + u32 channel; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + channel = par->dma_out; + else + channel = par->dma_in; + + s6dmac_set_terminal_count(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel), 0); + + spin_lock_irqsave(&prtd->lock, flags); + + s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel)); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + + ret = par->trigger(substream, cmd, 0); + if (ret < 0) + return ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = s6000_pcm_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = s6000_pcm_stop(substream); + break; + default: + ret = -EINVAL; + } + if (ret < 0) + return ret; + + return par->trigger(substream, cmd, 1); +} + +static int s6000_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + + prtd->period = 0; + + return 0; +} + +static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + unsigned long flags; + unsigned int offset; + dma_addr_t count; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out), + DMA_INDEX_CHNL(par->dma_out)); + else + count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in), + DMA_INDEX_CHNL(par->dma_in)); + + count -= runtime->dma_addr; + + spin_unlock_irqrestore(&prtd->lock, flags); + + offset = bytes_to_frames(runtime, count); + if (unlikely(offset >= runtime->buffer_size)) + offset = 0; + + return offset; +} + +static int s6000_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + if (par->same_rate) { + int rate; + spin_lock(&par->lock); /* needed? */ + rate = par->rate; + spin_unlock(&par->lock); + if (rate != -1) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_RATE, + rate, rate); + if (ret < 0) + return ret; + } + } + + prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int s6000_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n"); + return ret; + } + + if (par->same_rate) { + spin_lock(&par->lock); + if (par->rate == -1 || + !(par->in_use & ~(1 << substream->stream))) { + par->rate = params_rate(hw_params); + par->in_use |= 1 << substream->stream; + } else if (params_rate(hw_params) != par->rate) { + snd_pcm_lib_free_pages(substream); + par->in_use &= ~(1 << substream->stream); + ret = -EBUSY; + } + spin_unlock(&par->lock); + } + return ret; +} + +static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + + spin_lock(&par->lock); + par->in_use &= ~(1 << substream->stream); + if (!par->in_use) + par->rate = -1; + spin_unlock(&par->lock); + + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_pcm_ops s6000_pcm_ops = { + .open = s6000_pcm_open, + .close = s6000_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s6000_pcm_hw_params, + .hw_free = s6000_pcm_hw_free, + .trigger = s6000_pcm_trigger, + .prepare = s6000_pcm_prepare, + .pointer = s6000_pcm_pointer, +}; + +static void s6000_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + + free_irq(params->irq, pcm); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; + +static int s6000_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + int res; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s6000_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (params->dma_in) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + } + + if (params->dma_out) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + } + + res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED, + s6000_soc_platform.name, pcm); + if (res) { + printk(KERN_ERR "s6000-pcm couldn't get IRQ\n"); + return res; + } + + res = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, + S6_PCM_PREALLOCATE_SIZE, + S6_PCM_PREALLOCATE_MAX); + if (res) + printk(KERN_WARNING "s6000-pcm: preallocation failed\n"); + + spin_lock_init(¶ms->lock); + params->in_use = 0; + params->rate = -1; + return 0; +} + +struct snd_soc_platform s6000_soc_platform = { + .name = "s6000-audio", + .pcm_ops = &s6000_pcm_ops, + .pcm_new = s6000_pcm_new, + .pcm_free = s6000_pcm_free, +}; +EXPORT_SYMBOL_GPL(s6000_soc_platform); + +static int __init s6000_pcm_init(void) +{ + return snd_soc_register_platform(&s6000_soc_platform); +} +module_init(s6000_pcm_init); + +static void __exit s6000_pcm_exit(void) +{ + snd_soc_unregister_platform(&s6000_soc_platform); +} +module_exit(s6000_pcm_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h new file mode 100644 index 0000000..96f23f6 --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.h @@ -0,0 +1,35 @@ +/* + * ALSA PCM interface for the Stretch s6000 family + * + * Author: Daniel Gloeckner, + * Copyright: (C) 2009 emlix GmbH + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _S6000_PCM_H +#define _S6000_PCM_H + +struct snd_soc_dai; +struct snd_pcm_substream; + +struct s6000_pcm_dma_params { + unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai); + int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after); + dma_addr_t sif_in; + dma_addr_t sif_out; + u32 dma_in; + u32 dma_out; + int irq; + int same_rate; + + spinlock_t lock; + int in_use; + int rate; +}; + +extern struct snd_soc_platform s6000_soc_platform; + +#endif -- cgit v1.1 From 2b7dbbe0c9491e62b50978d1615193bec33a8291 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Daniel=20Gl=C3=B6ckner?= Date: Sat, 28 Mar 2009 19:47:02 +0100 Subject: ASoC: s6105 IP camera machine specific ASoC code MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds machine specific code for the audio part of the Stretch s6105 IP camera reference design. The device uses the tlv320aic31(01) codec to generate the clock for both I2S ports of the soc. While the master clock is generated by a configurable PLL chip, the code assumes the factory default settings. An additional kcontrol has been added to handle the special routing of the board, connecting both HPLCOM and HPROUT to the same pin of the audio jack. One of these should always be switched off. Signed-off-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/Kconfig | 9 ++ sound/soc/s6000/Makefile | 5 + sound/soc/s6000/s6105-ipcam.c | 244 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 258 insertions(+) create mode 100644 sound/soc/s6000/s6105-ipcam.c diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig index 4bfc8bc..c74eb3d 100644 --- a/sound/soc/s6000/Kconfig +++ b/sound/soc/s6000/Kconfig @@ -8,3 +8,12 @@ config SND_S6000_SOC config SND_S6000_SOC_I2S tristate + +config SND_S6000_SOC_S6IPCAM + tristate "SoC Audio support for Stretch 6105 IP Camera" + depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105 + select SND_S6000_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on the + Stretch s6105 IP Camera Reference Design. diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile index df15f87..7a61361 100644 --- a/sound/soc/s6000/Makefile +++ b/sound/soc/s6000/Makefile @@ -4,3 +4,8 @@ snd-soc-s6000-i2s-objs := s6000-i2s.o obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o + +# s6105 Machine Support +snd-soc-s6ipcam-objs := s6105-ipcam.o + +obj-$(CONFIG_SND_S6000_SOC_S6IPCAM) += snd-soc-s6ipcam.o diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c new file mode 100644 index 0000000..21c4f55 --- /dev/null +++ b/sound/soc/s6000/s6105-ipcam.c @@ -0,0 +1,244 @@ +/* + * ASoC driver for Stretch s6105 IP camera platform + * + * Author: Daniel Gloeckner, + * Copyright: (C) 2009 emlix GmbH + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "../codecs/tlv320aic3x.h" +#include "s6000-pcm.h" +#include "s6000-i2s.h" + +#define S6105_CAM_CODEC_CLOCK 12288000 + +static int s6105_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_IF); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, S6105_CAM_CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s6105_ops = { + .hw_params = s6105_hw_params, +}; + +/* s6105 machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Audio Out Differential", NULL), + SND_SOC_DAPM_LINE("Audio Out Stereo", NULL), + SND_SOC_DAPM_LINE("Audio In", NULL), +}; + +/* s6105 machine audio_mapnections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Audio Out connected to HPLOUT, HPLCOM, HPROUT */ + {"Audio Out Differential", NULL, "HPLOUT"}, + {"Audio Out Differential", NULL, "HPLCOM"}, + {"Audio Out Stereo", NULL, "HPLOUT"}, + {"Audio Out Stereo", NULL, "HPROUT"}, + + /* Audio In connected to LINE1L, LINE1R */ + {"LINE1L", NULL, "Audio In"}, + {"LINE1R", NULL, "Audio In"}, +}; + +static int output_type_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item) { + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, "HPLOUT/HPROUT"); + } else { + strcpy(uinfo->value.enumerated.name, "HPLOUT/HPLCOM"); + } + return 0; +} + +static int output_type_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +static int output_type_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + unsigned int val = (ucontrol->value.enumerated.item[0] != 0); + char *differential = "Audio Out Differential"; + char *stereo = "Audio Out Stereo"; + + if (kcontrol->private_value == val) + return 0; + kcontrol->private_value = val; + snd_soc_dapm_disable_pin(codec, val ? differential : stereo); + snd_soc_dapm_sync(codec); + snd_soc_dapm_enable_pin(codec, val ? stereo : differential); + snd_soc_dapm_sync(codec); + + return 1; +} + +static const struct snd_kcontrol_new audio_out_mux = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Output Mux", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = output_type_info, + .get = output_type_get, + .put = output_type_put, + .private_value = 1 /* default to stereo */ +}; + +/* Logic for a aic3x as connected on the s6105 ip camera ref design */ +static int s6105_aic3x_init(struct snd_soc_codec *codec) +{ + /* Add s6105 specific widgets */ + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); + + /* Set up s6105 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* not present */ + snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); + snd_soc_dapm_nc_pin(codec, "LINE2L"); + snd_soc_dapm_nc_pin(codec, "LINE2R"); + + /* not connected */ + snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */ + snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */ + snd_soc_dapm_nc_pin(codec, "LLOUT"); + snd_soc_dapm_nc_pin(codec, "RLOUT"); + snd_soc_dapm_nc_pin(codec, "HPRCOM"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Audio In"); + + /* must correspond to audio_out_mux.private_value initializer */ + snd_soc_dapm_disable_pin(codec, "Audio Out Differential"); + snd_soc_dapm_sync(codec); + snd_soc_dapm_enable_pin(codec, "Audio Out Stereo"); + + snd_soc_dapm_sync(codec); + + snd_ctl_add(codec->card, snd_ctl_new1(&audio_out_mux, codec)); + + return 0; +} + +/* s6105 digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link s6105_dai = { + .name = "TLV320AIC31", + .stream_name = "AIC31", + .cpu_dai = &s6000_i2s_dai, + .codec_dai = &aic3x_dai, + .init = s6105_aic3x_init, + .ops = &s6105_ops, +}; + +/* s6105 audio machine driver */ +static struct snd_soc_card snd_soc_card_s6105 = { + .name = "Stretch IP Camera", + .platform = &s6000_soc_platform, + .dai_link = &s6105_dai, + .num_links = 1, +}; + +/* s6105 audio private data */ +static struct aic3x_setup_data s6105_aic3x_setup = { + .i2c_bus = 0, + .i2c_address = 0x18, +}; + +/* s6105 audio subsystem */ +static struct snd_soc_device s6105_snd_devdata = { + .card = &snd_soc_card_s6105, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &s6105_aic3x_setup, +}; + +static struct s6000_snd_platform_data __initdata s6105_snd_data = { + .wide = 0, + .channel_in = 0, + .channel_out = 1, + .lines_in = 1, + .lines_out = 1, + .same_rate = 1, +}; + +static struct platform_device *s6105_snd_device; + +static int __init s6105_init(void) +{ + int ret; + + s6105_snd_device = platform_device_alloc("soc-audio", -1); + if (!s6105_snd_device) + return -ENOMEM; + + platform_set_drvdata(s6105_snd_device, &s6105_snd_devdata); + s6105_snd_devdata.dev = &s6105_snd_device->dev; + platform_device_add_data(s6105_snd_device, &s6105_snd_data, + sizeof(s6105_snd_data)); + + ret = platform_device_add(s6105_snd_device); + if (ret) + platform_device_put(s6105_snd_device); + + return ret; +} + +static void __exit s6105_exit(void) +{ + platform_device_unregister(s6105_snd_device); +} + +module_init(s6105_init); +module_exit(s6105_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6105 IP camera ASoC driver"); +MODULE_LICENSE("GPL"); -- cgit v1.1 From 80fbe6ac9b47cbc11e174a9bf853834dc281da35 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Daniel=20Gl=C3=B6ckner?= Date: Mon, 6 Apr 2009 11:50:22 +0200 Subject: ASoC: correct s6000 I2S clock polarity MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit According to the data sheet data is clocked out on the falling edge and latched on the rising edge of the bit clock. While the left sample is transmitted the word clock line is low. Signed-off-by: Daniel Glöckner Signed-off-by: Mark Brown --- sound/soc/s6000/s6000-i2s.c | 4 ++-- sound/soc/s6000/s6105-ipcam.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index dcc7904..c5cda18 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -252,10 +252,10 @@ static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_NB_NF: w |= S6_I2S_LEFT_FIRST; break; - case SND_SOC_DAIFMT_IB_NF: + case SND_SOC_DAIFMT_NB_IF: w |= S6_I2S_RIGHT_FIRST; break; default: diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 21c4f55..b5f95f9 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -43,7 +43,7 @@ static int s6105_hw_params(struct snd_pcm_substream *substream, /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_IF); + SND_SOC_DAIFMT_NB_NF); if (ret < 0) return ret; -- cgit v1.1 From 6553e192d48af88184029066c30c9464516ea0b7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Apr 2009 16:59:32 +0100 Subject: ASoC: Display return code when failing to add a DAPM kcontrol Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 735903a..46485de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -357,8 +357,9 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, path->long_name); ret = snd_ctl_add(codec->card, path->kcontrol); if (ret < 0) { - printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n", - path->long_name); + printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n", + path->long_name, + ret); kfree(path->long_name); path->long_name = NULL; return ret; -- cgit v1.1 From 06f409d76f1d382167eb1cadde2e23a73272865d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Apr 2009 18:10:13 +0100 Subject: ASoC: Provide core support for symmetric sample rates Many devices require symmetric configurations of capture and playback data formats, often due to shared clocking but sometimes also due to other shared playback and record configuration in the device. Start providing core support for this by allowing the DAIs or the machine to specify that the sample rates used should be kept symmetric. A flag symmetric_rates is provided in the snd_soc_dai and snd_soc_dai_link structures. If this is set in either of the DAIs or in the machine then a constraint will be applied when a stream is already open preventing any changes in sample rate. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 + include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 38 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 45 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1367647..22b729f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -208,6 +208,7 @@ struct snd_soc_dai { /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; + unsigned int symmetric_rates:1; /* DAI runtime info */ struct snd_pcm_runtime *runtime; diff --git a/include/sound/soc.h b/include/sound/soc.h index a40bc6f..b1f2f88 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -417,6 +417,12 @@ struct snd_soc_dai_link { /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_codec *codec); + /* Symmetry requirements */ + unsigned int symmetric_rates:1; + + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + /* DAI pcm */ struct snd_pcm *pcm; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f6..dd28009 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -113,6 +113,35 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai_link *machine = rtd->dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; + int ret; + + if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || + machine->symmetric_rates) { + dev_dbg(card->dev, "Symmetry forces %dHz rate\n", + machine->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + machine->rate, + machine->rate); + if (ret < 0) { + dev_err(card->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + } + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -221,6 +250,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto machine_err; + } + pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, @@ -521,6 +557,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } + machine->rate = params_rate(params); + out: mutex_unlock(&pcm_mutex); return ret; -- cgit v1.1 From 5409fb4e327a84972483047ecf4fb41f279453e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Apr 2009 18:45:21 +0100 Subject: ASoC: Add WM8988 CODEC driver The WM8988 is a low power, high quality stereo CODEC designed for portable digital audio applications. The device integrates complete interfaces to 2 stereo headphone or line out ports. External component requirements are drastically reduced as no separate headphone amplifiers are required. Advanced on-chip digital signal processing performs graphic equaliser, 3-D sound enhancement and automatic level control for the microphone or line input. The WM8988 can operate as a master or a slave, with various master clock frequencies including 12 or 24MHz for USB devices, or standard 256fs rates like 12.288MHz and 24.576MHz. Different audio sample rates such as 96kHz, 48kHz, 44.1kHz are generated directly from the master clock without the need for an external PLL. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8988.c | 1097 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8988.h | 60 +++ 4 files changed, 1163 insertions(+) create mode 100644 sound/soc/codecs/wm8988.c create mode 100644 sound/soc/codecs/wm8988.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a..ab36485 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C + select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS @@ -141,6 +142,9 @@ config SND_SOC_WM8903 config SND_SOC_WM8971 tristate +config SND_SOC_WM8988 + tristate + config SND_SOC_WM8990 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d245..a72548d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -24,6 +24,7 @@ snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o +snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o @@ -55,6 +56,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c new file mode 100644 index 0000000..c05f718 --- /dev/null +++ b/sound/soc/codecs/wm8988.c @@ -0,0 +1,1097 @@ +/* + * wm8988.c -- WM8988 ALSA SoC audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2005 Openedhand Ltd. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8988.h" + +/* + * wm8988 register cache + * We can't read the WM8988 register space when we + * are using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8988_reg[] = { + 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ + 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ + 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ + 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ + 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ + 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ + 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ + 0x0079, 0x0079, 0x0079, /* 40 */ +}; + +/* codec private data */ +struct wm8988_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + u16 reg_cache[WM8988_NUM_REG]; +}; + + +/* + * read wm8988 register cache + */ +static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg > WM8988_NUM_REG) + return -1; + return cache[reg]; +} + +/* + * write wm8988 register cache + */ +static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg > WM8988_NUM_REG) + return; + cache[reg] = value; +} + +static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8753 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8988_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0) + +/* + * WM8988 Controls + */ + +static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"}; +static const struct soc_enum bass_boost = + SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt); + +static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; +static const struct soc_enum bass_filter = + SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt); + +static const char *treble_txt[] = {"8kHz", "4kHz"}; +static const struct soc_enum treble = + SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt); + +static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"}; +static const struct soc_enum stereo_3d_lc = + SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt); + +static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"}; +static const struct soc_enum stereo_3d_uc = + SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt); + +static const char *stereo_3d_func_txt[] = {"Capture", "Playback"}; +static const struct soc_enum stereo_3d_func = + SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt); + +static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"}; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt); + +static const char *ng_type_txt[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt); + +static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const struct soc_enum deemph = + SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt); + +static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new wm8988_snd_controls[] = { + +SOC_ENUM("Bass Boost", bass_boost), +SOC_ENUM("Bass Filter", bass_filter), +SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1), + +SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0), +SOC_ENUM("Treble Cut-off", treble), + +SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0), +SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0), +SOC_ENUM("3D Lower Cut-off", stereo_3d_lc), +SOC_ENUM("3D Upper Cut-off", stereo_3d_uc), +SOC_ENUM("3D Mode", stereo_3d_func), + +SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0), +SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0), +SOC_ENUM("ALC Capture Function", alc_func), +SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0), +SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0), +SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0), +SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0), +SOC_ENUM("ALC Capture NG Type", ng_type), +SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0), + +SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC, + 0, 255, 0, adc_tlv), +SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL, + 0, 63, 0, pga_tlv), +SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0), +SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1), + +SOC_ENUM("Playback De-emphasis", deemph), + +SOC_ENUM("Capture Polarity", adcpol), +SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0), +SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0), + +SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv), + +SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1, + bypass_tlv), + +SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V, + WM8988_ROUT1V, 7, 1, 0), +SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V, + 0, 127, 0, out_tlv), + +SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V, + WM8988_ROUT2V, 7, 1, 0), +SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V, + 0, 127, 0, out_tlv), + +}; + +/* + * DAPM Controls + */ + +static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2); + + /* Use the DAC to gate LRC if active, otherwise use ADC */ + if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180) + adctl2 &= ~0x4; + else + adctl2 |= 0x4; + + return wm8988_write(codec, WM8988_ADCTL2, adctl2); +} + +static const char *wm8988_line_texts[] = { + "Line 1", "Line 2", "PGA", "Differential"}; + +static const unsigned int wm8988_line_values[] = { + 0, 1, 3, 4}; + +static const struct soc_enum wm8988_lline_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7, + ARRAY_SIZE(wm8988_line_texts), + wm8988_line_texts, + wm8988_line_values); +static const struct snd_kcontrol_new wm8988_left_line_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum); + +static const struct soc_enum wm8988_rline_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7, + ARRAY_SIZE(wm8988_line_texts), + wm8988_line_texts, + wm8988_line_values); +static const struct snd_kcontrol_new wm8988_right_line_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum); + +/* Left Mixer */ +static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0), +}; + +static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"}; +static const unsigned int wm8988_pga_val[] = { 0, 1, 3 }; + +/* Left PGA Mux */ +static const struct soc_enum wm8988_lpga_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3, + ARRAY_SIZE(wm8988_pga_sel), + wm8988_pga_sel, + wm8988_pga_val); +static const struct snd_kcontrol_new wm8988_left_pga_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum); + +/* Right PGA Mux */ +static const struct soc_enum wm8988_rpga_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3, + ARRAY_SIZE(wm8988_pga_sel), + wm8988_pga_sel, + wm8988_pga_val); +static const struct snd_kcontrol_new wm8988_right_pga_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum); + +/* Differential Mux */ +static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"}; +static const struct soc_enum diffmux = + SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel); +static const struct snd_kcontrol_new wm8988_diffmux_controls = + SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const struct soc_enum monomux = + SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux); +static const struct snd_kcontrol_new wm8988_monomux_controls = + SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = { + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0), + + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &wm8988_diffmux_controls), + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8988_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8988_monomux_controls), + + SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0, + &wm8988_left_pga_controls), + SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0, + &wm8988_right_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &wm8988_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &wm8988_right_line_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0), + + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &wm8988_left_mixer_controls[0], + ARRAY_SIZE(wm8988_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &wm8988_right_mixer_controls[0], + ARRAY_SIZE(wm8988_right_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0), + + SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("VREF"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left PGA Mux", "Line 1", "LINPUT1" }, + { "Left PGA Mux", "Line 2", "LINPUT2" }, + { "Left PGA Mux", "Differential", "Differential Mux" }, + + { "Right PGA Mux", "Line 1", "RINPUT1" }, + { "Right PGA Mux", "Line 2", "RINPUT2" }, + { "Right PGA Mux", "Differential", "Differential Mux" }, + + { "Differential Mux", "Line 1", "LINPUT1" }, + { "Differential Mux", "Line 1", "RINPUT1" }, + { "Differential Mux", "Line 2", "LINPUT2" }, + { "Differential Mux", "Line 2", "RINPUT2" }, + + { "Left ADC Mux", "Stereo", "Left PGA Mux" }, + { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" }, + { "Left ADC Mux", "Digital Mono", "Left PGA Mux" }, + + { "Right ADC Mux", "Stereo", "Right PGA Mux" }, + { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" }, + { "Right ADC Mux", "Digital Mono", "Right PGA Mux" }, + + { "Left ADC", NULL, "Left ADC Mux" }, + { "Right ADC", NULL, "Right ADC Mux" }, + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Left Mixer", "Right Playback Switch", "Right DAC" }, + { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Right Mixer", "Left Playback Switch", "Left DAC" }, + { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Left Out 1", NULL, "Left Mixer" }, + { "LOUT1", NULL, "Left Out 1" }, + { "Right Out 1", NULL, "Right Mixer" }, + { "ROUT1", NULL, "Right Out 1" }, + + { "Left Out 2", NULL, "Left Mixer" }, + { "LOUT2", NULL, "Left Out 2" }, + { "Right Out 2", NULL, "Right Mixer" }, + { "ROUT2", NULL, "Right Out 2" }, +}; + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:5; + u8 usb:1; +}; + +/* codec hifi mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 8k */ + {12288000, 8000, 1536, 0x6, 0x0}, + {11289600, 8000, 1408, 0x16, 0x0}, + {18432000, 8000, 2304, 0x7, 0x0}, + {16934400, 8000, 2112, 0x17, 0x0}, + {12000000, 8000, 1500, 0x6, 0x1}, + + /* 11.025k */ + {11289600, 11025, 1024, 0x18, 0x0}, + {16934400, 11025, 1536, 0x19, 0x0}, + {12000000, 11025, 1088, 0x19, 0x1}, + + /* 16k */ + {12288000, 16000, 768, 0xa, 0x0}, + {18432000, 16000, 1152, 0xb, 0x0}, + {12000000, 16000, 750, 0xa, 0x1}, + + /* 22.05k */ + {11289600, 22050, 512, 0x1a, 0x0}, + {16934400, 22050, 768, 0x1b, 0x0}, + {12000000, 22050, 544, 0x1b, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0xc, 0x0}, + {18432000, 32000, 576, 0xd, 0x0}, + {12000000, 32000, 375, 0xa, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x10, 0x0}, + {16934400, 44100, 384, 0x11, 0x0}, + {12000000, 44100, 272, 0x11, 0x1}, + + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0}, + {18432000, 48000, 384, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0x1e, 0x0}, + {16934400, 88200, 192, 0x1f, 0x0}, + {12000000, 88200, 136, 0x1f, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0xe, 0x0}, + {18432000, 96000, 192, 0xf, 0x0}, + {12000000, 96000, 125, 0xe, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +/* The set of rates we can generate from the above for each SYSCLK */ + +static unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + +/* + * Note that this should be called from init rather than from hw_params. + */ +static int wm8988_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8988_priv *wm8988 = codec->private_data; + + switch (freq) { + case 11289600: + case 18432000: + case 22579200: + case 36864000: + wm8988->sysclk_constraints = &constraints_112896; + wm8988->sysclk = freq; + return 0; + + case 12288000: + case 16934400: + case 24576000: + case 33868800: + wm8988->sysclk_constraints = &constraints_12288; + wm8988->sysclk = freq; + return 0; + + case 12000000: + case 24000000: + wm8988->sysclk_constraints = &constraints_12; + wm8988->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + wm8988_write(codec, WM8988_IFACE, iface); + return 0; +} + +static int wm8988_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8988_priv *wm8988 = codec->private_data; + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8988->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8988->sysclk_constraints); + + return 0; +} + +static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8988_priv *wm8988 = codec->private_data; + u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3; + u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180; + int coeff; + + coeff = get_coeff(wm8988->sysclk, params_rate(params)); + if (coeff < 0) { + coeff = get_coeff(wm8988->sysclk / 2, params_rate(params)); + srate |= 0x40; + } + if (coeff < 0) { + dev_err(codec->dev, + "Unable to configure sample rate %dHz with %dHz MCLK\n", + params_rate(params), wm8988->sysclk); + return coeff; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + + /* set iface & srate */ + wm8988_write(codec, WM8988_IFACE, iface); + if (coeff >= 0) + wm8988_write(codec, WM8988_SRATE, srate | + (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); + + return 0; +} + +static int wm8988_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7; + + if (mute) + wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8); + else + wm8988_write(codec, WM8988_ADCDAC, mute_reg); + return 0; +} + +static int wm8988_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VREF, VMID=2x50k, digital enabled */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* VREF, VMID=2x5k */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); + + /* Charge caps */ + msleep(100); + } + + /* VREF, VMID=2*500k, digital stopped */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141); + break; + + case SND_SOC_BIAS_OFF: + wm8988_write(codec, WM8988_PWR1, 0x0000); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8988_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8988_ops = { + .startup = wm8988_pcm_startup, + .hw_params = wm8988_pcm_hw_params, + .set_fmt = wm8988_set_dai_fmt, + .set_sysclk = wm8988_set_dai_sysclk, + .digital_mute = wm8988_mute, +}; + +struct snd_soc_dai wm8988_dai = { + .name = "WM8988", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8988_RATES, + .formats = WM8988_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8988_RATES, + .formats = WM8988_FORMATS, + }, + .ops = &wm8988_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8988_dai); + +static int wm8988_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8988_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < WM8988_NUM_REG; i++) { + if (i == WM8988_RESET) + continue; + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec *wm8988_codec; + +static int wm8988_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8988_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8988_codec; + codec = wm8988_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8988_snd_controls, + ARRAY_SIZE(wm8988_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, + ARRAY_SIZE(wm8988_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8988_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8988 = { + .probe = wm8988_probe, + .remove = wm8988_remove, + .suspend = wm8988_suspend, + .resume = wm8988_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988); + +static int wm8988_register(struct wm8988_priv *wm8988) +{ + struct snd_soc_codec *codec = &wm8988->codec; + int ret; + u16 reg; + + if (wm8988_codec) { + dev_err(codec->dev, "Another WM8988 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8988; + codec->name = "WM8988"; + codec->owner = THIS_MODULE; + codec->read = wm8988_read_reg_cache; + codec->write = wm8988_write; + codec->dai = &wm8988_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache); + codec->reg_cache = &wm8988->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8988_set_bias_level; + + memcpy(codec->reg_cache, wm8988_reg, + sizeof(wm8988_reg)); + + ret = wm8988_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + /* set the update bits (we always update left then right) */ + reg = wm8988_read_reg_cache(codec, WM8988_RADC); + wm8988_write(codec, WM8988_RADC, reg | 0x100); + reg = wm8988_read_reg_cache(codec, WM8988_RDAC); + wm8988_write(codec, WM8988_RDAC, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V); + wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V); + wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_RINVOL); + wm8988_write(codec, WM8988_RINVOL, reg | 0x0100); + + wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY); + + wm8988_dai.dev = codec->dev; + + wm8988_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8988_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm8988); + return ret; +} + +static void wm8988_unregister(struct wm8988_priv *wm8988) +{ + wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8988_dai); + snd_soc_unregister_codec(&wm8988->codec); + kfree(wm8988); + wm8988_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int wm8988_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8988_priv *wm8988; + struct snd_soc_codec *codec; + + wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + if (wm8988 == NULL) + return -ENOMEM; + + codec = &wm8988->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8988); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8988_register(wm8988); +} + +static int wm8988_i2c_remove(struct i2c_client *client) +{ + struct wm8988_priv *wm8988 = i2c_get_clientdata(client); + wm8988_unregister(wm8988); + return 0; +} + +static const struct i2c_device_id wm8988_i2c_id[] = { + { "wm8988", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); + +static struct i2c_driver wm8988_i2c_driver = { + .driver = { + .name = "WM8988", + .owner = THIS_MODULE, + }, + .probe = wm8988_i2c_probe, + .remove = wm8988_i2c_remove, + .id_table = wm8988_i2c_id, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int wm8988_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8988_spi_probe(struct spi_device *spi) +{ + struct wm8988_priv *wm8988; + struct snd_soc_codec *codec; + + wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + if (wm8988 == NULL) + return -ENOMEM; + + codec = &wm8988->codec; + codec->hw_write = (hw_write_t)wm8988_spi_write; + codec->control_data = spi; + codec->dev = &spi->dev; + + spi->dev.driver_data = wm8988; + + return wm8988_register(wm8988); +} + +static int __devexit wm8988_spi_remove(struct spi_device *spi) +{ + struct wm8988_priv *wm8988 = spi->dev.driver_data; + + wm8988_unregister(wm8988); + + return 0; +} + +static struct spi_driver wm8988_spi_driver = { + .driver = { + .name = "wm8988", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8988_spi_probe, + .remove = __devexit_p(wm8988_spi_remove), +}; +#endif + +static int __init wm8988_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8988_i2c_driver); + if (ret != 0) + pr_err("WM8988: Unable to register I2C driver: %d\n", ret); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8988_spi_driver); + if (ret != 0) + pr_err("WM8988: Unable to register SPI driver: %d\n", ret); +#endif + return ret; +} +module_init(wm8988_modinit); + +static void __exit wm8988_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8988_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8988_spi_driver); +#endif +} +module_exit(wm8988_exit); + + +MODULE_DESCRIPTION("ASoC WM8988 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8988.h b/sound/soc/codecs/wm8988.h new file mode 100644 index 0000000..4552d37 --- /dev/null +++ b/sound/soc/codecs/wm8988.h @@ -0,0 +1,60 @@ +/* + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie + * + * Based on WM8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _WM8988_H +#define _WM8988_H + +/* WM8988 register space */ + +#define WM8988_LINVOL 0x00 +#define WM8988_RINVOL 0x01 +#define WM8988_LOUT1V 0x02 +#define WM8988_ROUT1V 0x03 +#define WM8988_ADCDAC 0x05 +#define WM8988_IFACE 0x07 +#define WM8988_SRATE 0x08 +#define WM8988_LDAC 0x0a +#define WM8988_RDAC 0x0b +#define WM8988_BASS 0x0c +#define WM8988_TREBLE 0x0d +#define WM8988_RESET 0x0f +#define WM8988_3D 0x10 +#define WM8988_ALC1 0x11 +#define WM8988_ALC2 0x12 +#define WM8988_ALC3 0x13 +#define WM8988_NGATE 0x14 +#define WM8988_LADC 0x15 +#define WM8988_RADC 0x16 +#define WM8988_ADCTL1 0x17 +#define WM8988_ADCTL2 0x18 +#define WM8988_PWR1 0x19 +#define WM8988_PWR2 0x1a +#define WM8988_ADCTL3 0x1b +#define WM8988_ADCIN 0x1f +#define WM8988_LADCIN 0x20 +#define WM8988_RADCIN 0x21 +#define WM8988_LOUTM1 0x22 +#define WM8988_LOUTM2 0x23 +#define WM8988_ROUTM1 0x24 +#define WM8988_ROUTM2 0x25 +#define WM8988_LOUT2V 0x28 +#define WM8988_ROUT2V 0x29 +#define WM8988_LPPB 0x43 +#define WM8988_NUM_REG 0x44 + +#define WM8988_SYSCLK 0 + +extern struct snd_soc_dai wm8988_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8988; + +#endif -- cgit v1.1 From 894bf92fdec9909fefcfe907786c6c6944a22052 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Apr 2009 12:34:40 +0300 Subject: ASoC: tlv320aic23: add DSP_A format support Add DSP_A interface format support by setting the LRP bit in DSP mode. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3f4afb..21f69df 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -523,6 +523,8 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: iface_reg |= TLV320AIC23_FOR_I2S; break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_LRP_ON; case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; -- cgit v1.1 From f4c1724f3437ac70d8330968379148c954ca34c7 Mon Sep 17 00:00:00 2001 From: Alexander Beregalov Date: Sun, 12 Apr 2009 05:04:43 +0400 Subject: ASoC: n810: replace BUG() with BUG_ON() Signed-off-by: Alexander Beregalov Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a6d1178..e54e1c2 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -383,10 +383,9 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) - BUG(); - if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) - BUG(); + BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) || + (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); -- cgit v1.1 From f4976116a98f108bf385f217332aadb3ca98fe66 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 10:53:02 +0100 Subject: ASoC: WM9713 requires symmetric rates on the voice DAI Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 523bad0..aa94cc6 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1069,6 +1069,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = &wm9713_dai_ops_voice, + .symmetric_rates = 1, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); -- cgit v1.1 From 025756eca458b4a3d5e3d76baaffb2e8e3df79db Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 11:09:18 +0100 Subject: ASoC: Factor out application of power for generic widgets This is simple code motion, intended to support future refactoring of the DAPM algorithms and (more immediately) the additon of events for DACs and ADCs. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 110 ++++++++++++++++++++++++++++----------------------- 1 file changed, 60 insertions(+), 50 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 46485de..713d125 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -522,6 +522,65 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +/* Standard power change method, used to apply power changes to most + * widgets. + */ +static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) +{ + int ret; + + /* call any power change event handlers */ + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !w->power) + dapm_set_pga(w, w->power); + + dapm_update_bits(w); + + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && w->power) + dapm_set_pga(w, w->power); + + /* power up post event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; + } + + return 0; +} + /* * Scan a single DAPM widget for a complete audio path and update the * power status appropriately. @@ -601,56 +660,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, if (!power_change) return 0; - /* call any power change event handlers */ - if (w->event) - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", - w->name, w->event_flags); - - /* power up pre event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - - /* power down pre event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - - /* Lower PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && !power) - dapm_set_pga(w, power); - - dapm_update_bits(w); - - /* Raise PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && power) - dapm_set_pga(w, power); - - /* power up post event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - - /* power down post event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - - return 0; + return dapm_generic_apply_power(w); } /* -- cgit v1.1 From f6d655a6e6974e474a11b25052c29d10b80814b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 11:27:03 +0100 Subject: ASoC: Support DAPM events for DACs and ADCs Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 10 ++++++++++ sound/soc/soc-dapm.c | 16 ++++++++++------ 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a7def6a..fcc929d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -140,9 +140,19 @@ #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ + .shift = wshift, .invert = winvert, \ + .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ + .shift = wshift, .invert = winvert, \ + .event = wevent, .event_flags = wflags} /* generic register modifier widget */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 713d125..a6d7337 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -598,18 +598,22 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, if (w->id == snd_soc_dapm_adc && w->active) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); - w->power = (in != 0) ? 1 : 0; - dapm_update_bits(w); - return 0; + power = (in != 0) ? 1 : 0; + if (power == w->power) + return 0; + w->power = power; + return dapm_generic_apply_power(w); } /* active DAC */ if (w->id == snd_soc_dapm_dac && w->active) { out = is_connected_output_ep(w); dapm_clear_walk(w->codec); - w->power = (out != 0) ? 1 : 0; - dapm_update_bits(w); - return 0; + power = (out != 0) ? 1 : 0; + if (power == w->power) + return 0; + w->power = power; + return dapm_generic_apply_power(w); } /* pre and post event widgets */ -- cgit v1.1 From 6bbcb459cd50807511491ddf96bca1ef92006bf8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 11:29:10 +0100 Subject: ASoC: Move the WM9713 voice DAC powerdown to a DAPM event This ensures that we sync with the DAPM powerdown sequencing properly and don't need to bounce the power on the voice DAC so often. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aa94cc6..a6feb784 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; +static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 status, rate; + + BUG_ON(event != SND_SOC_DAPM_PRE_PMD); + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; + rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); + ac97_write(codec, AC97_EXTENDED_MID, status); + + return 0; +} + + /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1, + wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0), @@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - u16 status, rate; - - /* Gracefully shut down the voice interface. */ - status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; - rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); - schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); - ac97_write(codec, AC97_EXTENDED_MID, status); -} - static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = { static struct snd_soc_dai_ops wm9713_dai_ops_voice = { .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, .set_fmt = wm9713_set_dai_fmt, -- cgit v1.1 From a820532002e70e3a06f1ea7133e9b02443d07382 Mon Sep 17 00:00:00 2001 From: Daniel Ribeiro Date: Wed, 8 Apr 2009 10:51:24 -0300 Subject: ASoC: pxa-ssp.c fix clock/frame invert SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low) SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low) SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High) SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High) SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0). This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and DSP_B modes. Signed-off-by: Daniel Ribeiro Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c7c1996..176af7ff 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -568,7 +568,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_IF: break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(3); + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; @@ -585,7 +588,13 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_NF: sspsp |= SSPSP_SFRMP; break; + case SND_SOC_DAIFMT_NB_IF: + break; case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; -- cgit v1.1 From f2644a2c00a06236a9c5e85488b0680825bad39c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Apr 2009 19:20:14 +0100 Subject: ASoC: Add WM8960 CODEC driver The WM8960 is a low power, high quality stereo codec designed for portable digital audio applications. Stereo class D speaker drivers provide 1W per channel into 8W loads. Guaranteed low leakage, excellent PSRR and pop/click suppression mechanisms enable direct battery connection for the speaker supply. The device also integrates a complete microphone interface and a stereo headphone driver. External component requirements are drastically reduced as no separate microphone, speaker or headphone amplifiers are required. Advanced on-chip digital signal processing performs automatic level control for the microphone or line input. Stereo 24-bit sigma-delta ADCs and DACs are used with low power over-sampling digital interpolation and decimation filters and a flexible digital audio interface. The master clock can be input directly or generated internally by an onboard PLL, supporting most commonly-used clocking schemes. This driver was originally written by Liam Girdwood, with substantial subsequent additions and updates for feature completeness and changes in the ASoC framework from me. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8960.c | 969 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8960.h | 127 ++++++ 4 files changed, 1102 insertions(+) create mode 100644 sound/soc/codecs/wm8960.c create mode 100644 sound/soc/codecs/wm8960.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ab36485..121d63f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8960 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C @@ -139,6 +140,9 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8960 + tristate + config SND_SOC_WM8971 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a72548d..d8e15a4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c new file mode 100644 index 0000000..e224d8a --- /dev/null +++ b/sound/soc/codecs/wm8960.c @@ -0,0 +1,969 @@ +/* + * wm8960.c -- WM8960 ALSA SoC Audio driver + * + * Author: Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8960.h" + +#define AUDIO_NAME "wm8960" + +struct snd_soc_codec_device soc_codec_dev_wm8960; + +/* R25 - Power 1 */ +#define WM8960_VREF 0x40 + +/* R28 - Anti-pop 1 */ +#define WM8960_POBCTRL 0x80 +#define WM8960_BUFDCOPEN 0x10 +#define WM8960_BUFIOEN 0x08 +#define WM8960_SOFT_ST 0x04 +#define WM8960_HPSTBY 0x01 + +/* R29 - Anti-pop 2 */ +#define WM8960_DISOP 0x40 + +/* + * wm8960 register cache + * We can't read the WM8960 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { + 0x0097, 0x0097, 0x0000, 0x0000, + 0x0000, 0x0008, 0x0000, 0x000a, + 0x01c0, 0x0000, 0x00ff, 0x00ff, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x007b, 0x0100, 0x0032, + 0x0000, 0x00c3, 0x00c3, 0x01c0, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0100, 0x0100, 0x0050, 0x0050, + 0x0050, 0x0050, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0040, 0x0000, + 0x0000, 0x0050, 0x0050, 0x0000, + 0x0002, 0x0037, 0x004d, 0x0080, + 0x0008, 0x0031, 0x0026, 0x00e9, +}; + +struct wm8960_priv { + u16 reg_cache[WM8960_CACHEREGNUM]; + struct snd_soc_codec codec; +}; + +/* + * read wm8960 register cache + */ +static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8960_RESET) + return 0; + if (reg >= WM8960_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8960 register cache + */ +static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8960_CACHEREGNUM) + return; + cache[reg] = value; +} + +static inline unsigned int wm8960_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + return wm8960_read_reg_cache(codec, reg); +} + +/* + * write to the WM8960 register space + */ +static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8960 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8960_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0) + +/* enumerated controls */ +static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", + "Right Inverted", "Stereo Inversion"}; +static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; +static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; +static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; +static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; + +static const struct soc_enum wm8960_enum[] = { + SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph), + SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity), + SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff), + SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), + SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), + SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), +}; + +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + +static const struct snd_kcontrol_new wm8960_snd_controls[] = { +SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, + 0, 63, 0, adc_tlv), +SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, + 6, 1, 0), +SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC, + 0, 255, 0, dac_tlv), + +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2, + 7, 1, 0), +SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0), +SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0), + +SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), +SOC_ENUM("ADC Polarity", wm8960_enum[1]), +SOC_ENUM("Playback De-emphasis", wm8960_enum[0]), +SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), + +SOC_ENUM("DAC Polarity", wm8960_enum[2]), + +SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]), +SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]), +SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0), +SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0), + +SOC_ENUM("ALC Function", wm8960_enum[5]), +SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0), +SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1), +SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0), +SOC_ENUM("ALC Mode", wm8960_enum[6]), +SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), + +SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), +SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), + +SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, + 0, 127, 0), + +SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", + WM8960_BYPASS1, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume", + WM8960_LOUTMIX, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", + WM8960_BYPASS2, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", + WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), +}; + +static const struct snd_kcontrol_new wm8960_lin_boost[] = { +SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0), +SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_lin[] = { +SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_rin_boost[] = { +SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0), +SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_rin[] = { +SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_loutput_mixer[] = { +SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0), +SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_routput_mixer[] = { +SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0), +SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_mono_out[] = { +SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("LINPUT1"), +SND_SOC_DAPM_INPUT("RINPUT1"), +SND_SOC_DAPM_INPUT("LINPUT2"), +SND_SOC_DAPM_INPUT("RINPUT2"), +SND_SOC_DAPM_INPUT("LINPUT3"), +SND_SOC_DAPM_INPUT("RINPUT3"), + +SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0), + +SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0, + wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)), +SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0, + wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)), + +SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0, + wm8960_lin, ARRAY_SIZE(wm8960_lin)), +SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0, + wm8960_rin, ARRAY_SIZE(wm8960_rin)), + +SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0), +SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0), + +SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), +SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), + +SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0, + &wm8960_loutput_mixer[0], + ARRAY_SIZE(wm8960_loutput_mixer)), +SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0, + &wm8960_routput_mixer[0], + ARRAY_SIZE(wm8960_routput_mixer)), + +SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0, + &wm8960_mono_out[0], + ARRAY_SIZE(wm8960_mono_out)), + +SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0), + +SND_SOC_DAPM_OUTPUT("SPK_LP"), +SND_SOC_DAPM_OUTPUT("SPK_LN"), +SND_SOC_DAPM_OUTPUT("HP_L"), +SND_SOC_DAPM_OUTPUT("HP_R"), +SND_SOC_DAPM_OUTPUT("SPK_RP"), +SND_SOC_DAPM_OUTPUT("SPK_RN"), +SND_SOC_DAPM_OUTPUT("OUT3"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" }, + { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" }, + { "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" }, + + { "Left Input Mixer", "Boost Switch", "Left Boost Mixer", }, + { "Left Input Mixer", NULL, "LINPUT1", }, /* Really Boost Switch */ + { "Left Input Mixer", NULL, "LINPUT2" }, + { "Left Input Mixer", NULL, "LINPUT3" }, + + { "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" }, + { "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" }, + { "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" }, + + { "Right Input Mixer", "Boost Switch", "Right Boost Mixer", }, + { "Right Input Mixer", NULL, "RINPUT1", }, /* Really Boost Switch */ + { "Right Input Mixer", NULL, "RINPUT2" }, + { "Right Input Mixer", NULL, "LINPUT3" }, + + { "Left ADC", NULL, "Left Input Mixer" }, + { "Right ADC", NULL, "Right Input Mixer" }, + + { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" }, + { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} , + { "Left Output Mixer", "PCM Playback Switch", "Left DAC" }, + + { "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" }, + { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } , + { "Right Output Mixer", "PCM Playback Switch", "Right DAC" }, + + { "Mono Output Mixer", "Left Switch", "Left Output Mixer" }, + { "Mono Output Mixer", "Right Switch", "Right Output Mixer" }, + + { "LOUT1 PGA", NULL, "Left Output Mixer" }, + { "ROUT1 PGA", NULL, "Right Output Mixer" }, + + { "HP_L", NULL, "LOUT1 PGA" }, + { "HP_R", NULL, "ROUT1 PGA" }, + + { "Left Speaker PGA", NULL, "Left Output Mixer" }, + { "Right Speaker PGA", NULL, "Right Output Mixer" }, + + { "Left Speaker Output", NULL, "Left Speaker PGA" }, + { "Right Speaker Output", NULL, "Right Speaker PGA" }, + + { "SPK_LN", NULL, "Left Speaker Output" }, + { "SPK_LP", NULL, "Left Speaker Output" }, + { "SPK_RN", NULL, "Right Speaker Output" }, + { "SPK_RP", NULL, "Right Speaker Output" }, + + { "OUT3", NULL, "Mono Output Mixer", } +}; + +static int wm8960_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets, + ARRAY_SIZE(wm8960_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + wm8960_write(codec, WM8960_IFACE1, iface); + return 0; +} + +static int wm8960_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + /* set iface */ + wm8960_write(codec, WM8960_IFACE1, iface); + return 0; +} + +static int wm8960_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7; + + if (mute) + wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + else + wm8960_write(codec, WM8960_DACCTL1, mute_reg); + return 0; +} + +static int wm8960_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8960_data *pdata = codec->dev->platform_data; + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Set VMID to 2x50k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg &= ~0x180; + reg |= 0x80; + wm8960_write(codec, WM8960_POWER1, reg); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); + + /* Discharge HP output */ + reg = WM8960_DISOP; + if (pdata) + reg |= pdata->dres << 4; + wm8960_write(codec, WM8960_APOP2, reg); + + msleep(400); + + wm8960_write(codec, WM8960_APOP2, 0); + + /* Enable & ramp VMID at 2x50k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg |= 0x80; + wm8960_write(codec, WM8960_POWER1, reg); + msleep(100); + + /* Enable VREF */ + wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF); + + /* Disable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN); + } + + /* Set VMID to 2x250k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg &= ~0x180; + reg |= 0x100; + wm8960_write(codec, WM8960_POWER1, reg); + break; + + case SND_SOC_BIAS_OFF: + /* Enable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); + + /* Disable VMID and VREF, let them discharge */ + wm8960_write(codec, WM8960_POWER1, 0); + msleep(600); + + wm8960_write(codec, WM8960_APOP1, 0); + break; + } + + codec->bias_level = level; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 pre_div:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static int pll_factors(unsigned int source, unsigned int target, + struct _pll_div *pll_div) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target); + + /* Scale up target to PLL operating frequency */ + target *= 4; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->pre_div = 1; + Ndiv = target / source; + } else + pll_div->pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) { + pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv); + return -EINVAL; + } + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; + + pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n", + pll_div->n, pll_div->k, pll_div->pre_div); + + return 0; +} + +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + static struct _pll_div pll_div; + int ret; + + if (freq_in && freq_out) { + ret = pll_factors(freq_in, freq_out, &pll_div); + if (ret != 0) + return ret; + } + + /* Disable the PLL: even if we are changing the frequency the + * PLL needs to be disabled while we do so. */ + wm8960_write(codec, WM8960_CLOCK1, + wm8960_read(codec, WM8960_CLOCK1) & ~1); + wm8960_write(codec, WM8960_POWER2, + wm8960_read(codec, WM8960_POWER2) & ~1); + + if (!freq_in || !freq_out) + return 0; + + reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f; + reg |= pll_div.pre_div << 4; + reg |= pll_div.n; + + if (pll_div.k) { + reg |= 0x20; + + wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); + wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); + wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + } + wm8960_write(codec, WM8960_PLL1, reg); + + /* Turn it on */ + wm8960_write(codec, WM8960_POWER2, + wm8960_read(codec, WM8960_POWER2) | 1); + msleep(250); + wm8960_write(codec, WM8960_CLOCK1, + wm8960_read(codec, WM8960_CLOCK1) | 1); + + return 0; +} + +static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8960_SYSCLKSEL: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_SYSCLKDIV: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_DACDIV: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_OPCLKDIV: + reg = wm8960_read(codec, WM8960_PLL1) & 0x03f; + wm8960_write(codec, WM8960_PLL1, reg | div); + break; + case WM8960_DCLKDIV: + reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f; + wm8960_write(codec, WM8960_CLOCK2, reg | div); + break; + case WM8960_TOCLKSEL: + reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd; + wm8960_write(codec, WM8960_ADDCTL1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +#define WM8960_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8960_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8960_dai_ops = { + .hw_params = wm8960_hw_params, + .digital_mute = wm8960_mute, + .set_fmt = wm8960_set_dai_fmt, + .set_clkdiv = wm8960_set_dai_clkdiv, + .set_pll = wm8960_set_dai_pll, +}; + +struct snd_soc_dai wm8960_dai = { + .name = "WM8960", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8960_RATES, + .formats = WM8960_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8960_RATES, + .formats = WM8960_FORMATS,}, + .ops = &wm8960_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8960_dai); + +static int wm8960_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8960_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8960_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static struct snd_soc_codec *wm8960_codec; + +static int wm8960_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8960_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8960_codec; + codec = wm8960_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8960_snd_controls, + ARRAY_SIZE(wm8960_snd_controls)); + wm8960_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8960_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8960 = { + .probe = wm8960_probe, + .remove = wm8960_remove, + .suspend = wm8960_suspend, + .resume = wm8960_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960); + +static int wm8960_register(struct wm8960_priv *wm8960) +{ + struct wm8960_data *pdata = wm8960->codec.dev->platform_data; + struct snd_soc_codec *codec = &wm8960->codec; + int ret; + u16 reg; + + if (wm8960_codec) { + dev_err(codec->dev, "Another WM8960 is registered\n"); + return -EINVAL; + } + + if (!pdata) { + dev_warn(codec->dev, "No platform data supplied\n"); + } else { + if (pdata->dres > WM8960_DRES_MAX) { + dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); + pdata->dres = 0; + } + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8960; + codec->name = "WM8960"; + codec->owner = THIS_MODULE; + codec->read = wm8960_read_reg_cache; + codec->write = wm8960_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8960_set_bias_level; + codec->dai = &wm8960_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8960_CACHEREGNUM; + codec->reg_cache = &wm8960->reg_cache; + + memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg)); + + ret = wm8960_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8960_dai.dev = codec->dev; + + wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = wm8960_read(codec, WM8960_LINVOL); + wm8960_write(codec, WM8960_LINVOL, reg | 0x100); + reg = wm8960_read(codec, WM8960_RINVOL); + wm8960_write(codec, WM8960_RINVOL, reg | 0x100); + reg = wm8960_read(codec, WM8960_LADC); + wm8960_write(codec, WM8960_LADC, reg | 0x100); + reg = wm8960_read(codec, WM8960_RADC); + wm8960_write(codec, WM8960_RADC, reg | 0x100); + reg = wm8960_read(codec, WM8960_LDAC); + wm8960_write(codec, WM8960_LDAC, reg | 0x100); + reg = wm8960_read(codec, WM8960_RDAC); + wm8960_write(codec, WM8960_RDAC, reg | 0x100); + reg = wm8960_read(codec, WM8960_LOUT1); + wm8960_write(codec, WM8960_LOUT1, reg | 0x100); + reg = wm8960_read(codec, WM8960_ROUT1); + wm8960_write(codec, WM8960_ROUT1, reg | 0x100); + reg = wm8960_read(codec, WM8960_LOUT2); + wm8960_write(codec, WM8960_LOUT2, reg | 0x100); + reg = wm8960_read(codec, WM8960_ROUT2); + wm8960_write(codec, WM8960_ROUT2, reg | 0x100); + + wm8960_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8960_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static void wm8960_unregister(struct wm8960_priv *wm8960) +{ + wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8960_dai); + snd_soc_unregister_codec(&wm8960->codec); + kfree(wm8960); + wm8960_codec = NULL; +} + +static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8960_priv *wm8960; + struct snd_soc_codec *codec; + + wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL); + if (wm8960 == NULL) + return -ENOMEM; + + codec = &wm8960->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8960); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8960_register(wm8960); +} + +static __devexit int wm8960_i2c_remove(struct i2c_client *client) +{ + struct wm8960_priv *wm8960 = i2c_get_clientdata(client); + wm8960_unregister(wm8960); + return 0; +} + +static const struct i2c_device_id wm8960_i2c_id[] = { + { "wm8960", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); + +static struct i2c_driver wm8960_i2c_driver = { + .driver = { + .name = "WM8960 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8960_i2c_probe, + .remove = __devexit_p(wm8960_i2c_remove), + .id_table = wm8960_i2c_id, +}; + +static int __init wm8960_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8960_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm8960_modinit); + +static void __exit wm8960_exit(void) +{ + i2c_del_driver(&wm8960_i2c_driver); +} +module_exit(wm8960_exit); + + +MODULE_DESCRIPTION("ASoC WM8960 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h new file mode 100644 index 0000000..c9af56c --- /dev/null +++ b/sound/soc/codecs/wm8960.h @@ -0,0 +1,127 @@ +/* + * wm8960.h -- WM8960 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8960_H +#define _WM8960_H + +/* WM8960 register space */ + + +#define WM8960_CACHEREGNUM 56 + +#define WM8960_LINVOL 0x0 +#define WM8960_RINVOL 0x1 +#define WM8960_LOUT1 0x2 +#define WM8960_ROUT1 0x3 +#define WM8960_CLOCK1 0x4 +#define WM8960_DACCTL1 0x5 +#define WM8960_DACCTL2 0x6 +#define WM8960_IFACE1 0x7 +#define WM8960_CLOCK2 0x8 +#define WM8960_IFACE2 0x9 +#define WM8960_LDAC 0xa +#define WM8960_RDAC 0xb + +#define WM8960_RESET 0xf +#define WM8960_3D 0x10 +#define WM8960_ALC1 0x11 +#define WM8960_ALC2 0x12 +#define WM8960_ALC3 0x13 +#define WM8960_NOISEG 0x14 +#define WM8960_LADC 0x15 +#define WM8960_RADC 0x16 +#define WM8960_ADDCTL1 0x17 +#define WM8960_ADDCTL2 0x18 +#define WM8960_POWER1 0x19 +#define WM8960_POWER2 0x1a +#define WM8960_ADDCTL3 0x1b +#define WM8960_APOP1 0x1c +#define WM8960_APOP2 0x1d + +#define WM8960_LINPATH 0x20 +#define WM8960_RINPATH 0x21 +#define WM8960_LOUTMIX 0x22 + +#define WM8960_ROUTMIX 0x25 +#define WM8960_MONOMIX1 0x26 +#define WM8960_MONOMIX2 0x27 +#define WM8960_LOUT2 0x28 +#define WM8960_ROUT2 0x29 +#define WM8960_MONO 0x2a +#define WM8960_INBMIX1 0x2b +#define WM8960_INBMIX2 0x2c +#define WM8960_BYPASS1 0x2d +#define WM8960_BYPASS2 0x2e +#define WM8960_POWER3 0x2f +#define WM8960_ADDCTL4 0x30 +#define WM8960_CLASSD1 0x31 + +#define WM8960_CLASSD3 0x33 +#define WM8960_PLL1 0x34 +#define WM8960_PLL2 0x35 +#define WM8960_PLL3 0x36 +#define WM8960_PLL4 0x37 + + +/* + * WM8960 Clock dividers + */ +#define WM8960_SYSCLKDIV 0 +#define WM8960_DACDIV 1 +#define WM8960_OPCLKDIV 2 +#define WM8960_DCLKDIV 3 +#define WM8960_TOCLKSEL 4 +#define WM8960_SYSCLKSEL 5 + +#define WM8960_SYSCLK_DIV_1 (0 << 1) +#define WM8960_SYSCLK_DIV_2 (2 << 1) + +#define WM8960_SYSCLK_MCLK (0 << 0) +#define WM8960_SYSCLK_PLL (1 << 0) + +#define WM8960_DAC_DIV_1 (0 << 3) +#define WM8960_DAC_DIV_1_5 (1 << 3) +#define WM8960_DAC_DIV_2 (2 << 3) +#define WM8960_DAC_DIV_3 (3 << 3) +#define WM8960_DAC_DIV_4 (4 << 3) +#define WM8960_DAC_DIV_5_5 (5 << 3) +#define WM8960_DAC_DIV_6 (6 << 3) + +#define WM8960_DCLK_DIV_1_5 (0 << 6) +#define WM8960_DCLK_DIV_2 (1 << 6) +#define WM8960_DCLK_DIV_3 (2 << 6) +#define WM8960_DCLK_DIV_4 (3 << 6) +#define WM8960_DCLK_DIV_6 (4 << 6) +#define WM8960_DCLK_DIV_8 (5 << 6) +#define WM8960_DCLK_DIV_12 (6 << 6) +#define WM8960_DCLK_DIV_16 (7 << 6) + +#define WM8960_TOCLK_F19 (0 << 1) +#define WM8960_TOCLK_F21 (1 << 1) + +#define WM8960_OPCLK_DIV_1 (0 << 0) +#define WM8960_OPCLK_DIV_2 (1 << 0) +#define WM8960_OPCLK_DIV_3 (2 << 0) +#define WM8960_OPCLK_DIV_4 (3 << 0) +#define WM8960_OPCLK_DIV_5_5 (4 << 0) +#define WM8960_OPCLK_DIV_6 (5 << 0) + +extern struct snd_soc_dai wm8960_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8960; + +#define WM8960_DRES_400R 0 +#define WM8960_DRES_200R 1 +#define WM8960_DRES_600R 2 +#define WM8960_DRES_150R 3 +#define WM8960_DRES_MAX 3 + +struct wm8960_data { + int dres; +}; + +#endif -- cgit v1.1 From 02bec490450836ebbd628e97ec03f10b57def8ce Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Tue, 24 Mar 2009 12:24:35 +0100 Subject: ALSA: lx6464es - driver for the digigram lx6464es interface prototype of a driver for the digigram lx6464es 64 channel ethersound interface. Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 5 + sound/pci/Kconfig | 10 + sound/pci/Makefile | 1 + sound/pci/lx6464es/Makefile | 2 + sound/pci/lx6464es/lx6464es.c | 1152 ++++++++++++++++++++++++++++++++ sound/pci/lx6464es/lx6464es.h | 114 ++++ sound/pci/lx6464es/lx_core.c | 1442 +++++++++++++++++++++++++++++++++++++++++ sound/pci/lx6464es/lx_core.h | 242 +++++++ sound/pci/lx6464es/lx_defs.h | 376 +++++++++++ 9 files changed, 3344 insertions(+) create mode 100644 sound/pci/lx6464es/Makefile create mode 100644 sound/pci/lx6464es/lx6464es.c create mode 100644 sound/pci/lx6464es/lx6464es.h create mode 100644 sound/pci/lx6464es/lx_core.c create mode 100644 sound/pci/lx6464es/lx_core.h create mode 100644 sound/pci/lx6464es/lx_defs.h diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ee98cd5..2b1a695 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -1005,6 +1005,7 @@ #define PCI_DEVICE_ID_PLX_PCI200SYN 0x3196 #define PCI_DEVICE_ID_PLX_9030 0x9030 #define PCI_DEVICE_ID_PLX_9050 0x9050 +#define PCI_DEVICE_ID_PLX_9056 0x9056 #define PCI_DEVICE_ID_PLX_9080 0x9080 #define PCI_DEVICE_ID_PLX_GTEK_SERIAL2 0xa001 @@ -1847,6 +1848,10 @@ #define PCI_SUBDEVICE_ID_HYPERCOPE_METRO 0x0107 #define PCI_SUBDEVICE_ID_HYPERCOPE_CHAMP2 0x0108 +#define PCI_VENDOR_ID_DIGIGRAM 0x1369 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM 0xc001 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM 0xc002 + #define PCI_VENDOR_ID_KAWASAKI 0x136b #define PCI_DEVICE_ID_MCHIP_KL5A72002 0xff01 diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93422e3..e912b70 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -622,6 +622,16 @@ config SND_KORG1212 To compile this driver as a module, choose M here: the module will be called snd-korg1212. +config SND_LX6464ES + tristate "Digigram LX6464ES" + select SND_PCM + help + Say Y here to include support for Digigram LX6464ES boards. + + To compile this driver as a module, choose M here: the module + will be called snd-lx6464es. + + config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" select SND_AC97_CODEC diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 65b25d2..7d83e08 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -62,6 +62,7 @@ obj-$(CONFIG_SND) += \ ca0106/ \ cs46xx/ \ cs5535audio/ \ + lx6464es/ \ echoaudio/ \ emu10k1/ \ hda/ \ diff --git a/sound/pci/lx6464es/Makefile b/sound/pci/lx6464es/Makefile new file mode 100644 index 0000000..eb04a6c --- /dev/null +++ b/sound/pci/lx6464es/Makefile @@ -0,0 +1,2 @@ +snd-lx6464es-objs := lx6464es.o lx_core.o +obj-$(CONFIG_SND_LX6464ES) += snd-lx6464es.o diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c new file mode 100644 index 0000000..7bc8b8c --- /dev/null +++ b/sound/pci/lx6464es/lx6464es.c @@ -0,0 +1,1152 @@ +/* -*- linux-c -*- * + * + * ALSA driver for the digigram lx6464es interface + * + * Copyright (c) 2008, 2009 Tim Blechmann + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + */ + +#include +#include +#include +#include + +#include +#include +#include + +#include "lx6464es.h" + +MODULE_AUTHOR("Tim Blechmann"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("digigram lx6464es"); +MODULE_SUPPORTED_DEVICE("{digigram lx6464es{}}"); + + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static const char card_name[] = "LX6464ES"; + + +#define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 + +static struct pci_device_id snd_lx6464es_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), + .subvendor = PCI_VENDOR_ID_DIGIGRAM, + .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM + }, /* LX6464ES */ + { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), + .subvendor = PCI_VENDOR_ID_DIGIGRAM, + .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM + }, /* LX6464ES-CAE */ + { 0, }, +}; + +MODULE_DEVICE_TABLE(pci, snd_lx6464es_ids); + + + +/* PGO pour USERo dans le registre pci_0x06/loc_0xEC */ +#define CHIPSC_RESET_XILINX (1L<<16) + + +/* alsa callbacks */ +static struct snd_pcm_hardware lx_caps = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_3BE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_192000), + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 64, + .buffer_bytes_max = 64*2*3*MICROBLAZE_IBL_MAX*MAX_STREAM_BUFFER, + .period_bytes_min = (2*2*MICROBLAZE_IBL_MIN*2), + .period_bytes_max = (4*64*MICROBLAZE_IBL_MAX*MAX_STREAM_BUFFER), + .periods_min = 2, + .periods_max = MAX_STREAM_BUFFER, +}; + +static int lx_set_granularity(struct lx6464es *chip, u32 gran); + + +static int lx_hardware_open(struct lx6464es *chip, + struct snd_pcm_substream *substream) +{ + int err = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + int channels = runtime->channels; + int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + snd_pcm_uframes_t period_size = runtime->period_size; + + snd_printd(LXP "allocating pipe for %d channels\n", channels); + err = lx_pipe_allocate(chip, 0, is_capture, channels); + if (err < 0) { + snd_printk(KERN_ERR LXP "allocating pipe failed\n"); + return err; + } + + err = lx_set_granularity(chip, period_size); + if (err < 0) { + snd_printk(KERN_ERR LXP "setting granularity to %ld failed\n", + period_size); + return err; + } + + return 0; +} + +static int lx_hardware_start(struct lx6464es *chip, + struct snd_pcm_substream *substream) +{ + int err = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + snd_printd(LXP "setting stream format\n"); + err = lx_stream_set_format(chip, runtime, 0, is_capture); + if (err < 0) { + snd_printk(KERN_ERR LXP "setting stream format failed\n"); + return err; + } + + snd_printd(LXP "starting pipe\n"); + err = lx_pipe_start(chip, 0, is_capture); + if (err < 0) { + snd_printk(KERN_ERR LXP "starting pipe failed\n"); + return err; + } + + snd_printd(LXP "waiting for pipe to start\n"); + err = lx_pipe_wait_for_start(chip, 0, is_capture); + if (err < 0) { + snd_printk(KERN_ERR LXP "waiting for pipe failed\n"); + return err; + } + + return err; +} + + +static int lx_hardware_stop(struct lx6464es *chip, + struct snd_pcm_substream *substream) +{ + int err = 0; + int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + snd_printd(LXP "pausing pipe\n"); + err = lx_pipe_pause(chip, 0, is_capture); + if (err < 0) { + snd_printk(KERN_ERR LXP "pausing pipe failed\n"); + return err; + } + + snd_printd(LXP "waiting for pipe to become idle\n"); + err = lx_pipe_wait_for_idle(chip, 0, is_capture); + if (err < 0) { + snd_printk(KERN_ERR LXP "waiting for pipe failed\n"); + return err; + } + + snd_printd(LXP "stopping pipe\n"); + err = lx_pipe_stop(chip, 0, is_capture); + if (err < 0) { + snd_printk(LXP "stopping pipe failed\n"); + return err; + } + + return err; +} + + +static int lx_hardware_close(struct lx6464es *chip, + struct snd_pcm_substream *substream) +{ + int err = 0; + int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + snd_printd(LXP "releasing pipe\n"); + err = lx_pipe_release(chip, 0, is_capture); + if (err < 0) { + snd_printk(LXP "releasing pipe failed\n"); + return err; + } + + return err; +} + + +static int lx_pcm_open(struct snd_pcm_substream *substream) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int err = 0; + int board_rate; + + snd_printdd("->lx_pcm_open\n"); + mutex_lock(&chip->setup_mutex); + + /* copy the struct snd_pcm_hardware struct */ + runtime->hw = lx_caps; + +#if 0 + /* buffer-size should better be multiple of period-size */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + snd_printk(KERN_WARNING LXP "could not constrain periods\n"); + goto exit; + } +#endif + + /* the clock rate cannot be changed */ + board_rate = chip->board_sample_rate; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, + board_rate, board_rate); + + if (err < 0) { + snd_printk(KERN_WARNING LXP "could not constrain periods\n"); + goto exit; + } + + /* constrain period size */ + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + MICROBLAZE_IBL_MIN, + MICROBLAZE_IBL_MAX); + if (err < 0) { + snd_printk(KERN_WARNING LXP + "could not constrain period size\n"); + goto exit; + } + + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); + + snd_pcm_set_sync(substream); + err = 0; + +exit: + runtime->private_data = chip; + + mutex_unlock(&chip->setup_mutex); + snd_printdd("<-lx_pcm_open, %d\n", err); + return err; +} + +static int lx_pcm_close(struct snd_pcm_substream *substream) +{ + int err = 0; + snd_printdd("->lx_pcm_close\n"); + return err; +} + +static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream + *substream) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + snd_pcm_uframes_t pos; + unsigned long flags; + int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + struct lx_stream *lx_stream = is_capture ? &chip->capture_stream : + &chip->playback_stream; + + snd_printdd("->lx_pcm_stream_pointer\n"); + + spin_lock_irqsave(&chip->lock, flags); + pos = lx_stream->frame_pos * substream->runtime->period_size; + spin_unlock_irqrestore(&chip->lock, flags); + + snd_printdd(LXP "stream_pointer at %ld\n", pos); + return pos; +} + +static int lx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + int err = 0; + const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + snd_printdd("->lx_pcm_prepare\n"); + + mutex_lock(&chip->setup_mutex); + + if (chip->hardware_running[is_capture]) { + err = lx_hardware_stop(chip, substream); + if (err < 0) { + snd_printk(KERN_ERR LXP "failed to stop hardware. " + "Error code %d\n", err); + goto exit; + } + + err = lx_hardware_close(chip, substream); + if (err < 0) { + snd_printk(KERN_ERR LXP "failed to close hardware. " + "Error code %d\n", err); + goto exit; + } + } + + snd_printd(LXP "opening hardware\n"); + err = lx_hardware_open(chip, substream); + if (err < 0) { + snd_printk(KERN_ERR LXP "failed to open hardware. " + "Error code %d\n", err); + goto exit; + } + + err = lx_hardware_start(chip, substream); + if (err < 0) { + snd_printk(KERN_ERR LXP "failed to start hardware. " + "Error code %d\n", err); + goto exit; + } + + chip->hardware_running[is_capture] = 1; + + if (chip->board_sample_rate != substream->runtime->rate) { + if (!err) + chip->board_sample_rate = substream->runtime->rate; + } + +exit: + mutex_unlock(&chip->setup_mutex); + return err; +} + +static int lx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, int is_capture) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + int err = 0; + + snd_printdd("->lx_pcm_hw_params\n"); + + mutex_lock(&chip->setup_mutex); + + /* set dma buffer */ + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + + if (is_capture) + chip->capture_stream.stream = substream; + else + chip->playback_stream.stream = substream; + + mutex_unlock(&chip->setup_mutex); + return err; +} + +static int lx_pcm_hw_params_playback(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return lx_pcm_hw_params(substream, hw_params, 0); +} + +static int lx_pcm_hw_params_capture(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return lx_pcm_hw_params(substream, hw_params, 1); +} + +static int lx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + int err = 0; + int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + + snd_printdd("->lx_pcm_hw_free\n"); + mutex_lock(&chip->setup_mutex); + + if (chip->hardware_running[is_capture]) { + err = lx_hardware_stop(chip, substream); + if (err < 0) { + snd_printk(KERN_ERR LXP "failed to stop hardware. " + "Error code %d\n", err); + goto exit; + } + + err = lx_hardware_close(chip, substream); + if (err < 0) { + snd_printk(KERN_ERR LXP "failed to close hardware. " + "Error code %d\n", err); + goto exit; + } + + chip->hardware_running[is_capture] = 0; + } + + err = snd_pcm_lib_free_pages(substream); + + if (is_capture) + chip->capture_stream.stream = 0; + else + chip->playback_stream.stream = 0; + +exit: + mutex_unlock(&chip->setup_mutex); + return err; +} + +static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream) +{ + struct snd_pcm_substream *substream = lx_stream->stream; + const int is_capture = lx_stream->is_capture; + + int err; + + const u32 channels = substream->runtime->channels; + const u32 bytes_per_frame = channels * 3; + const u32 period_size = substream->runtime->period_size; + const u32 periods = substream->runtime->periods; + const u32 period_bytes = period_size * bytes_per_frame; + + dma_addr_t buf = substream->dma_buffer.addr; + int i; + + u32 needed, freed; + u32 size_array[5]; + + for (i = 0; i != periods; ++i) { + u32 buffer_index = 0; + + err = lx_buffer_ask(chip, 0, is_capture, &needed, &freed, + size_array); + snd_printdd(LXP "starting: needed %d, freed %d\n", + needed, freed); + + err = lx_buffer_give(chip, 0, is_capture, period_bytes, + lower_32_bits(buf), upper_32_bits(buf), + &buffer_index); + + snd_printdd(LXP "starting: buffer index %x on %p (%d bytes)\n", + buffer_index, (void *)buf, period_bytes); + buf += period_bytes; + } + + err = lx_buffer_ask(chip, 0, is_capture, &needed, &freed, size_array); + snd_printdd(LXP "starting: needed %d, freed %d\n", needed, freed); + + snd_printd(LXP "starting: starting stream\n"); + err = lx_stream_start(chip, 0, is_capture); + if (err < 0) + snd_printk(KERN_ERR LXP "couldn't start stream\n"); + else + lx_stream->status = LX_STREAM_STATUS_RUNNING; + + lx_stream->frame_pos = 0; +} + +static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream) +{ + const int is_capture = lx_stream->is_capture; + int err; + + snd_printd(LXP "stopping: stopping stream\n"); + err = lx_stream_stop(chip, 0, is_capture); + if (err < 0) + snd_printk(KERN_ERR LXP "couldn't stop stream\n"); + else + lx_stream->status = LX_STREAM_STATUS_FREE; + +} + +static void lx_trigger_tasklet_dispatch_stream(struct lx6464es *chip, + struct lx_stream *lx_stream) +{ + switch (lx_stream->status) { + case LX_STREAM_STATUS_SCHEDULE_RUN: + lx_trigger_start(chip, lx_stream); + break; + + case LX_STREAM_STATUS_SCHEDULE_STOP: + lx_trigger_stop(chip, lx_stream); + break; + + default: + break; + } +} + +static void lx_trigger_tasklet(unsigned long data) +{ + struct lx6464es *chip = (struct lx6464es *)data; + unsigned long flags; + + snd_printdd("->lx_trigger_tasklet\n"); + + spin_lock_irqsave(&chip->lock, flags); + lx_trigger_tasklet_dispatch_stream(chip, &chip->capture_stream); + lx_trigger_tasklet_dispatch_stream(chip, &chip->playback_stream); + spin_unlock_irqrestore(&chip->lock, flags); +} + +static int lx_pcm_trigger_dispatch(struct lx6464es *chip, + struct lx_stream *lx_stream, int cmd) +{ + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + lx_stream->status = LX_STREAM_STATUS_SCHEDULE_RUN; + break; + + case SNDRV_PCM_TRIGGER_STOP: + lx_stream->status = LX_STREAM_STATUS_SCHEDULE_STOP; + break; + + default: + err = -EINVAL; + goto exit; + } + tasklet_schedule(&chip->trigger_tasklet); + +exit: + return err; +} + + +static int lx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + struct lx_stream *stream = is_capture ? &chip->capture_stream : + &chip->playback_stream; + + snd_printdd("->lx_pcm_trigger\n"); + + return lx_pcm_trigger_dispatch(chip, stream, cmd); +} + +static int snd_lx6464es_free(struct lx6464es *chip) +{ + snd_printdd("->snd_lx6464es_free\n"); + + lx_irq_disable(chip); + + if (chip->irq >= 0) + free_irq(chip->irq, chip); + + iounmap(chip->port_dsp_bar); + ioport_unmap(chip->port_plx_remapped); + + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + + kfree(chip); + + return 0; +} + +static int snd_lx6464es_dev_free(struct snd_device *device) +{ + return snd_lx6464es_free(device->device_data); +} + +/* reset the dsp during initialization */ +static int __devinit lx_init_xilinx_reset(struct lx6464es *chip) +{ + int i; + u32 plx_reg = lx_plx_reg_read(chip, ePLX_CHIPSC); + + snd_printdd("->lx_init_xilinx_reset\n"); + + /* activate reset of xilinx */ + plx_reg &= ~CHIPSC_RESET_XILINX; + + lx_plx_reg_write(chip, ePLX_CHIPSC, plx_reg); + msleep(1); + + lx_plx_reg_write(chip, ePLX_MBOX3, 0); + msleep(1); + + plx_reg |= CHIPSC_RESET_XILINX; + lx_plx_reg_write(chip, ePLX_CHIPSC, plx_reg); + + /* deactivate reset of xilinx */ + for (i = 0; i != 100; ++i) { + u32 reg_mbox3; + msleep(10); + reg_mbox3 = lx_plx_reg_read(chip, ePLX_MBOX3); + if (reg_mbox3) { + snd_printd(LXP "xilinx reset done\n"); + snd_printdd(LXP "xilinx took %d loops\n", i); + break; + } + } + + /* todo: add some error handling? */ + + /* clear mr */ + lx_dsp_reg_write(chip, eReg_CSM, 0); + + /* le xilinx ES peut ne pas etre encore pret, on attend. */ + msleep(600); + + return 0; +} + +static int __devinit lx_init_xilinx_test(struct lx6464es *chip) +{ + u32 reg; + + snd_printdd("->lx_init_xilinx_test\n"); + + /* TEST if we have access to Xilinx/MicroBlaze */ + lx_dsp_reg_write(chip, eReg_CSM, 0); + + reg = lx_dsp_reg_read(chip, eReg_CSM); + + if (reg) { + snd_printk(KERN_ERR LXP "Problem: Reg_CSM %x.\n", reg); + + /* PCI9056_SPACE0_REMAP */ + lx_plx_reg_write(chip, ePLX_PCICR, 1); + + reg = lx_dsp_reg_read(chip, eReg_CSM); + if (reg) { + snd_printk(KERN_ERR LXP "Error: Reg_CSM %x.\n", reg); + return -EAGAIN; /* seems to be appropriate */ + } + } + + snd_printd(LXP "Xilinx/MicroBlaze access test successful\n"); + + return 0; +} + +/* initialize ethersound */ +static int __devinit lx_init_ethersound_config(struct lx6464es *chip) +{ + int i; + u32 orig_conf_es = lx_dsp_reg_read(chip, eReg_CONFES); + + u32 default_conf_es = (64 << IOCR_OUTPUTS_OFFSET) | + (64 << IOCR_INPUTS_OFFSET) | + (FREQ_RATIO_SINGLE_MODE << FREQ_RATIO_OFFSET); + + u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) + | (default_conf_es & CONFES_WRITE_PART_MASK); + + snd_printdd("->lx_init_ethersound\n"); + + chip->freq_ratio = FREQ_RATIO_SINGLE_MODE; + + /* + * write it to the card ! + * this actually kicks the ES xilinx, the first time since poweron. + * the MAC address in the Reg_ADMACESMSB Reg_ADMACESLSB registers + * is not ready before this is done, and the bit 2 in Reg_CSES is set. + * */ + lx_dsp_reg_write(chip, eReg_CONFES, conf_es); + + for (i = 0; i != 1000; ++i) { + if (lx_dsp_reg_read(chip, eReg_CSES) & 4) { + snd_printd(LXP "ethersound initialized after %dms\n", + i); + goto ethersound_initialized; + } + msleep(1); + } + snd_printk(KERN_WARNING LXP + "ethersound could not be initialized after %dms\n", i); + return -ETIMEDOUT; + + ethersound_initialized: + snd_printd(LXP "ethersound initialized\n"); + return 0; +} + +static int __devinit lx_init_get_version_features(struct lx6464es *chip) +{ + u32 dsp_version; + + int err; + + snd_printdd("->lx_init_get_version_features\n"); + + err = lx_dsp_get_version(chip, &dsp_version); + + if (err == 0) { + u32 freq; + + snd_printk(LXP "DSP version: V%02d.%02d #%d\n", + (dsp_version>>16) & 0xff, (dsp_version>>8) & 0xff, + dsp_version & 0xff); + + /* later: what firmware version do we expect? */ + + /* retrieve Play/Rec features */ + /* done here because we may have to handle alternate + * DSP files. */ + /* later */ + + /* init the EtherSound sample rate */ + err = lx_dsp_get_clock_frequency(chip, &freq); + if (err == 0) + chip->board_sample_rate = freq; + snd_printd(LXP "actual clock frequency %d\n", freq); + } else { + snd_printk(KERN_ERR LXP "DSP corrupted \n"); + err = -EAGAIN; + } + + return err; +} + +static int lx_set_granularity(struct lx6464es *chip, u32 gran) +{ + int err = 0; + u32 snapped_gran = MICROBLAZE_IBL_MIN; + + snd_printdd("->lx_set_granularity\n"); + + /* blocksize is a power of 2 */ + while ((snapped_gran < gran) && + (snapped_gran < MICROBLAZE_IBL_MAX)) { + snapped_gran *= 2; + } + + if (snapped_gran == chip->pcm_granularity) + return 0; + + err = lx_dsp_set_granularity(chip, snapped_gran); + if (err < 0) { + snd_printk(KERN_WARNING LXP "could not set granularity\n"); + err = -EAGAIN; + } + + if (snapped_gran != gran) + snd_printk(LXP "snapped blocksize to %d\n", snapped_gran); + + snd_printd(LXP "set blocksize on board %d\n", snapped_gran); + chip->pcm_granularity = snapped_gran; + + return err; +} + +/* initialize and test the xilinx dsp chip */ +static int __devinit lx_init_dsp(struct lx6464es *chip) +{ + int err; + u8 mac_address[6]; + int i; + + snd_printdd("->lx_init_dsp\n"); + + snd_printd(LXP "initialize board\n"); + err = lx_init_xilinx_reset(chip); + if (err) + return err; + + snd_printd(LXP "testing board\n"); + err = lx_init_xilinx_test(chip); + if (err) + return err; + + snd_printd(LXP "initialize ethersound configuration\n"); + err = lx_init_ethersound_config(chip); + if (err) + return err; + + lx_irq_enable(chip); + + /** \todo the mac address should be ready by not, but it isn't, + * so we wait for it */ + for (i = 0; i != 1000; ++i) { + err = lx_dsp_get_mac(chip, mac_address); + if (err) + return err; + if (mac_address[0] || mac_address[1] || mac_address[2] || + mac_address[3] || mac_address[4] || mac_address[5]) + goto mac_ready; + msleep(1); + } + return -ETIMEDOUT; + +mac_ready: + snd_printd(LXP "mac address ready read after: %dms\n", i); + snd_printk(LXP "mac address: %02X.%02X.%02X.%02X.%02X.%02X\n", + mac_address[0], mac_address[1], mac_address[2], + mac_address[3], mac_address[4], mac_address[5]); + + err = lx_init_get_version_features(chip); + if (err) + return err; + + lx_set_granularity(chip, MICROBLAZE_IBL_DEFAULT); + + chip->playback_mute = 0; + + return err; +} + +static struct snd_pcm_ops lx_ops_playback = { + .open = lx_pcm_open, + .close = lx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = lx_pcm_prepare, + .hw_params = lx_pcm_hw_params_playback, + .hw_free = lx_pcm_hw_free, + .trigger = lx_pcm_trigger, + .pointer = lx_pcm_stream_pointer, +}; + +static struct snd_pcm_ops lx_ops_capture = { + .open = lx_pcm_open, + .close = lx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = lx_pcm_prepare, + .hw_params = lx_pcm_hw_params_capture, + .hw_free = lx_pcm_hw_free, + .trigger = lx_pcm_trigger, + .pointer = lx_pcm_stream_pointer, +}; + +static int __devinit lx_pcm_create(struct lx6464es *chip) +{ + int err; + struct snd_pcm *pcm; + + u32 size = 64 * /* channels */ + 3 * /* 24 bit samples */ + MAX_STREAM_BUFFER * /* periods */ + MICROBLAZE_IBL_MAX * /* frames per period */ + 2; /* duplex */ + + size = PAGE_ALIGN(size); + + /* hardcoded device name & channel count */ + err = snd_pcm_new(chip->card, (char *)card_name, 0, + 1, 1, &pcm); + + pcm->private_data = chip; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &lx_ops_playback); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &lx_ops_capture); + + pcm->info_flags = 0; + strcpy(pcm->name, card_name); + + err = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + size, size); + if (err < 0) + return err; + + chip->pcm = pcm; + chip->capture_stream.is_capture = 1; + + return 0; +} + +static int lx_control_playback_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int lx_control_playback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lx6464es *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = chip->playback_mute; + return 0; +} + +static int lx_control_playback_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lx6464es *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int current_value = chip->playback_mute; + + if (current_value != ucontrol->value.integer.value[0]) { + lx_level_unmute(chip, 0, !current_value); + chip->playback_mute = !current_value; + changed = 1; + } + return changed; +} + +static struct snd_kcontrol_new lx_control_playback_switch __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0, + .info = lx_control_playback_info, + .get = lx_control_playback_get, + .put = lx_control_playback_put +}; + + + +static void lx_proc_levels_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + u32 levels[64]; + int err; + int i, j; + struct lx6464es *chip = entry->private_data; + + snd_iprintf(buffer, "capture levels:\n"); + err = lx_level_peaks(chip, 1, 64, levels); + if (err < 0) + return; + + for (i = 0; i != 8; ++i) { + for (j = 0; j != 8; ++j) + snd_iprintf(buffer, "%08x ", levels[i*8+j]); + snd_iprintf(buffer, "\n"); + } + + snd_iprintf(buffer, "\nplayback levels:\n"); + + err = lx_level_peaks(chip, 0, 64, levels); + if (err < 0) + return; + + for (i = 0; i != 8; ++i) { + for (j = 0; j != 8; ++j) + snd_iprintf(buffer, "%08x ", levels[i*8+j]); + snd_iprintf(buffer, "\n"); + } + + snd_iprintf(buffer, "\n"); +} + +static int __devinit lx_proc_create(struct snd_card *card, struct lx6464es *chip) +{ + struct snd_info_entry *entry; + int err = snd_card_proc_new(card, "levels", &entry); + if (err < 0) + return err; + + snd_info_set_text_ops(entry, chip, lx_proc_levels_read); + return 0; +} + + +static int __devinit snd_lx6464es_create(struct snd_card *card, + struct pci_dev *pci, + struct lx6464es **rchip) +{ + struct lx6464es *chip; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_lx6464es_dev_free, + }; + + snd_printdd("->snd_lx6464es_create\n"); + + *rchip = NULL; + + /* enable PCI device */ + err = pci_enable_device(pci); + if (err < 0) + return err; + + pci_set_master(pci); + + /* check if we can restrict PCI DMA transfers to 32 bits */ + err = pci_set_dma_mask(pci, DMA_32BIT_MASK); + if (err < 0) { + snd_printk(KERN_ERR "architecture does not support " + "32bit PCI busmaster DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + err = -ENOMEM; + goto alloc_failed; + } + + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* initialize synchronization structs */ + spin_lock_init(&chip->lock); + spin_lock_init(&chip->msg_lock); + mutex_init(&chip->setup_mutex); + tasklet_init(&chip->trigger_tasklet, lx_trigger_tasklet, + (unsigned long)chip); + tasklet_init(&chip->tasklet_capture, lx_tasklet_capture, + (unsigned long)chip); + tasklet_init(&chip->tasklet_playback, lx_tasklet_playback, + (unsigned long)chip); + + /* request resources */ + err = pci_request_regions(pci, card_name); + if (err < 0) + goto request_regions_failed; + + /* plx port */ + chip->port_plx = pci_resource_start(pci, 1); + chip->port_plx_remapped = ioport_map(chip->port_plx, + pci_resource_len(pci, 1)); + + /* dsp port */ + chip->port_dsp_bar = pci_ioremap_bar(pci, 2); + + err = request_irq(pci->irq, lx_interrupt, IRQF_SHARED, + card_name, chip); + if (err) { + snd_printk(KERN_ERR LXP "unable to grab IRQ %d\n", pci->irq); + goto request_irq_failed; + } + chip->irq = pci->irq; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto device_new_failed; + + err = lx_init_dsp(chip); + if (err < 0) { + snd_printk(KERN_ERR LXP "error during DSP initialization\n"); + return err; + } + + err = lx_pcm_create(chip); + if (err < 0) + return err; + + err = lx_proc_create(card, chip); + if (err < 0) + return err; + + err = snd_ctl_add(card, snd_ctl_new1(&lx_control_playback_switch, + chip)); + if (err < 0) + return err; + + snd_card_set_dev(card, &pci->dev); + + *rchip = chip; + return 0; + +device_new_failed: + free_irq(pci->irq, chip); + +request_irq_failed: + pci_release_regions(pci); + +request_regions_failed: + kfree(chip); + +alloc_failed: + pci_disable_device(pci); + + return err; +} + +static int __devinit snd_lx6464es_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct lx6464es *chip; + int err; + + snd_printdd("->snd_lx6464es_probe\n"); + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = snd_lx6464es_create(card, pci, &chip); + if (err < 0) { + snd_printk(KERN_ERR LXP "error during snd_lx6464es_create\n"); + goto out_free; + } + + strcpy(card->driver, "lx6464es"); + strcpy(card->shortname, "Digigram LX6464ES"); + sprintf(card->longname, "%s at 0x%lx, 0x%p, irq %i", + card->shortname, chip->port_plx, + chip->port_dsp_bar, chip->irq); + + err = snd_card_register(card); + if (err < 0) + goto out_free; + + snd_printdd(LXP "initialization successful\n"); + pci_set_drvdata(pci, card); + dev++; + return 0; + +out_free: + snd_card_free(card); + return err; + +} + +static void __devexit snd_lx6464es_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + + +static struct pci_driver driver = { + .name = "Digigram LX6464ES", + .id_table = snd_lx6464es_ids, + .probe = snd_lx6464es_probe, + .remove = __devexit_p(snd_lx6464es_remove), +}; + + +/* module initialization */ +static int __init mod_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit mod_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(mod_init); +module_exit(mod_exit); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h new file mode 100644 index 0000000..012c010 --- /dev/null +++ b/sound/pci/lx6464es/lx6464es.h @@ -0,0 +1,114 @@ +/* -*- linux-c -*- * + * + * ALSA driver for the digigram lx6464es interface + * + * Copyright (c) 2009 Tim Blechmann + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + */ + +#ifndef LX6464ES_H +#define LX6464ES_H + +#include +#include + +#include +#include + +#include "lx_core.h" + +#define LXP "LX6464ES: " + +enum { + ES_cmd_free = 0, /* no command executing */ + ES_cmd_processing = 1, /* execution of a read/write command */ + ES_read_pending = 2, /* a asynchron read command is pending */ + ES_read_finishing = 3, /* a read command has finished waiting (set by + * Interrupt or CancelIrp) */ +}; + +enum lx_stream_status { + LX_STREAM_STATUS_FREE, +/* LX_STREAM_STATUS_OPEN, */ + LX_STREAM_STATUS_SCHEDULE_RUN, +/* LX_STREAM_STATUS_STARTED, */ + LX_STREAM_STATUS_RUNNING, + LX_STREAM_STATUS_SCHEDULE_STOP, +/* LX_STREAM_STATUS_STOPPED, */ +/* LX_STREAM_STATUS_PAUSED */ +}; + + +struct lx_stream { + struct snd_pcm_substream *stream; + snd_pcm_uframes_t frame_pos; + enum lx_stream_status status; /* free, open, running, draining + * pause */ + int is_capture:1; +}; + + +struct lx6464es { + struct snd_card *card; + struct pci_dev *pci; + int irq; + + spinlock_t lock; /* interrupt spinlock */ + struct mutex setup_mutex; /* mutex used in hw_params, open + * and close */ + + struct tasklet_struct trigger_tasklet; /* trigger tasklet */ + struct tasklet_struct tasklet_capture; + struct tasklet_struct tasklet_playback; + + /* ports */ + unsigned long port_plx; /* io port (size=256) */ + void __iomem *port_plx_remapped; /* remapped plx port */ + void __iomem *port_dsp_bar; /* memory port (32-bit, + * non-prefetchable, + * size=8K) */ + + /* messaging */ + spinlock_t msg_lock; /* message spinlock */ + atomic_t send_message_locked; + struct lx_rmh rmh; + + /* configuration */ + uint freq_ratio : 2; + uint playback_mute : 1; + uint hardware_running[2]; + u32 board_sample_rate; /* sample rate read from + * board */ + u32 sample_rate; /* our sample rate */ + u16 pcm_granularity; /* board blocksize */ + + /* dma */ + struct snd_dma_buffer capture_dma_buf; + struct snd_dma_buffer playback_dma_buf; + + /* pcm */ + struct snd_pcm *pcm; + + /* streams */ + struct lx_stream capture_stream; + struct lx_stream playback_stream; +}; + + +#endif /* LX6464ES_H */ diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c new file mode 100644 index 0000000..a9f8f88 --- /dev/null +++ b/sound/pci/lx6464es/lx_core.c @@ -0,0 +1,1442 @@ +/* -*- linux-c -*- * + * + * ALSA driver for the digigram lx6464es interface + * low-level interface + * + * Copyright (c) 2009 Tim Blechmann + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + */ + +/* #define RMH_DEBUG 1 */ + +#include +#include +#include + +#include "lx6464es.h" +#include "lx_core.h" + +/* low-level register access */ + +static const unsigned long dsp_port_offsets[] = { + 0, + 0x400, + 0x401, + 0x402, + 0x403, + 0x404, + 0x405, + 0x406, + 0x407, + 0x408, + 0x409, + 0x40a, + 0x40b, + 0x40c, + + 0x410, + 0x411, + 0x412, + 0x413, + 0x414, + 0x415, + 0x416, + + 0x420, + 0x430, + 0x431, + 0x432, + 0x433, + 0x434, + 0x440 +}; + +static void __iomem *lx_dsp_register(struct lx6464es *chip, int port) +{ + void __iomem *base_address = chip->port_dsp_bar; + return base_address + dsp_port_offsets[port]*4; +} + +unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port) +{ + void __iomem *address = lx_dsp_register(chip, port); + return ioread32(address); +} + +void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len) +{ + void __iomem *address = lx_dsp_register(chip, port); + memcpy_fromio(data, address, len*sizeof(u32)); +} + + +void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data) +{ + void __iomem *address = lx_dsp_register(chip, port); + iowrite32(data, address); +} + +void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, + u32 len) +{ + void __iomem *address = lx_dsp_register(chip, port); + memcpy_toio(address, data, len*sizeof(u32)); +} + + +static const unsigned long plx_port_offsets[] = { + 0x04, + 0x40, + 0x44, + 0x48, + 0x4c, + 0x50, + 0x54, + 0x58, + 0x5c, + 0x64, + 0x68, + 0x6C +}; + +static void __iomem *lx_plx_register(struct lx6464es *chip, int port) +{ + void __iomem *base_address = chip->port_plx_remapped; + return base_address + plx_port_offsets[port]; +} + +unsigned long lx_plx_reg_read(struct lx6464es *chip, int port) +{ + void __iomem *address = lx_plx_register(chip, port); + return ioread32(address); +} + +void lx_plx_reg_write(struct lx6464es *chip, int port, u32 data) +{ + void __iomem *address = lx_plx_register(chip, port); + iowrite32(data, address); +} + +u32 lx_plx_mbox_read(struct lx6464es *chip, int mbox_nr) +{ + int index; + + switch (mbox_nr) { + case 1: + index = ePLX_MBOX1; break; + case 2: + index = ePLX_MBOX2; break; + case 3: + index = ePLX_MBOX3; break; + case 4: + index = ePLX_MBOX4; break; + case 5: + index = ePLX_MBOX5; break; + case 6: + index = ePLX_MBOX6; break; + case 7: + index = ePLX_MBOX7; break; + case 0: /* reserved for HF flags */ + snd_BUG(); + default: + return 0xdeadbeef; + } + + return lx_plx_reg_read(chip, index); +} + +int lx_plx_mbox_write(struct lx6464es *chip, int mbox_nr, u32 value) +{ + int index = -1; + + switch (mbox_nr) { + case 1: + index = ePLX_MBOX1; break; + case 3: + index = ePLX_MBOX3; break; + case 4: + index = ePLX_MBOX4; break; + case 5: + index = ePLX_MBOX5; break; + case 6: + index = ePLX_MBOX6; break; + case 7: + index = ePLX_MBOX7; break; + case 0: /* reserved for HF flags */ + case 2: /* reserved for Pipe States + * the DSP keeps an image of it */ + snd_BUG(); + return -EBADRQC; + } + + lx_plx_reg_write(chip, index, value); + return 0; +} + + +/* rmh */ + +#ifdef CONFIG_SND_DEBUG +#define CMD_NAME(a) a +#else +#define CMD_NAME(a) NULL +#endif + +#define Reg_CSM_MR 0x00000002 +#define Reg_CSM_MC 0x00000001 + +struct dsp_cmd_info { + u32 dcCodeOp; /* Op Code of the command (usually 1st 24-bits + * word).*/ + u16 dcCmdLength; /* Command length in words of 24 bits.*/ + u16 dcStatusType; /* Status type: 0 for fixed length, 1 for + * random. */ + u16 dcStatusLength; /* Status length (if fixed).*/ + char *dcOpName; +}; + +/* + Initialization and control data for the Microblaze interface + - OpCode: + the opcode field of the command set at the proper offset + - CmdLength + the number of command words + - StatusType + offset in the status registers: 0 means that the return value may be + different from 0, and must be read + - StatusLength + the number of status words (in addition to the return value) +*/ + +static struct dsp_cmd_info dsp_commands[] = +{ + { (CMD_00_INFO_DEBUG << OPCODE_OFFSET) , 1 /*custom*/ + , 1 , 0 /**/ , CMD_NAME("INFO_DEBUG") }, + { (CMD_01_GET_SYS_CFG << OPCODE_OFFSET) , 1 /**/ + , 1 , 2 /**/ , CMD_NAME("GET_SYS_CFG") }, + { (CMD_02_SET_GRANULARITY << OPCODE_OFFSET) , 1 /**/ + , 1 , 0 /**/ , CMD_NAME("SET_GRANULARITY") }, + { (CMD_03_SET_TIMER_IRQ << OPCODE_OFFSET) , 1 /**/ + , 1 , 0 /**/ , CMD_NAME("SET_TIMER_IRQ") }, + { (CMD_04_GET_EVENT << OPCODE_OFFSET) , 1 /**/ + , 1 , 0 /*up to 10*/ , CMD_NAME("GET_EVENT") }, + { (CMD_05_GET_PIPES << OPCODE_OFFSET) , 1 /**/ + , 1 , 2 /*up to 4*/ , CMD_NAME("GET_PIPES") }, + { (CMD_06_ALLOCATE_PIPE << OPCODE_OFFSET) , 1 /**/ + , 0 , 0 /**/ , CMD_NAME("ALLOCATE_PIPE") }, + { (CMD_07_RELEASE_PIPE << OPCODE_OFFSET) , 1 /**/ + , 0 , 0 /**/ , CMD_NAME("RELEASE_PIPE") }, + { (CMD_08_ASK_BUFFERS << OPCODE_OFFSET) , 1 /**/ + , 1 , MAX_STREAM_BUFFER , CMD_NAME("ASK_BUFFERS") }, + { (CMD_09_STOP_PIPE << OPCODE_OFFSET) , 1 /**/ + , 0 , 0 /*up to 2*/ , CMD_NAME("STOP_PIPE") }, + { (CMD_0A_GET_PIPE_SPL_COUNT << OPCODE_OFFSET) , 1 /**/ + , 1 , 1 /*up to 2*/ , CMD_NAME("GET_PIPE_SPL_COUNT") }, + { (CMD_0B_TOGGLE_PIPE_STATE << OPCODE_OFFSET) , 1 /*up to 5*/ + , 1 , 0 /**/ , CMD_NAME("TOGGLE_PIPE_STATE") }, + { (CMD_0C_DEF_STREAM << OPCODE_OFFSET) , 1 /*up to 4*/ + , 1 , 0 /**/ , CMD_NAME("DEF_STREAM") }, + { (CMD_0D_SET_MUTE << OPCODE_OFFSET) , 3 /**/ + , 1 , 0 /**/ , CMD_NAME("SET_MUTE") }, + { (CMD_0E_GET_STREAM_SPL_COUNT << OPCODE_OFFSET) , 1/**/ + , 1 , 2 /**/ , CMD_NAME("GET_STREAM_SPL_COUNT") }, + { (CMD_0F_UPDATE_BUFFER << OPCODE_OFFSET) , 3 /*up to 4*/ + , 0 , 1 /**/ , CMD_NAME("UPDATE_BUFFER") }, + { (CMD_10_GET_BUFFER << OPCODE_OFFSET) , 1 /**/ + , 1 , 4 /**/ , CMD_NAME("GET_BUFFER") }, + { (CMD_11_CANCEL_BUFFER << OPCODE_OFFSET) , 1 /**/ + , 1 , 1 /*up to 4*/ , CMD_NAME("CANCEL_BUFFER") }, + { (CMD_12_GET_PEAK << OPCODE_OFFSET) , 1 /**/ + , 1 , 1 /**/ , CMD_NAME("GET_PEAK") }, + { (CMD_13_SET_STREAM_STATE << OPCODE_OFFSET) , 1 /**/ + , 1 , 0 /**/ , CMD_NAME("SET_STREAM_STATE") }, +}; + +static void lx_message_init(struct lx_rmh *rmh, enum cmd_mb_opcodes cmd) +{ + snd_BUG_ON(cmd >= CMD_14_INVALID); + + rmh->cmd[0] = dsp_commands[cmd].dcCodeOp; + rmh->cmd_len = dsp_commands[cmd].dcCmdLength; + rmh->stat_len = dsp_commands[cmd].dcStatusLength; + rmh->dsp_stat = dsp_commands[cmd].dcStatusType; + rmh->cmd_idx = cmd; + memset(&rmh->cmd[1], 0, (REG_CRM_NUMBER - 1) * sizeof(u32)); + +#ifdef CONFIG_SND_DEBUG + memset(rmh->stat, 0, REG_CRM_NUMBER * sizeof(u32)); +#endif +#ifdef RMH_DEBUG + rmh->cmd_idx = cmd; +#endif +} + +#ifdef RMH_DEBUG +#define LXRMH "lx6464es rmh: " +static void lx_message_dump(struct lx_rmh *rmh) +{ + u8 idx = rmh->cmd_idx; + int i; + + snd_printk(LXRMH "command %s\n", dsp_commands[idx].dcOpName); + + for (i = 0; i != rmh->cmd_len; ++i) + snd_printk(LXRMH "\tcmd[%d] %08x\n", i, rmh->cmd[i]); + + for (i = 0; i != rmh->stat_len; ++i) + snd_printk(LXRMH "\tstat[%d]: %08x\n", i, rmh->stat[i]); + snd_printk("\n"); +} +#else +static inline void lx_message_dump(struct lx_rmh *rmh) +{} +#endif + + + +/* sleep 500 - 100 = 400 times 100us -> the timeout is >= 40 ms */ +#define XILINX_TIMEOUT_MS 40 +#define XILINX_POLL_NO_SLEEP 100 +#define XILINX_POLL_ITERATIONS 150 + +static int lx_message_send(struct lx6464es *chip, struct lx_rmh *rmh) +{ + u32 reg = ED_DSP_TIMED_OUT; + int dwloop; + int answer_received; + + if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) { + snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg); + return -EBUSY; + } + + /* write command */ + lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len); + + snd_BUG_ON(atomic_read(&chip->send_message_locked) != 0); + atomic_set(&chip->send_message_locked, 1); + + /* MicoBlaze gogogo */ + lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC); + + /* wait for interrupt to answer */ + for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS; ++dwloop) { + answer_received = atomic_read(&chip->send_message_locked); + if (answer_received == 0) + break; + msleep(1); + } + + if (answer_received == 0) { + /* in Debug mode verify Reg_CSM_MR */ + snd_BUG_ON(!(lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)); + + /* command finished, read status */ + if (rmh->dsp_stat == 0) + reg = lx_dsp_reg_read(chip, eReg_CRM1); + else + reg = 0; + } else { + int i; + snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! " + "Interrupts disabled?\n"); + + /* attente bit Reg_CSM_MR */ + for (i = 0; i != XILINX_POLL_ITERATIONS; i++) { + if ((lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)) { + if (rmh->dsp_stat == 0) + reg = lx_dsp_reg_read(chip, eReg_CRM1); + else + reg = 0; + goto polling_successful; + } + + if (i > XILINX_POLL_NO_SLEEP) + msleep(1); + } + snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! " + "polling failed\n"); + +polling_successful: + atomic_set(&chip->send_message_locked, 0); + } + + if ((reg & ERROR_VALUE) == 0) { + /* read response */ + if (rmh->stat_len) { + snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1)); + + lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, + rmh->stat_len); + } + } else + snd_printk(KERN_WARNING LXP "lx_message_send: error_value %x\n", + reg); + + /* clear Reg_CSM_MR */ + lx_dsp_reg_write(chip, eReg_CSM, 0); + + switch (reg) { + case ED_DSP_TIMED_OUT: + snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); + return -ETIMEDOUT; + + case ED_DSP_CRASHED: + snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); + return -EAGAIN; + } + + lx_message_dump(rmh); + return 0; +} + +static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh) +{ + u32 reg = ED_DSP_TIMED_OUT; + int dwloop; + + if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) { + snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg); + return -EBUSY; + } + + /* write command */ + lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len); + + /* MicoBlaze gogogo */ + lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC); + + /* wait for interrupt to answer */ + for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS * 1000; ++dwloop) { + if (lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR) { + if (rmh->dsp_stat == 0) + reg = lx_dsp_reg_read(chip, eReg_CRM1); + else + reg = 0; + goto polling_successful; + } else + udelay(1); + } + snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send_atomic! " + "polling failed\n"); + +polling_successful: + if ((reg & ERROR_VALUE) == 0) { + /* read response */ + if (rmh->stat_len) { + snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1)); + lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, + rmh->stat_len); + } + } else + snd_printk(LXP "rmh error: %08x\n", reg); + + /* clear Reg_CSM_MR */ + lx_dsp_reg_write(chip, eReg_CSM, 0); + + switch (reg) { + case ED_DSP_TIMED_OUT: + snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); + return -ETIMEDOUT; + + case ED_DSP_CRASHED: + snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); + return -EAGAIN; + } + + lx_message_dump(rmh); + + return reg; +} + + +/* low-level dsp access */ +int __devinit lx_dsp_get_version(struct lx6464es *chip, u32 *rdsp_version) +{ + u16 ret; + unsigned long flags; + + spin_lock_irqsave(&chip->msg_lock, flags); + + lx_message_init(&chip->rmh, CMD_01_GET_SYS_CFG); + ret = lx_message_send_atomic(chip, &chip->rmh); + + *rdsp_version = chip->rmh.stat[1]; + spin_unlock_irqrestore(&chip->msg_lock, flags); + return ret; +} + +int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq) +{ + u16 ret = 0; + unsigned long flags; + u32 freq_raw = 0; + u32 freq = 0; + u32 frequency = 0; + + spin_lock_irqsave(&chip->msg_lock, flags); + + lx_message_init(&chip->rmh, CMD_01_GET_SYS_CFG); + ret = lx_message_send_atomic(chip, &chip->rmh); + + if (ret == 0) { + freq_raw = chip->rmh.stat[0] >> FREQ_FIELD_OFFSET; + freq = freq_raw & XES_FREQ_COUNT8_MASK; + + if ((freq < XES_FREQ_COUNT8_48_MAX) || + (freq > XES_FREQ_COUNT8_44_MIN)) + frequency = 0; /* unknown */ + else if (freq >= XES_FREQ_COUNT8_44_MAX) + frequency = 44100; + else + frequency = 48000; + } + + spin_unlock_irqrestore(&chip->msg_lock, flags); + + *rfreq = frequency * chip->freq_ratio; + + return ret; +} + +int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address) +{ + u32 macmsb, maclsb; + + macmsb = lx_dsp_reg_read(chip, eReg_ADMACESMSB) & 0x00FFFFFF; + maclsb = lx_dsp_reg_read(chip, eReg_ADMACESLSB) & 0x00FFFFFF; + + /* todo: endianess handling */ + mac_address[5] = ((u8 *)(&maclsb))[0]; + mac_address[4] = ((u8 *)(&maclsb))[1]; + mac_address[3] = ((u8 *)(&maclsb))[2]; + mac_address[2] = ((u8 *)(&macmsb))[0]; + mac_address[1] = ((u8 *)(&macmsb))[1]; + mac_address[0] = ((u8 *)(&macmsb))[2]; + + return 0; +} + + +int lx_dsp_set_granularity(struct lx6464es *chip, u32 gran) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&chip->msg_lock, flags); + + lx_message_init(&chip->rmh, CMD_02_SET_GRANULARITY); + chip->rmh.cmd[0] |= gran; + + ret = lx_message_send_atomic(chip, &chip->rmh); + spin_unlock_irqrestore(&chip->msg_lock, flags); + return ret; +} + +int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&chip->msg_lock, flags); + + lx_message_init(&chip->rmh, CMD_04_GET_EVENT); + chip->rmh.stat_len = 9; /* we don't necessarily need the full length */ + + ret = lx_message_send_atomic(chip, &chip->rmh); + + if (!ret) + memcpy(data, chip->rmh.stat, chip->rmh.stat_len * sizeof(u32)); + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return ret; +} + +#define CSES_TIMEOUT 100 /* microseconds */ +#define CSES_CE 0x0001 +#define CSES_BROADCAST 0x0002 +#define CSES_UPDATE_LDSV 0x0004 + +int lx_dsp_es_check_pipeline(struct lx6464es *chip) +{ + int i; + + for (i = 0; i != CSES_TIMEOUT; ++i) { + /* + * le bit CSES_UPDATE_LDSV est à 1 dés que le macprog + * est pret. il re-passe à 0 lorsque le premier read a + * été fait. pour l'instant on retire le test car ce bit + * passe a 1 environ 200 à 400 ms aprés que le registre + * confES à été écrit (kick du xilinx ES). + * + * On ne teste que le bit CE. + * */ + + u32 cses = lx_dsp_reg_read(chip, eReg_CSES); + + if ((cses & CSES_CE) == 0) + return 0; + + udelay(1); + } + + return -ETIMEDOUT; +} + + +#define PIPE_INFO_TO_CMD(capture, pipe) \ + ((u32)((u32)(pipe) | ((capture) ? ID_IS_CAPTURE : 0L)) << ID_OFFSET) + + + +/* low-level pipe handling */ +int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, + int channels) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_06_ALLOCATE_PIPE); + + chip->rmh.cmd[0] |= pipe_cmd; + chip->rmh.cmd[0] |= channels; + + err = lx_message_send_atomic(chip, &chip->rmh); + spin_unlock_irqrestore(&chip->msg_lock, flags); + + if (err != 0) + snd_printk(KERN_ERR "lx6464es: could not allocate pipe\n"); + + return err; +} + +int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_07_RELEASE_PIPE); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + spin_unlock_irqrestore(&chip->msg_lock, flags); + + return err; +} + +int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, + u32 *r_needed, u32 *r_freed, u32 *size_array) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + +#ifdef CONFIG_SND_DEBUG + if (size_array) + memset(size_array, 0, sizeof(u32)*MAX_STREAM_BUFFER); +#endif + + *r_needed = 0; + *r_freed = 0; + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_08_ASK_BUFFERS); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + + if (!err) { + int i; + for (i = 0; i < MAX_STREAM_BUFFER; ++i) { + u32 stat = chip->rmh.stat[i]; + if (stat & (BF_EOB << BUFF_FLAGS_OFFSET)) { + /* finished */ + *r_freed += 1; + if (size_array) + size_array[i] = stat & MASK_DATA_SIZE; + } else if ((stat & (BF_VALID << BUFF_FLAGS_OFFSET)) + == 0) + /* free */ + *r_needed += 1; + } + +#if 0 + snd_printdd(LXP "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", + *r_needed, *r_freed); + for (i = 0; i < MAX_STREAM_BUFFER; ++i) { + for (i = 0; i != chip->rmh.stat_len; ++i) + snd_printdd(" stat[%d]: %x, %x\n", i, + chip->rmh.stat[i], + chip->rmh.stat[i] & MASK_DATA_SIZE); + } +#endif + } + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + + +int lx_pipe_stop(struct lx6464es *chip, u32 pipe, int is_capture) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_09_STOP_PIPE); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +static int lx_pipe_toggle_state(struct lx6464es *chip, u32 pipe, int is_capture) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0B_TOGGLE_PIPE_STATE); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + + +int lx_pipe_start(struct lx6464es *chip, u32 pipe, int is_capture) +{ + int err; + + err = lx_pipe_wait_for_idle(chip, pipe, is_capture); + if (err < 0) + return err; + + err = lx_pipe_toggle_state(chip, pipe, is_capture); + + return err; +} + +int lx_pipe_pause(struct lx6464es *chip, u32 pipe, int is_capture) +{ + int err = 0; + + err = lx_pipe_wait_for_start(chip, pipe, is_capture); + if (err < 0) + return err; + + err = lx_pipe_toggle_state(chip, pipe, is_capture); + + return err; +} + + +int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture, + u64 *rsample_count) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT); + + chip->rmh.cmd[0] |= pipe_cmd; + chip->rmh.stat_len = 2; /* need all words here! */ + + err = lx_message_send_atomic(chip, &chip->rmh); /* don't sleep! */ + + if (err != 0) + snd_printk(KERN_ERR + "lx6464es: could not query pipe's sample count\n"); + else { + *rsample_count = ((u64)(chip->rmh.stat[0] & MASK_SPL_COUNT_HI) + << 24) /* hi part */ + + chip->rmh.stat[1]; /* lo part */ + } + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + + if (err != 0) + snd_printk(KERN_ERR "lx6464es: could not query pipe's state\n"); + else + *rstate = (chip->rmh.stat[0] >> PSTATE_OFFSET) & 0x0F; + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, + int is_capture, u16 state) +{ + int i; + + /* max 2*PCMOnlyGranularity = 2*1024 at 44100 = < 50 ms: + * timeout 50 ms */ + for (i = 0; i != 50; ++i) { + u16 current_state; + int err = lx_pipe_state(chip, pipe, is_capture, ¤t_state); + + if (err < 0) + return err; + + if (current_state == state) + return 0; + + mdelay(1); + } + + return -ETIMEDOUT; +} + +int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture) +{ + return lx_pipe_wait_for_state(chip, pipe, is_capture, PSTATE_RUN); +} + +int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture) +{ + return lx_pipe_wait_for_state(chip, pipe, is_capture, PSTATE_IDLE); +} + +/* low-level stream handling */ +int lx_stream_set_state(struct lx6464es *chip, u32 pipe, + int is_capture, enum stream_state_t state) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_13_SET_STREAM_STATE); + + chip->rmh.cmd[0] |= pipe_cmd; + chip->rmh.cmd[0] |= state; + + err = lx_message_send_atomic(chip, &chip->rmh); + spin_unlock_irqrestore(&chip->msg_lock, flags); + + return err; +} + +int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, + u32 pipe, int is_capture) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + u32 channels = runtime->channels; + + if (runtime->channels != channels) + snd_printk(KERN_ERR LXP "channel count mismatch: %d vs %d", + runtime->channels, channels); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0C_DEF_STREAM); + + chip->rmh.cmd[0] |= pipe_cmd; + + if (runtime->sample_bits == 16) + /* 16 bit format */ + chip->rmh.cmd[0] |= (STREAM_FMT_16b << STREAM_FMT_OFFSET); + + if (snd_pcm_format_little_endian(runtime->format)) + /* little endian/intel format */ + chip->rmh.cmd[0] |= (STREAM_FMT_intel << STREAM_FMT_OFFSET); + + chip->rmh.cmd[0] |= channels-1; + + err = lx_message_send_atomic(chip, &chip->rmh); + spin_unlock_irqrestore(&chip->msg_lock, flags); + + return err; +} + +int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, + int *rstate) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + + *rstate = (chip->rmh.stat[0] & SF_START) ? START_STATE : PAUSE_STATE; + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture, + u64 *r_bytepos) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT); + + chip->rmh.cmd[0] |= pipe_cmd; + + err = lx_message_send_atomic(chip, &chip->rmh); + + *r_bytepos = ((u64) (chip->rmh.stat[0] & MASK_SPL_COUNT_HI) + << 32) /* hi part */ + + chip->rmh.stat[1]; /* lo part */ + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +/* low-level buffer handling */ +int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, + u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi, + u32 *r_buffer_index) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0F_UPDATE_BUFFER); + + chip->rmh.cmd[0] |= pipe_cmd; + chip->rmh.cmd[0] |= BF_NOTIFY_EOB; /* request interrupt notification */ + + /* todo: pause request, circular buffer */ + + chip->rmh.cmd[1] = buffer_size & MASK_DATA_SIZE; + chip->rmh.cmd[2] = buf_address_lo; + + if (buf_address_hi) { + chip->rmh.cmd_len = 4; + chip->rmh.cmd[3] = buf_address_hi; + chip->rmh.cmd[0] |= BF_64BITS_ADR; + } + + err = lx_message_send_atomic(chip, &chip->rmh); + + if (err == 0) { + *r_buffer_index = chip->rmh.stat[0]; + goto done; + } + + if (err == EB_RBUFFERS_TABLE_OVERFLOW) + snd_printk(LXP "lx_buffer_give EB_RBUFFERS_TABLE_OVERFLOW\n"); + + if (err == EB_INVALID_STREAM) + snd_printk(LXP "lx_buffer_give EB_INVALID_STREAM\n"); + + if (err == EB_CMD_REFUSED) + snd_printk(LXP "lx_buffer_give EB_CMD_REFUSED\n"); + + done: + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture, + u32 *r_buffer_size) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER); + + chip->rmh.cmd[0] |= pipe_cmd; + chip->rmh.cmd[0] |= MASK_BUFFER_ID; /* ask for the current buffer: the + * microblaze will seek for it */ + + err = lx_message_send_atomic(chip, &chip->rmh); + + if (err == 0) + *r_buffer_size = chip->rmh.stat[0] & MASK_DATA_SIZE; + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture, + u32 buffer_index) +{ + int err; + unsigned long flags; + + u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER); + + chip->rmh.cmd[0] |= pipe_cmd; + chip->rmh.cmd[0] |= buffer_index; + + err = lx_message_send_atomic(chip, &chip->rmh); + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + + +/* low-level gain/peak handling + * + * \todo: can we unmute capture/playback channels independently? + * + * */ +int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute) +{ + int err; + unsigned long flags; + + /* bit set to 1: channel muted */ + u64 mute_mask = unmute ? 0 : 0xFFFFFFFFFFFFFFFFLLU; + + spin_lock_irqsave(&chip->msg_lock, flags); + lx_message_init(&chip->rmh, CMD_0D_SET_MUTE); + + chip->rmh.cmd[0] |= PIPE_INFO_TO_CMD(is_capture, 0); + + chip->rmh.cmd[1] = (u32)(mute_mask >> (u64)32); /* hi part */ + chip->rmh.cmd[2] = (u32)(mute_mask & (u64)0xFFFFFFFF); /* lo part */ + + snd_printk("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], + chip->rmh.cmd[2]); + + err = lx_message_send_atomic(chip, &chip->rmh); + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +static u32 peak_map[] = { + 0x00000109, /* -90.308dB */ + 0x0000083B, /* -72.247dB */ + 0x000020C4, /* -60.205dB */ + 0x00008273, /* -48.030dB */ + 0x00020756, /* -36.005dB */ + 0x00040C37, /* -30.001dB */ + 0x00081385, /* -24.002dB */ + 0x00101D3F, /* -18.000dB */ + 0x0016C310, /* -15.000dB */ + 0x002026F2, /* -12.001dB */ + 0x002D6A86, /* -9.000dB */ + 0x004026E6, /* -6.004dB */ + 0x005A9DF6, /* -3.000dB */ + 0x0065AC8B, /* -2.000dB */ + 0x00721481, /* -1.000dB */ + 0x007FFFFF, /* FS */ +}; + +int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels, + u32 *r_levels) +{ + int err = 0; + unsigned long flags; + int i; + spin_lock_irqsave(&chip->msg_lock, flags); + + for (i = 0; i < channels; i += 4) { + u32 s0, s1, s2, s3; + + lx_message_init(&chip->rmh, CMD_12_GET_PEAK); + chip->rmh.cmd[0] |= PIPE_INFO_TO_CMD(is_capture, i); + + err = lx_message_send_atomic(chip, &chip->rmh); + + if (err == 0) { + s0 = peak_map[chip->rmh.stat[0] & 0x0F]; + s1 = peak_map[(chip->rmh.stat[0] >> 4) & 0xf]; + s2 = peak_map[(chip->rmh.stat[0] >> 8) & 0xf]; + s3 = peak_map[(chip->rmh.stat[0] >> 12) & 0xf]; + } else + s0 = s1 = s2 = s3 = 0; + + r_levels[0] = s0; + r_levels[1] = s1; + r_levels[2] = s2; + r_levels[3] = s3; + + r_levels += 4; + } + + spin_unlock_irqrestore(&chip->msg_lock, flags); + return err; +} + +/* interrupt handling */ +#define PCX_IRQ_NONE 0 +#define IRQCS_ACTIVE_PCIDB 0x00002000L /* Bit nø 13 */ +#define IRQCS_ENABLE_PCIIRQ 0x00000100L /* Bit nø 08 */ +#define IRQCS_ENABLE_PCIDB 0x00000200L /* Bit nø 09 */ + +static u32 lx_interrupt_test_ack(struct lx6464es *chip) +{ + u32 irqcs = lx_plx_reg_read(chip, ePLX_IRQCS); + + /* Test if PCI Doorbell interrupt is active */ + if (irqcs & IRQCS_ACTIVE_PCIDB) { + u32 temp; + irqcs = PCX_IRQ_NONE; + + while ((temp = lx_plx_reg_read(chip, ePLX_L2PCIDB))) { + /* RAZ interrupt */ + irqcs |= temp; + lx_plx_reg_write(chip, ePLX_L2PCIDB, temp); + } + + return irqcs; + } + return PCX_IRQ_NONE; +} + +static int lx_interrupt_ack(struct lx6464es *chip, u32 *r_irqsrc, + int *r_async_pending, int *r_async_escmd) +{ + u32 irq_async; + u32 irqsrc = lx_interrupt_test_ack(chip); + + if (irqsrc == PCX_IRQ_NONE) + return 0; + + *r_irqsrc = irqsrc; + + irq_async = irqsrc & MASK_SYS_ASYNC_EVENTS; /* + EtherSound response + * (set by xilinx) + EOB */ + + if (irq_async & MASK_SYS_STATUS_ESA) { + irq_async &= ~MASK_SYS_STATUS_ESA; + *r_async_escmd = 1; + } + + if (irqsrc & MASK_SYS_STATUS_CMD_DONE) + /* xilinx command notification */ + atomic_set(&chip->send_message_locked, 0); + + if (irq_async) { + /* snd_printd("interrupt: async event pending\n"); */ + *r_async_pending = 1; + } + + return 1; +} + +static int lx_interrupt_handle_async_events(struct lx6464es *chip, u32 irqsrc, + int *r_freq_changed, + u64 *r_notified_in_pipe_mask, + u64 *r_notified_out_pipe_mask) +{ + int err; + u32 stat[9]; /* answer from CMD_04_GET_EVENT */ + + /* On peut optimiser pour ne pas lire les evenements vides + * les mots de réponse sont dans l'ordre suivant : + * Stat[0] mot de status général + * Stat[1] fin de buffer OUT pF + * Stat[2] fin de buffer OUT pf + * Stat[3] fin de buffer IN pF + * Stat[4] fin de buffer IN pf + * Stat[5] underrun poid fort + * Stat[6] underrun poid faible + * Stat[7] overrun poid fort + * Stat[8] overrun poid faible + * */ + + u64 orun_mask; + u64 urun_mask; +#if 0 + int has_underrun = (irqsrc & MASK_SYS_STATUS_URUN) ? 1 : 0; + int has_overrun = (irqsrc & MASK_SYS_STATUS_ORUN) ? 1 : 0; +#endif + int eb_pending_out = (irqsrc & MASK_SYS_STATUS_EOBO) ? 1 : 0; + int eb_pending_in = (irqsrc & MASK_SYS_STATUS_EOBI) ? 1 : 0; + + *r_freq_changed = (irqsrc & MASK_SYS_STATUS_FREQ) ? 1 : 0; + + err = lx_dsp_read_async_events(chip, stat); + if (err < 0) + return err; + + if (eb_pending_in) { + *r_notified_in_pipe_mask = ((u64)stat[3] << 32) + + stat[4]; + snd_printdd(LXP "interrupt: EOBI pending %llx\n", + *r_notified_in_pipe_mask); + } + if (eb_pending_out) { + *r_notified_out_pipe_mask = ((u64)stat[1] << 32) + + stat[2]; + snd_printdd(LXP "interrupt: EOBO pending %llx\n", + *r_notified_out_pipe_mask); + } + + orun_mask = ((u64)stat[7] << 32) + stat[8]; + urun_mask = ((u64)stat[5] << 32) + stat[6]; + + /* todo: handle xrun notification */ + + return err; +} + +static int lx_interrupt_request_new_buffer(struct lx6464es *chip, + struct lx_stream *lx_stream) +{ + struct snd_pcm_substream *substream = lx_stream->stream; + int is_capture = lx_stream->is_capture; + int err; + unsigned long flags; + + const u32 channels = substream->runtime->channels; + const u32 bytes_per_frame = channels * 3; + const u32 period_size = substream->runtime->period_size; + const u32 period_bytes = period_size * bytes_per_frame; + const u32 pos = lx_stream->frame_pos; + const u32 next_pos = ((pos+1) == substream->runtime->periods) ? + 0 : pos + 1; + + dma_addr_t buf = substream->dma_buffer.addr + pos * period_bytes; + u32 buf_hi = 0; + u32 buf_lo = 0; + u32 buffer_index = 0; + + u32 needed, freed; + u32 size_array[MAX_STREAM_BUFFER]; + + snd_printdd("->lx_interrupt_request_new_buffer\n"); + + spin_lock_irqsave(&chip->lock, flags); + + err = lx_buffer_ask(chip, 0, is_capture, &needed, &freed, size_array); + snd_printdd(LXP "interrupt: needed %d, freed %d\n", needed, freed); + + unpack_pointer(buf, &buf_lo, &buf_hi); + err = lx_buffer_give(chip, 0, is_capture, period_bytes, buf_lo, buf_hi, + &buffer_index); + snd_printdd(LXP "interrupt: gave buffer index %x on %p (%d bytes)\n", + buffer_index, (void *)buf, period_bytes); + + lx_stream->frame_pos = next_pos; + spin_unlock_irqrestore(&chip->lock, flags); + + return err; +} + +void lx_tasklet_playback(unsigned long data) +{ + struct lx6464es *chip = (struct lx6464es *)data; + struct lx_stream *lx_stream = &chip->playback_stream; + int err; + + snd_printdd("->lx_tasklet_playback\n"); + + err = lx_interrupt_request_new_buffer(chip, lx_stream); + if (err < 0) + snd_printk(KERN_ERR LXP + "cannot request new buffer for playback\n"); + + snd_pcm_period_elapsed(lx_stream->stream); +} + +void lx_tasklet_capture(unsigned long data) +{ + struct lx6464es *chip = (struct lx6464es *)data; + struct lx_stream *lx_stream = &chip->capture_stream; + int err; + + snd_printdd("->lx_tasklet_capture\n"); + err = lx_interrupt_request_new_buffer(chip, lx_stream); + if (err < 0) + snd_printk(KERN_ERR LXP + "cannot request new buffer for capture\n"); + + snd_pcm_period_elapsed(lx_stream->stream); +} + + + +static int lx_interrupt_handle_audio_transfer(struct lx6464es *chip, + u64 notified_in_pipe_mask, + u64 notified_out_pipe_mask) +{ + int err = 0; + + if (notified_in_pipe_mask) { + snd_printdd(LXP "requesting audio transfer for capture\n"); + tasklet_hi_schedule(&chip->tasklet_capture); + } + + if (notified_out_pipe_mask) { + snd_printdd(LXP "requesting audio transfer for playback\n"); + tasklet_hi_schedule(&chip->tasklet_playback); + } + + return err; +} + + +irqreturn_t lx_interrupt(int irq, void *dev_id) +{ + struct lx6464es *chip = dev_id; + int async_pending, async_escmd; + u32 irqsrc; + + spin_lock(&chip->lock); + + snd_printdd("**************************************************\n"); + + if (!lx_interrupt_ack(chip, &irqsrc, &async_pending, &async_escmd)) { + spin_unlock(&chip->lock); + snd_printdd("IRQ_NONE\n"); + return IRQ_NONE; /* this device did not cause the interrupt */ + } + + if (irqsrc & MASK_SYS_STATUS_CMD_DONE) + goto exit; + +#if 0 + if (irqsrc & MASK_SYS_STATUS_EOBI) + snd_printdd(LXP "interrupt: EOBI\n"); + + if (irqsrc & MASK_SYS_STATUS_EOBO) + snd_printdd(LXP "interrupt: EOBO\n"); + + if (irqsrc & MASK_SYS_STATUS_URUN) + snd_printdd(LXP "interrupt: URUN\n"); + + if (irqsrc & MASK_SYS_STATUS_ORUN) + snd_printdd(LXP "interrupt: ORUN\n"); +#endif + + if (async_pending) { + u64 notified_in_pipe_mask = 0; + u64 notified_out_pipe_mask = 0; + int freq_changed; + int err; + + /* handle async events */ + err = lx_interrupt_handle_async_events(chip, irqsrc, + &freq_changed, + ¬ified_in_pipe_mask, + ¬ified_out_pipe_mask); + if (err) + snd_printk(KERN_ERR LXP + "error handling async events\n"); + + err = lx_interrupt_handle_audio_transfer(chip, + notified_in_pipe_mask, + notified_out_pipe_mask + ); + if (err) + snd_printk(KERN_ERR LXP + "error during audio transfer\n"); + } + + if (async_escmd) { +#if 0 + /* backdoor for ethersound commands + * + * for now, we do not need this + * + * */ + + snd_printdd("lx6464es: interrupt requests escmd handling\n"); +#endif + } + +exit: + spin_unlock(&chip->lock); + return IRQ_HANDLED; /* this device caused the interrupt */ +} + + +static void lx_irq_set(struct lx6464es *chip, int enable) +{ + u32 reg = lx_plx_reg_read(chip, ePLX_IRQCS); + + /* enable/disable interrupts + * + * Set the Doorbell and PCI interrupt enable bits + * + * */ + if (enable) + reg |= (IRQCS_ENABLE_PCIIRQ | IRQCS_ENABLE_PCIDB); + else + reg &= ~(IRQCS_ENABLE_PCIIRQ | IRQCS_ENABLE_PCIDB); + lx_plx_reg_write(chip, ePLX_IRQCS, reg); +} + +void lx_irq_enable(struct lx6464es *chip) +{ + snd_printdd("->lx_irq_enable\n"); + lx_irq_set(chip, 1); +} + +void lx_irq_disable(struct lx6464es *chip) +{ + snd_printdd("->lx_irq_disable\n"); + lx_irq_set(chip, 0); +} diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h new file mode 100644 index 0000000..6bd9cbb --- /dev/null +++ b/sound/pci/lx6464es/lx_core.h @@ -0,0 +1,242 @@ +/* -*- linux-c -*- * + * + * ALSA driver for the digigram lx6464es interface + * low-level interface + * + * Copyright (c) 2009 Tim Blechmann + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + */ + +#ifndef LX_CORE_H +#define LX_CORE_H + +#include + +#include "lx_defs.h" + +#define REG_CRM_NUMBER 12 + +struct lx6464es; + +/* low-level register access */ + +/* dsp register access */ +enum { + eReg_BASE, + eReg_CSM, + eReg_CRM1, + eReg_CRM2, + eReg_CRM3, + eReg_CRM4, + eReg_CRM5, + eReg_CRM6, + eReg_CRM7, + eReg_CRM8, + eReg_CRM9, + eReg_CRM10, + eReg_CRM11, + eReg_CRM12, + + eReg_ICR, + eReg_CVR, + eReg_ISR, + eReg_RXHTXH, + eReg_RXMTXM, + eReg_RHLTXL, + eReg_RESETDSP, + + eReg_CSUF, + eReg_CSES, + eReg_CRESMSB, + eReg_CRESLSB, + eReg_ADMACESMSB, + eReg_ADMACESLSB, + eReg_CONFES, + + eMaxPortLx +}; + +unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port); +void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len); +void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data); +void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, + u32 len); + +/* plx register access */ +enum { + ePLX_PCICR, + + ePLX_MBOX0, + ePLX_MBOX1, + ePLX_MBOX2, + ePLX_MBOX3, + ePLX_MBOX4, + ePLX_MBOX5, + ePLX_MBOX6, + ePLX_MBOX7, + + ePLX_L2PCIDB, + ePLX_IRQCS, + ePLX_CHIPSC, + + eMaxPort +}; + +unsigned long lx_plx_reg_read(struct lx6464es *chip, int port); +void lx_plx_reg_write(struct lx6464es *chip, int port, u32 data); + +/* rhm */ +struct lx_rmh { + u16 cmd_len; /* length of the command to send (WORDs) */ + u16 stat_len; /* length of the status received (WORDs) */ + u16 dsp_stat; /* status type, RMP_SSIZE_XXX */ + u16 cmd_idx; /* index of the command */ + u32 cmd[REG_CRM_NUMBER]; + u32 stat[REG_CRM_NUMBER]; +}; + + +/* low-level dsp access */ +int __devinit lx_dsp_get_version(struct lx6464es *chip, u32 *rdsp_version); +int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq); +int lx_dsp_set_granularity(struct lx6464es *chip, u32 gran); +int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data); +int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address); + + +/* low-level pipe handling */ +int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, + int channels); +int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture); +int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture, + u64 *rsample_count); +int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate); +int lx_pipe_stop(struct lx6464es *chip, u32 pipe, int is_capture); +int lx_pipe_start(struct lx6464es *chip, u32 pipe, int is_capture); +int lx_pipe_pause(struct lx6464es *chip, u32 pipe, int is_capture); + +int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture); +int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture); + +/* low-level stream handling */ +int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, + u32 pipe, int is_capture); +int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, + int *rstate); +int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture, + u64 *r_bytepos); + +int lx_stream_set_state(struct lx6464es *chip, u32 pipe, + int is_capture, enum stream_state_t state); + +static inline int lx_stream_start(struct lx6464es *chip, u32 pipe, + int is_capture) +{ + snd_printdd("->lx_stream_start\n"); + return lx_stream_set_state(chip, pipe, is_capture, SSTATE_RUN); +} + +static inline int lx_stream_pause(struct lx6464es *chip, u32 pipe, + int is_capture) +{ + snd_printdd("->lx_stream_pause\n"); + return lx_stream_set_state(chip, pipe, is_capture, SSTATE_PAUSE); +} + +static inline int lx_stream_stop(struct lx6464es *chip, u32 pipe, + int is_capture) +{ + snd_printdd("->lx_stream_stop\n"); + return lx_stream_set_state(chip, pipe, is_capture, SSTATE_STOP); +} + +/* low-level buffer handling */ +int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, + u32 *r_needed, u32 *r_freed, u32 *size_array); +int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, + u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi, + u32 *r_buffer_index); +int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture, + u32 *r_buffer_size); +int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture, + u32 buffer_index); + +/* low-level gain/peak handling */ +int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute); +int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels, + u32 *r_levels); + + +/* interrupt handling */ +irqreturn_t lx_interrupt(int irq, void *dev_id); +void lx_irq_enable(struct lx6464es *chip); +void lx_irq_disable(struct lx6464es *chip); + +void lx_tasklet_capture(unsigned long data); +void lx_tasklet_playback(unsigned long data); + + +/* Stream Format Header Defines (for LIN and IEEE754) */ +#define HEADER_FMT_BASE HEADER_FMT_BASE_LIN +#define HEADER_FMT_BASE_LIN 0xFED00000 +#define HEADER_FMT_BASE_FLOAT 0xFAD00000 +#define HEADER_FMT_MONO 0x00000080 /* bit 23 in header_lo. WARNING: old + * bit 22 is ignored in float + * format */ +#define HEADER_FMT_INTEL 0x00008000 +#define HEADER_FMT_16BITS 0x00002000 +#define HEADER_FMT_24BITS 0x00004000 +#define HEADER_FMT_UPTO11 0x00000200 /* frequency is less or equ. to 11k. + * */ +#define HEADER_FMT_UPTO32 0x00000100 /* frequency is over 11k and less + * then 32k.*/ + + +#define BIT_FMP_HEADER 23 +#define BIT_FMP_SD 22 +#define BIT_FMP_MULTICHANNEL 19 + +#define START_STATE 1 +#define PAUSE_STATE 0 + + + + + +/* from PcxAll_e.h */ +/* Start/Pause condition for pipes (PCXStartPipe, PCXPausePipe) */ +#define START_PAUSE_IMMEDIATE 0 +#define START_PAUSE_ON_SYNCHRO 1 +#define START_PAUSE_ON_TIME_CODE 2 + + +/* Pipe / Stream state */ +#define START_STATE 1 +#define PAUSE_STATE 0 + +static inline void unpack_pointer(dma_addr_t ptr, u32 *r_low, u32 *r_high) +{ + *r_low = (u32)(ptr & 0xffffffff); +#if BITS_PER_LONG == 32 + *r_high = 0; +#else + *r_high = (u32)((u64)ptr>>32); +#endif +} + +#endif /* LX_CORE_H */ diff --git a/sound/pci/lx6464es/lx_defs.h b/sound/pci/lx6464es/lx_defs.h new file mode 100644 index 0000000..49d36bd --- /dev/null +++ b/sound/pci/lx6464es/lx_defs.h @@ -0,0 +1,376 @@ +/* -*- linux-c -*- * + * + * ALSA driver for the digigram lx6464es interface + * adapted upstream headers + * + * Copyright (c) 2009 Tim Blechmann + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; see the file COPYING. If not, write to + * the Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * + */ + +#ifndef LX_DEFS_H +#define LX_DEFS_H + +/* code adapted from ethersound.h */ +#define XES_FREQ_COUNT8_MASK 0x00001FFF /* compteur 25MHz entre 8 ech. */ +#define XES_FREQ_COUNT8_44_MIN 0x00001288 /* 25M / + * [ 44k - ( 44.1k + 48k ) / 2 ] + * * 8 */ +#define XES_FREQ_COUNT8_44_MAX 0x000010F0 /* 25M / [ ( 44.1k + 48k ) / 2 ] + * * 8 */ +#define XES_FREQ_COUNT8_48_MAX 0x00000F08 /* 25M / + * [ 48k + ( 44.1k + 48k ) / 2 ] + * * 8 */ + +/* code adapted from LXES_registers.h */ + +#define IOCR_OUTPUTS_OFFSET 0 /* (rw) offset for the number of OUTs in the + * ConfES register. */ +#define IOCR_INPUTS_OFFSET 8 /* (rw) offset for the number of INs in the + * ConfES register. */ +#define FREQ_RATIO_OFFSET 19 /* (rw) offset for frequency ratio in the + * ConfES register. */ +#define FREQ_RATIO_SINGLE_MODE 0x01 /* value for single mode frequency ratio: + * sample rate = frequency rate. */ + +#define CONFES_READ_PART_MASK 0x00070000 +#define CONFES_WRITE_PART_MASK 0x00F80000 + +/* code adapted from if_drv_mb.h */ + +#define MASK_SYS_STATUS_ERROR (1L << 31) /* events that lead to a PCI irq if + * not yet pending */ +#define MASK_SYS_STATUS_URUN (1L << 30) +#define MASK_SYS_STATUS_ORUN (1L << 29) +#define MASK_SYS_STATUS_EOBO (1L << 28) +#define MASK_SYS_STATUS_EOBI (1L << 27) +#define MASK_SYS_STATUS_FREQ (1L << 26) +#define MASK_SYS_STATUS_ESA (1L << 25) /* reserved, this is set by the + * XES */ +#define MASK_SYS_STATUS_TIMER (1L << 24) + +#define MASK_SYS_ASYNC_EVENTS (MASK_SYS_STATUS_ERROR | \ + MASK_SYS_STATUS_URUN | \ + MASK_SYS_STATUS_ORUN | \ + MASK_SYS_STATUS_EOBO | \ + MASK_SYS_STATUS_EOBI | \ + MASK_SYS_STATUS_FREQ | \ + MASK_SYS_STATUS_ESA) + +#define MASK_SYS_PCI_EVENTS (MASK_SYS_ASYNC_EVENTS | \ + MASK_SYS_STATUS_TIMER) + +#define MASK_SYS_TIMER_COUNT 0x0000FFFF + +#define MASK_SYS_STATUS_EOT_PLX (1L << 22) /* event that remains + * internal: reserved fo end + * of plx dma */ +#define MASK_SYS_STATUS_XES (1L << 21) /* event that remains + * internal: pending XES + * IRQ */ +#define MASK_SYS_STATUS_CMD_DONE (1L << 20) /* alternate command + * management: notify driver + * instead of polling */ + + +#define MAX_STREAM_BUFFER 5 /* max amount of stream buffers. */ + +#define MICROBLAZE_IBL_MIN 32 +#define MICROBLAZE_IBL_DEFAULT 128 +#define MICROBLAZE_IBL_MAX 512 +/* #define MASK_GRANULARITY (2*MICROBLAZE_IBL_MAX-1) */ + + + +/* command opcodes, see reference for details */ + +/* + the capture bit position in the object_id field in driver commands + depends upon the number of managed channels. For now, 64 IN + 64 OUT are + supported. HOwever, the communication protocol forsees 1024 channels, hence + bit 10 indicates a capture (input) object). +*/ +#define ID_IS_CAPTURE (1L << 10) +#define ID_OFFSET 13 /* object ID is at the 13th bit in the + * 1st command word.*/ +#define ID_CH_MASK 0x3F +#define OPCODE_OFFSET 24 /* offset of the command opcode in the first + * command word.*/ + +enum cmd_mb_opcodes { + CMD_00_INFO_DEBUG = 0x00, + CMD_01_GET_SYS_CFG = 0x01, + CMD_02_SET_GRANULARITY = 0x02, + CMD_03_SET_TIMER_IRQ = 0x03, + CMD_04_GET_EVENT = 0x04, + CMD_05_GET_PIPES = 0x05, + + CMD_06_ALLOCATE_PIPE = 0x06, + CMD_07_RELEASE_PIPE = 0x07, + CMD_08_ASK_BUFFERS = 0x08, + CMD_09_STOP_PIPE = 0x09, + CMD_0A_GET_PIPE_SPL_COUNT = 0x0a, + CMD_0B_TOGGLE_PIPE_STATE = 0x0b, + + CMD_0C_DEF_STREAM = 0x0c, + CMD_0D_SET_MUTE = 0x0d, + CMD_0E_GET_STREAM_SPL_COUNT = 0x0e, + CMD_0F_UPDATE_BUFFER = 0x0f, + CMD_10_GET_BUFFER = 0x10, + CMD_11_CANCEL_BUFFER = 0x11, + CMD_12_GET_PEAK = 0x12, + CMD_13_SET_STREAM_STATE = 0x13, + CMD_14_INVALID = 0x14, +}; + +/* pipe states */ +enum pipe_state_t { + PSTATE_IDLE = 0, /* the pipe is not processed in the XES_IRQ + * (free or stopped, or paused). */ + PSTATE_RUN = 1, /* sustained play/record state. */ + PSTATE_PURGE = 2, /* the ES channels are now off, render pipes do + * not DMA, record pipe do a last DMA. */ + PSTATE_ACQUIRE = 3, /* the ES channels are now on, render pipes do + * not yet increase their sample count, record + * pipes do not DMA. */ + PSTATE_CLOSING = 4, /* the pipe is releasing, and may not yet + * receive an "alloc" command. */ +}; + +/* stream states */ +enum stream_state_t { + SSTATE_STOP = 0x00, /* setting to stop resets the stream spl + * count.*/ + SSTATE_RUN = (0x01 << 0), /* start DMA and spl count handling. */ + SSTATE_PAUSE = (0x01 << 1), /* pause DMA and spl count handling. */ +}; + +/* buffer flags */ +enum buffer_flags { + BF_VALID = 0x80, /* set if the buffer is valid, clear if free.*/ + BF_CURRENT = 0x40, /* set if this is the current buffer (there is + * always a current buffer).*/ + BF_NOTIFY_EOB = 0x20, /* set if this buffer must cause a PCI event + * when finished.*/ + BF_CIRCULAR = 0x10, /* set if buffer[1] must be copied to buffer[0] + * by the end of this buffer.*/ + BF_64BITS_ADR = 0x08, /* set if the hi part of the address is valid.*/ + BF_xx = 0x04, /* future extension.*/ + BF_EOB = 0x02, /* set if finished, but not yet free.*/ + BF_PAUSE = 0x01, /* pause stream at buffer end.*/ + BF_ZERO = 0x00, /* no flags (init).*/ +}; + +/** +* Stream Flags definitions +*/ +enum stream_flags { + SF_ZERO = 0x00000000, /* no flags (stream invalid). */ + SF_VALID = 0x10000000, /* the stream has a valid DMA_conf + * info (setstreamformat). */ + SF_XRUN = 0x20000000, /* the stream is un x-run state. */ + SF_START = 0x40000000, /* the DMA is running.*/ + SF_ASIO = 0x80000000, /* ASIO.*/ +}; + + +#define MASK_SPL_COUNT_HI 0x00FFFFFF /* 4 MSBits are status bits */ +#define PSTATE_OFFSET 28 /* 4 MSBits are status bits */ + + +#define MASK_STREAM_HAS_MAPPING (1L << 12) +#define MASK_STREAM_IS_ASIO (1L << 9) +#define STREAM_FMT_OFFSET 10 /* the stream fmt bits start at the 10th + * bit in the command word. */ + +#define STREAM_FMT_16b 0x02 +#define STREAM_FMT_intel 0x01 + +#define FREQ_FIELD_OFFSET 15 /* offset of the freq field in the response + * word */ + +#define BUFF_FLAGS_OFFSET 24 /* offset of the buffer flags in the + * response word. */ +#define MASK_DATA_SIZE 0x00FFFFFF /* this must match the field size of + * datasize in the buffer_t structure. */ + +#define MASK_BUFFER_ID 0xFF /* the cancel command awaits a buffer ID, + * may be 0xFF for "current". */ + + +/* code adapted from PcxErr_e.h */ + +/* Bits masks */ + +#define ERROR_MASK 0x8000 + +#define SOURCE_MASK 0x7800 + +#define E_SOURCE_BOARD 0x4000 /* 8 >> 1 */ +#define E_SOURCE_DRV 0x2000 /* 4 >> 1 */ +#define E_SOURCE_API 0x1000 /* 2 >> 1 */ +/* Error tools */ +#define E_SOURCE_TOOLS 0x0800 /* 1 >> 1 */ +/* Error pcxaudio */ +#define E_SOURCE_AUDIO 0x1800 /* 3 >> 1 */ +/* Error virtual pcx */ +#define E_SOURCE_VPCX 0x2800 /* 5 >> 1 */ +/* Error dispatcher */ +#define E_SOURCE_DISPATCHER 0x3000 /* 6 >> 1 */ +/* Error from CobraNet firmware */ +#define E_SOURCE_COBRANET 0x3800 /* 7 >> 1 */ + +#define E_SOURCE_USER 0x7800 + +#define CLASS_MASK 0x0700 + +#define CODE_MASK 0x00FF + +/* Bits values */ + +/* Values for the error/warning bit */ +#define ERROR_VALUE 0x8000 +#define WARNING_VALUE 0x0000 + +/* Class values */ +#define E_CLASS_GENERAL 0x0000 +#define E_CLASS_INVALID_CMD 0x0100 +#define E_CLASS_INVALID_STD_OBJECT 0x0200 +#define E_CLASS_RSRC_IMPOSSIBLE 0x0300 +#define E_CLASS_WRONG_CONTEXT 0x0400 +#define E_CLASS_BAD_SPECIFIC_PARAMETER 0x0500 +#define E_CLASS_REAL_TIME_ERROR 0x0600 +#define E_CLASS_DIRECTSHOW 0x0700 +#define E_CLASS_FREE 0x0700 + + +/* Complete DRV error code for the general class */ +#define ED_GN (ERROR_VALUE | E_SOURCE_DRV | E_CLASS_GENERAL) +#define ED_CONCURRENCY (ED_GN | 0x01) +#define ED_DSP_CRASHED (ED_GN | 0x02) +#define ED_UNKNOWN_BOARD (ED_GN | 0x03) +#define ED_NOT_INSTALLED (ED_GN | 0x04) +#define ED_CANNOT_OPEN_SVC_MANAGER (ED_GN | 0x05) +#define ED_CANNOT_READ_REGISTRY (ED_GN | 0x06) +#define ED_DSP_VERSION_MISMATCH (ED_GN | 0x07) +#define ED_UNAVAILABLE_FEATURE (ED_GN | 0x08) +#define ED_CANCELLED (ED_GN | 0x09) +#define ED_NO_RESPONSE_AT_IRQA (ED_GN | 0x10) +#define ED_INVALID_ADDRESS (ED_GN | 0x11) +#define ED_DSP_CORRUPTED (ED_GN | 0x12) +#define ED_PENDING_OPERATION (ED_GN | 0x13) +#define ED_NET_ALLOCATE_MEMORY_IMPOSSIBLE (ED_GN | 0x14) +#define ED_NET_REGISTER_ERROR (ED_GN | 0x15) +#define ED_NET_THREAD_ERROR (ED_GN | 0x16) +#define ED_NET_OPEN_ERROR (ED_GN | 0x17) +#define ED_NET_CLOSE_ERROR (ED_GN | 0x18) +#define ED_NET_NO_MORE_PACKET (ED_GN | 0x19) +#define ED_NET_NO_MORE_BUFFER (ED_GN | 0x1A) +#define ED_NET_SEND_ERROR (ED_GN | 0x1B) +#define ED_NET_RECEIVE_ERROR (ED_GN | 0x1C) +#define ED_NET_WRONG_MSG_SIZE (ED_GN | 0x1D) +#define ED_NET_WAIT_ERROR (ED_GN | 0x1E) +#define ED_NET_EEPROM_ERROR (ED_GN | 0x1F) +#define ED_INVALID_RS232_COM_NUMBER (ED_GN | 0x20) +#define ED_INVALID_RS232_INIT (ED_GN | 0x21) +#define ED_FILE_ERROR (ED_GN | 0x22) +#define ED_INVALID_GPIO_CMD (ED_GN | 0x23) +#define ED_RS232_ALREADY_OPENED (ED_GN | 0x24) +#define ED_RS232_NOT_OPENED (ED_GN | 0x25) +#define ED_GPIO_ALREADY_OPENED (ED_GN | 0x26) +#define ED_GPIO_NOT_OPENED (ED_GN | 0x27) +#define ED_REGISTRY_ERROR (ED_GN | 0x28) /* <- NCX */ +#define ED_INVALID_SERVICE (ED_GN | 0x29) /* <- NCX */ + +#define ED_READ_FILE_ALREADY_OPENED (ED_GN | 0x2a) /* <- Decalage + * pour RCX + * (old 0x28) + * */ +#define ED_READ_FILE_INVALID_COMMAND (ED_GN | 0x2b) /* ~ */ +#define ED_READ_FILE_INVALID_PARAMETER (ED_GN | 0x2c) /* ~ */ +#define ED_READ_FILE_ALREADY_CLOSED (ED_GN | 0x2d) /* ~ */ +#define ED_READ_FILE_NO_INFORMATION (ED_GN | 0x2e) /* ~ */ +#define ED_READ_FILE_INVALID_HANDLE (ED_GN | 0x2f) /* ~ */ +#define ED_READ_FILE_END_OF_FILE (ED_GN | 0x30) /* ~ */ +#define ED_READ_FILE_ERROR (ED_GN | 0x31) /* ~ */ + +#define ED_DSP_CRASHED_EXC_DSPSTACK_OVERFLOW (ED_GN | 0x32) /* <- Decalage pour + * PCX (old 0x14) */ +#define ED_DSP_CRASHED_EXC_SYSSTACK_OVERFLOW (ED_GN | 0x33) /* ~ */ +#define ED_DSP_CRASHED_EXC_ILLEGAL (ED_GN | 0x34) /* ~ */ +#define ED_DSP_CRASHED_EXC_TIMER_REENTRY (ED_GN | 0x35) /* ~ */ +#define ED_DSP_CRASHED_EXC_FATAL_ERROR (ED_GN | 0x36) /* ~ */ + +#define ED_FLASH_PCCARD_NOT_PRESENT (ED_GN | 0x37) + +#define ED_NO_CURRENT_CLOCK (ED_GN | 0x38) + +/* Complete DRV error code for real time class */ +#define ED_RT (ERROR_VALUE | E_SOURCE_DRV | E_CLASS_REAL_TIME_ERROR) +#define ED_DSP_TIMED_OUT (ED_RT | 0x01) +#define ED_DSP_CHK_TIMED_OUT (ED_RT | 0x02) +#define ED_STREAM_OVERRUN (ED_RT | 0x03) +#define ED_DSP_BUSY (ED_RT | 0x04) +#define ED_DSP_SEMAPHORE_TIME_OUT (ED_RT | 0x05) +#define ED_BOARD_TIME_OUT (ED_RT | 0x06) +#define ED_XILINX_ERROR (ED_RT | 0x07) +#define ED_COBRANET_ITF_NOT_RESPONDING (ED_RT | 0x08) + +/* Complete BOARD error code for the invaid standard object class */ +#define EB_ISO (ERROR_VALUE | E_SOURCE_BOARD | \ + E_CLASS_INVALID_STD_OBJECT) +#define EB_INVALID_EFFECT (EB_ISO | 0x00) +#define EB_INVALID_PIPE (EB_ISO | 0x40) +#define EB_INVALID_STREAM (EB_ISO | 0x80) +#define EB_INVALID_AUDIO (EB_ISO | 0xC0) + +/* Complete BOARD error code for impossible resource allocation class */ +#define EB_RI (ERROR_VALUE | E_SOURCE_BOARD | E_CLASS_RSRC_IMPOSSIBLE) +#define EB_ALLOCATE_ALL_STREAM_TRANSFERT_BUFFERS_IMPOSSIBLE (EB_RI | 0x01) +#define EB_ALLOCATE_PIPE_SAMPLE_BUFFER_IMPOSSIBLE (EB_RI | 0x02) + +#define EB_ALLOCATE_MEM_STREAM_IMPOSSIBLE \ + EB_ALLOCATE_ALL_STREAM_TRANSFERT_BUFFERS_IMPOSSIBLE +#define EB_ALLOCATE_MEM_PIPE_IMPOSSIBLE \ + EB_ALLOCATE_PIPE_SAMPLE_BUFFER_IMPOSSIBLE + +#define EB_ALLOCATE_DIFFERED_CMD_IMPOSSIBLE (EB_RI | 0x03) +#define EB_TOO_MANY_DIFFERED_CMD (EB_RI | 0x04) +#define EB_RBUFFERS_TABLE_OVERFLOW (EB_RI | 0x05) +#define EB_ALLOCATE_EFFECTS_IMPOSSIBLE (EB_RI | 0x08) +#define EB_ALLOCATE_EFFECT_POS_IMPOSSIBLE (EB_RI | 0x09) +#define EB_RBUFFER_NOT_AVAILABLE (EB_RI | 0x0A) +#define EB_ALLOCATE_CONTEXT_LIII_IMPOSSIBLE (EB_RI | 0x0B) +#define EB_STATUS_DIALOG_IMPOSSIBLE (EB_RI | 0x1D) +#define EB_CONTROL_CMD_IMPOSSIBLE (EB_RI | 0x1E) +#define EB_STATUS_SEND_IMPOSSIBLE (EB_RI | 0x1F) +#define EB_ALLOCATE_PIPE_IMPOSSIBLE (EB_RI | 0x40) +#define EB_ALLOCATE_STREAM_IMPOSSIBLE (EB_RI | 0x80) +#define EB_ALLOCATE_AUDIO_IMPOSSIBLE (EB_RI | 0xC0) + +/* Complete BOARD error code for wrong call context class */ +#define EB_WCC (ERROR_VALUE | E_SOURCE_BOARD | E_CLASS_WRONG_CONTEXT) +#define EB_CMD_REFUSED (EB_WCC | 0x00) +#define EB_START_STREAM_REFUSED (EB_WCC | 0xFC) +#define EB_SPC_REFUSED (EB_WCC | 0xFD) +#define EB_CSN_REFUSED (EB_WCC | 0xFE) +#define EB_CSE_REFUSED (EB_WCC | 0xFF) + + + + +#endif /* LX_DEFS_H */ -- cgit v1.1 From 7852fd08fdc78bf43150137bdbfdfdccdefffe0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Apr 2009 12:25:52 +0200 Subject: ALSA: lx6464es - Use snd_card_create() Use snd_card_create() instead of the obsoleted snd_card_new(). Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 7bc8b8c..870bfc5 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1091,9 +1091,9 @@ static int __devinit snd_lx6464es_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_lx6464es_create(card, pci, &chip); if (err < 0) { -- cgit v1.1 From d7dee4d7744d039bf28e4f6d4f5674f44820265a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Apr 2009 12:27:12 +0200 Subject: ALSA: lx6464es - Disable lx_message_send() Disable lx_message_send() function temporarily as it's not used anywhere. Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx_core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index a9f8f88..5812780 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -314,6 +314,7 @@ static inline void lx_message_dump(struct lx_rmh *rmh) #define XILINX_POLL_NO_SLEEP 100 #define XILINX_POLL_ITERATIONS 150 +#if 0 /* not used now */ static int lx_message_send(struct lx6464es *chip, struct lx_rmh *rmh) { u32 reg = ED_DSP_TIMED_OUT; @@ -404,6 +405,7 @@ polling_successful: lx_message_dump(rmh); return 0; } +#endif /* not used now */ static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh) { -- cgit v1.1 From 8338c300642f67af5a8cc279ca5e088c7055b7eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Apr 2009 12:30:36 +0200 Subject: ALSA: Add missing description of lx6464es to ALSA-Configuration.txt Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..dfd266e 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1093,6 +1093,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards. The driver requires the firmware loader support on kernel. + Module snd-lx6464es + ------------------- + + Module for Digigram LX6464ES boards + + This module supports multiple cards. + Module snd-maestro3 ------------------- -- cgit v1.1 From c282866101bfde888a44da3babd2f9ab265ca6f9 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 4 Apr 2009 14:48:32 +0200 Subject: ALSA: sc6000: add support for SC-6600 and SC-7000 Add support for later cards based on CompuMedia ASC-9408 chipsets. These cards were produced by Gallant. This patch make the OSS aedsp16 driver redundant. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 7 ++- sound/isa/sc6000.c | 127 +++++++++++++++++++++++++++++++++++++++++------------ 2 files changed, 104 insertions(+), 30 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index c5c9a92..64129bf 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -177,15 +177,18 @@ config SND_ES18XX will be called snd-es18xx. config SND_SC6000 - tristate "Gallant SC-6000, Audio Excel DSP 16" + tristate "Gallant SC-6000/6600/7000 and Audio Excel DSP 16" depends on HAS_IOPORT select SND_WSS_LIB select SND_OPL3_LIB select SND_MPU401_UART help - Say Y here to include support for Gallant SC-6000 card and clones: + Say Y here to include support for Gallant SC-6000, SC-6600, SC-7000 + cards and clones: Audio Excel DSP 16 and Zoltrix AV302. + These cards are based on CompuMedia ASC-9308 or ASC-9408 chips. + To compile this driver as a module, choose M here: the module will be called snd-sc6000. diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 7820106..983ab7e 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -2,6 +2,8 @@ * Driver for Gallant SC-6000 soundcard. This card is also known as * Audio Excel DSP 16 or Zoltrix AV302. * These cards use CompuMedia ASC-9308 chip + AD1848 codec. + * SC-6600 and SC-7000 cards are also supported. They are based on + * CompuMedia ASC-9408 chip and CS4231 codec. * * Copyright (C) 2007 Krzysztof Helt * @@ -191,7 +193,7 @@ static __devinit unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) return val; } -static __devinit int sc6000_wait_data(char __iomem *vport) +static int sc6000_wait_data(char __iomem *vport) { int loop = 1000; unsigned char val = 0; @@ -206,7 +208,7 @@ static __devinit int sc6000_wait_data(char __iomem *vport) return -EAGAIN; } -static __devinit int sc6000_read(char __iomem *vport) +static int sc6000_read(char __iomem *vport) { if (sc6000_wait_data(vport)) return -EBUSY; @@ -215,7 +217,7 @@ static __devinit int sc6000_read(char __iomem *vport) } -static __devinit int sc6000_write(char __iomem *vport, int cmd) +static int sc6000_write(char __iomem *vport, int cmd) { unsigned char val; int loop = 500000; @@ -276,8 +278,33 @@ static int __devinit sc6000_dsp_reset(char __iomem *vport) } /* detection and initialization */ -static int __devinit sc6000_cfg_write(char __iomem *vport, - unsigned char softcfg) +static int __devinit sc6000_hw_cfg_write(char __iomem *vport, const int *cfg) +{ + if (sc6000_write(vport, COMMAND_6C) < 0) { + snd_printk(KERN_WARNING "CMD 0x%x: failed!\n", COMMAND_6C); + return -EIO; + } + if (sc6000_write(vport, COMMAND_5C) < 0) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_5C); + return -EIO; + } + if (sc6000_write(vport, cfg[0]) < 0) { + snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[0]); + return -EIO; + } + if (sc6000_write(vport, cfg[1]) < 0) { + snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[1]); + return -EIO; + } + if (sc6000_write(vport, COMMAND_C5) < 0) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_C5); + return -EIO; + } + + return 0; +} + +static int sc6000_cfg_write(char __iomem *vport, unsigned char softcfg) { if (sc6000_write(vport, WRITE_MDIRQ_CFG)) { @@ -291,7 +318,7 @@ static int __devinit sc6000_cfg_write(char __iomem *vport, return 0; } -static int __devinit sc6000_setup_board(char __iomem *vport, int config) +static int sc6000_setup_board(char __iomem *vport, int config) { int loop = 10; @@ -334,16 +361,38 @@ static int __devinit sc6000_init_mss(char __iomem *vport, int config, return 0; } -static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, - char __iomem *vmss_port, int mpu_irq) +static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, + long xport, long xmpu, + long xmss_port) +{ + cfg[0] = 0; + cfg[1] = 0; + if (xport == 0x240) + cfg[0] |= 1; + if (xmpu != SNDRV_AUTO_PORT) { + cfg[0] |= (xmpu & 0x30) >> 2; + cfg[1] |= 0x20; + } + if (xmss_port == 0xe80) + cfg[0] |= 0x10; + cfg[0] |= 0x40; /* always set */ + cfg[1] |= 0x80; /* enable WSS system */ + cfg[1] &= ~0x40; /* disable IDE */ + snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); +} + +static int __devinit sc6000_init_board(char __iomem *vport, + char __iomem *vmss_port, int dev) { char answer[15]; char version[2]; - int mss_config = sc6000_irq_to_softcfg(irq) | - sc6000_dma_to_softcfg(dma); + int mss_config = sc6000_irq_to_softcfg(irq[dev]) | + sc6000_dma_to_softcfg(dma[dev]); int config = mss_config | - sc6000_mpu_irq_to_softcfg(mpu_irq); + sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); int err; + int cfg[2]; + int old = 0; err = sc6000_dsp_reset(vport); if (err < 0) { @@ -360,7 +409,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, /* * My SC-6000 card return "SC-6000" in DSPCopyright, so * if we have something different, we have to be warned. - * Mine returns "SC-6000A " - KH */ if (strncmp("SC-6000", answer, 7)) snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); @@ -372,13 +420,29 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n", answer, version[0], version[1]); - /* - * 0x0A == (IRQ 7, DMA 1, MIRQ 0) - */ - err = sc6000_cfg_write(vport, 0x0a); + /* set configuration */ + sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], + mss_port[dev]); + if (sc6000_hw_cfg_write(vport, cfg) < 0) { + snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); + return -EIO; + } + err = sc6000_setup_board(vport, config); if (err < 0) { - snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); - return -EFAULT; + snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + sc6000_dsp_reset(vport); + sc6000_write(vport, COMMAND_5C); + if (sc6000_read(vport) < 0) + old = 1; + sc6000_dsp_reset(vport); + + if (!old) { + sc6000_write(vport, COMMAND_60); + sc6000_write(vport, 0x02); + sc6000_dsp_reset(vport); } err = sc6000_setup_board(vport, config); @@ -386,10 +450,9 @@ static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma, snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); return -ENODEV; } - err = sc6000_init_mss(vport, config, vmss_port, mss_config); if (err < 0) { - snd_printk(KERN_ERR "Can not initialize " + snd_printk(KERN_ERR "Cannot initialize " "Microsoft Sound System mode.\n"); return -ENODEV; } @@ -485,14 +548,16 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) struct snd_card *card; struct snd_wss *chip; struct snd_opl3 *opl3; - char __iomem *vport; + char __iomem **vport; char __iomem *vmss_port; - err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(vport), + &card); if (err < 0) return err; + vport = card->private_data; if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); if (xirq < 0) { @@ -517,8 +582,8 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) err = -EBUSY; goto err_exit; } - vport = devm_ioport_map(devptr, port[dev], 0x10); - if (!vport) { + *vport = devm_ioport_map(devptr, port[dev], 0x10); + if (*vport == NULL) { snd_printk(KERN_ERR PFX "I/O port cannot be iomaped.\n"); err = -EBUSY; @@ -533,7 +598,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) goto err_unmap1; } vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); - if (!vport) { + if (!vmss_port) { snd_printk(KERN_ERR PFX "MSS port I/O cannot be iomaped.\n"); err = -EBUSY; @@ -544,7 +609,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) port[dev], xirq, xdma, mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]); - err = sc6000_init_board(vport, xirq, xdma, vmss_port, mpu_irq[dev]); + err = sc6000_init_board(*vport, vmss_port, dev); if (err < 0) goto err_unmap2; @@ -552,7 +617,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) WSS_HW_DETECT, 0, &chip); if (err < 0) goto err_unmap2; - card->private_data = chip; err = snd_wss_pcm(chip, 0, NULL); if (err < 0) { @@ -608,6 +672,7 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) return 0; err_unmap2: + sc6000_setup_board(*vport, 0); release_region(mss_port[dev], 4); err_unmap1: release_region(port[dev], 0x10); @@ -618,11 +683,17 @@ err_exit: static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) { + struct snd_card *card = dev_get_drvdata(devptr); + char __iomem **vport = card->private_data; + + if (sc6000_setup_board(*vport, 0) < 0) + snd_printk(KERN_WARNING "sc6000_setup_board failed on exit!\n"); + release_region(port[dev], 0x10); release_region(mss_port[dev], 4); - snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); + snd_card_free(card); return 0; } -- cgit v1.1 From 4e01f54bfd3f423db8fd6c91c4f0471f18aa0c50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Apr 2009 08:53:34 +0200 Subject: ALSA: hda - Add Creative CA0110-IBG support Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode. In the HD-audio mode, no multiple streams are supported by just it behaves like a normal HD-audio device. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 13 + sound/pci/hda/Makefile | 4 + sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_intel.c | 5 + sound/pci/hda/patch_ca0110.c | 574 +++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 597 insertions(+) create mode 100644 sound/pci/hda/patch_ca0110.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index eb2a19b..c710150 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -139,6 +139,19 @@ config SND_HDA_CODEC_CONEXANT snd-hda-codec-conexant. This module is automatically loaded at probing. +config SND_HDA_CODEC_CA0110 + bool "Build Creative CA0110-IBG codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Creative CA0110-IBG codec support in + snd-hda-intel driver, found on some Creative X-Fi cards. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-ca0110. + This module is automatically loaded at probing. + config SND_HDA_CODEC_CMEDIA bool "Build C-Media HD-audio codec support" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 50f9d09..e3081d4 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -13,6 +13,7 @@ snd-hda-codec-analog-objs := patch_analog.o snd-hda-codec-idt-objs := patch_sigmatel.o snd-hda-codec-si3054-objs := patch_si3054.o snd-hda-codec-atihdmi-objs := patch_atihdmi.o +snd-hda-codec-ca0110-objs := patch_ca0110.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o @@ -40,6 +41,9 @@ endif ifdef CONFIG_SND_HDA_CODEC_ATIHDMI obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o endif +ifdef CONFIG_SND_HDA_CODEC_CA0110 +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o +endif ifdef CONFIG_SND_HDA_CODEC_CONEXANT obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fd6e6f3..37f24ce 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -48,6 +48,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, { 0x11c1, "LSI" }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21e99cf..21a3092 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2513,6 +2513,11 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, + /* Creative X-Fi (CA0110-IBG) */ + { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GENERIC }, /* AMD Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c new file mode 100644 index 0000000..7ec41da --- /dev/null +++ b/sound/pci/hda/patch_ca0110.c @@ -0,0 +1,574 @@ +/* + * HD audio interface patch for Creative X-Fi CA0110-IBG chip + * + * Copyright (c) 2008 Takashi Iwai + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + +/* + */ + +struct ca0110_spec { + struct auto_pin_cfg autocfg; + struct hda_multi_out multiout; + hda_nid_t out_pins[AUTO_CFG_MAX_OUTS]; + hda_nid_t dacs[AUTO_CFG_MAX_OUTS]; + hda_nid_t hp_dac; + hda_nid_t input_pins[AUTO_PIN_LAST]; + hda_nid_t adcs[AUTO_PIN_LAST]; + hda_nid_t dig_out; + hda_nid_t dig_in; + unsigned int num_inputs; + const char *input_labels[AUTO_PIN_LAST]; + struct hda_pcm pcm_rec[2]; /* PCM information */ +}; + +/* + * PCM callbacks + */ +static int ca0110_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); +} + +static int ca0110_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + +static int ca0110_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); +} + +/* + * Digital out + */ +static int ca0110_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int ca0110_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int ca0110_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +/* + * Analog capture + */ +static int ca0110_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, spec->adcs[substream->number], + stream_tag, 0, format); + return 0; +} + +static int ca0110_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0110_spec *spec = codec->spec; + + snd_hda_codec_cleanup_stream(codec, spec->adcs[substream->number]); + return 0; +} + +/* + */ + +static char *dirstr[2] = { "Playback", "Capture" }; + +static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); +} + +static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, + int chan, int dir) +{ + char namestr[44]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); + sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); +} + +#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0) +#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0) +#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1) +#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1) +#define add_mono_switch(codec, nid, pfx, chan) \ + _add_switch(codec, nid, pfx, chan, 0) +#define add_mono_volume(codec, nid, pfx, chan) \ + _add_volume(codec, nid, pfx, chan, 0) + +static int ca0110_build_controls(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + static char *prefix[AUTO_CFG_MAX_OUTS] = { + "Front", "Surround", NULL, "Side", "Multi" + }; + hda_nid_t mutenid; + int i, err; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (get_wcaps(codec, spec->out_pins[i]) & AC_WCAP_OUT_AMP) + mutenid = spec->out_pins[i]; + else + mutenid = spec->multiout.dac_nids[i]; + if (!prefix[i]) { + err = add_mono_switch(codec, mutenid, + "Center", 1); + if (err < 0) + return err; + err = add_mono_switch(codec, mutenid, + "LFE", 1); + if (err < 0) + return err; + err = add_mono_volume(codec, spec->multiout.dac_nids[i], + "Center", 1); + if (err < 0) + return err; + err = add_mono_volume(codec, spec->multiout.dac_nids[i], + "LFE", 1); + if (err < 0) + return err; + } else { + err = add_out_switch(codec, mutenid, + prefix[i]); + if (err < 0) + return err; + err = add_out_volume(codec, spec->multiout.dac_nids[i], + prefix[i]); + if (err < 0) + return err; + } + } + if (cfg->hp_outs) { + if (get_wcaps(codec, cfg->hp_pins[0]) & AC_WCAP_OUT_AMP) + mutenid = cfg->hp_pins[0]; + else + mutenid = spec->multiout.dac_nids[i]; + + err = add_out_switch(codec, mutenid, "Headphone"); + if (err < 0) + return err; + if (spec->hp_dac) { + err = add_out_volume(codec, spec->hp_dac, "Headphone"); + if (err < 0) + return err; + } + } + for (i = 0; i < spec->num_inputs; i++) { + const char *label = spec->input_labels[i]; + if (get_wcaps(codec, spec->input_pins[i]) & AC_WCAP_IN_AMP) + mutenid = spec->input_pins[i]; + else + mutenid = spec->adcs[i]; + err = add_in_switch(codec, mutenid, label); + if (err < 0) + return err; + err = add_in_volume(codec, spec->adcs[i], label); + if (err < 0) + return err; + } + + if (spec->dig_out) { + err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out); + if (err < 0) + return err; + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; + } + if (spec->dig_in) { + err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in); + if (err < 0) + return err; + err = add_in_volume(codec, spec->dig_in, "IEC958"); + } + return 0; +} + +/* + */ +static struct hda_pcm_stream ca0110_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .ops = { + .open = ca0110_playback_pcm_open, + .prepare = ca0110_playback_pcm_prepare, + .cleanup = ca0110_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0110_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = ca0110_capture_pcm_prepare, + .cleanup = ca0110_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream ca0110_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .open = ca0110_dig_playback_pcm_open, + .close = ca0110_dig_playback_pcm_close, + .prepare = ca0110_dig_playback_pcm_prepare + }, +}; + +static struct hda_pcm_stream ca0110_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +static int ca0110_build_pcms(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->pcm_info = info; + codec->num_pcms = 0; + + info->name = "CA0110 Analog"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.num_dacs * 2; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; + codec->num_pcms++; + + if (!spec->dig_out && !spec->dig_in) + return 0; + + info++; + info->name = "CA0110 Digital"; + info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + ca0110_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out; + } + if (spec->dig_in) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0110_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; + } + codec->num_pcms++; + + return 0; +} + +static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + if (dac) + snd_hda_codec_write(codec, dac, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO); +} + +static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc) +{ + if (pin) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } + if (adc) + snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); +} + +static int ca0110_init(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) + init_output(codec, spec->out_pins[i], + spec->multiout.dac_nids[i]); + init_output(codec, cfg->hp_pins[0], spec->hp_dac); + init_output(codec, cfg->dig_out_pins[0], spec->dig_out); + + for (i = 0; i < spec->num_inputs; i++) + init_input(codec, spec->input_pins[i], spec->adcs[i]); + init_input(codec, cfg->dig_in_pin, spec->dig_in); + return 0; +} + +static void ca0110_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +static struct hda_codec_ops ca0110_patch_ops = { + .build_controls = ca0110_build_controls, + .build_pcms = ca0110_build_pcms, + .init = ca0110_init, + .free = ca0110_free, +}; + + +static void parse_line_outs(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, n; + unsigned int def_conf; + hda_nid_t nid; + + n = 0; + for (i = 0; i < cfg->line_outs; i++) { + nid = cfg->line_out_pins[i]; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (!def_conf) + continue; /* invalid pin */ + if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1) + continue; + spec->out_pins[n++] = nid; + } + spec->multiout.dac_nids = spec->dacs; + spec->multiout.num_dacs = n; +} + +static void parse_hp_out(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + unsigned int def_conf; + hda_nid_t nid, dac; + + if (!cfg->hp_outs) + return; + nid = cfg->hp_pins[0]; + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + if (!def_conf) { + cfg->hp_outs = 0; + return; + } + if (snd_hda_get_connections(codec, nid, &dac, 1) != 1) + return; + + for (i = 0; i < cfg->line_outs; i++) + if (dac == spec->dacs[i]) + break; + if (i >= cfg->line_outs) { + spec->hp_dac = dac; + spec->multiout.hp_nid = dac; + } +} + +static void parse_input(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid, pin; + int n, i, j; + + n = 0; + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int type = (wcaps & AC_WCAP_TYPE) >> + AC_WCAP_TYPE_SHIFT; + if (type != AC_WID_AUD_IN) + continue; + if (snd_hda_get_connections(codec, nid, &pin, 1) != 1) + continue; + if (pin == cfg->dig_in_pin) { + spec->dig_in = nid; + continue; + } + for (j = 0; j < AUTO_PIN_LAST; j++) + if (cfg->input_pins[j] == pin) + break; + if (j >= AUTO_PIN_LAST) + continue; + spec->input_pins[n] = pin; + spec->input_labels[n] = auto_pin_cfg_labels[j]; + spec->adcs[n] = nid; + n++; + } + spec->num_inputs = n; +} + +static void parse_digital(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (cfg->dig_outs && + snd_hda_get_connections(codec, cfg->dig_out_pins[0], + &spec->dig_out, 1) == 1) + spec->multiout.dig_out_nid = cfg->dig_out_pins[0]; +} + +static int ca0110_parse_auto_config(struct hda_codec *codec) +{ + struct ca0110_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + parse_line_outs(codec); + parse_hp_out(codec); + parse_digital(codec); + parse_input(codec); + return 0; +} + + +int patch_ca0110(struct hda_codec *codec) +{ + struct ca0110_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + codec->bus->needs_damn_long_delay = 1; + + err = ca0110_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = ca0110_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + + +/* + * patch entries + */ +static struct hda_codec_preset snd_hda_preset_ca0110[] = { + { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 }, + { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 }, + { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 }, + {} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:1102000a"); +MODULE_ALIAS("snd-hda-codec-id:1102000b"); +MODULE_ALIAS("snd-hda-codec-id:1102000d"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec"); + +static struct hda_codec_preset_list ca0110_list = { + .preset = snd_hda_preset_ca0110, + .owner = THIS_MODULE, +}; + +static int __init patch_ca0110_init(void) +{ + return snd_hda_add_codec_preset(&ca0110_list); +} + +static void __exit patch_ca0110_exit(void) +{ + snd_hda_delete_codec_preset(&ca0110_list); +} + +module_init(patch_ca0110_init) +module_exit(patch_ca0110_exit) -- cgit v1.1 From 18cb7109d3e83195b605ff2905981020e86f72ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Apr 2009 10:22:24 +0200 Subject: ALSA: hda - Check strcpy length Check the length to copy via strlen() beforehand to avoid the stack corruption, or use strlcpy() to be safe in HD-audio codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/hda_intel.c | 10 ++++++---- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 37f24ce..48f0cea 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1431,6 +1431,8 @@ _snd_hda_find_mixer_ctl(struct hda_codec *codec, memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; id.index = idx; + if (snd_BUG_ON(strlen(name) >= sizeof(id.name))) + return NULL; strcpy(id.name, name); return snd_ctl_find_id(codec->bus->card, &id); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21a3092..41db5d4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1830,7 +1830,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, &pcm); if (err < 0) return err; - strcpy(pcm->name, cpcm->name); + strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); if (apcm == NULL) return -ENOMEM; @@ -2358,9 +2358,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } strcpy(card->driver, "HDA-Intel"); - strcpy(card->shortname, driver_short_names[chip->driver_type]); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->addr, chip->irq); + strlcpy(card->shortname, driver_short_names[chip->driver_type], + sizeof(card->shortname)); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); *rchip = chip; return 0; -- cgit v1.1 From 3f1a4d826751d9759fc95da4e47d08d2745e0055 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 15 Apr 2009 21:35:26 +0100 Subject: ASoC: Check we have DAI ops when calling via accessor functions Also make sure we're checking for the right operation while we're here. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dd28009..9250392 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2100,7 +2100,7 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops->set_sysclk) + if (dai->ops && dai->ops->set_sysclk) return dai->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; @@ -2120,7 +2120,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops->set_clkdiv) + if (dai->ops && dai->ops->set_clkdiv) return dai->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; @@ -2139,7 +2139,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops->set_pll) + if (dai->ops && dai->ops->set_pll) return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; @@ -2155,7 +2155,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops->set_fmt) + if (dai->ops && dai->ops->set_fmt) return dai->ops->set_fmt(dai, fmt); else return -EINVAL; @@ -2174,7 +2174,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->ops->set_sysclk) + if (dai->ops && dai->ops->set_tdm_slot) return dai->ops->set_tdm_slot(dai, mask, slots); else return -EINVAL; @@ -2190,7 +2190,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops->set_sysclk) + if (dai->ops && dai->ops->set_tristate) return dai->ops->set_tristate(dai, tristate); else return -EINVAL; @@ -2206,7 +2206,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops->digital_mute) + if (dai->ops && dai->ops->digital_mute) return dai->ops->digital_mute(dai, mute); else return -EINVAL; -- cgit v1.1 From fd5dfad9cf51bc3575b5e50193403de4a3de23bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 15 Apr 2009 21:37:46 +0100 Subject: ASoC: Volume controls are never of boolean type Some limited volume controls (mostly simple attenuations) have only two settings so the ASoC info functions misreport them as booleans. Since we currently have no better information check for " Volume" in the control name and always report any controls matching as being integer. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9250392..af11791 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1779,7 +1779,7 @@ int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, { int max = kcontrol->private_value; - if (max == 1) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1809,7 +1809,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; - if (max == 1) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1916,7 +1916,7 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - if (max == 1) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- cgit v1.1 From 0d960e8891459f5af85e5781bce3f1da5f7db0fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Apr 2009 10:08:39 +0100 Subject: ASoC: Request shared rates for WM8903 It has a shared LRCLK. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 8cf571f..c539184 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1523,6 +1523,7 @@ struct snd_soc_dai wm8903_dai = { .formats = WM8903_FORMATS, }, .ops = &wm8903_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(wm8903_dai); -- cgit v1.1 From c29b206ffd0700acb2dc1fdb70856cc4b907216c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 15 Apr 2009 15:38:55 +0300 Subject: ASoC: OMAP: Use single-phase for DSP mode Use single-phase mode for the DSP mode and keep the dual phase mode for the I2S mode. The mono (1 channel) mode already used single phase mode, now it is more cleaner. There is no need to configure the second phase, when the single phase is used. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 9c09b94..402a1eb 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -214,8 +214,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; - int wlen, channels; + int wlen, channels, wpf; unsigned long port; + unsigned int format; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -243,18 +244,23 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } - channels = params_channels(params); + format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + wpf = channels = params_channels(params); switch (channels) { case 2: - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; + if (format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); + } case 1: - /* Set 1 word per (McBSP) frame */ - regs->rcr2 |= RFRLEN2(1 - 1); - regs->rcr1 |= RFRLEN1(1 - 1); - regs->xcr2 |= XFRLEN2(1 - 1); - regs->xcr1 |= XFRLEN1(1 - 1); + /* Set word per (McBSP) frame for phase1 */ + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); break; default: /* Unsupported number of channels */ @@ -276,9 +282,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } /* Set FS period and length in terms of bit clock periods */ - switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr2 |= FPER(wlen * channels - 1); regs->srgr1 |= FWID(wlen - 1); break; case SND_SOC_DAIFMT_DSP_B: -- cgit v1.1 From 3ba191ce051a32b20757f063120496e860ea8f9d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 15 Apr 2009 15:38:56 +0300 Subject: ASoC: OMAP: Add DSP_A mode support for mcbsp DSP_A mode is similar to the DSP_B, but the MSB is delayed with one bclk (appears after the FS pulse and not under it). Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 402a1eb..2b4a8da 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->srgr2 |= FPER(wlen * channels - 1); regs->srgr1 |= FWID(wlen - 1); break; + case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: regs->srgr2 |= FPER(wlen * channels - 1); regs->srgr1 |= FWID(wlen * channels - 2); @@ -330,6 +331,13 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 1-bit data delay */ + regs->rcr2 |= RDATDLY(1); + regs->xcr2 |= XDATDLY(1); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + break; case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); -- cgit v1.1 From 6b87a91f5417226c7fe62100b0e7217e7096b789 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 17 Apr 2009 15:55:08 +0300 Subject: ASoC: TWL4030: Fix for the constraint handling The original implementation of the constraints were good against sane applications. If the opening sequence is: stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the constraints are set correctly for stream2. But if the sequence is: stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2 would receive constraint rate = 0, sample_bits = 0, since the stream1 has not yet called hw_params... The command to trigger this event: gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false This patch does some 'black magic' in order to always set the correct constraints and sets it only when it is needed for the other stream. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 85 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 66 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205..a1b76d7 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -125,6 +125,11 @@ struct twl4030_priv { struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + + unsigned int configured; + unsigned int rate; + unsigned int sample_bits; + unsigned int channels; }; /* @@ -1220,6 +1225,36 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } +static void twl4030_constraints(struct twl4030_priv *twl4030, + struct snd_pcm_substream *mst_substream) +{ + struct snd_pcm_substream *slv_substream; + + /* Pick the stream, which need to be constrained */ + if (mst_substream == twl4030->master_substream) + slv_substream = twl4030->slave_substream; + else if (mst_substream == twl4030->slave_substream) + slv_substream = twl4030->master_substream; + else /* This should not happen.. */ + return; + + /* Set the constraints according to the already configured stream */ + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + twl4030->rate, + twl4030->rate); + + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + twl4030->sample_bits, + twl4030->sample_bits); + + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + twl4030->channels, + twl4030->channels); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1228,26 +1263,16 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - /* If we already have a playback or capture going then constrain - * this substream to match it. - */ if (twl4030->master_substream) { - struct snd_pcm_runtime *master_runtime; - master_runtime = twl4030->master_substream->runtime; - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - twl4030->slave_substream = substream; - } else + /* The DAI has one configuration for playback and capture, so + * if the DAI has been already configured then constrain this + * substream to match it. */ + if (twl4030->configured) + twl4030_constraints(twl4030, twl4030->master_substream); + } else { twl4030->master_substream = substream; + } return 0; } @@ -1264,6 +1289,13 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream, twl4030->master_substream = twl4030->slave_substream; twl4030->slave_substream = NULL; + + /* If all streams are closed, or the remaining stream has not yet + * been configured than set the DAI as not configured. */ + if (!twl4030->master_substream) + twl4030->configured = 0; + else if (!twl4030->master_substream->runtime->channels) + twl4030->configured = 0; } static int twl4030_hw_params(struct snd_pcm_substream *substream, @@ -1276,8 +1308,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; - if (substream == twl4030->slave_substream) - /* Ignoring hw_params for slave substream */ + if (twl4030->configured) + /* Ignoring hw_params for already configured DAI */ return 0; /* bit rate */ @@ -1357,6 +1389,21 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* set CODECPDZ afterwards */ twl4030_codec_enable(codec, 1); } + + /* Store the important parameters for the DAI configuration and set + * the DAI as configured */ + twl4030->configured = 1; + twl4030->rate = params_rate(params); + twl4030->sample_bits = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + twl4030->channels = params_channels(params); + + /* If both playback and capture streams are open, and one of them + * is setting the hw parameters right now (since we are here), set + * constraints to the other stream to match the current one. */ + if (twl4030->slave_substream) + twl4030_constraints(twl4030, substream); + return 0; } -- cgit v1.1 From 67667263674663767ddf4250bab2437a00ee780e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Apr 2009 10:49:25 +0200 Subject: ALSA: hda - Fix channels_max setting for CA0110 Added the missing definition of max channels for CA0110, which resulted in an error at opening PCM devices. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 7ec41da..9398d92 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -309,7 +309,7 @@ static int ca0110_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = - spec->multiout.num_dacs * 2; + spec->multiout.max_channels; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; @@ -418,6 +418,7 @@ static void parse_line_outs(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = n; + spec->multiout.max_channels = n * 2; } static void parse_hp_out(struct hda_codec *codec) -- cgit v1.1 From 7670dc41b51983b369f9adfb8694a580e7b0cef2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Apr 2009 10:51:11 +0200 Subject: ALSA: hda - Use snd_hda_codec_get_pincfg() in patch_ca0110.c Use the new function to reduce the access and allow the user setup via sysfs, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 9398d92..392d108 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -408,8 +408,7 @@ static void parse_line_outs(struct hda_codec *codec) n = 0; for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (!def_conf) continue; /* invalid pin */ if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1) @@ -432,8 +431,7 @@ static void parse_hp_out(struct hda_codec *codec) if (!cfg->hp_outs) return; nid = cfg->hp_pins[0]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (!def_conf) { cfg->hp_outs = 0; return; -- cgit v1.1 From 7154b3e80203ee91f9ba7d0a43d3daa05c49d9e9 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Mon, 20 Apr 2009 19:21:35 +0900 Subject: ASoC: TWL4030: Add support Voice DAI Add Voice DAI to support the PCM voice interface of the twl4030 codec. The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX. The PCM voice interface has two modes - PCM mode1 : This uses the normal FS polarity and the rising edge of the clock signal. - PCM mode2 : This uses the FS polarity inverted and the falling edge of the clock signal. If the system master clock is not 26MHz or the twl4030 codec mode is not option2, the voice PCM interface is not available. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 173 ++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 18 ++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 191 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a1b76d7..cc2968c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1484,6 +1484,144 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + u8 mode; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { + printk(KERN_ERR "TWL4030 voice startup: " + "MCLK is not 26MHz, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* If the codec mode is not option2, the voice PCM interface is not + * avilable. + */ + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & TWL4030_OPT_MODE; + + if (mode != TWL4030_OPTION_2) { + printk(KERN_ERR "TWL4030 voice startup: " + "the codec mode is not option2\n"); + return -EINVAL; + } + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & ~(TWL4030_CODECPDZ); + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + default: + printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_NB_IF: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1495,7 +1633,15 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_sysclk = twl4030_voice_set_dai_sysclk, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", @@ -1510,6 +1656,23 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ + .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1550,8 +1713,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1645,13 +1808,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..981ec60 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,6 +113,8 @@ #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +#define TWL4030_OPTION_1 (1 << 0) +#define TWL4030_OPTION_2 (0 << 0) /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ @@ -171,6 +173,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +249,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- cgit v1.1 From cd0f2d4736ae8efabc60e54ecc8f677d0eddce02 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 Apr 2009 16:56:59 +0100 Subject: ASoC: Factor out generic widget power checks This will form a basis for further power check refactoring: the overall goal of these changes is to allow us to check power separately to applying it, allowing improvements in the power sequencing algorithms. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a6d7337..28e6e32 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -581,6 +581,19 @@ static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) return 0; } +/* Generic check to see if a widget should be powered. + */ +static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) +{ + int in, out; + + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + return out != 0 && in != 0; +} + /* * Scan a single DAPM widget for a complete audio path and update the * power status appropriately. @@ -653,11 +666,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, } /* all other widgets */ - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - power = (out != 0 && in != 0) ? 1 : 0; + power = dapm_generic_check_power(w); power_change = (w->power == power) ? 0 : 1; w->power = power; -- cgit v1.1 From 6ea31b9f0a0307e16656af27fcda3160e2a64a1b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 Apr 2009 17:15:41 +0100 Subject: ASoC: Factor out DAPM power checks for DACs and ADCs This also switches us to using a switch statement for the widget type in dapm_power_widget(). Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 81 +++++++++++++++++++++++++++++++--------------------- 1 file changed, 48 insertions(+), 33 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 28e6e32..22522e2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -594,6 +594,34 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } +/* Check to see if an ADC has power */ +static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) +{ + int in; + + if (w->active) { + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + return in != 0; + } else { + return dapm_generic_check_power(w); + } +} + +/* Check to see if a DAC has power */ +static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) +{ + int out; + + if (w->active) { + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + return out != 0; + } else { + return dapm_generic_check_power(w); + } +} + /* * Scan a single DAPM widget for a complete audio path and update the * power status appropriately. @@ -601,36 +629,23 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) static int dapm_power_widget(struct snd_soc_codec *codec, int event, struct snd_soc_dapm_widget *w) { - int in, out, power_change, power, ret; + int power, ret; - /* vmid - no action */ - if (w->id == snd_soc_dapm_vmid) + /* Work out the new power state */ + switch (w->id) { + case snd_soc_dapm_vmid: + /* No action required */ return 0; - /* active ADC */ - if (w->id == snd_soc_dapm_adc && w->active) { - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - power = (in != 0) ? 1 : 0; - if (power == w->power) - return 0; - w->power = power; - return dapm_generic_apply_power(w); - } + case snd_soc_dapm_adc: + power = dapm_adc_check_power(w); + break; - /* active DAC */ - if (w->id == snd_soc_dapm_dac && w->active) { - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - power = (out != 0) ? 1 : 0; - if (power == w->power) - return 0; - w->power = power; - return dapm_generic_apply_power(w); - } + case snd_soc_dapm_dac: + power = dapm_dac_check_power(w); + break; - /* pre and post event widgets */ - if (w->id == snd_soc_dapm_pre) { + case snd_soc_dapm_pre: if (!w->event) return 0; @@ -646,8 +661,8 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, return ret; } return 0; - } - if (w->id == snd_soc_dapm_post) { + + case snd_soc_dapm_post: if (!w->event) return 0; @@ -663,15 +678,15 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, return ret; } return 0; - } - /* all other widgets */ - power = dapm_generic_check_power(w); - power_change = (w->power == power) ? 0 : 1; - w->power = power; + default: + power = dapm_generic_check_power(w); + break; + } - if (!power_change) + if (w->power == power) return 0; + w->power = power; return dapm_generic_apply_power(w); } -- cgit v1.1 From b75576d76d4be50196773f36709cb7a4f5ac2ab7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 Apr 2009 17:56:13 +0100 Subject: ASoC: Make the DAPM power check an operation on the widget Rather than having switch statements at point of use make the DAPM power check a member of the widget structure and set it when we instantiate the widget. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 27 +++++++++++++-------------- 2 files changed, 15 insertions(+), 14 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index fcc929d..839a97b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -367,6 +367,8 @@ struct snd_soc_dapm_widget { unsigned char suspend:1; /* was active before suspend */ unsigned char pmdown:1; /* waiting for timeout */ + int (*power_check)(struct snd_soc_dapm_widget *w); + /* external events */ unsigned short event_flags; /* flags to specify event types */ int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 22522e2..d3d1735 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -631,20 +631,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, { int power, ret; - /* Work out the new power state */ switch (w->id) { - case snd_soc_dapm_vmid: - /* No action required */ - return 0; - - case snd_soc_dapm_adc: - power = dapm_adc_check_power(w); - break; - - case snd_soc_dapm_dac: - power = dapm_dac_check_power(w); - break; - case snd_soc_dapm_pre: if (!w->event) return 0; @@ -680,10 +667,13 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, return 0; default: - power = dapm_generic_check_power(w); break; } + if (!w->power_check) + return 0; + + power = w->power_check(w); if (w->power == power) return 0; w->power = power; @@ -1147,15 +1137,22 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: + w->power_check = dapm_generic_check_power; dapm_new_mixer(codec, w); break; case snd_soc_dapm_mux: case snd_soc_dapm_value_mux: + w->power_check = dapm_generic_check_power; dapm_new_mux(codec, w); break; case snd_soc_dapm_adc: + w->power_check = dapm_adc_check_power; + break; case snd_soc_dapm_dac: + w->power_check = dapm_dac_check_power; + break; case snd_soc_dapm_pga: + w->power_check = dapm_generic_check_power; dapm_new_pga(codec, w); break; case snd_soc_dapm_input: @@ -1165,6 +1162,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_line: + w->power_check = dapm_generic_check_power; + break; case snd_soc_dapm_vmid: case snd_soc_dapm_pre: case snd_soc_dapm_post: -- cgit v1.1 From cd474f2d548af3c0eb932d9d47ec11483861aa6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 08:53:08 +0200 Subject: ALSA: Remove deprecated snd_card_new() Signed-off-by: Takashi Iwai --- include/sound/core.h | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..a26bbdc 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -300,16 +300,6 @@ int snd_card_create(int idx, const char *id, struct module *module, int extra_size, struct snd_card **card_ret); -static inline __deprecated -struct snd_card *snd_card_new(int idx, const char *id, - struct module *module, int extra_size) -{ - struct snd_card *card; - if (snd_card_create(idx, id, module, extra_size, &card) < 0) - return NULL; - return card; -} - int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); -- cgit v1.1 From ef9dfa4b1052af23a603de382d4665b2d1fccc61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 08:53:41 +0200 Subject: ALSA: Remove deprecated include/sound/driver.h Signed-off-by: Takashi Iwai --- include/sound/driver.h | 1 - 1 file changed, 1 deletion(-) delete mode 100644 include/sound/driver.h diff --git a/include/sound/driver.h b/include/sound/driver.h deleted file mode 100644 index f035943..0000000 --- a/include/sound/driver.h +++ /dev/null @@ -1 +0,0 @@ -#warning "This file is deprecated" -- cgit v1.1 From 92c7c8a7d6e03eb4c0a3c5888e35dbc45f24744c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Mar 2009 07:32:14 +0100 Subject: ALSA: hda - Cache PCM and STREAM parameters queries Cache quries for PCM and STREAM parameters as well as ampcap and pincap sharing the hash table. This will reduce the superfluous access of the same codec verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 97 +++++++++++++++++++++++++++++------------------ 1 file changed, 61 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a4e5e59..3d8bf39 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1053,6 +1053,8 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* FIXME: more better hash key? */ #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) +#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) +#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) #define INFO_AMP_CAPS (1<<0) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) @@ -1143,19 +1145,32 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); -u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +static unsigned int +query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key, + unsigned int (*func)(struct hda_codec *, hda_nid_t)) { struct hda_amp_info *info; - info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + info = get_alloc_amp_hash(codec, key); if (!info) return 0; if (!info->head.val) { - info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); info->head.val |= INFO_AMP_CAPS; + info->amp_caps = func(codec, nid); } return info->amp_caps; } + +static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); +} + +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), + read_pin_cap); +} EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); /* @@ -2538,6 +2553,41 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, } EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); +static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int val = 0; + if (nid != codec->afg && + (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) + val = snd_hda_param_read(codec, nid, AC_PAR_PCM); + if (!val || val == -1) + val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + if (!val || val == -1) + return 0; + return val; +} + +static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) +{ + return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid), + get_pcm_param); +} + +static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); + if (!streams || streams == -1) + streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); + if (!streams || streams == -1) + return 0; + return streams; +} + +static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) +{ + return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid), + get_stream_param); +} + /** * snd_hda_query_supported_pcm - query the supported PCM rates and formats * @codec: the HDA codec @@ -2556,15 +2606,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, { unsigned int i, val, wcaps; - val = 0; wcaps = get_wcaps(codec, nid); - if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); - if (val == -1) - return -EIO; - } - if (!val) - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); + val = query_pcm_param(codec, nid); if (ratesp) { u32 rates = 0; @@ -2586,15 +2629,9 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u64 formats = 0; unsigned int streams, bps; - streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); - if (streams == -1) + streams = query_stream_param(codec, nid); + if (!streams) return -EIO; - if (!streams) { - streams = snd_hda_param_read(codec, codec->afg, - AC_PAR_STREAM); - if (streams == -1) - return -EIO; - } bps = 0; if (streams & AC_SUPFMT_PCM) { @@ -2668,17 +2705,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, int i; unsigned int val = 0, rate, stream; - if (nid != codec->afg && - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); - if (val == -1) - return 0; - } - if (!val) { - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); - if (val == -1) - return 0; - } + val = query_pcm_param(codec, nid); + if (!val) + return 0; rate = format & 0xff00; for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++) @@ -2690,12 +2719,8 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, if (i >= AC_PAR_PCM_RATE_BITS) return 0; - stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); - if (stream == -1) - return 0; - if (!stream && nid != codec->afg) - stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); - if (!stream || stream == -1) + stream = query_stream_param(codec, nid); + if (!stream) return 0; if (stream & AC_SUPFMT_PCM) { -- cgit v1.1 From b613291fb21a2d74eb8323d97fe9aa5d281b306c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Mar 2009 07:36:09 +0100 Subject: ALSA: hda - Retry codec-verbs at errors The current error-recovery scheme for the codec communication errors doesn't work always well. Especially falling back to the single-command mode causes the fatal problem on many systems. In this patch, the problematic verb is re-issued again after the error (even with polling mode) instead of the single-cmd mode. The single-cmd mode will be used only when specified via the command option explicitly, mainly just for testing. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 17 +++++++++++++---- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 19 ++++++++++--------- 3 files changed, 24 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3d8bf39..1736ccb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -174,14 +174,23 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, unsigned int verb, unsigned int parm) { struct hda_bus *bus = codec->bus; - unsigned int res; + unsigned int cmd, res; + int repeated = 0; - res = make_codec_cmd(codec, nid, direct, verb, parm); + cmd = make_codec_cmd(codec, nid, direct, verb, parm); snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); - if (!bus->ops.command(bus, res)) + again: + if (!bus->ops.command(bus, cmd)) { res = bus->ops.get_response(bus); - else + if (res == -1 && bus->rirb_error) { + if (repeated++ < 1) { + snd_printd(KERN_WARNING "hda_codec: " + "Trying verb 0x%08x again\n", cmd); + goto again; + } + } + } else res = (unsigned int)-1; mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2fdecf4..cd8979c 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -623,6 +623,7 @@ struct hda_bus { /* misc op flags */ unsigned int needs_damn_long_delay :1; unsigned int shutdown :1; /* being unloaded */ + unsigned int rirb_error:1; /* error in codec communication */ }; /* diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 30829ee..803b720 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -604,6 +604,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) } if (!chip->rirb.cmds) { smp_rmb(); + bus->rirb_error = 0; return chip->rirb.res; /* the last value */ } if (time_after(jiffies, timeout)) @@ -623,8 +624,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) chip->irq = -1; pci_disable_msi(chip->pci); chip->msi = 0; - if (azx_acquire_irq(chip, 1) < 0) + if (azx_acquire_irq(chip, 1) < 0) { + bus->rirb_error = 1; return -1; + } goto again; } @@ -644,14 +647,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) return -1; } - snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " - "switching to single_cmd mode: last cmd=0x%08x\n", - chip->last_cmd); - chip->rirb.rp = azx_readb(chip, RIRBWP); - chip->rirb.cmds = 0; - /* switch to single_cmd mode */ - chip->single_cmd = 1; - azx_free_cmd_io(chip); + snd_printk(KERN_ERR "hda_intel: azx_get_response timeout (ERROR): " + "last cmd=0x%08x\n", chip->last_cmd); + spin_lock_irq(&chip->reg_lock); + chip->rirb.cmds = 0; /* reset the index */ + bus->rirb_error = 1; + spin_unlock_irq(&chip->reg_lock); return -1; } -- cgit v1.1 From 586be3fcf97eec22fbc0ef6d67e823706aea7167 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Mar 2009 07:43:24 +0100 Subject: ALSA: hda - Add debug prints for Realtek auto-init Added a couple of debug prints to show the checked id numbers in alc_subsystem_id(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8209779..ee92c73 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1022,6 +1022,9 @@ static void alc_subsystem_id(struct hda_codec *codec, if (codec->vendor_id == 0x10ec0260) nid = 0x17; ass = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("realtek: No valid SSID, " + "checking pincfg 0x%08x for NID 0x%x\n", + nid, ass); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ @@ -1036,6 +1039,8 @@ static void alc_subsystem_id(struct hda_codec *codec, if (((ass >> 16) & 0xf) != tmp) return; do_sku: + snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", + ass & 0xffff, codec->vendor_id); /* * 0 : override * 1 : Swap Jack -- cgit v1.1 From a3b48c88f2d5a34c0e25aec0a3dab8069e5a9a72 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 13:37:29 +0200 Subject: ALSA: hda - minor optimization in hda_set_power_state() Check the target power-state before checking EAPD exception to reduce unneeded verb executions. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b649033..b91f6ed 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2348,7 +2348,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (wcaps & AC_WCAP_POWER) { unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wid_type == AC_WID_PIN) { + if (power_state == AC_PWRST_D3 && + wid_type == AC_WID_PIN) { unsigned int pincap; /* * don't power down the widget if it controls @@ -2360,7 +2361,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, nid, 0, AC_VERB_GET_EAPD_BTLENABLE, 0); eapd &= 0x02; - if (power_state == AC_PWRST_D3 && eapd) + if (eapd) continue; } } -- cgit v1.1 From dfed0ef9b3ff9e37903920b6938ed33344ad0b3d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 18:33:12 +0200 Subject: ALSA: hda - Fix a typo in debug print for realtek auto-detection The NID and ASS numbers were swapped... Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a6ec87a..8877120 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1026,7 +1026,7 @@ static void alc_subsystem_id(struct hda_codec *codec, ass = snd_hda_codec_get_pincfg(codec, nid); snd_printd("realtek: No valid SSID, " "checking pincfg 0x%08x for NID 0x%x\n", - nid, ass); + ass, nid); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ -- cgit v1.1 From 1b4246a1fc487370665bf6c55aac8eae379642c0 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Wed, 22 Apr 2009 10:56:50 +0900 Subject: ASoC: OMAP: Add checking to detect bufferless pcms Add checking in hw_params and prepare to detect bufferless pcms(i.e. BT <--> codec). Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 07cf7f4..6454e15 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -87,8 +87,10 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; int err = 0; + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma_data) - return -ENODEV; + return 0; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); @@ -134,6 +136,11 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->dma_data) + return 0; + memset(&dma_params, 0, sizeof(dma_params)); /* * Note: Regardless of interface data formats supported by OMAP McBSP -- cgit v1.1 From 246d0a17f5e09af0794960164269fc8988a8811c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 18:24:55 +0100 Subject: ASoC: Add power supply widget to DAPM Many modern CODECs have shared resources on chip which must be enabled for portions of the chip to work but which can be disabled at other times in order to achieve power savings. Examples of such resources include power supplies and some internal clocks. Since these widgets are dependencies for the audio path but do not carry audio signals they require slightly different handling to most widgets - they do not contribute to the audio path and so should not be counted as either inputs or outputs during path walks. Cases where one supply provides a supply for another will require additional work. There is also room for more optimisation of the graph walking to avoid repeated checks for the same thing. Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/dapm.txt | 1 + include/sound/soc-dapm.h | 7 +++++- sound/soc/soc-dapm.c | 44 +++++++++++++++++++++++++++++++---- 3 files changed, 46 insertions(+), 6 deletions(-) diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 9e67632..9ac842b 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -62,6 +62,7 @@ Audio DAPM widgets fall into a number of types:- o Mic - Mic (and optional Jack) o Line - Line Input/Output (and optional Jack) o Speaker - Speaker + o Supply - Power or clock supply widget used by other widgets. o Pre - Special PRE widget (exec before all others) o Post - Special POST widget (exec after all others) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 839a97b..533f9f2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -154,12 +154,16 @@ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} -/* generic register modifier widget */ +/* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ { .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \ .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} +#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \ +{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .event = wevent, \ + .event_flags = wflags} /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ @@ -308,6 +312,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ + snd_soc_dapm_supply, /* power/clock supply */ }; /* diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d3d1735..7847f80 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,17 +52,19 @@ /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { - snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, - snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac, - snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga, - snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post + snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias, + snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux, + snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, + snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, + snd_soc_dapm_post }; static int dapm_down_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias, - snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post + snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply, + snd_soc_dapm_post }; static int dapm_status = 1; @@ -165,6 +167,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_dac: case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: + case snd_soc_dapm_supply: p->connect = 1; break; /* does effect routing - dynamically connected */ @@ -435,6 +438,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->id == snd_soc_dapm_supply) + return 0; + if (widget->id == snd_soc_dapm_adc && widget->active) return 1; @@ -471,6 +477,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + if (widget->id == snd_soc_dapm_supply) + return 0; + /* active stream ? */ if (widget->id == snd_soc_dapm_dac && widget->active) return 1; @@ -622,6 +631,26 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) } } +/* Check to see if a power supply is needed */ +static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *path; + int power = 0; + + /* Check if one of our outputs is connected */ + list_for_each_entry(path, &w->sinks, list_source) { + if (path->sink && path->sink->power_check && + path->sink->power_check(path->sink)) { + power = 1; + break; + } + } + + dapm_clear_walk(w->codec); + + return power; +} + /* * Scan a single DAPM widget for a complete audio path and update the * power status appropriately. @@ -752,6 +781,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_pga: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: + case snd_soc_dapm_supply: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -880,6 +910,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_pga: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: + case snd_soc_dapm_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -1044,6 +1075,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_vmid: case snd_soc_dapm_pre: case snd_soc_dapm_post: + case snd_soc_dapm_supply: list_add(&path->list, &codec->dapm_paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -1164,6 +1196,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) case snd_soc_dapm_line: w->power_check = dapm_generic_check_power; break; + case snd_soc_dapm_supply: + w->power_check = dapm_supply_check_power; case snd_soc_dapm_vmid: case snd_soc_dapm_pre: case snd_soc_dapm_post: -- cgit v1.1 From 42768a12822c3a0a6d7db69445281db975938294 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 18:39:39 +0100 Subject: ASoC: Use DAPM supply widget for WM8903 charge pump Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 53 ++++++++++++++++++----------------------------- 1 file changed, 20 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c539184..a3a489d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -217,7 +217,6 @@ struct wm8903_priv { int sysclk; /* Reference counts */ - int charge_pump_users; int class_w_users; int playback_active; int capture_active; @@ -373,6 +372,15 @@ static void wm8903_reset(struct snd_soc_codec *codec) #define WM8903_OUTPUT_INT 0x2 #define WM8903_OUTPUT_IN 0x1 +static int wm8903_cp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + WARN_ON(event != SND_SOC_DAPM_POST_PMU); + mdelay(4); + + return 0; +} + /* * Event for headphone and line out amplifier power changes. Special * power up/down sequences are required in order to maximise pop/click @@ -382,12 +390,9 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8903_priv *wm8903 = codec->private_data; - struct i2c_client *i2c = codec->control_data; u16 val; u16 reg; int shift; - u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0); switch (w->reg) { case WM8903_POWER_MANAGEMENT_2: @@ -419,18 +424,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, /* Short the output */ val &= ~(WM8903_OUTPUT_SHORT << shift); wm8903_write(codec, reg, val); - - wm8903->charge_pump_users++; - - dev_dbg(&i2c->dev, "Charge pump use count now %d\n", - wm8903->charge_pump_users); - - if (wm8903->charge_pump_users == 1) { - dev_dbg(&i2c->dev, "Enabling charge pump\n"); - wm8903_write(codec, WM8903_CHARGE_PUMP_0, - cp_reg | WM8903_CP_ENA); - mdelay(4); - } } if (event & SND_SOC_DAPM_POST_PMU) { @@ -464,19 +457,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, wm8903_write(codec, reg, val); } - if (event & SND_SOC_DAPM_POST_PMD) { - wm8903->charge_pump_users--; - - dev_dbg(&i2c->dev, "Charge pump use count now %d\n", - wm8903->charge_pump_users); - - if (wm8903->charge_pump_users == 0) { - dev_dbg(&i2c->dev, "Disabling charge pump\n"); - wm8903_write(codec, WM8903_CHARGE_PUMP_0, - cp_reg & ~WM8903_CP_ENA); - } - } - return 0; } @@ -844,26 +824,28 @@ SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, 1, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, 0, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0, NULL, 0, wm8903_output_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0, + wm8903_cp_event, SND_SOC_DAPM_POST_PMU), }; static const struct snd_soc_dapm_route intercon[] = { @@ -951,6 +933,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "ROP", NULL, "Right Speaker PGA" }, { "RON", NULL, "Right Speaker PGA" }, + + { "Left Headphone Output PGA", NULL, "Charge Pump" }, + { "Right Headphone Output PGA", NULL, "Charge Pump" }, + { "Left Line Output PGA", NULL, "Charge Pump" }, + { "Right Line Output PGA", NULL, "Charge Pump" }, }; static int wm8903_add_widgets(struct snd_soc_codec *codec) -- cgit v1.1 From c2aef4ffd24dab5c8e94c66e4042ad39d38bcf39 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 20:04:44 +0100 Subject: ASoC: Support CLK_DSP in WM8903 CLK_DSP provides a master clock for the DAC and ADC related functionality on the device. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3a489d..27c8b94 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -846,6 +846,7 @@ SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0, wm8903_cp_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0), }; static const struct snd_soc_dapm_route intercon[] = { @@ -891,7 +892,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Input PGA", NULL, "Right Input Mode Mux" }, { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "CLK_DSP" }, + + { "DACL", NULL, "CLK_DSP" }, + { "DACR", NULL, "CLK_DSP" }, { "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" }, { "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" }, -- cgit v1.1 From 4dbfe8097157fde1f8054f48f991ea45833852cd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 20:32:40 +0100 Subject: ASoC: Optimise configuration of WM8903 DC servo Modify the default startup sequence in the chip to set the DC servo dither level for optimal performance. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 27c8b94..de0a585 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -978,6 +978,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); + /* Change DC servo dither level in startup sequence */ + wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11); + wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257); + wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2); + wm8903_run_sequence(codec, 0); wm8903_sync_reg_cache(codec, codec->reg_cache); -- cgit v1.1 From d7d5c5476a12333a33b7a14ebb10eccc729c01cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 21:03:50 +0100 Subject: ASoC: Actively manage the DC servo for WM8903 Save a little extra power by enabling the DC servo offset correction for the output channels only when the relevant channels are enabled. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index de0a585..0bab5c6 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -392,14 +392,18 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; u16 val; u16 reg; + u16 dcs_reg; + u16 dcs_bit; int shift; switch (w->reg) { case WM8903_POWER_MANAGEMENT_2: reg = WM8903_ANALOGUE_HP_0; + dcs_bit = 0 + w->shift; break; case WM8903_POWER_MANAGEMENT_3: reg = WM8903_ANALOGUE_LINEOUT_0; + dcs_bit = 2 + w->shift; break; default: BUG(); @@ -439,6 +443,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, val |= (WM8903_OUTPUT_OUT << shift); wm8903_write(codec, reg, val); + /* Enable the DC servo */ + dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0); + dcs_reg |= dcs_bit; + wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg); + /* Remove the short */ val |= (WM8903_OUTPUT_SHORT << shift); wm8903_write(codec, reg, val); @@ -451,6 +460,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, val &= ~(WM8903_OUTPUT_SHORT << shift); wm8903_write(codec, reg, val); + /* Disable the DC servo */ + dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0); + dcs_reg &= ~dcs_bit; + wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg); + /* Then disable the intermediate and output stages */ val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | WM8903_OUTPUT_IN) << shift); -- cgit v1.1 From 727fb909e541ebd09d5b552afef02a147978c151 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 21:06:14 +0100 Subject: ASoC: Remove redundant rate constraint for WM8903 This is now handled by symmetric_rates. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0bab5c6..bec418a 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1289,14 +1289,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream, if (wm8903->master_substream) { master_runtime = wm8903->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); + dev_dbg(&i2c->dev, "Constraining to %d bits\n", + master_runtime->sample_bits); snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -- cgit v1.1 From 291ce18ceb84aca79368369885eec2d329ae16c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 21:36:14 +0100 Subject: ASoC: Implement WM8903 digital sidetone support Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bec418a..d8a9222 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -533,6 +533,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, /* ALSA can only do steps of .01dB */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0); @@ -651,6 +652,16 @@ static const struct soc_enum rinput_inv_enum = SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text); +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum lsidetone_enum = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text); + +static const struct soc_enum rsidetone_enum = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); + static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ @@ -694,6 +705,9 @@ SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT, SOC_ENUM("ADC Companding Mode", adc_companding), SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0), +SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8, + 12, 0, digital_sidetone_tlv), + /* DAC */ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT, WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv), @@ -756,6 +770,12 @@ static const struct snd_kcontrol_new rinput_mux = static const struct snd_kcontrol_new rinput_inv_mux = SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum); +static const struct snd_kcontrol_new lsidetone_mux = + SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum); + +static const struct snd_kcontrol_new rsidetone_mux = + SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); + static const struct snd_kcontrol_new left_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0), @@ -822,6 +842,9 @@ SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0), +SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux), +SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux), + SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0), SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0), @@ -910,7 +933,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "ADCR", NULL, "Right Input PGA" }, { "ADCR", NULL, "CLK_DSP" }, + { "DACL Sidetone", "Left", "ADCL" }, + { "DACL Sidetone", "Right", "ADCR" }, + { "DACR Sidetone", "Left", "ADCL" }, + { "DACR Sidetone", "Right", "ADCR" }, + + { "DACL", NULL, "DACL Sidetone" }, { "DACL", NULL, "CLK_DSP" }, + { "DACR", NULL, "DACR Sidetone" }, { "DACR", NULL, "CLK_DSP" }, { "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" }, -- cgit v1.1 From 1a787e7ad242312af0afb2156596d42ee5e0c6bc Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Wed, 22 Apr 2009 13:13:34 +0900 Subject: ASoC: TWL4030: Add VDL path support Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 256 ++++++++++++++++++++++----------------------- 1 file changed, 126 insertions(+), 130 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index cc2968c..fdf88df 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -321,104 +321,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -430,7 +386,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -829,6 +785,12 @@ static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1); static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0); /* + * Voice Downlink GAIN volume control: + * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB) + */ +static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1); + +/* * Analog playback gain * -24 dB to 12 dB in 2 dB steps */ @@ -892,6 +854,16 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, 1, 1, 0), + /* Common voice downlink gain controls */ + SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume", + TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv), + + SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume", + TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv), + + SOC_SINGLE("DAC Voice Analog Downlink Switch", + TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), + /* Separate output gain controls */ SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, @@ -956,6 +928,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", + TWL4030_REG_AVDAC_CTL, 4, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -966,6 +940,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), /* Analog bypasses */ SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, @@ -998,26 +974,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1082,50 +1067,61 @@ static const struct snd_soc_dapm_route intercon[] = { {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, + {"VDL_APGA", NULL, "DAC Voice"}, + /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, - {"EARPIECE", NULL, "Earpiece Mux"}, - {"PREDRIVEL", NULL, "PredriveL Mux"}, - {"PREDRIVER", NULL, "PredriveR Mux"}, - {"HSOL", NULL, "HeadsetL Mux"}, - {"HSOR", NULL, "HeadsetR Mux"}, - {"CARKITL", NULL, "CarkitL Mux"}, - {"CARKITR", NULL, "CarkitR Mux"}, + {"EARPIECE", NULL, "Earpiece Mixer"}, + {"PREDRIVEL", NULL, "PredriveL Mixer"}, + {"PREDRIVER", NULL, "PredriveR Mixer"}, + {"HSOL", NULL, "HeadsetL Mixer"}, + {"HSOR", NULL, "HeadsetR Mixer"}, + {"CARKITL", NULL, "CarkitL Mixer"}, + {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, -- cgit v1.1 From 2d7e71fa231035d69faffbfe506ef23638385994 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Thu, 23 Apr 2009 17:05:38 +0800 Subject: ASoC: simplify the SSP DMA parameters settings by run-time generation The SSP DMA parameters can actually be easily generated at run-time since they are almost similar except for the FIFO width and direction. Another benefit is the re-use of information from 'struct ssp_device', like SSDR physical FIFO address and DRCMR register index for both directions. Signed-off-by: Eric Miao Signed-off-by: Mark Brown Reviewed-by: pHilipp Zabel --- sound/soc/pxa/pxa-ssp.c | 203 ++++++++++-------------------------------------- 1 file changed, 43 insertions(+), 160 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b9b61dd..fb8cacc 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -50,139 +50,6 @@ struct ssp_priv { #endif }; -#define PXA2xx_SSP1_BASE 0x41000000 -#define PXA27x_SSP2_BASE 0x41700000 -#define PXA27x_SSP3_BASE 0x41900000 -#define PXA3xx_SSP4_BASE 0x41a00000 - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = { - .name = "SSP1 PCM Mono out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = { - .name = "SSP1 PCM Mono in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = { - .name = "SSP1 PCM Stereo out", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(14), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = { - .name = "SSP1 PCM Stereo in", - .dev_addr = PXA2xx_SSP1_BASE + SSDR, - .drcmr = &DRCMR(13), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = { - .name = "SSP2 PCM Mono out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = { - .name = "SSP2 PCM Mono in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = { - .name = "SSP2 PCM Stereo out", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(16), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = { - .name = "SSP2 PCM Stereo in", - .dev_addr = PXA27x_SSP2_BASE + SSDR, - .drcmr = &DRCMR(15), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = { - .name = "SSP3 PCM Mono out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = { - .name = "SSP3 PCM Mono in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = { - .name = "SSP3 PCM Stereo out", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = { - .name = "SSP3 PCM Stereo in", - .dev_addr = PXA27x_SSP3_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = { - .name = "SSP4 PCM Mono out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = { - .name = "SSP4 PCM Mono in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH2, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = { - .name = "SSP4 PCM Stereo out", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(67), - .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | - DCMD_BURST16 | DCMD_WIDTH4, -}; - -static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = { - .name = "SSP4 PCM Stereo in", - .dev_addr = PXA3xx_SSP4_BASE + SSDR, - .drcmr = &DRCMR(66), - .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | - DCMD_BURST16 | DCMD_WIDTH4, -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -194,25 +61,33 @@ static void dump_registers(struct ssp_device *ssp) ssp_read_reg(ssp, SSACD)); } -static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { - { - &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in, - &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in, - }, - { - &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in, - &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in, - }, - { - &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in, - &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in, - }, - { - &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in, - &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in, - }, +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; }; +static struct pxa2xx_pcm_dma_params * +ssp_get_dma_params(struct ssp_device *ssp, int stereo, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + stereo ? "Stereo" : "Mono", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (stereo ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} + static int pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -227,6 +102,11 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } return ret; } @@ -241,6 +121,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, ssp_disable(&priv->dev); clk_disable(priv->dev.ssp->clk); } + + if (cpu_dai->dma_data) { + kfree(cpu_dai->dma_data); + cpu_dai->dma_data = NULL; + } } #ifdef CONFIG_PM @@ -653,25 +538,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; - int dma = 0, chn = params_channels(params); + int chn = params_channels(params); u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; - /* select correct DMA params */ - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - dma = 1; /* capture DMA offset is 1,3 */ + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); + /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - if (((chn == 2) && (ttsa != 1)) || (width == 32)) - dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ - - cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; - - dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); + cpu_dai->dma_data = ssp_get_dma_params(ssp, + ((chn == 2) && (ttsa != 1)) || (width == 32), + substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) -- cgit v1.1 From 8eb9feabe566d8272510d5fb33f55a72e3ab3ce4 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Thu, 23 Apr 2009 17:57:46 +0800 Subject: ASoC: change stereo/mono to 32-bit/16-bit for pxa-ssp The original idea came from pHilipp, and this makes the code looks more consistent. Signed-off-by: Eric Miao Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fb8cacc..6fc7876 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -67,7 +67,7 @@ struct pxa2xx_pcm_dma_data { }; static struct pxa2xx_pcm_dma_params * -ssp_get_dma_params(struct ssp_device *ssp, int stereo, int out) +ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -76,13 +76,13 @@ ssp_get_dma_params(struct ssp_device *ssp, int stereo, int out) return NULL; snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, - stereo ? "Stereo" : "Mono", out ? "out" : "in"); + width4 ? "32-bit" : "16-bit", out ? "out" : "in"); dma->params.name = dma->name; dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | - (stereo ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; dma->params.dev_addr = ssp->phys_base + SSDR; return &dma->params; -- cgit v1.1 From 31a00c6b7c0c4f01be49f02660de920c8b82b613 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Apr 2009 14:36:48 +0300 Subject: ASoC: OMAP: Add 4 channel support to mcbsp Add 4 channel support to omap-mcbsp. This mode is going to be used by the twl4030 codec, when it is configured in Option1 mode. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 495192a..a5d46a7 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -259,6 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->xcr2 |= XFRLEN2(wpf - 1); } case 1: + case 4: /* Set word per (McBSP) frame for phase1 */ regs->rcr1 |= RFRLEN1(wpf - 1); regs->xcr1 |= XFRLEN1(wpf - 1); @@ -506,13 +507,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .id = (link_id), \ .playback = { \ .channels_min = 1, \ - .channels_max = 2, \ + .channels_max = 4, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ .channels_min = 1, \ - .channels_max = 2, \ + .channels_max = 4, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ -- cgit v1.1 From 8a1f936acdfd53cb0a981f3f80483863dcd84fa9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Apr 2009 14:36:49 +0300 Subject: ASoC: TWL4030: Add 4 channel TDM support Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl4030 codec. The channel allocations are: Playback: TDM i2s TWL RX Channel 1 Left SDRL2 Channel 3 Right SDRR2 Channel 2 -- SDRL1 Channel 4 -- SDRR1 Capture: TDM i2s TWL TX Channel 1 Left TXL1 Channel 3 Right TXR1 Channel 2 -- TXL2 Channel 4 -- TXR2 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 52 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 11 ++++++++++ 2 files changed, 61 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fdf88df..e23c20c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1251,6 +1251,28 @@ static void twl4030_constraints(struct twl4030_priv *twl4030, twl4030->channels); } +/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for + * capture has to be enabled/disabled. */ +static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1267,6 +1289,15 @@ static int twl4030_startup(struct snd_pcm_substream *substream, if (twl4030->configured) twl4030_constraints(twl4030, twl4030->master_substream); } else { + if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) { + /* In option2 4 channel is not supported, set the + * constraint for the first stream for channels, the + * second stream will 'inherit' this cosntraint */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, 2); + } twl4030->master_substream = substream; } @@ -1292,6 +1323,10 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream, twl4030->configured = 0; else if (!twl4030->master_substream->runtime->channels) twl4030->configured = 0; + + /* If the closing substream had 4 channel, do the necessary cleanup */ + if (substream->runtime->channels == 4) + twl4030_tdm_enable(codec, substream->stream, 0); } static int twl4030_hw_params(struct snd_pcm_substream *substream, @@ -1304,6 +1339,16 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; + /* If the substream has 4 channel, do the necessary setup */ + if (params_channels(params) == 4) { + /* Safety check: are we in the correct operating mode? */ + if ((twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & + TWL4030_OPTION_1)) + twl4030_tdm_enable(codec, substream->stream, 1); + else + return -EINVAL; + } + if (twl4030->configured) /* Ignoring hw_params for already configured DAI */ return 0; @@ -1461,6 +1506,9 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: format |= TWL4030_AIF_FORMAT_CODEC; break; + case SND_SOC_DAIFMT_DSP_A: + format |= TWL4030_AIF_FORMAT_TDM; + break; default: return -EINVAL; } @@ -1642,13 +1690,13 @@ struct snd_soc_dai twl4030_dai[] = { .playback = { .stream_name = "Playback", .channels_min = 2, - .channels_max = 2, + .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, - .channels_max = 2, + .channels_max = 4, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 981ec60..3441115 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -116,6 +116,17 @@ #define TWL4030_OPTION_1 (1 << 0) #define TWL4030_OPTION_2 (0 << 0) +/* TWL4030_OPTION (0x02) Fields */ + +#define TWL4030_ATXL1_EN (1 << 0) +#define TWL4030_ATXR1_EN (1 << 1) +#define TWL4030_ATXL2_VTXL_EN (1 << 2) +#define TWL4030_ATXR2_VTXR_EN (1 << 3) +#define TWL4030_ARXL1_VRX_EN (1 << 4) +#define TWL4030_ARXR1_EN (1 << 5) +#define TWL4030_ARXL2_EN (1 << 6) +#define TWL4030_ARXR2_EN (1 << 7) + /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ #define TWL4030_MICBIAS2_CTL 0x40 -- cgit v1.1 From 0cfcdedaddf2468cb53e3cff9c3abfef14b4d784 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 23 Apr 2009 21:46:19 +0200 Subject: ALSA: sc6000: fix older card initialization The last patch to handle newer cards like SC7000 broke initialization of the SC6000. Fix this. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sc6000.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 983ab7e..c803b2e 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -391,7 +391,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, int config = mss_config | sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); int err; - int cfg[2]; int old = 0; err = sc6000_dsp_reset(vport); @@ -421,11 +420,18 @@ static int __devinit sc6000_init_board(char __iomem *vport, answer, version[0], version[1]); /* set configuration */ - sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); - if (sc6000_hw_cfg_write(vport, cfg) < 0) { - snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); - return -EIO; + sc6000_write(vport, COMMAND_5C); + if (sc6000_read(vport) < 0) + old = 1; + + if (!old) { + int cfg[2]; + sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], + mss_port[dev]); + if (sc6000_hw_cfg_write(vport, cfg) < 0) { + snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); + return -EIO; + } } err = sc6000_setup_board(vport, config); if (err < 0) { @@ -434,10 +440,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, } sc6000_dsp_reset(vport); - sc6000_write(vport, COMMAND_5C); - if (sc6000_read(vport) < 0) - old = 1; - sc6000_dsp_reset(vport); if (!old) { sc6000_write(vport, COMMAND_60); -- cgit v1.1 From a8353a57299f965ca8747b1b062490aef2c9ca50 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Apr 2009 11:03:21 +0300 Subject: ASoC: Beagle: Add support for 4 channel This patch adds support for the four channel TDM mode on Beagle board. Depending on the channel count, the interface needs to be configured differently (I2S for stereo DSP_A for four channels) Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap3beagle.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index 6aa428e..b0cff9f 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -41,23 +41,33 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt; int ret; + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; } /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); return ret; -- cgit v1.1 From 7629ad24f2b3df95c8b4cd8869e3c04e1df6c442 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Apr 2009 16:37:44 +0200 Subject: ASoC: add SOC_DOUBLE_EXT macro Add a macro for double controls with special callback functions. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index b1f2f88..6ab80bf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -118,6 +118,14 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- cgit v1.1 From 172fd9e26200668ebaf3e1d6d09b36d5d531bfa6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Apr 2009 16:33:10 +0100 Subject: ASoC: Fix S3C64xx IIS device registration and support both ports The S3C64xx IIS code had a number of problems with device registration. The hardware has two IIS ports of which the driver supported only one at once via a single exported DAI, attempting to identify the DAI to use based on the dev->id of the ASoC platform device. As well as limiting the driver to only supporting one IIS port at once this also meant that the ID of the soc-audio device (or in future the card device) had to match the IIS ID. Fix both problems by converting the driver to register the DAIs based on probing of platform devices registered by the arch/arm code, using those platform devices to interact with the clock API. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 146 ++++++++++++++++++++++++++-------------- sound/soc/s3c24xx/s3c64xx-i2s.h | 2 +- 2 files changed, 98 insertions(+), 50 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 33c5de7..a84c4be 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -120,36 +120,8 @@ EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate); static int s3c64xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - struct device *dev = &pdev->dev; - struct s3c_i2sv2_info *i2s; - int ret; - - dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); - - if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { - dev_err(dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - i2s = &s3c64xx_i2s[pdev->id]; - - ret = s3c_i2sv2_probe(pdev, dai, i2s, - pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); - if (ret) - return ret; - - i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; - i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; - - i2s->iis_cclk = clk_get(dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(dev, "failed to get audio-bus"); - iounmap(i2s->regs); - return -ENODEV; - } - /* configure GPIO for i2s port */ - switch (pdev->id) { + switch (dai->id) { case 0: s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); @@ -181,35 +153,114 @@ static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { .set_sysclk = s3c64xx_i2s_set_sysclk, }; -struct snd_soc_dai s3c64xx_i2s_dai = { - .name = "s3c64xx-i2s", - .id = 0, - .probe = s3c64xx_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, +struct snd_soc_dai s3c64xx_i2s_dai[] = { + { + .name = "s3c64xx-i2s", + .id = 0, + .probe = s3c64xx_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .ops = &s3c64xx_i2s_dai_ops, }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C64XX_I2S_RATES, - .formats = S3C64XX_I2S_FMTS, + { + .name = "s3c64xx-i2s", + .id = 1, + .probe = s3c64xx_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .ops = &s3c64xx_i2s_dai_ops, }, - .ops = &s3c64xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); +static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + dai = &s3c64xx_i2s_dai[pdev->id]; + dai->dev = &pdev->dev; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + ret = s3c_i2sv2_probe(pdev, dai, i2s, + dai->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); + if (ret) + goto err_clk; + + ret = snd_soc_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_iis_driver = { + .probe = s3c64xx_iis_dev_probe, + .remove = s3c64xx_iis_dev_remove, + .driver = { + .name = "s3c64xx-iis", + .owner = THIS_MODULE, + }, +}; + static int __init s3c64xx_i2s_init(void) { - return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai); + return platform_driver_register(&s3c64xx_iis_driver); } module_init(s3c64xx_i2s_init); static void __exit s3c64xx_i2s_exit(void) { - snd_soc_unregister_dai(&s3c64xx_i2s_dai); + platform_driver_unregister(&s3c64xx_iis_driver); } module_exit(s3c64xx_i2s_exit); @@ -217,6 +268,3 @@ module_exit(s3c64xx_i2s_exit); MODULE_AUTHOR("Ben Dooks, "); MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); MODULE_LICENSE("GPL"); - - - diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index b7ffe3c..597822a 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -24,7 +24,7 @@ #define S3C64XX_CLKSRC_PCLK (0) #define S3C64XX_CLKSRC_MUX (1) -extern struct snd_soc_dai s3c64xx_i2s_dai; +extern struct snd_soc_dai s3c64xx_i2s_dai[]; extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai); -- cgit v1.1 From 008bec397cdabd22a6f4e4c16a746a86a046f8af Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Apr 2009 16:27:09 +0100 Subject: ASoC: S3C2412: Failing to get the I2S clock is an error Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index b7e0b3f..168a088 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -120,7 +120,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); if (s3c2412_i2s.iis_cclk == NULL) { - pr_debug("failed to get i2sclk clock\n"); + pr_err("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); return -ENODEV; } -- cgit v1.1 From 4a79ba34cada6a5a4ee86ed53aa8a73ba1e6fc51 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Apr 2009 16:31:35 +0200 Subject: ALSA: hda - Add amp initialization for realtek auto mode In the realtek auto-probing mode, the initialization of amp with some magic COEF or EAPD verbs is applied only when the codec SSID has valid values to satisfy the realtek's definition. However, many devices don't provide in that way, thus the device doesn't work as is. This patch allows the same initialization code even if the SSID doesn't pass the bit test. Also, alc_subsystem_id() is changed just to check and define the type, so that it's called in the parser, instead of the initializer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 239 +++++++++++++++++++++++++----------------- 1 file changed, 145 insertions(+), 94 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 583603f..3a63063 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -253,6 +253,15 @@ enum { /* for GPIO Poll */ #define GPIO_MASK 0x03 +/* extra amp-initialization sequence types */ +enum { + ALC_INIT_NONE, + ALC_INIT_DEFAULT, + ALC_INIT_GPIO1, + ALC_INIT_GPIO2, + ALC_INIT_GPIO3, +}; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -322,6 +331,7 @@ struct alc_spec { /* other flags */ unsigned int no_analog :1; /* digital I/O only */ + int init_amp; /* for virtual master */ hda_nid_t vmaster_nid; @@ -994,74 +1004,21 @@ static void alc888_coef_init(struct hda_codec *codec) AC_VERB_SET_PROC_COEF, 0x3030); } -/* 32-bit subsystem ID for BIOS loading in HD Audio codec. - * 31 ~ 16 : Manufacture ID - * 15 ~ 8 : SKU ID - * 7 ~ 0 : Assembly ID - * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36 - */ -static void alc_subsystem_id(struct hda_codec *codec, - unsigned int porta, unsigned int porte, - unsigned int portd) +static void alc_auto_init_amp(struct hda_codec *codec, int type) { - unsigned int ass, tmp, i; - unsigned nid; - struct alc_spec *spec = codec->spec; - - ass = codec->subsystem_id & 0xffff; - if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) - goto do_sku; - - /* - * 31~30 : port conetcivity - * 29~21 : reserve - * 20 : PCBEEP input - * 19~16 : Check sum (15:1) - * 15~1 : Custom - * 0 : override - */ - nid = 0x1d; - if (codec->vendor_id == 0x10ec0260) - nid = 0x17; - ass = snd_hda_codec_get_pincfg(codec, nid); - snd_printd("realtek: No valid SSID, " - "checking pincfg 0x%08x for NID 0x%x\n", - ass, nid); - if (!(ass & 1) && !(ass & 0x100000)) - return; - if ((ass >> 30) != 1) /* no physical connection */ - return; + unsigned int tmp; - /* check sum */ - tmp = 0; - for (i = 1; i < 16; i++) { - if ((ass >> i) & 1) - tmp++; - } - if (((ass >> 16) & 0xf) != tmp) - return; -do_sku: - snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", - ass & 0xffff, codec->vendor_id); - /* - * 0 : override - * 1 : Swap Jack - * 2 : 0 --> Desktop, 1 --> Laptop - * 3~5 : External Amplifier control - * 7~6 : Reserved - */ - tmp = (ass & 0x38) >> 3; /* external Amp control */ - switch (tmp) { - case 1: + switch (type) { + case ALC_INIT_GPIO1: snd_hda_sequence_write(codec, alc_gpio1_init_verbs); break; - case 3: + case ALC_INIT_GPIO2: snd_hda_sequence_write(codec, alc_gpio2_init_verbs); break; - case 7: + case ALC_INIT_GPIO3: snd_hda_sequence_write(codec, alc_gpio3_init_verbs); break; - case 5: /* set EAPD output high */ + case ALC_INIT_DEFAULT: switch (codec->vendor_id) { case 0x10ec0260: snd_hda_codec_write(codec, 0x0f, 0, @@ -1115,7 +1072,7 @@ do_sku: tmp | 0x2010); break; case 0x10ec0888: - /*alc888_coef_init(codec);*/ /* called in alc_init() */ + alc888_coef_init(codec); break; case 0x10ec0267: case 0x10ec0268: @@ -1130,7 +1087,104 @@ do_sku: tmp | 0x3000); break; } - default: + break; + } +} + +static void alc_init_auto_hp(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->autocfg.hp_pins[0]) + return; + + if (!spec->autocfg.speaker_pins[0]) { + if (spec->autocfg.line_out_pins[0]) + spec->autocfg.speaker_pins[0] = + spec->autocfg.line_out_pins[0]; + else + return; + } + + snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_HP_EVENT); + spec->unsol_event = alc_sku_unsol_event; +} + +/* check subsystem ID and set up device-specific initialization; + * return 1 if initialized, 0 if invalid SSID + */ +/* 32-bit subsystem ID for BIOS loading in HD Audio codec. + * 31 ~ 16 : Manufacture ID + * 15 ~ 8 : SKU ID + * 7 ~ 0 : Assembly ID + * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36 + */ +static int alc_subsystem_id(struct hda_codec *codec, + hda_nid_t porta, hda_nid_t porte, + hda_nid_t portd) +{ + unsigned int ass, tmp, i; + unsigned nid; + struct alc_spec *spec = codec->spec; + + ass = codec->subsystem_id & 0xffff; + if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) + goto do_sku; + + /* invalid SSID, check the special NID pin defcfg instead */ + /* + * 31~30 : port conetcivity + * 29~21 : reserve + * 20 : PCBEEP input + * 19~16 : Check sum (15:1) + * 15~1 : Custom + * 0 : override + */ + nid = 0x1d; + if (codec->vendor_id == 0x10ec0260) + nid = 0x17; + ass = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("realtek: No valid SSID, " + "checking pincfg 0x%08x for NID 0x%x\n", + nid, ass); + if (!(ass & 1) && !(ass & 0x100000)) + return 0; + if ((ass >> 30) != 1) /* no physical connection */ + return 0; + + /* check sum */ + tmp = 0; + for (i = 1; i < 16; i++) { + if ((ass >> i) & 1) + tmp++; + } + if (((ass >> 16) & 0xf) != tmp) + return 0; +do_sku: + snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n", + ass & 0xffff, codec->vendor_id); + /* + * 0 : override + * 1 : Swap Jack + * 2 : 0 --> Desktop, 1 --> Laptop + * 3~5 : External Amplifier control + * 7~6 : Reserved + */ + tmp = (ass & 0x38) >> 3; /* external Amp control */ + switch (tmp) { + case 1: + spec->init_amp = ALC_INIT_GPIO1; + break; + case 3: + spec->init_amp = ALC_INIT_GPIO2; + break; + case 7: + spec->init_amp = ALC_INIT_GPIO3; + break; + case 5: + spec->init_amp = ALC_INIT_DEFAULT; break; } @@ -1138,7 +1192,7 @@ do_sku: * when the external headphone out jack is plugged" */ if (!(ass & 0x8000)) - return; + return 1; /* * 10~8 : Jack location * 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered @@ -1146,14 +1200,6 @@ do_sku: * 15 : 1 --> enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ - if (!spec->autocfg.speaker_pins[0]) { - if (spec->autocfg.line_out_pins[0]) - spec->autocfg.speaker_pins[0] = - spec->autocfg.line_out_pins[0]; - else - return; - } - if (!spec->autocfg.hp_pins[0]) { tmp = (ass >> 11) & 0x3; /* HP to chassis */ if (tmp == 0) @@ -1163,23 +1209,23 @@ do_sku: else if (tmp == 2) spec->autocfg.hp_pins[0] = portd; else - return; + return 1; } - if (spec->autocfg.hp_pins[0]) - snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC880_HP_EVENT); -#if 0 /* it's broken in some acses -- temporarily disabled */ - if (spec->autocfg.input_pins[AUTO_PIN_MIC] && - spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]) - snd_hda_codec_write(codec, - spec->autocfg.input_pins[AUTO_PIN_MIC], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC880_MIC_EVENT); -#endif /* disabled */ + alc_init_auto_hp(codec); + return 1; +} - spec->unsol_event = alc_sku_unsol_event; +static void alc_ssid_check(struct hda_codec *codec, + hda_nid_t porta, hda_nid_t porte, hda_nid_t portd) +{ + if (!alc_subsystem_id(codec, porta, porte, portd)) { + struct alc_spec *spec = codec->spec; + snd_printd("realtek: " + "Enable default setup for auto mode as fallback\n"); + spec->init_amp = ALC_INIT_DEFAULT; + alc_init_auto_hp(codec); + } } /* @@ -2923,8 +2969,7 @@ static int alc_init(struct hda_codec *codec) unsigned int i; alc_fix_pll(codec); - if (codec->vendor_id == 0x10ec0888) - alc888_coef_init(codec); + alc_auto_init_amp(codec, spec->init_amp); for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); @@ -4198,7 +4243,6 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i < spec->autocfg.line_outs; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -4303,6 +4347,8 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -5678,7 +5724,6 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t nid; - alc_subsystem_id(codec, 0x10, 0x15, 0x0f); nid = spec->autocfg.line_out_pins[0]; if (nid) { int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -5788,6 +5833,8 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + alc_ssid_check(codec, 0x10, 0x15, 0x0f); + return 1; } @@ -7013,7 +7060,6 @@ static void alc882_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -9154,7 +9200,6 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -9317,6 +9362,7 @@ static int patch_alc883(struct hda_codec *codec) if (!spec->capsrc_nids) spec->capsrc_nids = alc883_capsrc_nids; spec->capture_style = CAPT_MIX; /* matrix-style capture */ + spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; @@ -10842,6 +10888,8 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x14, 0x1b); + return 1; } @@ -13925,7 +13973,6 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x0e, 0x0f, 0x0b); for (i = 0; i < spec->autocfg.line_outs; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -14008,6 +14055,8 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(spec); + alc_ssid_check(codec, 0x0e, 0x0f, 0x0b); + return 1; } @@ -14889,7 +14938,6 @@ static void alc861vd_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -15107,6 +15155,8 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -16931,7 +16981,6 @@ static void alc662_auto_init_multi_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int i; - alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); @@ -17028,6 +17077,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } -- cgit v1.1 From 4bc4d8998a472cd64aa66a4abad3d833be901028 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 14:28:44 +0100 Subject: ASoC: Enforce symmetric rates for S3C64xx I2S interface There is only one LRCLK pin on each interface. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index a84c4be..c335248 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -171,6 +171,7 @@ struct snd_soc_dai s3c64xx_i2s_dai[] = { .formats = S3C64XX_I2S_FMTS, }, .ops = &s3c64xx_i2s_dai_ops, + .symmetric_rates = 1, }, { .name = "s3c64xx-i2s", @@ -189,6 +190,7 @@ struct snd_soc_dai s3c64xx_i2s_dai[] = { .formats = S3C64XX_I2S_FMTS, }, .ops = &s3c64xx_i2s_dai_ops, + .symmetric_rates = 1, }, }; EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); -- cgit v1.1 From 008db442efa542357314593c71ab9966be909855 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 19:17:08 +0100 Subject: ASoC: Include WM8350 register definitions in CODEC header It's expected behaviour for the CODEC header to provide them but the WM8350 doesn't due to having all the registers together under drivers/mfd. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h index d11bd92..d088eb4 100644 --- a/sound/soc/codecs/wm8350.h +++ b/sound/soc/codecs/wm8350.h @@ -13,6 +13,7 @@ #define _WM8350_H #include +#include extern struct snd_soc_dai wm8350_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8350; -- cgit v1.1 From 5c556a6e190897a0f1ff14e13722591828412031 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 20:23:19 +0100 Subject: ASoC: s3c-i2s-v2 diagnostic improvements Say what invalid values we're seeing when we see an invalid value and ensure that errors are displayed by default. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index ab680aa..aeea49c 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -105,7 +105,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; } writel(con, regs + S3C2412_IISCON); @@ -132,7 +134,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "TXDIS: Invalid MODE %xin IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; } writel(mod, regs + S3C2412_IISMOD); @@ -175,7 +179,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); } writel(mod, regs + S3C2412_IISMOD); @@ -199,7 +204,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); } writel(con, regs + S3C2412_IISCON); @@ -281,7 +287,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= IISMOD_MASTER; break; default: - pr_debug("unknwon master/slave format\n"); + pr_err("unknwon master/slave format\n"); return -EINVAL; } @@ -298,7 +304,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= S3C2412_IISMOD_SDF_IIS; break; default: - pr_debug("Unknown data format\n"); + pr_err("Unknown data format\n"); return -EINVAL; } -- cgit v1.1 From a7be4d92d989fc53d840d24cba2ebea9e5ad8480 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 20:24:15 +0100 Subject: ASoC: Use our registration function for S3C64xx Make sure we get the DAI operations initialised. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 18 ++++++------------ sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- 2 files changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index aeea49c..ab680aa 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -105,9 +105,7 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; + dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); } writel(con, regs + S3C2412_IISCON); @@ -134,9 +132,7 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "TXDIS: Invalid MODE %xin IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; + dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); } writel(mod, regs + S3C2412_IISMOD); @@ -179,8 +175,7 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); } writel(mod, regs + S3C2412_IISMOD); @@ -204,8 +199,7 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); } writel(con, regs + S3C2412_IISCON); @@ -287,7 +281,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= IISMOD_MASTER; break; default: - pr_err("unknwon master/slave format\n"); + pr_debug("unknwon master/slave format\n"); return -EINVAL; } @@ -304,7 +298,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= S3C2412_IISMOD_SDF_IIS; break; default: - pr_err("Unknown data format\n"); + pr_debug("Unknown data format\n"); return -EINVAL; } diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index c335248..1345fbd 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -225,7 +225,7 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) if (ret) goto err_clk; - ret = snd_soc_register_dai(dai); + ret = s3c_i2sv2_register_dai(dai); if (ret != 0) goto err_i2sv2; -- cgit v1.1 From 0b5e92c5e020ee7437fa5304a8451d6dd08d1a26 Mon Sep 17 00:00:00 2001 From: Jonathan Cameron Date: Mon, 27 Apr 2009 13:49:44 +0000 Subject: ASoC WM8940 Driver Signed-off-by: Jonathan Cameron Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8940.c | 955 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8940.h | 104 +++++ 4 files changed, 1065 insertions(+) create mode 100644 sound/soc/codecs/wm8940.c create mode 100644 sound/soc/codecs/wm8940.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 121d63f..1c19ad5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8940 if I2C select SND_SOC_WM8960 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI @@ -140,6 +141,9 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8940 + tristate + config SND_SOC_WM8960 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8116968..3d31b6b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8940-objs := wm8940.o snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o @@ -57,6 +58,7 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c new file mode 100644 index 0000000..26987dc --- /dev/null +++ b/sound/soc/codecs/wm8940.c @@ -0,0 +1,955 @@ +/* + * wm8940.c -- WM8940 ALSA Soc Audio driver + * + * Author: Jonathan Cameron + * + * Based on wm8510.c + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Not currently handled: + * Notch filter control + * AUXMode (inverting vs mixer) + * No means to obtain current gain if alc enabled. + * No use made of gpio + * Fast VMID discharge for power down + * Soft Start + * DLR and ALR Swaps not enabled + * Digital Sidetone not supported + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8940.h" + +struct wm8940_priv { + unsigned int sysclk; + u16 reg_cache[WM8940_CACHEREGNUM]; + struct snd_soc_codec codec; +}; + +static u16 wm8940_reg_defaults[] = { + 0x8940, /* Soft Reset */ + 0x0000, /* Power 1 */ + 0x0000, /* Power 2 */ + 0x0000, /* Power 3 */ + 0x0010, /* Interface Control */ + 0x0000, /* Companding Control */ + 0x0140, /* Clock Control */ + 0x0000, /* Additional Controls */ + 0x0000, /* GPIO Control */ + 0x0002, /* Auto Increment Control */ + 0x0000, /* DAC Control */ + 0x00FF, /* DAC Volume */ + 0, + 0, + 0x0100, /* ADC Control */ + 0x00FF, /* ADC Volume */ + 0x0000, /* Notch Filter 1 Control 1 */ + 0x0000, /* Notch Filter 1 Control 2 */ + 0x0000, /* Notch Filter 2 Control 1 */ + 0x0000, /* Notch Filter 2 Control 2 */ + 0x0000, /* Notch Filter 3 Control 1 */ + 0x0000, /* Notch Filter 3 Control 2 */ + 0x0000, /* Notch Filter 4 Control 1 */ + 0x0000, /* Notch Filter 4 Control 2 */ + 0x0032, /* DAC Limit Control 1 */ + 0x0000, /* DAC Limit Control 2 */ + 0, + 0, + 0, + 0, + 0, + 0, + 0x0038, /* ALC Control 1 */ + 0x000B, /* ALC Control 2 */ + 0x0032, /* ALC Control 3 */ + 0x0000, /* Noise Gate */ + 0x0041, /* PLLN */ + 0x000C, /* PLLK1 */ + 0x0093, /* PLLK2 */ + 0x00E9, /* PLLK3 */ + 0, + 0, + 0x0030, /* ALC Control 4 */ + 0, + 0x0002, /* Input Control */ + 0x0050, /* PGA Gain */ + 0, + 0x0002, /* ADC Boost Control */ + 0, + 0x0002, /* Output Control */ + 0x0000, /* Speaker Mixer Control */ + 0, + 0, + 0, + 0x0079, /* Speaker Volume */ + 0, + 0x0000, /* Mono Mixer Control */ +}; + +static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg >= ARRAY_SIZE(wm8940_reg_defaults)) + return -1; + + return cache[reg]; +} + +static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + + if (reg >= ARRAY_SIZE(wm8940_reg_defaults)) + return -1; + + cache[reg] = value; + + return 0; +} + +static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 data[3] = { reg, + (value & 0xff00) >> 8, + (value & 0x00ff) + }; + + wm8940_write_reg_cache(codec, reg, value); + + ret = codec->hw_write(codec->control_data, data, 3); + + if (ret < 0) + return ret; + else if (ret != 3) + return -EIO; + return 0; +} + +static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const struct soc_enum wm8940_adc_companding_enum += SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding); +static const struct soc_enum wm8940_dac_companding_enum += SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding); + +static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"}; +static const struct soc_enum wm8940_alc_mode_enum += SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text); + +static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"}; +static const struct soc_enum wm8940_mic_bias_level_enum += SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text); + +static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; +static const struct soc_enum wm8940_filter_mode_enum += SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text); + +DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); +DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); +DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); +DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); +DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); +DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); +DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1); +DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0); + +static const struct snd_kcontrol_new wm8940_snd_controls[] = { + SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL, + 6, 1, 0), + SOC_ENUM("DAC Companding", wm8940_dac_companding_enum), + SOC_ENUM("ADC Companding", wm8940_adc_companding_enum), + + SOC_ENUM("ALC Mode", wm8940_alc_mode_enum), + SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0), + SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1, + 3, 7, 1, wm8940_alc_max_tlv), + SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1, + 0, 7, 0, wm8940_alc_min_tlv), + SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2, + 0, 14, 0, wm8940_alc_tar_tlv), + SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0), + SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0), + SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE, + 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE, + 0, 7, 0), + + SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0), + SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0), + SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2, + 4, 9, 1, wm8940_lim_thresh_tlv), + SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2, + 0, 12, 0, wm8940_lim_boost_tlv), + + SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0), + SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN, + 0, 63, 0, wm8940_pga_vol_tlv), + SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL, + 0, 255, 0, wm8940_adc_tlv), + SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL, + 0, 255, 0, wm8940_adc_tlv), + SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum), + SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST, + 8, 1, 0, wm8940_capture_boost_vol_tlv), + SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL, + 0, 63, 0, wm8940_spk_vol_tlv), + SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL, 6, 1, 1), + + SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL, + 8, 1, 1, wm8940_att_tlv), + SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0), + + SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1), + SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX, + 7, 1, 1, wm8940_att_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0), + SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum), + SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0), + SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0), + SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0), + SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0), + SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0), +}; + +DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); +static const struct snd_kcontrol_new wm8940_input_boost_controls[] = { + SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1), + SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST, + 0, 7, 0, wm8940_boost_vol_tlv), + SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST, + 4, 7, 0, wm8940_boost_vol_tlv), +}; + +static const struct snd_kcontrol_new wm8940_micpga_controls[] = { + SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0), + SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0), + SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0, + &wm8940_speaker_mixer_controls[0], + ARRAY_SIZE(wm8940_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0, + &wm8940_mono_mixer_controls[0], + ARRAY_SIZE(wm8940_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0), + + SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), + + SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0), + SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0, + &wm8940_micpga_controls[0], + ARRAY_SIZE(wm8940_micpga_controls)), + SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0, + &wm8940_input_boost_controls[0], + ARRAY_SIZE(wm8940_input_boost_controls)), + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_INPUT("AUX"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + {"Mic PGA", "AUX Switch", "AUX"}, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8940_add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets, + ARRAY_SIZE(wm8940_dapm_widgets)); + if (ret) + goto error_ret; + ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + if (ret) + goto error_ret; + ret = snd_soc_dapm_new_widgets(codec); + +error_ret: + return ret; +} + +#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0); + +static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67; + u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + wm8940_write(codec, WM8940_CLOCK, clk); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= (2 << 3); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= (1 << 3); + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= (3 << 3); + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= (3 << 3) | (1 << 7); + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= (1 << 7); + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= (1 << 8); + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= (1 << 8) | (1 << 7); + break; + } + + wm8940_write(codec, WM8940_IFACE, iface); + + return 0; +} + +static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F; + u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1; + u16 companding = wm8940_read_reg_cache(codec, + WM8940_COMPANDINGCTL) & 0xFFDF; + int ret; + + /* LoutR control */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + && params_channels(params) == 2) + iface |= (1 << 9); + + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + addcntrl |= (0x5 << 1); + break; + case SNDRV_PCM_RATE_11025: + addcntrl |= (0x4 << 1); + break; + case SNDRV_PCM_RATE_16000: + addcntrl |= (0x3 << 1); + break; + case SNDRV_PCM_RATE_22050: + addcntrl |= (0x2 << 1); + break; + case SNDRV_PCM_RATE_32000: + addcntrl |= (0x1 << 1); + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl); + if (ret) + goto error_ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + companding = companding | (1 << 5); + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= (1 << 5); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= (2 << 5); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= (3 << 5); + break; + } + ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding); + if (ret) + goto error_ret; + ret = wm8940_write(codec, WM8940_IFACE, iface); + +error_ret: + return ret; +} + +static int wm8940_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf; + + if (mute) + mute_reg |= 0x40; + + return wm8940_write(codec, WM8940_DAC, mute_reg); +} + +static int wm8940_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* Enable thermal shutdown */ + val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); + ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2); + if (ret) + break; + /* set vmid to 75k */ + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + break; + case SND_SOC_BIAS_PREPARE: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + break; + case SND_SOC_BIAS_STANDBY: + /* ensure bufioen and biasen */ + pwr_reg |= (1 << 2) | (1 << 3); + /* set vmid to 300k for standby */ + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2); + break; + case SND_SOC_BIAS_OFF: + ret = wm8940_write(codec, WM8940_POWER1, pwr_reg); + break; + } + + return ret; +} + +struct pll_ { + unsigned int pre_scale:2; + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + /* The left shift ist to avoid accuracy loss when right shifting */ + Ndiv = target / source; + + if (Ndiv > 12) { + source <<= 1; + /* Multiply by 2 */ + pll_div.pre_scale = 0; + Ndiv = target / source; + } else if (Ndiv < 3) { + source >>= 2; + /* Divide by 4 */ + pll_div.pre_scale = 3; + Ndiv = target / source; + } else if (Ndiv < 6) { + source >>= 1; + /* divide by 2 */ + pll_div.pre_scale = 2; + Ndiv = target / source; + } else + pll_div.pre_scale = 1; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8940 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +/* Untested at the moment */ +static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + /* Turn off PLL */ + reg = wm8940_read_reg_cache(codec, WM8940_POWER1); + wm8940_write(codec, WM8940_POWER1, reg & 0x1df); + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); + wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff); + /* Pll power down */ + wm8940_write(codec, WM8940_PLLN, (1 << 7)); + return 0; + } + + /* Pll is followed by a frequency divide by 4 */ + pll_factors(freq_out*4, freq_in); + if (pll_div.k) + wm8940_write(codec, WM8940_PLLN, + (pll_div.pre_scale << 4) | pll_div.n | (1 << 6)); + else /* No factional component */ + wm8940_write(codec, WM8940_PLLN, + (pll_div.pre_scale << 4) | pll_div.n); + wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18); + wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff); + /* Enable the PLL */ + reg = wm8940_read_reg_cache(codec, WM8940_POWER1); + wm8940_write(codec, WM8940_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); + wm8940_write(codec, WM8940_CLOCK, reg | 0x100); + + return 0; +} + +static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8940_priv *wm8940 = codec->private_data; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8940->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + int ret = 0; + + switch (div_id) { + case WM8940_BCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3; + ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2)); + break; + case WM8940_MCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F; + ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5)); + break; + case WM8940_OPCLKDIV: + reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF; + ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + break; + } + return ret; +} + +#define WM8940_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8940_dai_ops = { + .hw_params = wm8940_i2s_hw_params, + .set_sysclk = wm8940_set_dai_sysclk, + .digital_mute = wm8940_mute, + .set_fmt = wm8940_set_dai_fmt, + .set_clkdiv = wm8940_set_dai_clkdiv, + .set_pll = wm8940_set_dai_pll, +}; + +struct snd_soc_dai wm8940_dai = { + .name = "WM8940", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8940_RATES, + .formats = WM8940_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8940_RATES, + .formats = WM8940_FORMATS, + }, + .ops = &wm8940_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8940_dai); + +static int wm8940_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int wm8940_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + int ret; + u8 data[3]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware + * Could use auto incremented writes to speed this up + */ + for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) { + data[0] = i; + data[1] = (cache[i] & 0xFF00) >> 8; + data[2] = cache[i] & 0x00FF; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret < 0) + goto error_ret; + else if (ret != 3) { + ret = -EIO; + goto error_ret; + } + } + ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto error_ret; + ret = wm8940_set_bias_level(codec, codec->suspend_bias_level); + +error_ret: + return ret; +} + +static struct snd_soc_codec *wm8940_codec; + +static int wm8940_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + int ret = 0; + + if (wm8940_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8940_codec; + codec = wm8940_codec; + + mutex_init(&codec->mutex); + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + ret = snd_soc_add_controls(codec, wm8940_snd_controls, + ARRAY_SIZE(wm8940_snd_controls)); + if (ret) + goto error_free_pcms; + ret = wm8940_add_widgets(codec); + if (ret) + goto error_free_pcms; + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto error_free_pcms; + } + + return ret; + +error_free_pcms: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8940_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8940 = { + .probe = wm8940_probe, + .remove = wm8940_remove, + .suspend = wm8940_suspend, + .resume = wm8940_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940); + +static int wm8940_register(struct wm8940_priv *wm8940) +{ + struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data; + struct snd_soc_codec *codec = &wm8940->codec; + int ret; + u16 reg; + if (wm8940_codec) { + dev_err(codec->dev, "Another WM8940 is registered\n"); + return -EINVAL; + } + + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8940; + codec->name = "WM8940"; + codec->owner = THIS_MODULE; + codec->read = wm8940_read_reg_cache; + codec->write = wm8940_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8940_set_bias_level; + codec->dai = &wm8940_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults); + codec->reg_cache = &wm8940->reg_cache; + + memcpy(codec->reg_cache, wm8940_reg_defaults, + sizeof(wm8940_reg_defaults)); + + ret = wm8940_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8940_dai.dev = codec->dev; + + wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = wm8940_write(codec, WM8940_POWER1, 0x180); + if (ret < 0) + return ret; + + if (!pdata) + dev_warn(codec->dev, "No platform data supplied\n"); + else { + reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); + ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi); + if (ret < 0) + return ret; + } + + + wm8940_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8940_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static void wm8940_unregister(struct wm8940_priv *wm8940) +{ + wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8940_dai); + snd_soc_unregister_codec(&wm8940->codec); + kfree(wm8940); + wm8940_codec = NULL; +} + +static int wm8940_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8940_priv *wm8940; + struct snd_soc_codec *codec; + + wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL); + if (wm8940 == NULL) + return -ENOMEM; + + codec = &wm8940->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + i2c_set_clientdata(i2c, wm8940); + codec->control_data = i2c; + codec->dev = &i2c->dev; + + return wm8940_register(wm8940); +} + +static int wm8940_i2c_remove(struct i2c_client *client) +{ + struct wm8940_priv *wm8940 = i2c_get_clientdata(client); + + wm8940_unregister(wm8940); + + return 0; +} + +static const struct i2c_device_id wm8940_i2c_id[] = { + { "wm8940", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); + +static struct i2c_driver wm8940_i2c_driver = { + .driver = { + .name = "WM8940 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8940_i2c_probe, + .remove = __devexit_p(wm8940_i2c_remove), + .id_table = wm8940_i2c_id, +}; + +static int __init wm8940_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8940_i2c_driver); + if (ret) + printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n", + ret); + return ret; +} +module_init(wm8940_modinit); + +static void __exit wm8940_exit(void) +{ + i2c_del_driver(&wm8940_i2c_driver); +} +module_exit(wm8940_exit); + +MODULE_DESCRIPTION("ASoC WM8940 driver"); +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h new file mode 100644 index 0000000..8410eed --- /dev/null +++ b/sound/soc/codecs/wm8940.h @@ -0,0 +1,104 @@ +/* + * wm8940.h -- WM8940 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8940_H +#define _WM8940_H + +struct wm8940_setup_data { + /* Vref to analogue output resistance */ +#define WM8940_VROI_1K 0 +#define WM8940_VROI_30K 1 + unsigned int vroi:1; +}; +extern struct snd_soc_dai wm8940_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8940; + +/* WM8940 register space */ +#define WM8940_SOFTRESET 0x00 +#define WM8940_POWER1 0x01 +#define WM8940_POWER2 0x02 +#define WM8940_POWER3 0x03 +#define WM8940_IFACE 0x04 +#define WM8940_COMPANDINGCTL 0x05 +#define WM8940_CLOCK 0x06 +#define WM8940_ADDCNTRL 0x07 +#define WM8940_GPIO 0x08 +#define WM8940_CTLINT 0x09 +#define WM8940_DAC 0x0A +#define WM8940_DACVOL 0x0B + +#define WM8940_ADC 0x0E +#define WM8940_ADCVOL 0x0F +#define WM8940_NOTCH1 0x10 +#define WM8940_NOTCH2 0x11 +#define WM8940_NOTCH3 0x12 +#define WM8940_NOTCH4 0x13 +#define WM8940_NOTCH5 0x14 +#define WM8940_NOTCH6 0x15 +#define WM8940_NOTCH7 0x16 +#define WM8940_NOTCH8 0x17 +#define WM8940_DACLIM1 0x18 +#define WM8940_DACLIM2 0x19 + +#define WM8940_ALC1 0x20 +#define WM8940_ALC2 0x21 +#define WM8940_ALC3 0x22 +#define WM8940_NOISEGATE 0x23 +#define WM8940_PLLN 0x24 +#define WM8940_PLLK1 0x25 +#define WM8940_PLLK2 0x26 +#define WM8940_PLLK3 0x27 + +#define WM8940_ALC4 0x2A + +#define WM8940_INPUTCTL 0x2C +#define WM8940_PGAGAIN 0x2D + +#define WM8940_ADCBOOST 0x2F + +#define WM8940_OUTPUTCTL 0x31 +#define WM8940_SPKMIX 0x32 + +#define WM8940_SPKVOL 0x36 + +#define WM8940_MONOMIX 0x38 + +#define WM8940_CACHEREGNUM 0x57 + + +/* Clock divider Id's */ +#define WM8940_BCLKDIV 0 +#define WM8940_MCLKDIV 1 +#define WM8940_OPCLKDIV 2 + +/* MCLK clock dividers */ +#define WM8940_MCLKDIV_1 0 +#define WM8940_MCLKDIV_1_5 1 +#define WM8940_MCLKDIV_2 2 +#define WM8940_MCLKDIV_3 3 +#define WM8940_MCLKDIV_4 4 +#define WM8940_MCLKDIV_6 5 +#define WM8940_MCLKDIV_8 6 +#define WM8940_MCLKDIV_12 7 + +/* BCLK clock dividers */ +#define WM8940_BCLKDIV_1 0 +#define WM8940_BCLKDIV_2 1 +#define WM8940_BCLKDIV_4 2 +#define WM8940_BCLKDIV_8 3 +#define WM8940_BCLKDIV_16 4 +#define WM8940_BCLKDIV_32 5 + +/* PLL Out Dividers */ +#define WM8940_OPCLKDIV_1 0 +#define WM8940_OPCLKDIV_2 1 +#define WM8940_OPCLKDIV_3 2 +#define WM8940_OPCLKDIV_4 3 + +#endif /* _WM8940_H */ + -- cgit v1.1 From 9c935386512a3faa1be1c3d81cba38b7259a43f5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Apr 2009 15:00:25 +0200 Subject: ASoC: cs4270: fix Master Capture Switch polarity The control modifies the MUTE register, hence the polarity must be inverted. Signed-off-by: Daniel Mack Acked-By: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 7fa09a3..3c34fe6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -486,7 +486,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), - SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1) }; /* -- cgit v1.1 From 1a4ba05ec8369d62c10155a8931e81267bfbd31c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Apr 2009 16:37:45 +0200 Subject: ASoC: cs4270: add Master Playback Switch This adds a new control named 'Master Playback Switch' for cs4270 codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch the put function and store the information about manually set mute controls from userspace. When a manual mute is set, we don't want the soc core to un-mute the outputs. Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 44 +++++++++++++++++++++++++++++++++++++++----- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3c34fe6..ece6ed6 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -109,6 +109,7 @@ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; + unsigned int manual_mute; }; /** @@ -453,7 +454,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } /** - * cs4270_mute - enable/disable the CS4270 external mute + * cs4270_dai_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI * @mute: 0 = disable mute, 1 = enable mute * @@ -462,21 +463,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_dai *dai, int mute) +static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4270_private *cs4270 = codec->private_data; int reg6; reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; - else + else { reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + reg6 |= cs4270->manual_mute; + } return snd_soc_write(codec, CS4270_MUTE, reg6); } +/** + * cs4270_soc_put_mute - put callback for the 'Master Playback switch' + * alsa control. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * This function basically passes the arguments on to the generic + * snd_soc_put_volsw() function and saves the mute information in + * our private data structure. This is because we want to prevent + * cs4270_dai_mute() neglecting the user's decision to manually + * mute the codec's output. + * + * Returns 0 for success. + */ +static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4270_private *cs4270 = codec->private_data; + int left = !ucontrol->value.integer.value[0]; + int right = !ucontrol->value.integer.value[1]; + + cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) | + (right ? CS4270_MUTE_DAC_B : 0); + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", @@ -486,7 +518,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), - SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1) + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1), + SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1, + snd_soc_get_volsw, cs4270_soc_put_mute), }; /* @@ -506,7 +540,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = { .hw_params = cs4270_hw_params, .set_sysclk = cs4270_set_dai_sysclk, .set_fmt = cs4270_set_dai_fmt, - .digital_mute = cs4270_mute, + .digital_mute = cs4270_dai_mute, }; struct snd_soc_dai cs4270_dai = { -- cgit v1.1 From 6be01cfb854818298753bfce65543dbc81d51d5a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 20:57:42 +0100 Subject: ASoC: Staticise TLV values in WM8940 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 26987dc..a66dacc 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -168,16 +168,16 @@ static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; static const struct soc_enum wm8940_filter_mode_enum = SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text); -DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); -DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); -DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); -DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); -DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); -DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); -DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); -DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); -DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1); -DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); +static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); +static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0); +static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0); +static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0); +static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0); +static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0); +static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1); +static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0); static const struct snd_kcontrol_new wm8940_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL, @@ -253,7 +253,7 @@ static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = { SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0), }; -DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); +static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1); static const struct snd_kcontrol_new wm8940_input_boost_controls[] = { SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1), SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST, -- cgit v1.1 From 2a2ed0dfc9ec44a899c7d4672f73f2c045099118 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 13:01:26 +0200 Subject: ALSA: hda - Don't enable auto-mute but for speakers in patch_realtek.c Enable auto-muting in model=auto only for devices with HP and speakers. For devices with HP and line-outs, don't enable the auto-muting. Also, add a debug print to show the auto-mute feature. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a63063..96475dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1099,13 +1099,16 @@ static void alc_init_auto_hp(struct hda_codec *codec) return; if (!spec->autocfg.speaker_pins[0]) { - if (spec->autocfg.line_out_pins[0]) + if (spec->autocfg.line_out_pins[0] && + spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) spec->autocfg.speaker_pins[0] = spec->autocfg.line_out_pins[0]; else return; } + snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", + spec->autocfg.hp_pins[0]); snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); -- cgit v1.1 From cb6605c1e4d2a2eaffdde433fbfe1567ca688458 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 13:03:19 +0200 Subject: ALSA: hda - Fix a typo in patch_realtek.c again The commmit dfed0ef9b3ff9e37903920b6938ed33344ad0b3d was reverted accidentally by the merge of auto-detection fix patch. Fixed again now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96475dc..3e7207b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1151,7 +1151,7 @@ static int alc_subsystem_id(struct hda_codec *codec, ass = snd_hda_codec_get_pincfg(codec, nid); snd_printd("realtek: No valid SSID, " "checking pincfg 0x%08x for NID 0x%x\n", - nid, ass); + ass, nid); if (!(ass & 1) && !(ass & 0x100000)) return 0; if ((ass >> 30) != 1) /* no physical connection */ -- cgit v1.1 From 09aa60df92a9c5ff00e156c0dbc79f166d406a7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Apr 2009 18:51:48 +0100 Subject: ASoC: Fix error message formatting in s3c64xx-i2s driver Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 1345fbd..7679f7b 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -215,7 +215,7 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus"); + dev_err(&pdev->dev, "failed to get audio-bus\n"); ret = PTR_ERR(i2s->iis_cclk); goto err; } -- cgit v1.1 From 8a0f62b842e2f189e36d9f4c575ee15da9c605ff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Apr 2009 20:28:47 +0100 Subject: ASoC: Check for supported CPUs when building s3c-i2s-v2 Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index ab680aa..3b9201c 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -37,6 +37,20 @@ #include "s3c-i2s-v2.h" +#undef S3C_IIS_V2_SUPPORTED + +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifdef CONFIG_PLAT_S3C64XX +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifndef S3C_IIS_V2_SUPPORTED +#error Unsupported CPU model +#endif + #define S3C2412_I2S_DEBUG_CON 0 static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -- cgit v1.1 From 51438449e717db54550b4676f38208092eb654da Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Apr 2009 20:30:39 +0100 Subject: ASoC: Make S3C64xx clock export function to return struct clk This makes the interface usable with the s3c-iis-v2 rate calculator and consistent with S3C2412. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 7 +++---- sound/soc/s3c24xx/s3c64xx-i2s.h | 4 +++- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 7679f7b..cb11f78 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -108,14 +108,13 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, return 0; } - -unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai) +struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) { struct s3c_i2sv2_info *i2s = to_info(dai); - return clk_get_rate(i2s->iis_cclk); + return i2s->iis_cclk; } -EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate); +EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); static int s3c64xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index 597822a..02148ce 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -15,6 +15,8 @@ #ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H #define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ +struct clk; + #include "s3c-i2s-v2.h" #define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK @@ -26,6 +28,6 @@ extern struct snd_soc_dai s3c64xx_i2s_dai[]; -extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai); +extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai); #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ -- cgit v1.1 From 553b1dd58c5cf1abd6d0965041169400a3cff1ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Apr 2009 20:29:25 +0100 Subject: ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bit The data format configuration for S3C64xx IISv2 is completely different to that for S3C24xx. Instead of a single bit configuration in bit 0 of IISMOD we have format selection in bits 13 and 14 and bit clock rate selection in bits 1 and 2. While we're here add support for 24 bit samples in S3C64xx. At some point it may be desirable to expose the bit clock rate selection to users but given the limited configuration options that may not be required. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 22 +++++++++++++++++++++- sound/soc/s3c24xx/s3c64xx-i2s.c | 3 ++- 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 3b9201c..54f4119 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -280,7 +280,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, */ #define IISMOD_MASTER_MASK (1 << 11) #define IISMOD_SLAVE (1 << 11) -#define IISMOD_MASTER (0x0) +#define IISMOD_MASTER (0 << 11) #endif switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -341,6 +341,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod |= S3C2412_IISMOD_8BIT; @@ -349,6 +350,25 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, iismod &= ~S3C2412_IISMOD_8BIT; break; } +#endif + +#ifdef CONFIG_PLAT_S3C64XX + iismod &= ~0x606; + /* Sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + /* 8 bit sample, 16fs BCLK */ + iismod |= 0x2004; + break; + case SNDRV_PCM_FORMAT_S16_LE: + /* 16 bit sample, 32fs BCLK */ + break; + case SNDRV_PCM_FORMAT_S24_LE: + /* 24 bit sample, 48fs BCLK */ + iismod |= 0x4002; + break; + } +#endif writel(iismod, i2s->regs + S3C2412_IISMOD); pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index cb11f78..e0f4a16 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -146,7 +146,8 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define S3C64XX_I2S_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { .set_sysclk = s3c64xx_i2s_set_sysclk, -- cgit v1.1 From 07736d48051869c37838635b41850618aa63b9a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Apr 2009 13:13:14 +0100 Subject: ASoC: Fix boot warnings from S3C IISv2 On startup we try to make sure that the port is quiesced but if the port is already stopped then this will generate a warning about the RX/TX mode configuration. Configure the mode before doing the teardown to suppress these warnings. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 54f4119..34142c8 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -573,6 +573,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev, unsigned long base) { struct device *dev = &pdev->dev; + unsigned int iismod; i2s->dev = dev; @@ -594,12 +595,16 @@ int s3c_i2sv2_probe(struct platform_device *pdev, clk_enable(i2s->iis_pclk); + /* Mark ourselves as in TXRX mode so we can run through our cleanup + * process without warnings. */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + iismod |= S3C2412_IISMOD_MODE_TXRX; + writel(iismod, i2s->regs + S3C2412_IISMOD); s3c2412_snd_txctrl(i2s, 0); s3c2412_snd_rxctrl(i2s, 0); return 0; } - EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); #ifdef CONFIG_PM -- cgit v1.1 From c86bde54062a4d02c1b58203b7802797e4007a8a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Apr 2009 13:09:33 +0100 Subject: ASoC: Allow use of resource from the platform device for S3C IISv2 Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 34142c8..cb85498 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -580,6 +580,24 @@ int s3c_i2sv2_probe(struct platform_device *pdev, /* record our i2s structure for later use in the callbacks */ dai->private_data = i2s; + if (!base) { + struct resource *res = platform_get_resource(pdev, + IORESOURCE_MEM, + 0); + if (!res) { + dev_err(dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (!request_mem_region(res->start, resource_size(res), + "s3c64xx-i2s-v4")) { + dev_err(dev, "Unable to request register region\n"); + return -EBUSY; + } + + base = res->start; + } + i2s->regs = ioremap(base, 0x100); if (i2s->regs == NULL) { dev_err(dev, "cannot ioremap registers\n"); -- cgit v1.1 From af3ea7bdc77be000f69a41e7c41060f72b5a7111 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Apr 2009 13:13:55 +0100 Subject: ASoC: Display the clock rate used as the basis for rate calculation Aids debugging. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index cb85498..ad690b2 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -523,6 +523,8 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, unsigned int best_rate = 0; unsigned int best_deviation = INT_MAX; + pr_debug("Input clock rate %ldHz\n", clkrate); + if (fstab == NULL) fstab = iis_fs_tab; -- cgit v1.1 From 38e43c81a07de8ee8a757a9c93dd3a4937dd35e0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Apr 2009 13:14:38 +0100 Subject: ASoC: Display S3C IISv2 mode and MS errors by default Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index ad690b2..bc4e504 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -295,7 +295,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= IISMOD_MASTER; break; default: - pr_debug("unknwon master/slave format\n"); + pr_err("unknwon master/slave format\n"); return -EINVAL; } @@ -312,7 +312,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= S3C2412_IISMOD_SDF_IIS; break; default: - pr_debug("Unknown data format\n"); + pr_err("Unknown data format\n"); return -EINVAL; } -- cgit v1.1 From abbc82466967064e4eaafa367fc225a8c803569c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Apr 2009 13:21:52 +0100 Subject: ASoC: Staticise txctrl and rxctrl for S3C IISv2 They aren't used by anything external and aren't prototyped; if any users appear they can be exported again for them. Also report what modes we have a problem with when we encounter invalid mode configurations. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index bc4e504..972c276 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -89,7 +89,7 @@ static inline void dbg_showcon(const char *fn, u32 con) /* Turn on or off the transmission path. */ -void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) +static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) { void __iomem *regs = i2s->regs; u32 fic, con, mod; @@ -119,7 +119,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; } writel(con, regs + S3C2412_IISCON); @@ -146,7 +148,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; } writel(mod, regs + S3C2412_IISMOD); @@ -157,9 +161,8 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) dbg_showcon(__func__, con); pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } -EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl); -void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) +static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) { void __iomem *regs = i2s->regs; u32 fic, con, mod; @@ -189,7 +192,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); } writel(mod, regs + S3C2412_IISMOD); @@ -213,7 +217,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) break; default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); } writel(con, regs + S3C2412_IISCON); @@ -223,7 +228,6 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) fic = readl(regs + S3C2412_IISFIC); pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } -EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl); /* * Wait for the LR signal to allow synchronisation to the L/R clock -- cgit v1.1 From 71437552f2564c0d0c5cc4995045683051c5fe62 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Apr 2009 13:42:04 +0100 Subject: ASoC: Use platform device resource for S3C64xx IISv2 Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index e0f4a16..3c06c40 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -220,8 +220,7 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) goto err; } - ret = s3c_i2sv2_probe(pdev, dai, i2s, - dai->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); if (ret) goto err_clk; -- cgit v1.1 From 33f503c96c976fd585dedb76514ca6cb286e60d9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 May 2009 12:24:55 +0100 Subject: ASoC: Use a shared define for AC97 CODEC data formats The AC97 wire format is completely fixed so CODECs don't have any choice about the formats they accept but controllers accept a variety of data formats and render them down onto the bus. Have a shared define so all the CODEC drivers will interoperate with any of our controller drivers. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 +++ sound/soc/codecs/ac97.c | 4 ++-- sound/soc/codecs/ad1980.c | 4 ++-- sound/soc/codecs/wm9705.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- 6 files changed, 15 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 22b729f..ea07b4b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -96,6 +96,9 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 +#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai_ops; struct snd_soc_dai; struct snd_ac97_bus_ops; diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index b0d4af1..932299b 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -53,13 +53,13 @@ struct snd_soc_dai ac97_dai = { .channels_min = 1, .channels_max = 2, .rates = STD_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, .rates = STD_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; EXPORT_SYMBOL_GPL(ac97_dai); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index ddb3b08..d7440a9 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -137,13 +137,13 @@ struct snd_soc_dai ad1980_dai = { .channels_min = 2, .channels_max = 6, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .formats = SND_SOC_STD_AC97_FMTS, }, }; EXPORT_SYMBOL_GPL(ad1980_dai); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index c2d1a7a..fa88b46 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -282,14 +282,14 @@ struct snd_soc_dai wm9705_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM9705_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SND_SOC_STD_AC97_FMTS, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9705_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SND_SOC_STD_AC97_FMTS, }, .ops = &wm9705_dai_ops, }, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 765cf1e..550c903 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -534,13 +534,13 @@ struct snd_soc_dai wm9712_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9712_dai_ops_hifi, }, { @@ -550,7 +550,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM9712_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9712_dai_ops_aux, } }; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a6feb784..d1744e9 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1040,13 +1040,13 @@ struct snd_soc_dai wm9713_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM9713_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9713_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9713_dai_ops_hifi, }, { @@ -1056,7 +1056,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM9713_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9713_dai_ops_aux, }, { -- cgit v1.1 From 4072604b9dd18f25a98cc0f4d3d4553ed1ad4152 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 May 2009 12:28:25 +0100 Subject: ASoC: Remove unused DAI format defines The defines for TDM and synchronous clocks are not used - they are mostly a legacy of the automatic clocking configuration. TDM will require configuration of the number of timeslots and which ones to use so can't be fit into the DAI format and synchronous mode is handled by symmetric_rates (and needs to be done by constraints rather than when the DAI format is being configured). Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 18 ------------------ 1 file changed, 18 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ea07b4b..a997c2c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -45,24 +45,6 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ /* - * DAI Left/Right Clocks. - * - * Specifies whether the DAI can support different samples for similtanious - * playback and capture. This usually requires a seperate physical frame - * clock for playback and capture. - */ -#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ -#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ - -/* - * TDM - * - * Time Division Multiplexing. Allows PCM data to be multplexed with other - * data on the DAI. - */ -#define SND_SOC_DAIFMT_TDM (1 << 6) - -/* * DAI hardware signal inversions. * * Specifies whether the DAI can also support inverted clocks for the specified -- cgit v1.1 From 514bf54cd8c7f172816d3c003a6d022e9165a29b Mon Sep 17 00:00:00 2001 From: James Gardiner Date: Sun, 3 May 2009 04:00:44 -0400 Subject: ALSA: hda - Addition for HP dv4-1222nr laptop support Signed-off-by: James Gardiner Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 44 ++++++++++++++++++++++++---- 2 files changed, 39 insertions(+), 6 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8eec05b..36c9712 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -347,6 +347,7 @@ STAC92HD71B* hp-m4 HP mini 1000 hp-dv5 HP dv series hp-hdx HP HDX series + hp-dv4-1222nr HP dv4-1222nr (with LED support) auto BIOS setup (default) STAC92HD73* diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 917bc5d..76487de 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_HP_M4, STAC_HP_DV5, STAC_HP_HDX, + STAC_HP_DV4_1222NR, STAC_92HD71BXX_MODELS }; @@ -1836,6 +1837,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = NULL, [STAC_HP_DV5] = NULL, [STAC_HP_HDX] = NULL, + [STAC_HP_DV4_1222NR] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { @@ -1847,6 +1849,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", [STAC_HP_HDX] = "hp-hdx", + [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -1855,6 +1858,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, + "HP dv4-1222nr", STAC_HP_DV4_1222NR), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -4520,27 +4525,38 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } - /* - * using power check for controlling mute led of HP HDX notebooks + * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) * * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise * the LED is NOT working properly ! + * + * Changed name to reflect that it now works for any designated + * model, not just HP HDX. */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, +static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_bit = 0; /* gets rid of compiler warning */ + + switch (spec->board_config) { + case STAC_HP_DV4_1222NR: + gpio_bit = 0x01; + break; + case STAC_HP_HDX: + gpio_bit = 0x08; + } if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data &= ~0x08; /* orange */ + spec->gpio_data &= ~gpio_bit; /* orange */ else - spec->gpio_data |= 0x08; /* white */ + spec->gpio_data |= gpio_bit; /* white */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, @@ -5219,6 +5235,22 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 1; break; + case STAC_HP_DV4_1222NR: + spec->num_dmics = 1; + /* I don't know if it needs 1 or 2 smuxes - will wait for + * bug reports to fix if needed + */ + spec->num_smuxes = 1; + spec->num_dmuxes = 1; +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* This controls MUTE LED */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + spec->gpio_data |= 0x01; + codec->patch_ops.check_power_status = + stac92xx_hp_check_power_status; +#endif + /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); @@ -5239,7 +5271,7 @@ again: /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_hdx_check_power_status; + stac92xx_hp_check_power_status; #endif break; }; -- cgit v1.1 From b0ec3a30bc01c15cc6277b223fae136f7b71e90c Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 3 May 2009 10:39:19 +0200 Subject: ALSA: sc6000: enable joystick port Add module parameter to enable or disable joystick port (gameport) on the SC6600 and later cards. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 4 +++- sound/isa/sc6000.c | 9 +++++++-- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..11a80a9c71 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1543,13 +1543,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sc6000 ----------------- - Module for Gallant SC-6000 soundcard. + Module for Gallant SC-6000 soundcard and later models: SC-6600 + and SC-7000. port - Port # (0x220 or 0x240) mss_port - MSS Port # (0x530 or 0xe80) irq - IRQ # (5,7,9,10,11) mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq dma - DMA # (1,3,0) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c803b2e..9a8bbf6 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -56,6 +56,7 @@ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x300, 0x310, 0x320, 0x330 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ +static bool joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = false }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); @@ -75,6 +76,8 @@ module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); /* * Commands of SC6000's DSP (SBPRO+special). @@ -363,7 +366,7 @@ static int __devinit sc6000_init_mss(char __iomem *vport, int config, static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, long xport, long xmpu, - long xmss_port) + long xmss_port, int joystick) { cfg[0] = 0; cfg[1] = 0; @@ -376,6 +379,8 @@ static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, if (xmss_port == 0xe80) cfg[0] |= 0x10; cfg[0] |= 0x40; /* always set */ + if (!joystick) + cfg[0] |= 0x02; cfg[1] |= 0x80; /* enable WSS system */ cfg[1] &= ~0x40; /* disable IDE */ snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); @@ -427,7 +432,7 @@ static int __devinit sc6000_init_board(char __iomem *vport, if (!old) { int cfg[2]; sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); + mss_port[dev], joystick[dev]); if (sc6000_hw_cfg_write(vport, cfg) < 0) { snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); return -EIO; -- cgit v1.1 From fcd274a345875b05c348ba19bc6b3dd48ecbb7d0 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 30 Apr 2009 21:47:22 -0500 Subject: ASoC: TWL4030: Add VDL analog bypass This patch adds voice downlink analog bypass switch. It follows the same approach as in other analog bypass switches. DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer', that will also allow voice DAC to be powered in digital voice loopback (sidetone). Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++------ 1 file changed, 31 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80..efb371f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -468,6 +468,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); +/* Analog bypass for Voice */ +static const struct snd_kcontrol_new twl4030_dapm_abypassv_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0); + /* Digital bypass gain, 0 mutes the bypass */ static const unsigned int twl4030_dapm_dbypass_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -585,7 +589,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -597,14 +601,28 @@ static int bypass_event(struct snd_soc_dapm_widget *w, else twl4030->bypass_state &= ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { + /* Analog voice bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= (1 << 4); + else + twl4030->bypass_state &= ~(1 << 4); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0x1F) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -935,7 +953,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - TWL4030_REG_AVDAC_CTL, 4, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -962,6 +980,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassl2_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control, + bypass_event, SND_SOC_DAPM_POST_REG), /* Digital bypasses */ SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, @@ -979,6 +1000,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 2, 0, NULL, 0), SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, + 4, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1067,13 +1090,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - - {"VDL_APGA", NULL, "DAC Voice"}, + {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -1169,11 +1192,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"}, {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"}, /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, -- cgit v1.1 From ee8f6894f358b6a04d8190fd78990749de98a498 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 30 Apr 2009 21:48:08 -0500 Subject: ASoC: TWL4030: Add voice digital loopback: sidetone This patch add voice digital loopback (sidetone) to the twl4030 driver. It mixes voice uplink attenuated (by sidetone gain) with voice downlink when the codec is working in option2 (voice/audio mode). Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efb371f..23bae74 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -491,6 +491,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = TWL4030_REG_ATX2ARXPGA, 0, 7, 0, twl4030_dapm_dbypass_tlv); +/* + * Voice Sidetone GAIN volume control: + * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB) + */ +static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1); + +/* Digital bypass voice: sidetone (VUL -> VDL)*/ +static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_VSTPGA, 0, 0x29, 0, + twl4030_dapm_dbypassv_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -607,12 +619,18 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); + } else if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 6 : 5)); + twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); else - twl4030->bypass_state &= ~(1 << (m->shift ? 6 : 5)); + twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); } /* Enable master analog loopback mode if any analog switch is enabled*/ @@ -991,6 +1009,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control, bypass_event, + SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), @@ -1203,9 +1224,11 @@ static const struct snd_soc_dapm_route intercon[] = { /* Digital bypass routes */ {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- cgit v1.1 From a195b51bc5abc745f12b7b2fe0e3422f55c1953f Mon Sep 17 00:00:00 2001 From: Jonathan Cameron Date: Mon, 4 May 2009 14:54:11 +0000 Subject: ASoC: IMote2 ASoC Support This patch adds the ASoC side of the board support for the Crossbow IMB400 daughter board. Thanks to Crossbow for considerable assistance. Signed-off-by: Jonathan Cameron Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 9 ++++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/imote2.c | 114 +++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 125 insertions(+) create mode 100644 sound/soc/pxa/imote2.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..eb75a1c 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701 help Say Y if you want to add support for SoC audio on the MIO A701. + +config SND_PXA2XX_SOC_IMOTE2 + tristate "SoC Audio support for IMote 2" + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 + select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8940 + help + Say Y if you want to add support for SoC audio on the + IMote 2. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 4b90c3c..6e096b4 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o +snd-soc-imote2-objs := imote2.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -35,3 +36,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o +obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c new file mode 100644 index 0000000..405587a --- /dev/null +++ b/sound/soc/pxa/imote2.c @@ -0,0 +1,114 @@ + +#include +#include + +#include + +#include "../codecs/wm8940.h" +#include "pxa2xx-i2s.h" +#include "pxa2xx-pcm.h" + +static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* CPU should be clock master */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set the I2S system clock as input (unused) */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk, + SND_SOC_CLOCK_OUT); + + return ret; +} + +static struct snd_soc_ops imote2_asoc_ops = { + .hw_params = imote2_asoc_hw_params, +}; + +static struct snd_soc_dai_link imote2_dai = { + .name = "WM8940", + .stream_name = "WM8940", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &wm8940_dai, + .ops = &imote2_asoc_ops, +}; + +static struct snd_soc_card snd_soc_imote2 = { + .name = "Imote2", + .platform = &pxa2xx_soc_platform, + .dai_link = &imote2_dai, + .num_links = 1, +}; + +static struct snd_soc_device imote2_snd_devdata = { + .card = &snd_soc_imote2, + .codec_dev = &soc_codec_dev_wm8940, +}; + +static struct platform_device *imote2_snd_device; + +static int __init imote2_asoc_init(void) +{ + int ret; + + if (!machine_is_intelmote2()) + return -ENODEV; + imote2_snd_device = platform_device_alloc("soc-audio", -1); + if (!imote2_snd_device) + return -ENOMEM; + + platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata); + imote2_snd_devdata.dev = &imote2_snd_device->dev; + ret = platform_device_add(imote2_snd_device); + if (ret) + platform_device_put(imote2_snd_device); + + return ret; +} +module_init(imote2_asoc_init); + +static void __exit imote2_asoc_exit(void) +{ + platform_device_unregister(imote2_snd_device); +} +module_exit(imote2_asoc_exit); + +MODULE_AUTHOR("Jonathan Cameron"); +MODULE_DESCRIPTION("ALSA SoC Imote 2"); +MODULE_LICENSE("GPL"); -- cgit v1.1 From 376f7839b72ec526173cafb5d8eadfc61e2effdf Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 5 May 2009 08:55:47 +0300 Subject: ASoC: TWL4030: Add VIBRA output This patch adds support for the VIBRA output on TWL4030 codec. The VIBRA output can be driven with audio data or with local vibrator driver. Add the needed DAPM elements and routes for the VIBRA output and controls for the VIBRA driver configuration. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 61 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 61 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 23bae74..1a00e4b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -396,6 +396,31 @@ static const struct soc_enum twl4030_handsfreer_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); +/* Vibra */ +/* Vibra audio path selection */ +static const char *twl4030_vibra_texts[] = + {"AudioL1", "AudioR1", "AudioL2", "AudioR2"}; + +static const struct soc_enum twl4030_vibra_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2, + ARRAY_SIZE(twl4030_vibra_texts), + twl4030_vibra_texts); + +static const struct snd_kcontrol_new twl4030_dapm_vibra_control = +SOC_DAPM_ENUM("Route", twl4030_vibra_enum); + +/* Vibra path selection: local vibrator (PWM) or audio driven */ +static const char *twl4030_vibrapath_texts[] = + {"Local vibrator", "Audio"}; + +static const struct soc_enum twl4030_vibrapath_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4, + ARRAY_SIZE(twl4030_vibrapath_texts), + twl4030_vibrapath_texts); + +static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control = +SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); + /* Left analog microphone selection */ static const char *twl4030_analoglmic_texts[] = {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; @@ -867,6 +892,26 @@ static const struct soc_enum twl4030_rampdelay_enum = ARRAY_SIZE(twl4030_rampdelay_texts), twl4030_rampdelay_texts); +/* Vibra H-bridge direction mode */ +static const char *twl4030_vibradirmode_texts[] = { + "Vibra H-bridge direction", "Audio data MSB", +}; + +static const struct soc_enum twl4030_vibradirmode_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5, + ARRAY_SIZE(twl4030_vibradirmode_texts), + twl4030_vibradirmode_texts); + +/* Vibra H-bridge direction */ +static const char *twl4030_vibradir_texts[] = { + "Positive polarity", "Negative polarity", +}; + +static const struct soc_enum twl4030_vibradir_enum = + SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1, + ARRAY_SIZE(twl4030_vibradir_texts), + twl4030_vibradir_texts); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -933,6 +978,9 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 0, 3, 5, 0, input_gain_tlv), SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), + + SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), + SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum), }; static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { @@ -960,6 +1008,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("CARKITR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("VIBRA"), /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", @@ -1060,6 +1109,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, &twl4030_dapm_handsfreer_control, handsfree_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + /* Vibra */ + SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, + &twl4030_dapm_vibra_control), + SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_vibrapath_control), /* Introducing four virtual ADC, since TWL4030 have four channel for capture */ @@ -1161,6 +1215,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, + /* Vibra */ + {"Vibra Mux", "AudioL1", "DAC Left1"}, + {"Vibra Mux", "AudioR1", "DAC Right1"}, + {"Vibra Mux", "AudioL2", "DAC Left2"}, + {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, @@ -1174,6 +1233,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, + {"Vibra Route", "Audio", "Vibra Mux"}, + {"VIBRA", NULL, "Vibra Route"}, /* Capture path */ {"Analog Left Capture Route", "Main mic", "MAINMIC"}, -- cgit v1.1 From bbd993077d788589a86a718ba7a7895ba5e71a17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 May 2009 10:27:38 +0100 Subject: ASoC: Remove redundant codec pointer from DAIs The DAI structure has two pointers to the codec, one in the body of the DAI and one in a union for a parent pointer. Drop the parent pointer version. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index a997c2c..496dc30 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -205,11 +205,8 @@ struct snd_soc_dai { /* DAI private data */ void *private_data; - /* parent codec/platform */ - union { - struct snd_soc_codec *codec; - struct snd_soc_platform *platform; - }; + /* parent platform */ + struct snd_soc_platform *platform; struct list_head list; }; -- cgit v1.1 From 4bbe1ddf89a5ba3ec30fe5980912d8bda3a3cbb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Oct 2008 03:07:14 +0200 Subject: ALSA: Add extra delay count in PCM Added runtime->delay field to adjust the delayed samples for snd_pcm_delay(). Typically a hardware FIFO length is stored in this field, so that the extra delay between hwptr and applptr can be computed. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 + sound/core/pcm_native.c | 8 +++++--- 2 files changed, 6 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c172968..267effd 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -270,6 +270,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ + snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index fc6f98e..cb769d4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -587,14 +587,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { status->avail = snd_pcm_playback_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || - runtime->status->state == SNDRV_PCM_STATE_DRAINING) + runtime->status->state == SNDRV_PCM_STATE_DRAINING) { status->delay = runtime->buffer_size - status->avail; - else + status->delay += runtime->delay; + } else status->delay = 0; } else { status->avail = snd_pcm_capture_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - status->delay = status->avail; + status->delay = status->avail + runtime->delay; else status->delay = 0; } @@ -2404,6 +2405,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, n = snd_pcm_playback_hw_avail(runtime); else n = snd_pcm_capture_avail(runtime); + n += runtime->delay; break; case SNDRV_PCM_STATE_XRUN: err = -EPIPE; -- cgit v1.1 From ae1ec5e1e97f67d41e641a73380129e5905e41cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Oct 2008 03:08:53 +0200 Subject: ALSA: usbaudio - Add delay account Manage the PCM delay account based on the queued URBs. Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 823296d..6c25fa2 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -627,6 +627,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += offs; if (subs->hwptr_done >= runtime->buffer_size) subs->hwptr_done -= runtime->buffer_size; + runtime->delay += offs; spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = offs * stride; if (period_elapsed) @@ -636,12 +637,22 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, /* * process after playback data complete - * - nothing to do + * - decrease the delay count again */ static int retire_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { + unsigned long flags; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay = 0; + else + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); return 0; } @@ -1520,6 +1531,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->phase = 0; + runtime->delay = 0; /* clear urbs (to be sure) */ deactivate_urbs(subs, 0, 1); -- cgit v1.1 From e6e55122a54db87e22c67477de2a9978a3e4c81b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 May 2009 11:10:24 +0100 Subject: ASoC: Add headers to match patterns in MAINTAINERS Signed-off-by: Mark Brown --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index ef03abe..3cf4f0d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5277,6 +5277,7 @@ L: alsa-devel@alsa-project.org (subscribers-only) W: http://alsa-project.org/main/index.php/ASoC S: Supported F: sound/soc/ +F: include/sound/soc* SPARC + UltraSPARC (sparc/sparc64) P: David S. Miller -- cgit v1.1 From 80ab8817bf9b740df1f0778c41875e93151409bf Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 5 May 2009 11:25:00 +0200 Subject: ASoC: cs4270: introduce CS4270_I2C_INCR Replace the magic 0x80 value with a suitable macro definition. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ece6ed6..153124b 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -56,6 +56,7 @@ #define CS4270_FIRSTREG 0x01 #define CS4270_LASTREG 0x08 #define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) +#define CS4270_I2C_INCR 0x80 /* Bit masks for the CS4270 registers */ #define CS4270_CHIPID_ID 0xF0 @@ -296,7 +297,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) s32 length; length = i2c_smbus_read_i2c_block_data(i2c_client, - CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); + CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { dev_err(codec->dev, "i2c read failure, addr=0x%x\n", -- cgit v1.1 From 92d71005e2f305d6dca126d8b8497e885b842dba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 May 2009 17:18:34 +0200 Subject: ALSA: ice1724 - Check error in set_rate function The set_rate might return error but the current code doesn't check it. This patch adds a proper error check. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 128510e..5c5ef7f 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -626,7 +626,7 @@ static unsigned char stdclock_set_mclk(struct snd_ice1712 *ice, return 0; } -static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, +static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, int force) { unsigned long flags; @@ -634,17 +634,18 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, unsigned int i, old_rate; if (rate > ice->hw_rates->list[ice->hw_rates->count - 1]) - return; + return -EINVAL; + spin_lock_irqsave(&ice->reg_lock, flags); if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) || (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); - return; + return -EBUSY; } if (!force && is_pro_rate_locked(ice)) { spin_unlock_irqrestore(&ice->reg_lock, flags); - return; + return (rate == ice->cur_rate) ? 0 : -EBUSY; } old_rate = ice->get_rate(ice); @@ -652,7 +653,7 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, ice->set_rate(ice, rate); else if (rate == ice->cur_rate) { spin_unlock_irqrestore(&ice->reg_lock, flags); - return; + return 0; } ice->cur_rate = rate; @@ -674,13 +675,15 @@ static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, } if (ice->spdif.ops.setup_rate) ice->spdif.ops.setup_rate(ice, rate); + + return 0; } static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); - int i, chs; + int i, chs, err; chs = params_channels(hw_params); mutex_lock(&ice->open_mutex); @@ -715,7 +718,11 @@ static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream, } } mutex_unlock(&ice->open_mutex); - snd_vt1724_set_pro_rate(ice, params_rate(hw_params), 0); + + err = snd_vt1724_set_pro_rate(ice, params_rate(hw_params), 0); + if (err < 0) + return err; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } -- cgit v1.1 From a5b7b5c1d05387ffeaf0487482806ec6c5968ac7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 May 2009 17:22:07 +0200 Subject: ALSA: ice1724 - Clean up definitions of DMA records Rename some vt1724_pcm_reg records to more generic and consistent ones. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 37 +++++++++++++++++++++---------------- 1 file changed, 21 insertions(+), 16 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 5c5ef7f..8afa043 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -855,20 +855,39 @@ static snd_pcm_uframes_t snd_vt1724_pcm_pointer(struct snd_pcm_substream *substr #endif } -static const struct vt1724_pcm_reg vt1724_playback_pro_reg = { +static const struct vt1724_pcm_reg vt1724_pdma0_reg = { .addr = VT1724_MT_PLAYBACK_ADDR, .size = VT1724_MT_PLAYBACK_SIZE, .count = VT1724_MT_PLAYBACK_COUNT, .start = VT1724_PDMA0_START, }; -static const struct vt1724_pcm_reg vt1724_capture_pro_reg = { +static const struct vt1724_pcm_reg vt1724_pdma4_reg = { + .addr = VT1724_MT_PDMA4_ADDR, + .size = VT1724_MT_PDMA4_SIZE, + .count = VT1724_MT_PDMA4_COUNT, + .start = VT1724_PDMA4_START, +}; + +static const struct vt1724_pcm_reg vt1724_rdma0_reg = { .addr = VT1724_MT_CAPTURE_ADDR, .size = VT1724_MT_CAPTURE_SIZE, .count = VT1724_MT_CAPTURE_COUNT, .start = VT1724_RDMA0_START, }; +static const struct vt1724_pcm_reg vt1724_rdma1_reg = { + .addr = VT1724_MT_RDMA1_ADDR, + .size = VT1724_MT_RDMA1_SIZE, + .count = VT1724_MT_RDMA1_COUNT, + .start = VT1724_RDMA1_START, +}; + +#define vt1724_playback_pro_reg vt1724_pdma0_reg +#define vt1724_playback_spdif_reg vt1724_pdma4_reg +#define vt1724_capture_pro_reg vt1724_rdma0_reg +#define vt1724_capture_spdif_reg vt1724_rdma1_reg + static const struct snd_pcm_hardware snd_vt1724_playback_pro = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -1084,20 +1103,6 @@ static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) * SPDIF PCM */ -static const struct vt1724_pcm_reg vt1724_playback_spdif_reg = { - .addr = VT1724_MT_PDMA4_ADDR, - .size = VT1724_MT_PDMA4_SIZE, - .count = VT1724_MT_PDMA4_COUNT, - .start = VT1724_PDMA4_START, -}; - -static const struct vt1724_pcm_reg vt1724_capture_spdif_reg = { - .addr = VT1724_MT_RDMA1_ADDR, - .size = VT1724_MT_RDMA1_SIZE, - .count = VT1724_MT_RDMA1_COUNT, - .start = VT1724_RDMA1_START, -}; - /* update spdif control bits; call with reg_lock */ static void update_spdif_bits(struct snd_ice1712 *ice, unsigned int val) { -- cgit v1.1 From d82b64f4764755d765038cf95b1dbe7db039592a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 May 2009 17:25:42 +0200 Subject: ALSA: ice1724 - Add PCI postint to reset sequence Add the PCI posting to ensure the reset sequence in snd_vt1724_chip_reset(). Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 8afa043..1f05c59 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2258,8 +2258,10 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) { outb(VT1724_RESET , ICEREG1724(ice, CONTROL)); + inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */ msleep(10); outb(0, ICEREG1724(ice, CONTROL)); + inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */ msleep(10); } -- cgit v1.1 From 2bf864ac963c5126a9c7c3dce0958390fb612270 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 May 2009 17:30:17 +0200 Subject: ALSA: ice1724 - Allow spec driver to create own routing controls Added a new flag, own_routing, to allow spec drivers to create own routing controls. Also, the basic get/put calls are changed to be external for later use by maya44 driver. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.h | 12 ++++++++++-- sound/pci/ice1712/ice1724.c | 33 ++++++++++++++++++++------------- 2 files changed, 30 insertions(+), 15 deletions(-) diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index fdae6de..adc909e 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -335,6 +335,7 @@ struct snd_ice1712 { unsigned int force_rdma1:1; /* VT1720/4 - RDMA1 as non-spdif */ unsigned int midi_output:1; /* VT1720/4: MIDI output triggered */ unsigned int midi_input:1; /* VT1720/4: MIDI input triggered */ + unsigned int own_routing:1; /* VT1720/4: use own routing ctls */ unsigned int num_total_dacs; /* total DACs */ unsigned int num_total_adcs; /* total ADCs */ unsigned int cur_rate; /* current rate */ @@ -458,10 +459,17 @@ static inline int snd_ice1712_gpio_read_bits(struct snd_ice1712 *ice, return snd_ice1712_gpio_read(ice) & mask; } +/* route access functions */ +int snd_ice1724_get_route_val(struct snd_ice1712 *ice, int shift); +int snd_ice1724_put_route_val(struct snd_ice1712 *ice, unsigned int val, + int shift); + int snd_ice1712_spdif_build_controls(struct snd_ice1712 *ice); -int snd_ice1712_akm4xxx_init(struct snd_akm4xxx *ak, const struct snd_akm4xxx *template, - const struct snd_ak4xxx_private *priv, struct snd_ice1712 *ice); +int snd_ice1712_akm4xxx_init(struct snd_akm4xxx *ak, + const struct snd_akm4xxx *template, + const struct snd_ak4xxx_private *priv, + struct snd_ice1712 *ice); void snd_ice1712_akm4xxx_free(struct snd_ice1712 *ice); int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1f05c59..5a735ee 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1975,7 +1975,7 @@ static inline int digital_route_shift(int idx) return idx * 3; } -static int get_route_val(struct snd_ice1712 *ice, int shift) +int snd_ice1724_get_route_val(struct snd_ice1712 *ice, int shift) { unsigned long val; unsigned char eitem; @@ -1994,7 +1994,8 @@ static int get_route_val(struct snd_ice1712 *ice, int shift) return eitem; } -static int put_route_val(struct snd_ice1712 *ice, unsigned int val, int shift) +int snd_ice1724_put_route_val(struct snd_ice1712 *ice, unsigned int val, + int shift) { unsigned int old_val, nval; int change; @@ -2022,7 +2023,7 @@ static int snd_vt1724_pro_route_analog_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); ucontrol->value.enumerated.item[0] = - get_route_val(ice, analog_route_shift(idx)); + snd_ice1724_get_route_val(ice, analog_route_shift(idx)); return 0; } @@ -2031,8 +2032,9 @@ static int snd_vt1724_pro_route_analog_put(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - return put_route_val(ice, ucontrol->value.enumerated.item[0], - analog_route_shift(idx)); + return snd_ice1724_put_route_val(ice, + ucontrol->value.enumerated.item[0], + analog_route_shift(idx)); } static int snd_vt1724_pro_route_spdif_get(struct snd_kcontrol *kcontrol, @@ -2041,7 +2043,7 @@ static int snd_vt1724_pro_route_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); ucontrol->value.enumerated.item[0] = - get_route_val(ice, digital_route_shift(idx)); + snd_ice1724_get_route_val(ice, digital_route_shift(idx)); return 0; } @@ -2050,11 +2052,13 @@ static int snd_vt1724_pro_route_spdif_put(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - return put_route_val(ice, ucontrol->value.enumerated.item[0], - digital_route_shift(idx)); + return snd_ice1724_put_route_val(ice, + ucontrol->value.enumerated.item[0], + digital_route_shift(idx)); } -static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route __devinitdata = { +static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route __devinitdata = +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "H/W Playback Route", .info = snd_vt1724_pro_route_info, @@ -2291,9 +2295,12 @@ static int __devinit snd_vt1724_spdif_build_controls(struct snd_ice1712 *ice) if (snd_BUG_ON(!ice->pcm)) return -EIO; - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_spdif_route, ice)); - if (err < 0) - return err; + if (!ice->own_routing) { + err = snd_ctl_add(ice->card, + snd_ctl_new1(&snd_vt1724_mixer_pro_spdif_route, ice)); + if (err < 0) + return err; + } err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_spdif_switch, ice)); if (err < 0) @@ -2340,7 +2347,7 @@ static int __devinit snd_vt1724_build_controls(struct snd_ice1712 *ice) if (err < 0) return err; - if (ice->num_total_dacs > 0) { + if (!ice->own_routing && ice->num_total_dacs > 0) { struct snd_kcontrol_new tmp = snd_vt1724_mixer_pro_analog_route; tmp.count = ice->num_total_dacs; if (ice->vt1720 && tmp.count > 2) -- cgit v1.1 From 72cbfd45fac627de4bd38c203baaac40cd26477c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 May 2009 17:33:19 +0200 Subject: ALSA: ice1724 - Add ESI Maya44 support Added the support for ESI Maya44 board to ice1724 driver. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 +- Documentation/sound/alsa/README.maya44 | 163 +++++ sound/pci/ice1712/Makefile | 2 +- sound/pci/ice1712/ice1724.c | 3 + sound/pci/ice1712/maya44.c | 779 ++++++++++++++++++++++++ sound/pci/ice1712/maya44.h | 10 + 6 files changed, 958 insertions(+), 2 deletions(-) create mode 100644 Documentation/sound/alsa/README.maya44 create mode 100644 sound/pci/ice1712/maya44.c create mode 100644 sound/pci/ice1712/maya44.h diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..821a994 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -925,6 +925,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Onkyo SE-90PCI * Onkyo SE-200PCI * ESI Juli@ + * ESI Maya44 * Hercules Fortissimo IV * EGO-SYS WaveTerminal 192M @@ -933,7 +934,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, juli, aureon51, aureon71, universe, ap192, k8x800, phase22, phase28, ms300, av710, se200pci, se90pci, - fortissimo4, sn25p, WT192M + fortissimo4, sn25p, WT192M, maya44 This module supports multiple cards and autoprobe. diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44 new file mode 100644 index 0000000..0e41576 --- /dev/null +++ b/Documentation/sound/alsa/README.maya44 @@ -0,0 +1,163 @@ +NOTE: The following is the original document of Rainer's patch that the +current maya44 code based on. Some contents might be obsoleted, but I +keep here as reference -- tiwai + +---------------------------------------------------------------- + +STATE OF DEVELOPMENT: + +This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann. +Development is carried out by Rainer Zimmermann (mail@lightshed.de). + +ESI provided a sample Maya44 card for the development work. + +However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing. + +This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008). + + +The following functions work, as tested by Rainer Zimmermann and Piotr Makowski: + +- playback and capture at all sampling rates +- input/output level +- crossmixing +- line/mic switch +- phantom power switch +- analogue monitor a.k.a bypass + + +The following functions *should* work, but are not fully tested: + +- Channel 3+4 analogue - S/PDIF input switching +- S/PDIF output +- all inputs/outputs on the M/IO/DIO extension card +- internal/external clock selection + + +*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.* + + +Things that do not seem to work: + +- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code). + +- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down. + + +DRIVER DETAILS: + +the following files were added: + +pci/ice1724/maya44.c - Maya44 specific code +pci/ice1724/maya44.h +pci/ice1724/ice1724.patch +pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES) +i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs +include/wm8776.h + + +Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure. +This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately. + + +the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree: + +wtm.h +vt1720_mobo.h +revo.h +prodigy192.h +pontis.h +phase.h +maya44.h +juli.h +aureon.h +amp.h +envy24ht.h +se.h +prodigy_hifi.h + + +*I hope this is the correct way to do things.* + + +SAMPLING RATES: + +The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture. + +As the ICE1724 chip only allows one global sampling rate, this is handled as follows: + +* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels. + +* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices. + +*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality. + + +I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic. + +The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712). + + +SOUND DEVICES: + +PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0): + +hw:0,0 input - stereo, analog input 1+2 +hw:0,0 output - stereo, analog output 1+2 +hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input +hw:0,1 output - stereo, analog output 3+4 (and SPDIF out) + + +NAMING OF MIXER CONTROLS: + +(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software). + + +PCM: (digital) output level for channel 1+2 +PCM 1: same for channel 3+4 + +Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2. + Make sure this is not turned on while any other source is connected to input 1/2. + It might damage the source and/or the maya44 card. + +Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo). + +Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver. +Bypass 1: same for channel 3+4. + +Crossmix: cross-mixer from channels 1+2 to channels 3+4 +Crossmix 1: cross-mixer from channels 3+4 to channels 1+2 + +IEC958 Output: switch for S/PDIF output. + This is not supported by the ESI windows driver. + S/PDIF should output the same signal as channel 3+4. [untested!] + + +Digitial output selectors: + + These switches allow a direct digital routing from the ADCs to the DACs. + Each switch determines where the digital input data to one of the DACs comes from. + They are not supported by the ESI windows driver. + For normal operation, they should all be set to "PCM out". + +H/W: Output source channel 1 +H/W 1: Output source channel 2 +H/W 2: Output source channel 3 +H/W 3: Output source channel 4 + +H/W 4 ... H/W 9: unknown function, left in to enable testing. + Possibly some of these control S/PDIF output(s). + If these turn out to be unused, they will go away in later driver versions. + +Selectable values for each of the digital output selectors are: + "PCM out" -> DAC output of the corresponding channel (default setting) + "Input 1"... + "Input 4" -> direct routing from ADC output of the selected input channel + + +-------- + +Feb 14, 2008 +Rainer Zimmermann +mail@lightshed.de + diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index f99fe08..536eae2 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -5,7 +5,7 @@ snd-ice17xx-ak4xxx-objs := ak4xxx.o snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o +snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 5a735ee..36ade77 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -49,6 +49,7 @@ #include "prodigy192.h" #include "prodigy_hifi.h" #include "juli.h" +#include "maya44.h" #include "phase.h" #include "wtm.h" #include "se.h" @@ -65,6 +66,7 @@ MODULE_SUPPORTED_DEVICE("{" PRODIGY192_DEVICE_DESC PRODIGY_HIFI_DEVICE_DESC JULI_DEVICE_DESC + MAYA44_DEVICE_DESC PHASE_DEVICE_DESC WTM_DEVICE_DESC SE_DEVICE_DESC @@ -2125,6 +2127,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_prodigy_hifi_cards, snd_vt1724_prodigy192_cards, snd_vt1724_juli_cards, + snd_vt1724_maya44_cards, snd_vt1724_phase_cards, snd_vt1724_wtm_cards, snd_vt1724_se_cards, diff --git a/sound/pci/ice1712/maya44.c b/sound/pci/ice1712/maya44.c new file mode 100644 index 0000000..3e1c20a --- /dev/null +++ b/sound/pci/ice1712/maya44.c @@ -0,0 +1,779 @@ +/* + * ALSA driver for ICEnsemble VT1724 (Envy24HT) + * + * Lowlevel functions for ESI Maya44 cards + * + * Copyright (c) 2009 Takashi Iwai + * Based on the patches by Rainer Zimmermann + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "ice1712.h" +#include "envy24ht.h" +#include "maya44.h" + +/* WM8776 register indexes */ +#define WM8776_REG_HEADPHONE_L 0x00 +#define WM8776_REG_HEADPHONE_R 0x01 +#define WM8776_REG_HEADPHONE_MASTER 0x02 +#define WM8776_REG_DAC_ATTEN_L 0x03 +#define WM8776_REG_DAC_ATTEN_R 0x04 +#define WM8776_REG_DAC_ATTEN_MASTER 0x05 +#define WM8776_REG_DAC_PHASE 0x06 +#define WM8776_REG_DAC_CONTROL 0x07 +#define WM8776_REG_DAC_MUTE 0x08 +#define WM8776_REG_DAC_DEEMPH 0x09 +#define WM8776_REG_DAC_IF_CONTROL 0x0a +#define WM8776_REG_ADC_IF_CONTROL 0x0b +#define WM8776_REG_MASTER_MODE_CONTROL 0x0c +#define WM8776_REG_POWERDOWN 0x0d +#define WM8776_REG_ADC_ATTEN_L 0x0e +#define WM8776_REG_ADC_ATTEN_R 0x0f +#define WM8776_REG_ADC_ALC1 0x10 +#define WM8776_REG_ADC_ALC2 0x11 +#define WM8776_REG_ADC_ALC3 0x12 +#define WM8776_REG_ADC_NOISE_GATE 0x13 +#define WM8776_REG_ADC_LIMITER 0x14 +#define WM8776_REG_ADC_MUX 0x15 +#define WM8776_REG_OUTPUT_MUX 0x16 +#define WM8776_REG_RESET 0x17 + +#define WM8776_NUM_REGS 0x18 + +/* clock ratio identifiers for snd_wm8776_set_rate() */ +#define WM8776_CLOCK_RATIO_128FS 0 +#define WM8776_CLOCK_RATIO_192FS 1 +#define WM8776_CLOCK_RATIO_256FS 2 +#define WM8776_CLOCK_RATIO_384FS 3 +#define WM8776_CLOCK_RATIO_512FS 4 +#define WM8776_CLOCK_RATIO_768FS 5 + +enum { WM_VOL_HP, WM_VOL_DAC, WM_VOL_ADC, WM_NUM_VOLS }; +enum { WM_SW_DAC, WM_SW_BYPASS, WM_NUM_SWITCHES }; + +struct snd_wm8776 { + unsigned char addr; + unsigned short regs[WM8776_NUM_REGS]; + unsigned char volumes[WM_NUM_VOLS][2]; + unsigned int switch_bits; +}; + +struct snd_maya44 { + struct snd_ice1712 *ice; + struct snd_wm8776 wm[2]; + struct mutex mutex; +}; + + +/* write the given register and save the data to the cache */ +static void wm8776_write(struct snd_ice1712 *ice, struct snd_wm8776 *wm, + unsigned char reg, unsigned short val) +{ + /* + * WM8776 registers are up to 9 bits wide, bit 8 is placed in the LSB + * of the address field + */ + snd_vt1724_write_i2c(ice, wm->addr, + (reg << 1) | ((val >> 8) & 1), + val & 0xff); + wm->regs[reg] = val; +} + +/* + * update the given register with and/or mask and save the data to the cache + */ +static int wm8776_write_bits(struct snd_ice1712 *ice, struct snd_wm8776 *wm, + unsigned char reg, + unsigned short mask, unsigned short val) +{ + val |= wm->regs[reg] & ~mask; + if (val != wm->regs[reg]) { + wm8776_write(ice, wm, reg, val); + return 1; + } + return 0; +} + + +/* + * WM8776 volume controls + */ + +struct maya_vol_info { + unsigned int maxval; /* volume range: 0..maxval */ + unsigned char regs[2]; /* left and right registers */ + unsigned short mask; /* value mask */ + unsigned short offset; /* zero-value offset */ + unsigned short mute; /* mute bit */ + unsigned short update; /* update bits */ + unsigned char mux_bits[2]; /* extra bits for ADC mute */ +}; + +static struct maya_vol_info vol_info[WM_NUM_VOLS] = { + [WM_VOL_HP] = { + .maxval = 80, + .regs = { WM8776_REG_HEADPHONE_L, WM8776_REG_HEADPHONE_R }, + .mask = 0x7f, + .offset = 0x30, + .mute = 0x00, + .update = 0x180, /* update and zero-cross enable */ + }, + [WM_VOL_DAC] = { + .maxval = 255, + .regs = { WM8776_REG_DAC_ATTEN_L, WM8776_REG_DAC_ATTEN_R }, + .mask = 0xff, + .offset = 0x01, + .mute = 0x00, + .update = 0x100, /* zero-cross enable */ + }, + [WM_VOL_ADC] = { + .maxval = 91, + .regs = { WM8776_REG_ADC_ATTEN_L, WM8776_REG_ADC_ATTEN_R }, + .mask = 0xff, + .offset = 0xa5, + .mute = 0xa5, + .update = 0x100, /* update */ + .mux_bits = { 0x80, 0x40 }, /* ADCMUX bits */ + }, +}; + +/* + * dB tables + */ +/* headphone output: mute, -73..+6db (1db step) */ +static const DECLARE_TLV_DB_SCALE(db_scale_hp, -7400, 100, 1); +/* DAC output: mute, -127..0db (0.5db step) */ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -12750, 50, 1); +/* ADC gain: mute, -21..+24db (0.5db step) */ +static const DECLARE_TLV_DB_SCALE(db_scale_adc, -2100, 50, 1); + +static int maya_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int idx = kcontrol->private_value; + struct maya_vol_info *vol = &vol_info[idx]; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = vol->maxval; + return 0; +} + +static int maya_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wm8776 *wm = + &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)]; + unsigned int idx = kcontrol->private_value; + + mutex_lock(&chip->mutex); + ucontrol->value.integer.value[0] = wm->volumes[idx][0]; + ucontrol->value.integer.value[1] = wm->volumes[idx][1]; + mutex_unlock(&chip->mutex); + return 0; +} + +static int maya_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wm8776 *wm = + &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)]; + unsigned int idx = kcontrol->private_value; + struct maya_vol_info *vol = &vol_info[idx]; + unsigned int val, data; + int ch, changed = 0; + + mutex_lock(&chip->mutex); + for (ch = 0; ch < 2; ch++) { + val = ucontrol->value.integer.value[ch]; + if (val > vol->maxval) + val = vol->maxval; + if (val == wm->volumes[idx][ch]) + continue; + if (!val) + data = vol->mute; + else + data = (val - 1) + vol->offset; + data |= vol->update; + changed |= wm8776_write_bits(chip->ice, wm, vol->regs[ch], + vol->mask | vol->update, data); + if (vol->mux_bits[ch]) + wm8776_write_bits(chip->ice, wm, WM8776_REG_ADC_MUX, + vol->mux_bits[ch], + val ? 0 : vol->mux_bits[ch]); + wm->volumes[idx][ch] = val; + } + mutex_unlock(&chip->mutex); + return changed; +} + +/* + * WM8776 switch controls + */ + +#define COMPOSE_SW_VAL(idx, reg, mask) ((idx) | ((reg) << 8) | ((mask) << 16)) +#define GET_SW_VAL_IDX(val) ((val) & 0xff) +#define GET_SW_VAL_REG(val) (((val) >> 8) & 0xff) +#define GET_SW_VAL_MASK(val) (((val) >> 16) & 0xff) + +#define maya_sw_info snd_ctl_boolean_mono_info + +static int maya_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wm8776 *wm = + &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)]; + unsigned int idx = GET_SW_VAL_IDX(kcontrol->private_value); + + ucontrol->value.integer.value[0] = (wm->switch_bits >> idx) & 1; + return 0; +} + +static int maya_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wm8776 *wm = + &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)]; + unsigned int idx = GET_SW_VAL_IDX(kcontrol->private_value); + unsigned int mask, val; + int changed; + + mutex_lock(&chip->mutex); + mask = 1 << idx; + wm->switch_bits &= ~mask; + val = ucontrol->value.integer.value[0]; + if (val) + wm->switch_bits |= mask; + mask = GET_SW_VAL_MASK(kcontrol->private_value); + changed = wm8776_write_bits(chip->ice, wm, + GET_SW_VAL_REG(kcontrol->private_value), + mask, val ? mask : 0); + mutex_unlock(&chip->mutex); + return changed; +} + +/* + * GPIO pins (known ones for maya44) + */ +#define GPIO_PHANTOM_OFF 2 +#define GPIO_MIC_RELAY 4 +#define GPIO_SPDIF_IN_INV 5 +#define GPIO_MUST_BE_0 7 + +/* + * GPIO switch controls + */ + +#define COMPOSE_GPIO_VAL(shift, inv) ((shift) | ((inv) << 8)) +#define GET_GPIO_VAL_SHIFT(val) ((val) & 0xff) +#define GET_GPIO_VAL_INV(val) (((val) >> 8) & 1) + +static int maya_set_gpio_bits(struct snd_ice1712 *ice, unsigned int mask, + unsigned int bits) +{ + unsigned int data; + data = snd_ice1712_gpio_read(ice); + if ((data & mask) == bits) + return 0; + snd_ice1712_gpio_write(ice, (data & ~mask) | bits); + return 1; +} + +#define maya_gpio_sw_info snd_ctl_boolean_mono_info + +static int maya_gpio_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + unsigned int shift = GET_GPIO_VAL_SHIFT(kcontrol->private_value); + unsigned int val; + + val = (snd_ice1712_gpio_read(chip->ice) >> shift) & 1; + if (GET_GPIO_VAL_INV(kcontrol->private_value)) + val = !val; + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int maya_gpio_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + unsigned int shift = GET_GPIO_VAL_SHIFT(kcontrol->private_value); + unsigned int val, mask; + int changed; + + mutex_lock(&chip->mutex); + mask = 1 << shift; + val = ucontrol->value.integer.value[0]; + if (GET_GPIO_VAL_INV(kcontrol->private_value)) + val = !val; + val = val ? mask : 0; + changed = maya_set_gpio_bits(chip->ice, mask, val); + mutex_unlock(&chip->mutex); + return changed; +} + +/* + * capture source selection + */ + +/* known working input slots (0-4) */ +#define MAYA_LINE_IN 1 /* in-2 */ +#define MAYA_MIC_IN 4 /* in-5 */ + +static void wm8776_select_input(struct snd_maya44 *chip, int idx, int line) +{ + wm8776_write_bits(chip->ice, &chip->wm[idx], WM8776_REG_ADC_MUX, + 0x1f, 1 << line); +} + +static int maya_rec_src_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { "Line", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = ARRAY_SIZE(texts); + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int maya_rec_src_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + int sel; + + if (snd_ice1712_gpio_read(chip->ice) & (1 << GPIO_MIC_RELAY)) + sel = 1; + else + sel = 0; + ucontrol->value.enumerated.item[0] = sel; + return 0; +} + +static int maya_rec_src_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + int sel = ucontrol->value.enumerated.item[0]; + int changed; + + mutex_lock(&chip->mutex); + changed = maya_set_gpio_bits(chip->ice, GPIO_MIC_RELAY, + sel ? GPIO_MIC_RELAY : 0); + wm8776_select_input(chip, 0, sel ? MAYA_MIC_IN : MAYA_LINE_IN); + mutex_unlock(&chip->mutex); + return changed; +} + +/* + * Maya44 routing switch settings have different meanings than the standard + * ice1724 switches as defined in snd_vt1724_pro_route_info (ice1724.c). + */ +static int maya_pb_route_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { + "PCM Out", /* 0 */ + "Input 1", "Input 2", "Input 3", "Input 4" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = ARRAY_SIZE(texts); + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int maya_pb_route_shift(int idx) +{ + static const unsigned char shift[10] = + { 8, 20, 0, 3, 11, 23, 14, 26, 17, 29 }; + return shift[idx % 10]; +} + +static int maya_pb_route_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + ucontrol->value.enumerated.item[0] = + snd_ice1724_get_route_val(chip->ice, maya_pb_route_shift(idx)); + return 0; +} + +static int maya_pb_route_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol); + int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + return snd_ice1724_put_route_val(chip->ice, + ucontrol->value.enumerated.item[0], + maya_pb_route_shift(idx)); +} + + +/* + * controls to be added + */ + +static struct snd_kcontrol_new maya_controls[] __devinitdata = { + { + .name = "Crossmix Playback Volume", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = maya_vol_info, + .get = maya_vol_get, + .put = maya_vol_put, + .tlv = { .p = db_scale_hp }, + .private_value = WM_VOL_HP, + .count = 2, + }, + { + .name = "PCM Playback Volume", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = maya_vol_info, + .get = maya_vol_get, + .put = maya_vol_put, + .tlv = { .p = db_scale_dac }, + .private_value = WM_VOL_DAC, + .count = 2, + }, + { + .name = "Line Capture Volume", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = maya_vol_info, + .get = maya_vol_get, + .put = maya_vol_put, + .tlv = { .p = db_scale_adc }, + .private_value = WM_VOL_ADC, + .count = 2, + }, + { + .name = "PCM Playback Switch", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = maya_sw_info, + .get = maya_sw_get, + .put = maya_sw_put, + .private_value = COMPOSE_SW_VAL(WM_SW_DAC, + WM8776_REG_OUTPUT_MUX, 0x01), + .count = 2, + }, + { + .name = "Bypass Playback Switch", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = maya_sw_info, + .get = maya_sw_get, + .put = maya_sw_put, + .private_value = COMPOSE_SW_VAL(WM_SW_BYPASS, + WM8776_REG_OUTPUT_MUX, 0x04), + .count = 2, + }, + { + .name = "Capture Source", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = maya_rec_src_info, + .get = maya_rec_src_get, + .put = maya_rec_src_put, + }, + { + .name = "Mic Phantom Power Switch", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = maya_gpio_sw_info, + .get = maya_gpio_sw_get, + .put = maya_gpio_sw_put, + .private_value = COMPOSE_GPIO_VAL(GPIO_PHANTOM_OFF, 1), + }, + { + .name = "SPDIF Capture Switch", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = maya_gpio_sw_info, + .get = maya_gpio_sw_get, + .put = maya_gpio_sw_put, + .private_value = COMPOSE_GPIO_VAL(GPIO_SPDIF_IN_INV, 1), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "H/W Playback Route", + .info = maya_pb_route_info, + .get = maya_pb_route_get, + .put = maya_pb_route_put, + .count = 4, /* FIXME: do controls 5-9 have any meaning? */ + }, +}; + +static int __devinit maya44_add_controls(struct snd_ice1712 *ice) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(maya_controls); i++) { + err = snd_ctl_add(ice->card, snd_ctl_new1(&maya_controls[i], + ice->spec)); + if (err < 0) + return err; + } + return 0; +} + + +/* + * initialize a wm8776 chip + */ +static void __devinit wm8776_init(struct snd_ice1712 *ice, + struct snd_wm8776 *wm, unsigned int addr) +{ + static const unsigned short inits_wm8776[] = { + 0x02, 0x100, /* R2: headphone L+R muted + update */ + 0x05, 0x100, /* R5: DAC output L+R muted + update */ + 0x06, 0x000, /* R6: DAC output phase normal */ + 0x07, 0x091, /* R7: DAC enable zero cross detection, + normal output */ + 0x08, 0x000, /* R8: DAC soft mute off */ + 0x09, 0x000, /* R9: no deemph, DAC zero detect disabled */ + 0x0a, 0x022, /* R10: DAC I2C mode, std polarities, 24bit */ + 0x0b, 0x022, /* R11: ADC I2C mode, std polarities, 24bit, + highpass filter enabled */ + 0x0c, 0x042, /* R12: ADC+DAC slave, ADC+DAC 44,1kHz */ + 0x0d, 0x000, /* R13: all power up */ + 0x0e, 0x100, /* R14: ADC left muted, + enable zero cross detection */ + 0x0f, 0x100, /* R15: ADC right muted, + enable zero cross detection */ + /* R16: ALC...*/ + 0x11, 0x000, /* R17: disable ALC */ + /* R18: ALC...*/ + /* R19: noise gate...*/ + 0x15, 0x000, /* R21: ADC input mux init, mute all inputs */ + 0x16, 0x001, /* R22: output mux, select DAC */ + 0xff, 0xff + }; + + const unsigned short *ptr; + unsigned char reg; + unsigned short data; + + wm->addr = addr; + /* enable DAC output; mute bypass, aux & all inputs */ + wm->switch_bits = (1 << WM_SW_DAC); + + ptr = inits_wm8776; + while (*ptr != 0xff) { + reg = *ptr++; + data = *ptr++; + wm8776_write(ice, wm, reg, data); + } +} + + +/* + * change the rate on the WM8776 codecs. + * this assumes that the VT17xx's rate is changed by the calling function. + * NOTE: even though the WM8776's are running in slave mode and rate + * selection is automatic, we need to call snd_wm8776_set_rate() here + * to make sure some flags are set correctly. + */ +static void set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + struct snd_maya44 *chip = ice->spec; + unsigned int ratio, adc_ratio, val; + int i; + + switch (rate) { + case 192000: + ratio = WM8776_CLOCK_RATIO_128FS; + break; + case 176400: + ratio = WM8776_CLOCK_RATIO_128FS; + break; + case 96000: + ratio = WM8776_CLOCK_RATIO_256FS; + break; + case 88200: + ratio = WM8776_CLOCK_RATIO_384FS; + break; + case 48000: + ratio = WM8776_CLOCK_RATIO_512FS; + break; + case 44100: + ratio = WM8776_CLOCK_RATIO_512FS; + break; + case 32000: + ratio = WM8776_CLOCK_RATIO_768FS; + break; + case 0: + /* no hint - S/PDIF input is master, simply return */ + return; + default: + snd_BUG(); + return; + } + + /* + * this currently sets the same rate for ADC and DAC, but limits + * ADC rate to 256X (96kHz). For 256X mode (96kHz), this sets ADC + * oversampling to 64x, as recommended by WM8776 datasheet. + * Setting the rate is not really necessary in slave mode. + */ + adc_ratio = ratio; + if (adc_ratio < WM8776_CLOCK_RATIO_256FS) + adc_ratio = WM8776_CLOCK_RATIO_256FS; + + val = adc_ratio; + if (adc_ratio == WM8776_CLOCK_RATIO_256FS) + val |= 8; + val |= ratio << 4; + + mutex_lock(&chip->mutex); + for (i = 0; i < 2; i++) + wm8776_write_bits(ice, &chip->wm[i], + WM8776_REG_MASTER_MODE_CONTROL, + 0x180, val); + mutex_unlock(&chip->mutex); +} + +/* + * supported sample rates (to override the default one) + */ + +static unsigned int rates[] = { + 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000 +}; + +/* playback rates: 32..192 kHz */ +static struct snd_pcm_hw_constraint_list dac_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0 +}; + + +/* + * chip addresses on I2C bus + */ +static unsigned char wm8776_addr[2] __devinitdata = { + 0x34, 0x36, /* codec 0 & 1 */ +}; + +/* + * initialize the chip + */ +static int __devinit maya44_init(struct snd_ice1712 *ice) +{ + int i; + struct snd_maya44 *chip; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + mutex_init(&chip->mutex); + chip->ice = ice; + ice->spec = chip; + + /* initialise codecs */ + ice->num_total_dacs = 4; + ice->num_total_adcs = 4; + ice->akm_codecs = 0; + + for (i = 0; i < 2; i++) { + wm8776_init(ice, &chip->wm[i], wm8776_addr[i]); + wm8776_select_input(chip, i, MAYA_LINE_IN); + } + + /* set card specific rates */ + ice->hw_rates = &dac_rates; + + /* register change rate notifier */ + ice->gpio.set_pro_rate = set_rate; + + /* RDMA1 (2nd input channel) is used for ADC by default */ + ice->force_rdma1 = 1; + + /* have an own routing control */ + ice->own_routing = 1; + + return 0; +} + + +/* + * Maya44 boards don't provide the EEPROM data except for the vendor IDs. + * hence the driver needs to sets up it properly. + */ + +static unsigned char maya44_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x45, + /* clock xin1=49.152MHz, mpu401, 2 stereo ADCs+DACs */ + [ICE_EEP2_ACLINK] = 0x80, + /* I2S */ + [ICE_EEP2_I2S] = 0xf8, + /* vol, 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc3, + /* enable spdif out, spdif out supp, spdif-in, ext spdif out */ + [ICE_EEP2_GPIO_DIR] = 0xff, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0xff, + [ICE_EEP2_GPIO_MASK] = 0/*0x9f*/, + [ICE_EEP2_GPIO_MASK1] = 0/*0xff*/, + [ICE_EEP2_GPIO_MASK2] = 0/*0x7f*/, + [ICE_EEP2_GPIO_STATE] = (1 << GPIO_PHANTOM_OFF) | + (1 << GPIO_SPDIF_IN_INV), + [ICE_EEP2_GPIO_STATE1] = 0x00, + [ICE_EEP2_GPIO_STATE2] = 0x00, +}; + +/* entry point */ +struct snd_ice1712_card_info snd_vt1724_maya44_cards[] __devinitdata = { + { + .subvendor = VT1724_SUBDEVICE_MAYA44, + .name = "ESI Maya44", + .model = "maya44", + .chip_init = maya44_init, + .build_controls = maya44_add_controls, + .eeprom_size = sizeof(maya44_eeprom), + .eeprom_data = maya44_eeprom, + }, + { } /* terminator */ +}; diff --git a/sound/pci/ice1712/maya44.h b/sound/pci/ice1712/maya44.h new file mode 100644 index 0000000..eafd03a --- /dev/null +++ b/sound/pci/ice1712/maya44.h @@ -0,0 +1,10 @@ +#ifndef __SOUND_MAYA44_H +#define __SOUND_MAYA44_H + +#define MAYA44_DEVICE_DESC "{ESI,Maya44}," + +#define VT1724_SUBDEVICE_MAYA44 0x34315441 /* Maya44 */ + +extern struct snd_ice1712_card_info snd_vt1724_maya44_cards[]; + +#endif /* __SOUND_MAYA44_H */ -- cgit v1.1 From 5e7c03442574ed0376c0621bfb0c477d79c12c71 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 6 May 2009 01:26:01 +0200 Subject: ASoC: cs4270: add power management support Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 58 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 57 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 153124b..a32b822 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -18,7 +18,7 @@ * - The machine driver's 'startup' function must call * cs4270_set_dai_sysclk() with the value of MCLK. * - Only I2S and left-justified modes are supported - * - Power management is not supported + * - Power management is supported */ #include @@ -27,6 +27,7 @@ #include #include #include +#include #include "cs4270.h" @@ -65,6 +66,8 @@ #define CS4270_PWRCTL_PDN_ADC 0x20 #define CS4270_PWRCTL_PDN_DAC 0x02 #define CS4270_PWRCTL_PDN 0x01 +#define CS4270_PWRCTL_PDN_ALL \ + (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN) #define CS4270_MODE_SPEED_MASK 0x30 #define CS4270_MODE_1X 0x00 #define CS4270_MODE_2X 0x10 @@ -788,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +#ifdef CONFIG_PM + +/* This suspend/resume implementation can handle both - a simple standby + * where the codec remains powered, and a full suspend, where the voltage + * domain the codec is connected to is teared down and/or any other hardware + * reset condition is asserted. + * + * The codec's own power saving features are enabled in the suspend callback, + * and all registers are written back to the hardware when resuming. + */ + +static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_i2c_resume(struct i2c_client *client) +{ + struct cs4270_private *cs4270 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = &cs4270->codec; + int reg; + + /* In case the device was put to hard reset during sleep, we need to + * wait 500ns here before any I2C communication. */ + ndelay(500); + + /* first restore the entire register cache ... */ + for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { + u8 val = snd_soc_read(codec, reg); + + if (i2c_smbus_write_byte_data(client, reg, val)) { + dev_err(codec->dev, "i2c write failed\n"); + return -EIO; + } + } + + /* ... then disable the power-down bits */ + reg = snd_soc_read(codec, CS4270_PWRCTL); + reg &= ~CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} +#else +#define cs4270_i2c_suspend NULL +#define cs4270_i2c_resume NULL +#endif /* CONFIG_PM */ + /* * cs4270_i2c_driver - I2C device identification * @@ -802,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, + .suspend = cs4270_i2c_suspend, + .resume = cs4270_i2c_resume, }; /* -- cgit v1.1 From 41d5545d23d2ccf4d34725094dccebd37f15c1c4 Mon Sep 17 00:00:00 2001 From: Kacper Szczesniak Date: Thu, 7 May 2009 12:47:43 +0200 Subject: ALSA: hda - Add support for MacBook 5.1 (Aluminium) Signed-off-by: Kacper Szczesniak Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 73 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3e7207b..b949534 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -205,6 +205,7 @@ enum { ALC882_ASUS_A7M, ALC885_MACPRO, ALC885_MBP3, + ALC885_MB5, ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, @@ -6164,6 +6165,16 @@ static struct hda_input_mux alc882_capture_source = { { "CD", 0x4 }, }, }; + +static struct hda_input_mux mb5_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + /* * 2ch mode */ @@ -6293,6 +6304,20 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), { } /* end */ }; + +static struct snd_kcontrol_new alc885_mb5_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT), + { } /* end */ +}; static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6520,6 +6545,38 @@ static struct hda_verb alc882_macpro_init_verbs[] = { { } }; +/* Macbook 5,1 */ +static struct hda_verb alc885_mb5_init_verbs[] = { + /* Front mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LineOut mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin: output 0 (0x0d) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* HP Pin: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Front Mic pin: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -6864,6 +6921,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7J] = "asus-a7j", [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", + [ALC885_MB5] = "mb5", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", @@ -6944,6 +7002,18 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mbp3_unsol_event, .init_hook = alc885_mbp3_automute, }, + [ALC885_MB5] = { + .mixers = { alc885_mb5_mixer }, + .init_verbs = { alc885_mb5_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mbp_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .input_mux = &mb5_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, @@ -7249,6 +7319,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; + case 0x106b3f00: /* Macbook 5,1 */ + board_config = ALC885_MB5; + break; default: /* ALC889A is handled better as ALC888-compatible */ if (codec->revision_id == 0x100101 || -- cgit v1.1 From 9da29271bea5d831d745f3ceb7f6f6b2def13a5b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 May 2009 16:31:14 +0200 Subject: ALSA: hda - Fix secondary SPDIF on VT1708S and VT1702 codecs VIA VT1708S and VT1702 codecs can have two SPDIF outputs. One of them should have been handled as the extra digital out, but it's not properly accessed. This patch fixes the handling of the secondary SPDIF on these codecs with the slave dig-out as found in patch_sigmatel.c. This makes the use of such a device easier (for normal users). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 111 +++++++++++++++++++++------------------------- 1 file changed, 51 insertions(+), 60 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b25a5cc..8e004fb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -205,7 +205,7 @@ struct via_spec { /* playback */ struct hda_multi_out multiout; - hda_nid_t extra_dig_out_nid; + hda_nid_t slave_dig_outs[2]; /* capture */ unsigned int num_adc_nids; @@ -731,21 +731,6 @@ static int via_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_dig_close(codec, &spec->multiout); } -/* setup SPDIF output stream */ -static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid, - unsigned int stream_tag, unsigned int format) -{ - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_ctls & AC_DIG1_ENABLE) - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); - /* turn on again (if needed) */ - if (codec->spdif_ctls & AC_DIG1_ENABLE) - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); -} - static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -753,19 +738,16 @@ static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - hda_nid_t nid; - - /* 1st or 2nd S/PDIF */ - if (substream->number == 0) - nid = spec->multiout.dig_out_nid; - else if (substream->number == 1) - nid = spec->extra_dig_out_nid; - else - return -1; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} - mutex_lock(&codec->spdif_mutex); - setup_dig_playback_stream(codec, nid, stream_tag, format); - mutex_unlock(&codec->spdif_mutex); +static int via_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct via_spec *spec = codec->spec; + snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); return 0; } @@ -842,7 +824,8 @@ static struct hda_pcm_stream vt1708_pcm_digital_playback = { .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -874,13 +857,6 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; spec->multiout.share_spdif = 1; - - if (spec->extra_dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->extra_dig_out_nid); - if (err < 0) - return err; - } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -1013,10 +989,6 @@ static void via_unsol_event(struct hda_codec *codec, via_gpio_control(codec); } -static hda_nid_t slave_dig_outs[] = { - 0, -}; - static int via_init(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1051,8 +1023,9 @@ static int via_init(struct hda_codec *codec) snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); - /* no slave outs */ - codec->slave_dig_outs = slave_dig_outs; + /* assign slave outs */ + if (spec->slave_dig_outs[0]) + codec->slave_dig_outs = spec->slave_dig_outs; return 0; } @@ -2134,7 +2107,8 @@ static struct hda_pcm_stream vt1708B_pcm_digital_playback = { .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -2589,14 +2563,15 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = { }; static struct hda_pcm_stream vt1708S_pcm_digital_playback = { - .substreams = 2, + .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in via_build_pcms */ .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -2805,14 +2780,37 @@ static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, return 0; } +/* fill out digital output widgets; one for master and one for slave outputs */ +static void fill_dig_outs(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->autocfg.dig_outs; i++) { + hda_nid_t nid; + int conn; + + nid = spec->autocfg.dig_out_pins[i]; + if (!nid) + continue; + conn = snd_hda_get_connections(codec, nid, &nid, 1); + if (conn < 1) + continue; + if (!spec->multiout.dig_out_nid) + spec->multiout.dig_out_nid = nid; + else { + spec->slave_dig_outs[0] = nid; + break; /* at most two dig outs */ + } + } +} + static int vt1708S_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int err; - static hda_nid_t vt1708s_ignore[] = {0x21, 0}; - err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, - vt1708s_ignore); + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); if (err < 0) return err; err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg); @@ -2833,10 +2831,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; - - spec->extra_dig_out_nid = 0x15; + fill_dig_outs(codec); if (spec->kctls.list) spec->mixers[spec->num_mixers++] = spec->kctls.list; @@ -3000,7 +2995,8 @@ static struct hda_pcm_stream vt1702_pcm_digital_playback = { .ops = { .open = via_dig_playback_pcm_open, .close = via_dig_playback_pcm_close, - .prepare = via_dig_playback_pcm_prepare + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup }, }; @@ -3128,10 +3124,8 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int err; - static hda_nid_t vt1702_ignore[] = {0x1C, 0}; - err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, - vt1702_ignore); + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); if (err < 0) return err; err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg); @@ -3152,10 +3146,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = VT1702_DIGOUT_NID; - - spec->extra_dig_out_nid = 0x1B; + fill_dig_outs(codec); if (spec->kctls.list) spec->mixers[spec->num_mixers++] = spec->kctls.list; -- cgit v1.1 From c198d811812417961582d4e25360372ca1eccdae Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 May 2009 14:32:00 +0300 Subject: ASoC: TWL4030: Fix typo in twl4030_codec_mute function Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN It has not caused problems, since TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1a00e4b..fd392c6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -237,7 +237,7 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) TWL4030_REG_PRECKL_CTL); reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL); twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PRECKL_GAIN), + reg_val & (~TWL4030_PRECKR_GAIN), TWL4030_REG_PRECKR_CTL); /* Disable PLL */ -- cgit v1.1 From bec4c99e8637b5b8bd4b0513eacb51da25885e3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 May 2009 10:36:34 +0100 Subject: ASoC: Fix file patterns for PXA sound drivers The file matches for PXA sound drivers missed the generic AC97 support and were overly specific within sound/soc/pxa, omitting all machine drivers and the SSP driver. Signed-off-by: Mark Brown --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 3cf4f0d..17c8ec1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4544,7 +4544,8 @@ F: drivers/pcmcia/pxa2xx* F: drivers/spi/pxa2xx* F: drivers/usb/gadget/pxa2* F: include/sound/pxa2xx-lib.h -F: sound/soc/pxa/pxa2xx* +F: sound/arm/pxa* +F: sound/soc/pxa PXA168 SUPPORT P: Eric Miao -- cgit v1.1 From 42171c17f267d7fdc5167ad7b6b5fb9edfd04186 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 May 2009 14:11:43 +0200 Subject: ALSA: hda - Fix and clean up hippo-compat HP auto-muting The speaker auto-muting per HP plugging for ALC262 HIPPO and compatible devices is slightly buggy as the "Master" or "Front" mixer control can still toggle the speaker output even if the headphone is plugged. This patch fixes the issue, and clean up the hippo-related codes together with fixes of some inconsistent mixer names. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 372 ++++++++++++++++++------------------------ 1 file changed, 158 insertions(+), 214 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b949534..8b242c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9534,24 +9534,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_hippo1_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - { } /* end */ -}; - /* update HP, line and mono-out pins according to the master switch */ static void alc262_hp_master_update(struct hda_codec *codec) { @@ -9772,46 +9754,132 @@ static struct hda_input_mux alc262_hp_rp5700_capture_source = { }, }; -/* bind hp and internal speaker mute (with plug check) */ -static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* bind hp and internal speaker mute (with plug check) as master switch */ +static void alc262_hippo_master_update(struct hda_codec *codec) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; + struct alc_spec *spec = codec->spec; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; + hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0]; + unsigned int mute; - /* change hp mute */ - change = snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); - if (change) { - /* change speaker according to HP jack state */ - struct alc_spec *spec = codec->spec; - unsigned int mute; - if (spec->jack_present) - mute = HDA_AMP_MUTE; - else - mute = snd_hda_codec_amp_read(codec, 0x15, 0, - HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + /* HP */ + mute = spec->master_sw ? 0 : HDA_AMP_MUTE; + snd_hda_codec_amp_stereo(codec, hp_nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + /* mute internal speaker per jack sense */ + if (spec->jack_present) + mute = HDA_AMP_MUTE; + if (line_nid) + snd_hda_codec_amp_stereo(codec, line_nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + if (speaker_nid && speaker_nid != line_nid) + snd_hda_codec_amp_stereo(codec, speaker_nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, mute); +} + +#define alc262_hippo_master_sw_get alc262_hp_master_sw_get + +static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int val = !!*ucontrol->value.integer.value; + + if (val == spec->master_sw) + return 0; + spec->master_sw = val; + alc262_hippo_master_update(codec); + return 1; +} + +#define ALC262_HIPPO_MASTER_SWITCH \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Master Playback Switch", \ + .info = snd_ctl_boolean_mono_info, \ + .get = alc262_hippo_master_sw_get, \ + .put = alc262_hippo_master_sw_put, \ } - return change; + +static struct snd_kcontrol_new alc262_hippo_mixer[] = { + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc262_hippo1_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_hippo_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + unsigned int present; + + /* need to execute and sync at first */ + snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, hp_nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + alc262_hippo_master_update(codec); } +static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_hippo_automute(codec); +} + +static void alc262_hippo_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc262_hippo_automute(codec); +} + +static void alc262_hippo1_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc262_hippo_automute(codec); +} + + static struct snd_kcontrol_new alc262_sony_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc262_sony_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - }, + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -9820,8 +9888,8 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { }; static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -10076,69 +10144,6 @@ static void alc262_toshiba_s06_init_hook(struct hda_codec *codec) alc262_dmic_automute(codec); } -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hippo_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - unsigned int mute; - unsigned int present; - - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc262_hippo_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hippo_automute(codec); -} - -static void alc262_hippo1_automute(struct hda_codec *codec) -{ - unsigned int mute; - unsigned int present; - - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc262_hippo1_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hippo1_automute(codec); -} - /* * nec model * 0x15 = headphone @@ -10406,14 +10411,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc262_sony_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), - }, + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -11068,7 +11066,7 @@ static struct alc_config_preset alc262_presets[] = { .input_mux = &alc262_capture_source, }, [ALC262_HIPPO] = { - .mixers = { alc262_base_mixer }, + .mixers = { alc262_hippo_mixer }, .init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs}, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, @@ -11078,7 +11076,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_HIPPO_1] = { .mixers = { alc262_hippo1_mixer }, @@ -11090,8 +11088,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo1_unsol_event, - .init_hook = alc262_hippo1_automute, + .unsol_event = alc262_hippo_unsol_event, + .init_hook = alc262_hippo1_init_hook, }, [ALC262_FUJITSU] = { .mixers = { alc262_fujitsu_mixer }, @@ -11185,7 +11183,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, @@ -11197,7 +11195,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, @@ -11262,7 +11260,7 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_automute, + .init_hook = alc262_hippo_init_hook, }, [ALC262_TYAN] = { .mixers = { alc262_tyan_mixer }, @@ -11419,6 +11417,17 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { { } }; +static struct snd_kcontrol_new alc268_toshiba_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT), + { } +}; + /* bind Beep switches of both NID 0x0f and 0x10 */ static struct hda_bind_ctls alc268_bind_beep_sw = { .ops = &snd_hda_bind_sw, @@ -11442,8 +11451,6 @@ static struct hda_verb alc268_eapd_verbs[] = { }; /* Toshiba specific */ -#define alc268_toshiba_automute alc262_hippo_automute - static struct hda_verb alc268_toshiba_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } /* end */ @@ -11579,13 +11586,8 @@ static struct hda_verb alc268_acer_verbs[] = { }; /* unsolicited event for HP jack sensing */ -static void alc268_toshiba_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc268_toshiba_automute(codec); -} +#define alc268_toshiba_unsol_event alc262_hippo_unsol_event +#define alc268_toshiba_init_hook alc262_hippo_init_hook static void alc268_acer_unsol_event(struct hda_codec *codec, unsigned int res) @@ -12230,7 +12232,7 @@ static struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, + .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, @@ -12244,7 +12246,7 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc268_toshiba_unsol_event, - .init_hook = alc268_toshiba_automute, + .init_hook = alc268_toshiba_init_hook, }, [ALC268_ACER] = { .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, @@ -12327,7 +12329,7 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc268_toshiba_unsol_event, - .init_hook = alc268_toshiba_automute + .init_hook = alc268_toshiba_init_hook }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { @@ -15552,10 +15554,8 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { }; static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -15568,15 +15568,11 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { }; static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT), + ALC262_HIPPO_MASTER_SWITCH, + HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), @@ -16084,51 +16080,25 @@ static void alc662_eeepc_mic_automute(struct hda_codec *codec) static void alc662_eeepc_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc262_hippo1_automute( codec ); - if ((res >> 26) == ALC880_MIC_EVENT) alc662_eeepc_mic_automute(codec); + else + alc262_hippo_unsol_event(codec, res); } static void alc662_eeepc_inithook(struct hda_codec *codec) { - alc262_hippo1_automute( codec ); + alc262_hippo1_init_hook(codec); alc662_eeepc_mic_automute(codec); } -static void alc662_eeepc_ep20_automute(struct hda_codec *codec) -{ - unsigned int mute; - unsigned int present; - - snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc662_eeepc_ep20_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc662_eeepc_ep20_automute(codec); -} - static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) { - alc662_eeepc_ep20_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc262_hippo_master_update(codec); } static void alc663_m51va_speaker_automute(struct hda_codec *codec) @@ -16462,35 +16432,9 @@ static void alc663_g50v_inithook(struct hda_codec *codec) alc662_eeepc_mic_automute(codec); } -/* bind hp and internal speaker mute (with plug check) */ -static int alc662_ecs_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[0] ? 0 : HDA_AMP_MUTE); - change |= snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - valp[1] ? 0 : HDA_AMP_MUTE); - if (change) - alc262_hippo1_automute(codec); - return change; -} - static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc662_ecs_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - }, + ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -16682,7 +16626,7 @@ static struct alc_config_preset alc662_presets[] = { .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, - .unsol_event = alc662_eeepc_ep20_unsol_event, + .unsol_event = alc662_eeepc_unsol_event, .init_hook = alc662_eeepc_ep20_inithook, }, [ALC662_ECS] = { -- cgit v1.1 From b72519b518211ecb7c4c7b5e660ac6235ca7d1b7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 May 2009 14:31:55 +0200 Subject: ALSA: hda - Clean up for ALC262 HP model auto-mute functions Just clean up, no functional changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 54 +++++++++++++++---------------------------- 1 file changed, 19 insertions(+), 35 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b242c9..7c043b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9589,14 +9589,7 @@ static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec, alc262_hp_wildwest_automute(codec); } -static int alc262_hp_master_sw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - *ucontrol->value.integer.value = spec->master_sw; - return 0; -} +#define alc262_hp_master_sw_get alc260_hp_master_sw_get static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -9612,14 +9605,17 @@ static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } +#define ALC262_HP_MASTER_SWITCH \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Master Playback Switch", \ + .info = snd_ctl_boolean_mono_info, \ + .get = alc262_hp_master_sw_get, \ + .put = alc262_hp_master_sw_put, \ + } + static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = alc262_hp_master_sw_get, - .put = alc262_hp_master_sw_put, - }, + ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -9643,13 +9639,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { }; static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = alc262_hp_master_sw_get, - .put = alc262_hp_master_sw_put, - }, + ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9676,17 +9666,14 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hp_t5735_automute(struct hda_codec *codec, int force) +static void alc262_hp_t5735_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + unsigned int present; - if (force || !spec->sense_updated) { - unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; - spec->sense_updated = 1; - } + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE, spec->jack_present ? HDA_AMP_MUTE : 0); } @@ -9696,13 +9683,10 @@ static void alc262_hp_t5735_unsol_event(struct hda_codec *codec, { if ((res >> 26) != ALC880_HP_EVENT) return; - alc262_hp_t5735_automute(codec, 1); + alc262_hp_t5735_automute(codec); } -static void alc262_hp_t5735_init_hook(struct hda_codec *codec) -{ - alc262_hp_t5735_automute(codec, 1); -} +#define alc262_hp_t5735_init_hook alc262_hp_t5735_automute static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), -- cgit v1.1 From a9fd4f3fcdc7e5757424f7e5888f660cf34bb1a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 May 2009 15:57:59 +0200 Subject: ALSA: hda - Clean up Realtek auto-mute unsol routines Most of unsol handlers defined in patch_realtek.c can be classified to two types, mute via amp of pins and mute via ctl bits of pins. Thus there are a big room to generalize each implementation. This patch creates two generic functions, alc_automute_amp() and alc_automute_pin(). The latter is actually changed from the previous alc_sku_automute(). Each caller needs to initialize hp_pins and speaker_pins properly at own init_hook. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 901 +++++++++++++++--------------------------- 1 file changed, 311 insertions(+), 590 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c043b3..34f6fb7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -926,20 +926,26 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, alc_fix_pll(codec); } -static void alc_sku_automute(struct hda_codec *codec) +static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int present; - unsigned int hp_nid = spec->autocfg.hp_pins[0]; - unsigned int sp_nid = spec->autocfg.speaker_pins[0]; + unsigned int nid = spec->autocfg.hp_pins[0]; + int i; /* need to execute and sync at first */ - snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, hp_nid, 0, + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; - snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->jack_present ? 0 : PIN_OUT); + spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + nid = spec->autocfg.speaker_pins[i]; + if (!nid) + break; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->jack_present ? 0 : PIN_OUT); + } } #if 0 /* it's broken in some acses -- temporarily disabled */ @@ -974,16 +980,19 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 28; else res >>= 26; - if (res == ALC880_HP_EVENT) - alc_sku_automute(codec); - - if (res == ALC880_MIC_EVENT) + switch (res) { + case ALC880_HP_EVENT: + alc_automute_pin(codec); + break; + case ALC880_MIC_EVENT: alc_mic_automute(codec); + break; + } } static void alc_inithook(struct hda_codec *codec) { - alc_sku_automute(codec); + alc_automute_pin(codec); alc_mic_automute(codec); } @@ -1364,32 +1373,58 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { {} }; -static void alc888_fujitsu_xa3530_automute(struct hda_codec *codec) +static void alc_automute_amp(struct hda_codec *codec) { - unsigned int present; - unsigned int bits; - /* Line out presence */ - present = snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - /* HP out presence */ - present = present || snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; + struct alc_spec *spec = codec->spec; + unsigned int val, mute; + hda_nid_t nid; + int i; + + spec->jack_present = 0; + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + nid = spec->autocfg.hp_pins[i]; + if (!nid) + break; + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + if (val & AC_PINSENSE_PRESENCE) { + spec->jack_present = 1; + break; + } + } + + mute = spec->jack_present ? HDA_AMP_MUTE : 0; /* Toggle internal speakers muting */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - /* Toggle internal bass muting */ - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + nid = spec->autocfg.speaker_pins[i]; + if (!nid) + break; + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } } -static void alc888_fujitsu_xa3530_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc_automute_amp_unsol_event(struct hda_codec *codec, + unsigned int res) { - if (res >> 26 == ALC880_HP_EVENT) - alc888_fujitsu_xa3530_automute(codec); + if (codec->vendor_id == 0x10ec0880) + res >>= 28; + else + res >>= 26; + if (res == ALC880_HP_EVENT) + alc_automute_amp(codec); } +static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x17; /* line-out */ + spec->autocfg.hp_pins[1] = 0x1b; /* hp */ + spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ + spec->autocfg.speaker_pins[1] = 0x15; /* bass */ + alc_automute_amp(codec); +} /* * ALC888 Acer Aspire 4930G model @@ -1456,22 +1491,13 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; -static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) +static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned int bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if (res >> 26 == ALC880_HP_EVENT) - alc888_acer_aspire_4930g_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } /* @@ -2439,21 +2465,6 @@ static struct hda_verb alc880_beep_init_verbs[] = { { } }; -/* toggle speaker-output according to the hp-jack state */ -static void alc880_uniwill_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - /* auto-toggle front mic */ static void alc880_uniwill_mic_automute(struct hda_codec *codec) { @@ -2466,9 +2477,14 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec) snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } -static void alc880_uniwill_automute(struct hda_codec *codec) +static void alc880_uniwill_init_hook(struct hda_codec *codec) { - alc880_uniwill_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x16; + alc_automute_amp(codec); alc880_uniwill_mic_automute(codec); } @@ -2479,24 +2495,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, * definition. 4bit tag is placed at 28 bit! */ switch (res >> 28) { - case ALC880_HP_EVENT: - alc880_uniwill_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc880_uniwill_mic_automute(codec); break; + default: + alc_automute_amp_unsol_event(codec, res); + break; } } -static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) +static void alc880_uniwill_p53_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + struct alc_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -2518,10 +2532,10 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - if ((res >> 28) == ALC880_HP_EVENT) - alc880_uniwill_p53_hp_automute(codec); if ((res >> 28) == ALC880_DCVOL_EVENT) alc880_uniwill_p53_dcvol_automute(codec); + else + alc_automute_amp_unsol_event(codec, res); } /* @@ -2753,30 +2767,18 @@ static struct hda_verb alc880_lg_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* jack sense */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } }; /* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_automute(struct hda_codec *codec) +static void alc880_lg_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - if ((res >> 28) == 0x01) - alc880_lg_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x17; + alc_automute_amp(codec); } /* @@ -2850,30 +2852,18 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* jack sense */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } }; /* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_lw_automute(struct hda_codec *codec) +static void alc880_lg_lw_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - if ((res >> 28) == 0x01) - alc880_lg_lw_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -2920,16 +2910,10 @@ static struct hda_verb alc880_medion_rim_init_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc880_medion_rim_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - if (present) + struct alc_spec *spec = codec->spec; + alc_automute_amp(codec); + /* toggle EAPD */ + if (spec->jack_present) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); else snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); @@ -2945,6 +2929,15 @@ static void alc880_medion_rim_unsol_event(struct hda_codec *codec, alc880_medion_rim_automute(codec); } +static void alc880_medion_rim_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc880_medion_rim_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE static struct hda_amp_list alc880_loopbacks[] = { { 0x0b, HDA_INPUT, 0 }, @@ -3803,7 +3796,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_f1734_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_hp_automute, + .init_hook = alc880_uniwill_p53_init_hook, }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, @@ -3880,7 +3873,7 @@ static struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_unsol_event, - .init_hook = alc880_uniwill_automute, + .init_hook = alc880_uniwill_init_hook, }, [ALC880_UNIWILL_P53] = { .mixers = { alc880_uniwill_p53_mixer }, @@ -3892,7 +3885,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_hp_automute, + .init_hook = alc880_uniwill_p53_init_hook, }, [ALC880_FUJITSU] = { .mixers = { alc880_fujitsu_mixer }, @@ -3906,7 +3899,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_hp_automute, + .init_hook = alc880_uniwill_p53_init_hook, }, [ALC880_CLEVO] = { .mixers = { alc880_three_stack_mixer }, @@ -3931,8 +3924,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_ch_modes, .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, - .unsol_event = alc880_lg_unsol_event, - .init_hook = alc880_lg_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc880_lg_init_hook, #ifdef CONFIG_SND_HDA_POWER_SAVE .loopbacks = alc880_lg_loopbacks, #endif @@ -3947,8 +3940,8 @@ static struct alc_config_preset alc880_presets[] = { .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes), .channel_mode = alc880_lg_lw_modes, .input_mux = &alc880_lg_lw_capture_source, - .unsol_event = alc880_lg_lw_unsol_event, - .init_hook = alc880_lg_lw_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc880_lg_lw_init_hook, }, [ALC880_MEDION_RIM] = { .mixers = { alc880_medion_rim_mixer }, @@ -3962,7 +3955,7 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_medion_rim_capture_source, .unsol_event = alc880_medion_rim_unsol_event, - .init_hook = alc880_medion_rim_automute, + .init_hook = alc880_medion_rim_init_hook, }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { @@ -6666,45 +6659,23 @@ static struct hda_verb alc885_imac24_init_verbs[] = { }; /* Toggle speaker-output according to the hp-jack state */ -static void alc885_imac24_automute(struct hda_codec *codec) +static void alc885_imac24_automute_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -/* Processes unsolicited events. */ -static void alc885_imac24_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_imac24_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + spec->autocfg.speaker_pins[1] = 0x1a; + alc_automute_amp(codec); } -static void alc885_mbp3_automute(struct hda_codec *codec) +static void alc885_mbp3_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE); + struct alc_spec *spec = codec->spec; -} -static void alc885_mbp3_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Headphone insertion or removal. */ - if ((res >> 26) == ALC880_HP_EVENT) - alc885_mbp3_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } @@ -6729,24 +6700,25 @@ static struct hda_verb alc882_targa_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc882_targa_automute(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + struct alc_spec *spec = codec->spec; + alc_automute_amp(codec); snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - present ? 1 : 3); + spec->jack_present ? 1 : 3); +} + +static void alc882_targa_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc882_targa_automute(codec); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) { - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 26 bit! - */ - if (((res >> 26) == ALC880_HP_EVENT)) { + if ((res >> 26) == ALC880_HP_EVENT) alc882_targa_automute(codec); - } } static struct hda_verb alc882_asus_a7j_verbs[] = { @@ -6828,7 +6800,7 @@ static void alc885_macpro_init_hook(struct hda_codec *codec) static void alc885_imac24_init_hook(struct hda_codec *codec) { alc885_macpro_init_hook(codec); - alc885_imac24_automute(codec); + alc885_imac24_automute_init_hook(codec); } /* @@ -6999,8 +6971,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc885_mbp3_unsol_event, - .init_hook = alc885_mbp3_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc885_mbp3_init_hook, }, [ALC885_MB5] = { .mixers = { alc885_mb5_mixer }, @@ -7036,7 +7008,7 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, - .unsol_event = alc885_imac24_unsol_event, + .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc885_imac24_init_hook, }, [ALC882_TARGA] = { @@ -7053,7 +7025,7 @@ static struct alc_config_preset alc882_presets[] = { .need_dac_fix = 1, .input_mux = &alc882_capture_source, .unsol_event = alc882_targa_unsol_event, - .init_hook = alc882_targa_automute, + .init_hook = alc882_targa_init_hook, }, [ALC882_ASUS_A7J] = { .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, @@ -7903,8 +7875,6 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), @@ -8053,16 +8023,14 @@ static struct hda_verb alc883_init_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc883_mitac_hp_automute(struct hda_codec *codec) +static void alc883_mitac_init_hook(struct hda_codec *codec) { - unsigned int present; + struct alc_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x17; + alc_automute_amp(codec); } /* auto-toggle front mic */ @@ -8079,25 +8047,6 @@ static void alc883_mitac_mic_automute(struct hda_codec *codec) } */ -static void alc883_mitac_automute(struct hda_codec *codec) -{ - alc883_mitac_hp_automute(codec); - /* alc883_mitac_mic_automute(codec); */ -} - -static void alc883_mitac_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_mitac_hp_automute(codec); - break; - case ALC880_MIC_EVENT: - /* alc883_mitac_mic_automute(codec); */ - break; - } -} - static struct hda_verb alc883_mitac_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -8215,29 +8164,15 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; -static void alc888_3st_hp_front_automute(struct hda_codec *codec) +static void alc888_3st_hp_init_hook(struct hda_codec *codec) { - unsigned int present, bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc888_3st_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc888_3st_hp_front_automute(codec); - break; - } + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x18; + alc_automute_amp(codec); } static struct hda_verb alc888_3st_hp_verbs[] = { @@ -8334,56 +8269,18 @@ static struct hda_verb alc883_medion_md2_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc883_medion_md2_automute(struct hda_codec *codec) +static void alc883_medion_md2_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -static void alc883_medion_md2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_medion_md2_automute(codec); -} - -/* toggle speaker-output according to the hp-jack state */ -static void alc883_tagra_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - present ? 1 : 3); -} + struct alc_spec *spec = codec->spec; -static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_tagra_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } /* toggle speaker-output according to the hp-jack state */ -static void alc883_clevo_m720_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} +#define alc883_tagra_init_hook alc882_targa_init_hook +#define alc883_tagra_unsol_event alc882_targa_unsol_event static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { @@ -8395,9 +8292,13 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } -static void alc883_clevo_m720_automute(struct hda_codec *codec) +static void alc883_clevo_m720_init_hook(struct hda_codec *codec) { - alc883_clevo_m720_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); alc883_clevo_m720_mic_automute(codec); } @@ -8405,52 +8306,32 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_clevo_m720_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc883_clevo_m720_mic_automute(codec); break; + default: + alc_automute_amp_unsol_event(codec, res); + break; } } /* toggle speaker-output according to the hp-jack state */ -static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec) +static void alc883_2ch_fujitsu_pi2515_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_2ch_fujitsu_pi2515_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } -static void alc883_haier_w66_automute(struct hda_codec *codec) +static void alc883_haier_w66_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0x80 : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - 0x80, bits); -} + struct alc_spec *spec = codec->spec; -static void alc883_haier_w66_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_haier_w66_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) @@ -8489,23 +8370,14 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, } /* toggle speaker-output according to the hp-jack state */ -static void alc883_acer_aspire_automute(struct hda_codec *codec) +static void alc883_acer_aspire_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc883_acer_aspire_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_acer_aspire_automute(codec); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->autocfg.speaker_pins[1] = 0x16; + alc_automute_amp(codec); } static struct hda_verb alc883_acer_eapd_verbs[] = { @@ -8526,75 +8398,29 @@ static struct hda_verb alc883_acer_eapd_verbs[] = { { } }; -static void alc888_6st_dell_front_automute(struct hda_codec *codec) +static void alc888_6st_dell_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc888_6st_dell_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - /* printk(KERN_DEBUG "hp_event\n"); */ - alc888_6st_dell_front_automute(codec); - break; - } + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.speaker_pins[2] = 0x16; + spec->autocfg.speaker_pins[3] = 0x17; + alc_automute_amp(codec); } -static void alc888_lenovo_sky_front_automute(struct hda_codec *codec) +static void alc888_lenovo_sky_init_hook(struct hda_codec *codec) { - unsigned int mute; - unsigned int present; - - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} + struct alc_spec *spec = codec->spec; -static void alc883_lenovo_sky_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc888_lenovo_sky_front_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.speaker_pins[2] = 0x16; + spec->autocfg.speaker_pins[3] = 0x17; + spec->autocfg.speaker_pins[4] = 0x1a; + alc_automute_amp(codec); } /* @@ -8682,39 +8508,33 @@ static void alc883_nb_mic_automute(struct hda_codec *codec) 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); } -static void alc883_M90V_speaker_automute(struct hda_codec *codec) +static void alc883_M90V_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + struct alc_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); - snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.speaker_pins[2] = 0x16; + alc_automute_pin(codec); } static void alc883_mode2_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_M90V_speaker_automute(codec); - break; case ALC880_MIC_EVENT: alc883_nb_mic_automute(codec); break; + default: + alc_sku_unsol_event(codec, res); + break; } } static void alc883_mode2_inithook(struct hda_codec *codec) { - alc883_M90V_speaker_automute(codec); + alc883_M90V_init_hook(codec); alc883_nb_mic_automute(codec); } @@ -8731,32 +8551,13 @@ static struct hda_verb alc888_asus_eee1601_verbs[] = { { } /* end */ }; -static void alc883_eee1601_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); -} - -static void alc883_eee1601_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc883_eee1601_speaker_automute(codec); - break; - } -} - static void alc883_eee1601_inithook(struct hda_codec *codec) { - alc883_eee1601_speaker_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x1b; + alc_automute_pin(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -8969,7 +8770,7 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_automute, + .init_hook = alc883_tagra_init_hook, }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_tagra_2ch_mixer}, @@ -8983,7 +8784,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_automute, + .init_hook = alc883_tagra_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, @@ -9008,8 +8809,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_acer_aspire_unsol_event, - .init_hook = alc883_acer_aspire_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_acer_aspire_init_hook, }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_base_mixer, @@ -9028,8 +8829,8 @@ static struct alc_config_preset alc883_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, - .unsol_event = alc888_acer_aspire_4930g_unsol_event, - .init_hook = alc888_acer_aspire_4930g_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_4930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, @@ -9053,8 +8854,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_medion_md2_unsol_event, - .init_hook = alc883_medion_md2_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_medion_md2_init_hook, }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, @@ -9075,7 +8876,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_clevo_m720_unsol_event, - .init_hook = alc883_clevo_m720_automute, + .init_hook = alc883_clevo_m720_init_hook, }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, @@ -9099,8 +8900,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, .input_mux = &alc883_lenovo_nb0763_capture_source, - .unsol_event = alc883_medion_md2_unsol_event, - .init_hook = alc883_medion_md2_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_medion_md2_init_hook, }, [ALC888_LENOVO_MS7195_DIG] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9124,8 +8925,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_haier_w66_unsol_event, - .init_hook = alc883_haier_w66_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_haier_w66_init_hook, }, [ALC888_3ST_HP] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9136,8 +8937,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc888_3st_hp_unsol_event, - .init_hook = alc888_3st_hp_front_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_3st_hp_init_hook, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -9149,8 +8950,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc888_6st_dell_unsol_event, - .init_hook = alc888_6st_dell_front_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_6st_dell_init_hook, }, [ALC883_MITAC] = { .mixers = { alc883_mitac_mixer }, @@ -9160,8 +8961,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc883_mitac_unsol_event, - .init_hook = alc883_mitac_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_mitac_init_hook, }, [ALC883_FUJITSU_PI2515] = { .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, @@ -9173,8 +8974,8 @@ static struct alc_config_preset alc883_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_fujitsu_pi2515_capture_source, - .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, - .init_hook = alc883_2ch_fujitsu_pi2515_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_2ch_fujitsu_pi2515_init_hook, }, [ALC888_FUJITSU_XA3530] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -9191,8 +8992,8 @@ static struct alc_config_preset alc883_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, - .unsol_event = alc888_fujitsu_xa3530_unsol_event, - .init_hook = alc888_fujitsu_xa3530_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_fujitsu_xa3530_init_hook, }, [ALC888_LENOVO_SKY] = { .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, @@ -9204,8 +9005,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .need_dac_fix = 1, .input_mux = &alc883_lenovo_sky_capture_source, - .unsol_event = alc883_lenovo_sky_unsol_event, - .init_hook = alc888_lenovo_sky_front_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_lenovo_sky_init_hook, }, [ALC888_ASUS_M90V] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9233,7 +9034,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, .input_mux = &alc883_asus_eee1601_capture_source, - .unsol_event = alc883_eee1601_unsol_event, + .unsol_event = alc_sku_unsol_event, .init_hook = alc883_eee1601_inithook, }, [ALC1200_ASUS_P5Q] = { @@ -9666,28 +9467,15 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hp_t5735_automute(struct hda_codec *codec) +static void alc262_hp_t5735_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE, - spec->jack_present ? HDA_AMP_MUTE : 0); -} - -static void alc262_hp_t5735_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hp_t5735_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + alc_automute_amp(codec); } -#define alc262_hp_t5735_init_hook alc262_hp_t5735_automute - static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9914,34 +9702,15 @@ static struct hda_verb alc262_tyan_verbs[] = { }; /* unsolicited event for HP jack sensing */ -static void alc262_tyan_automute(struct hda_codec *codec) +static void alc262_tyan_init_hook(struct hda_codec *codec) { - unsigned int mute; - unsigned int present; + struct alc_spec *spec = codec->spec; - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - present = (present & 0x80000000) != 0; - if (present) { - /* mute line output on ATX panel */ - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute line output if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x15; + alc_automute_amp(codec); } -static void alc262_tyan_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_tyan_automute(codec); -} #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -10096,35 +9865,24 @@ static void alc262_dmic_automute(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09); } -/* toggle speaker-output according to the hp-jack state */ -static void alc262_toshiba_s06_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, bits); -} - - /* unsolicited event for HP jack sensing */ static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) - alc262_toshiba_s06_speaker_automute(codec); if ((res >> 26) == ALC880_MIC_EVENT) alc262_dmic_automute(codec); - + else + alc_sku_unsol_event(codec, res); } static void alc262_toshiba_s06_init_hook(struct hda_codec *codec) { - alc262_toshiba_s06_speaker_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_pin(codec); alc262_dmic_automute(codec); } @@ -11135,7 +10893,7 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hp_t5735_unsol_event, + .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc262_hp_t5735_init_hook, }, [ALC262_HP_RP5700] = { @@ -11256,8 +11014,8 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_tyan_unsol_event, - .init_hook = alc262_tyan_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc262_tyan_init_hook, }, }; @@ -11646,30 +11404,15 @@ static struct hda_verb alc268_dell_verbs[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc268_dell_automute(struct hda_codec *codec) +static void alc268_dell_init_hook(struct hda_codec *codec) { - unsigned int present; - unsigned int mute; - - present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0); - if (present & 0x80000000) - mute = HDA_AMP_MUTE; - else - mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); -} + struct alc_spec *spec = codec->spec; -static void alc268_dell_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc268_dell_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_pin(codec); } -#define alc268_dell_init_hook alc268_dell_automute - static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -11688,16 +11431,6 @@ static struct hda_verb alc267_quanta_il1_verbs[] = { { } }; -static void alc267_quanta_il1_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - present ? 0 : PIN_OUT); -} - static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -11709,9 +11442,13 @@ static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) present ? 0x00 : 0x01); } -static void alc267_quanta_il1_automute(struct hda_codec *codec) +static void alc267_quanta_il1_init_hook(struct hda_codec *codec) { - alc267_quanta_il1_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_pin(codec); alc267_quanta_il1_mic_automute(codec); } @@ -11719,12 +11456,12 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc267_quanta_il1_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc267_quanta_il1_mic_automute(codec); break; + default: + alc_sku_unsol_event(codec, res); + break; } } @@ -12198,7 +11935,7 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc267_quanta_il1_unsol_event, - .init_hook = alc267_quanta_il1_automute, + .init_hook = alc267_quanta_il1_init_hook, }, [ALC268_3ST] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, @@ -12293,7 +12030,7 @@ static struct alc_config_preset alc268_presets[] = { .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .unsol_event = alc268_dell_unsol_event, + .unsol_event = alc_sku_unsol_event, .init_hook = alc268_dell_init_hook, .input_mux = &alc268_capture_source, }, @@ -14727,19 +14464,6 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {} }; -/* toggle speaker-output according to the hp-jack state */ -static void alc861vd_lenovo_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -14752,9 +14476,13 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); } -static void alc861vd_lenovo_automute(struct hda_codec *codec) +static void alc861vd_lenovo_init_hook(struct hda_codec *codec) { - alc861vd_lenovo_hp_automute(codec); + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); alc861vd_lenovo_mic_automute(codec); } @@ -14762,12 +14490,12 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, unsigned int res) { switch (res >> 26) { - case ALC880_HP_EVENT: - alc861vd_lenovo_hp_automute(codec); - break; case ALC880_MIC_EVENT: alc861vd_lenovo_mic_automute(codec); break; + default: + alc_automute_amp_unsol_event(codec, res); + break; } } @@ -14817,20 +14545,13 @@ static struct hda_verb alc861vd_dallas_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc861vd_dallas_automute(struct hda_codec *codec) +static void alc861vd_dallas_init_hook(struct hda_codec *codec) { - unsigned int present; - - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc861vd_dallas_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + alc_automute_amp(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -14944,7 +14665,7 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, .unsol_event = alc861vd_lenovo_unsol_event, - .init_hook = alc861vd_lenovo_automute, + .init_hook = alc861vd_lenovo_init_hook, }, [ALC861VD_DALLAS] = { .mixers = { alc861vd_dallas_mixer }, @@ -14954,8 +14675,8 @@ static struct alc_config_preset alc861vd_presets[] = { .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_dallas_capture_source, - .unsol_event = alc861vd_dallas_unsol_event, - .init_hook = alc861vd_dallas_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc861vd_dallas_init_hook, }, [ALC861VD_HP] = { .mixers = { alc861vd_hp_mixer }, @@ -14966,8 +14687,8 @@ static struct alc_config_preset alc861vd_presets[] = { .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_hp_capture_source, - .unsol_event = alc861vd_dallas_unsol_event, - .init_hook = alc861vd_dallas_automute, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc861vd_dallas_init_hook, }, [ALC660VD_ASUS_V1S] = { .mixers = { alc861vd_lenovo_mixer }, @@ -14982,7 +14703,7 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, .unsol_event = alc861vd_lenovo_unsol_event, - .init_hook = alc861vd_lenovo_automute, + .init_hook = alc861vd_lenovo_init_hook, }, }; -- cgit v1.1 From b4df0a6c9d88cfff77c73d33873cd60f9ab909b6 Mon Sep 17 00:00:00 2001 From: Sergey Lapin Date: Fri, 8 May 2009 19:19:41 +0400 Subject: ASoC: AFEB9260 driver ASoC driver for AT91SAM9260-based AFEB9260 board Signed-off-by: Sergey Lapin Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 8 ++ sound/soc/atmel/Makefile | 1 + sound/soc/atmel/snd-soc-afeb9260.c | 203 +++++++++++++++++++++++++++++++++++++ 3 files changed, 212 insertions(+) create mode 100644 sound/soc/atmel/snd-soc-afeb9260.c diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a608d70..e720d5e 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE and FRAME signals on the PlayPaq. Unless you want to play with the AT32 as the SSC master, you probably want to say N here, as this will give you better sound quality. + +config SND_AT91_SOC_AFEB9260 + tristate "SoC Audio support for AFEB9260 board" + depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_TLV320AIC23 + help + Say Y here to support sound on AFEB9260 board. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index f54a7cc..e7ea56b 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o +obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c new file mode 100644 index 0000000..23349de --- /dev/null +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -0,0 +1,203 @@ +/* + * afeb9260.c -- SoC audio for AFEB9260 + * + * Copyright (C) 2009 Sergey Lapin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "../codecs/tlv320aic23.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + +#define CODEC_CLOCK 12000000 + +static int afeb9260_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S| + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops afeb9260_ops = { + .hw_params = afeb9260_hw_params, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add afeb9260 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up afeb9260 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link afeb9260_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = afeb9260_tlv320aic23_init, + .ops = &afeb9260_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_machine_afeb9260 = { + .name = "AFEB9260", + .platform = &atmel_soc_platform, + .dai_link = &afeb9260_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device afeb9260_snd_devdata = { + .card = &snd_soc_machine_afeb9260, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *afeb9260_snd_device; + +static int __init afeb9260_soc_init(void) +{ + int err; + struct device *dev; + struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + if (!(machine_is_afeb9260())) + return -ENODEV; + + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); + err = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + afeb9260_snd_device = platform_device_alloc("soc-audio", -1); + if (!afeb9260_snd_device) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata); + afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev; + err = platform_device_add(afeb9260_snd_device); + if (err) + goto err1; + + dev = &afeb9260_snd_device->dev; + + return 0; +err1: + platform_device_del(afeb9260_snd_device); + platform_device_put(afeb9260_snd_device); +err_ssc: + return err; + +} + +static void __exit afeb9260_soc_exit(void) +{ + platform_device_unregister(afeb9260_snd_device); +} + +module_init(afeb9260_soc_init); +module_exit(afeb9260_soc_exit); + +MODULE_AUTHOR("Sergey Lapin "); +MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); +MODULE_LICENSE("GPL"); + -- cgit v1.1 From 151ab22cf71b7a1b9dd696d65a1a41e13d90cd00 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 May 2009 16:22:58 +0100 Subject: ASoC: Fix up CODEC DAI formats for big endian CPUs ASoC uses the standard ALSA data format definitions to specify the wire format used between the CPU and CODEC. Since the ALSA data formats all include the endianess of the data but this information is not relevant by the time the data has been encoded onto the serial link to the CODEC this means that either all the CODEC drivers need to declare both big and little endian variants or the core needs to fix up the format constraints specified by CODEC drivers. For now take the latter approach - this will need to be revisited if any CODECs are endianness dependant. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index af11791..6ac68e4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2387,6 +2387,39 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform) } EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); +static u64 codec_format_map[] = { + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE, + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE, + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE, + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE, + SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE, + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE, + SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE, + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE, + SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE, + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE, + SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE, + SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE, + SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE, + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE + | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, +}; + +/* Fix up the DAI formats for endianness: codecs don't actually see + * the endianness of the data but we're using the CPU format + * definitions which do need to include endianness so we ensure that + * codec DAIs always have both big and little endian variants set. + */ +static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(codec_format_map); i++) + if (stream->formats & codec_format_map[i]) + stream->formats |= codec_format_map[i]; +} + /** * snd_soc_register_codec - Register a codec with the ASoC core * @@ -2394,6 +2427,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); */ int snd_soc_register_codec(struct snd_soc_codec *codec) { + int i; + if (!codec->name) return -EINVAL; @@ -2403,6 +2438,11 @@ int snd_soc_register_codec(struct snd_soc_codec *codec) INIT_LIST_HEAD(&codec->list); + for (i = 0; i < codec->num_dai; i++) { + fixup_codec_formats(&codec->dai[i].playback); + fixup_codec_formats(&codec->dai[i].capture); + } + mutex_lock(&client_mutex); list_add(&codec->list, &codec_list); snd_soc_instantiate_cards(); -- cgit v1.1 From 01f2bd48d08c6bbde12d86b66c760612e33e49a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 May 2009 08:12:43 +0200 Subject: ALSA: hda - Add missing models for Realtek codecs Added the missing descriptions and the model names for Realtek codecs to the documentation and the config table. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 10 ++++++++-- sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 36c9712..92c9d3a 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -36,6 +36,7 @@ ALC260 acer Acer TravelMate will Will laptops (PB V7900) replacer Replacer 672V + favorit100 Maxdata Favorit 100XS basic fixed pin assignment (old default model) test for testing/debugging purpose, almost all controls can adjusted. Appearing only when compiled with @@ -85,10 +86,11 @@ ALC269 eeepc-p703 ASUS Eeepc P703 P900A eeepc-p901 ASUS Eeepc P901 S101 fujitsu FSC Amilo + lifebook Fujitsu Lifebook S6420 auto auto-config reading BIOS (default) -ALC662/663 -========== +ALC662/663/272 +============== 3stack-dig 3-stack (2-channel) with SPDIF 3stack-6ch 3-stack (6-channel) 3stack-6ch-dig 3-stack (6-channel) with SPDIF @@ -107,6 +109,8 @@ ALC662/663 asus-mode4 ASUS asus-mode5 ASUS asus-mode6 ASUS + dell Dell with ALC272 + dell-zm1 Dell ZM1 with ALC272 auto auto-config reading BIOS (default) ALC882/885 @@ -118,6 +122,7 @@ ALC882/885 asus-a7j ASUS A7J asus-a7m ASUS A7M macpro MacPro support + mb5 Macbook 5,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC @@ -150,6 +155,7 @@ ALC883/888 fujitsu-pi2515 Fujitsu AMILO Pi2515 fujitsu-xa3530 Fujitsu AMILO XA3530 3stack-6ch-intel Intel DG33* boards + asus-p5q ASUS P5Q-EM boards auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34f6fb7..db89612 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16184,6 +16184,8 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC663_ASUS_MODE4] = "asus-mode4", [ALC663_ASUS_MODE5] = "asus-mode5", [ALC663_ASUS_MODE6] = "asus-mode6", + [ALC272_DELL] = "dell", + [ALC272_DELL_ZM1] = "dell-zm1", [ALC662_AUTO] = "auto", }; -- cgit v1.1 From 31cb31f76e030ae05ed45f409ce5eb68ef2987f6 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Mon, 11 May 2009 21:57:08 +0200 Subject: ASoC: remove driver_data direct access of struct device Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 4 ++-- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa6..e4547de 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index e043e3f..7a20587 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8731; + dev_set_drvdata(&spi->dev, wm8731); return wm8731_register(wm8731); } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi->dev.driver_data; + struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev); wm8731_unregister(wm8731); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f..d121e58 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; } -- cgit v1.1 From 9541ba1d665542c96f7c0b5b836bbc1fd9d961b6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Chris=20Pockel=C3=A9?= Date: Tue, 12 May 2009 08:08:53 +0200 Subject: ALSA: hda - Add support of Samsung NC10 mini notebook MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add specific configuration for Samsung NC10 mini notebook. Internal mic/speakers will be correctly muted when plugging in external ones. Mixer controls are added for speakers, headphones and PC beep. "Boost" is added for the microphones. Signed-off-by: Chris Pockelé Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 64 ++++++++++++++++++++++++++++ 2 files changed, 65 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 92c9d3a..14ed1d1 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -111,6 +111,7 @@ ALC662/663/272 asus-mode6 ASUS dell Dell with ALC272 dell-zm1 Dell ZM1 with ALC272 + samsung-nc10 Samsung NC10 mini notebook auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index db89612..c14020a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -190,6 +190,7 @@ enum { ALC663_ASUS_MODE6, ALC272_DELL, ALC272_DELL_ZM1, + ALC272_SAMSUNG_NC10, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -15120,6 +15121,38 @@ static struct hda_input_mux alc663_m51va_capture_source = { }, }; +#if 1 /* set to 0 for testing other input sources below */ +static struct hda_input_mux alc272_nc10_capture_source = { + .num_items = 2, + .items = { + { "Autoselect Mic", 0x0 }, + { "Internal Mic", 0x1 }, + }, +}; +#else +static struct hda_input_mux alc272_nc10_capture_source = { + .num_items = 16, + .items = { + { "Autoselect Mic", 0x0 }, + { "Internal Mic", 0x1 }, + { "In-0x02", 0x2 }, + { "In-0x03", 0x3 }, + { "In-0x04", 0x4 }, + { "In-0x05", 0x5 }, + { "In-0x06", 0x6 }, + { "In-0x07", 0x7 }, + { "In-0x08", 0x8 }, + { "In-0x09", 0x9 }, + { "In-0x0a", 0x0a }, + { "In-0x0b", 0x0b }, + { "In-0x0c", 0x0c }, + { "In-0x0d", 0x0d }, + { "In-0x0e", 0x0e }, + { "In-0x0f", 0x0f }, + }, +}; +#endif + /* * 2ch mode */ @@ -16151,6 +16184,23 @@ static struct snd_kcontrol_new alc662_ecs_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc272_nc10_mixer[] = { + /* Master Playback automatically created from Speaker and Headphone */ + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -16186,6 +16236,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC663_ASUS_MODE6] = "asus-mode6", [ALC272_DELL] = "dell", [ALC272_DELL_ZM1] = "dell-zm1", + [ALC272_SAMSUNG_NC10] = "samsung-nc10", [ALC662_AUTO] = "auto", }; @@ -16243,6 +16294,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), @@ -16514,6 +16566,18 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc663_m51va_unsol_event, .init_hook = alc663_m51va_inithook, }, + [ALC272_SAMSUNG_NC10] = { + .mixers = { alc272_nc10_mixer }, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc272_dac_nids), + .dac_nids = alc272_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc272_nc10_capture_source, + .unsol_event = alc663_mode4_unsol_event, + .init_hook = alc663_mode4_inithook, + }, }; -- cgit v1.1 From ae31c1fbdbb18d917b0a1139497c2dbd35886989 Mon Sep 17 00:00:00 2001 From: Greg Kroah-Hartman Date: Mon, 4 May 2009 12:40:54 -0700 Subject: sound: remove driver_data direct access of struct device In the near future, the driver core is going to not allow direct access to the driver_data pointer in struct device. Instead, the functions dev_get_drvdata() and dev_set_drvdata() should be used. These functions have been around since the beginning, so are backwards compatible with all older kernel versions. Signed-off-by: Greg Kroah-Hartman Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/aoa/fabrics/layout.c | 8 ++++---- sound/aoa/soundbus/i2sbus/core.c | 8 ++++---- sound/soc/codecs/wm8400.c | 4 ++-- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- 5 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index fbf5c93..586965f 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1037,7 +1037,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) } ldev->selfptr_headphone.ptr = ldev; ldev->selfptr_lineout.ptr = ldev; - sdev->ofdev.dev.driver_data = ldev; + dev_set_drvdata(&sdev->ofdev.dev, ldev); list_add(&ldev->list, &layouts_list); layouts_list_items++; @@ -1081,7 +1081,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) { - struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); int i; for (i=0; iofdev.dev.driver_data; + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) ldev->gpio.methods->all_amps_off(&ldev->gpio); @@ -1124,7 +1124,7 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta static int aoa_fabric_layout_resume(struct soundbus_dev *sdev) { - struct layout_dev *ldev = sdev->ofdev.dev.driver_data; + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) ldev->gpio.methods->all_amps_restore(&ldev->gpio); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 418c84c..4e3b819 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -358,14 +358,14 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - dev->ofdev.dev.driver_data = control; + dev_set_drvdata(&dev->ofdev.dev, control); return 0; } static int i2sbus_remove(struct macio_dev* dev) { - struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct i2sbus_dev *i2sdev, *tmp; list_for_each_entry_safe(i2sdev, tmp, &control->list, item) @@ -377,7 +377,7 @@ static int i2sbus_remove(struct macio_dev* dev) #ifdef CONFIG_PM static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state) { - struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct codec_info_item *cii; struct i2sbus_dev* i2sdev; int err, ret = 0; @@ -407,7 +407,7 @@ static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state) static int i2sbus_resume(struct macio_dev* dev) { - struct i2sbus_control *control = dev->ofdev.dev.driver_data; + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct codec_info_item *cii; struct i2sbus_dev* i2sdev; int err, ret = 0; diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 510efa6..e4547de 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev) codec = &priv->codec; codec->private_data = priv; - codec->control_data = dev->dev.driver_data; - priv->wm8400 = dev->dev.driver_data; + codec->control_data = dev_get_drvdata(&dev->dev); + priv->wm8400 = dev_get_drvdata(&dev->dev); ret = regulator_bulk_get(priv->wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index e043e3f..7a20587 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8731; + dev_set_drvdata(&spi->dev, wm8731); return wm8731_register(wm8731); } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct wm8731_priv *wm8731 = spi->dev.driver_data; + struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev); wm8731_unregister(wm8731); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a6e8f3f..d121e58 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8753_spi_write; codec->dev = &spi->dev; - spi->dev.driver_data = wm8753; + dev_set_drvdata(&spi->dev, wm8753); return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { - struct wm8753_priv *wm8753 = spi->dev.driver_data; + struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev); wm8753_unregister(wm8753); return 0; } -- cgit v1.1 From 8e365f9b6e583e82dff96ff415ff9cc2106f5b3c Mon Sep 17 00:00:00 2001 From: Raphael Doursenaud Date: Tue, 12 May 2009 11:49:39 +0200 Subject: ALSA: hdsp: allow firmware loading from inside the kernel Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and in-kernel driver. Also corrected a typo in the comment. Signed-off-by: Raphael Doursenaud Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 314e735..d0d4629 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -402,9 +402,9 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_DMA_AREA_BYTES ((HDSP_MAX_CHANNELS+1) * HDSP_CHANNEL_BUFFER_BYTES) #define HDSP_DMA_AREA_KILOBYTES (HDSP_DMA_AREA_BYTES/1024) -/* use hotplug firmeare loader? */ +/* use hotplug firmware loader? */ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP) +#if !defined(HDSP_USE_HWDEP_LOADER) #define HDSP_FW_LOADER #endif #endif -- cgit v1.1 From eaaa5328835d8085d24221a0e5ceaacbe14a7523 Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 11 May 2009 15:05:29 +0300 Subject: ASoC: em-x270: make the driver support also eXeda and CM-X300 machines Signed-off-by: Mike Rapoport Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 4 ++-- sound/soc/pxa/em-x270.c | 9 +++++---- 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index eb75a1c..dcd163a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800 Toshiba e800 PDA config SND_PXA2XX_SOC_EM_X270 - tristate "SoC Audio support for CompuLab EM-x270" + tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" depends on SND_PXA2XX_SOC && MACH_EM_X270 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help Say Y if you want to add support for SoC audio on - CompuLab EM-x270. + CompuLab EM-x270, eXeda and CM-X300 machines. config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5 and LifeDrive" diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 949be9c..f4756e4 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -1,7 +1,7 @@ /* - * em-x270.c -- SoC audio for EM-X270 + * SoC audio driver for EM-X270, eXeda and CM-X300 * - * Copyright 2007 CompuLab, Ltd. + * Copyright 2007, 2009 CompuLab, Ltd. * * Author: Mike Rapoport * @@ -68,7 +68,8 @@ static int __init em_x270_init(void) { int ret; - if (!machine_is_em_x270()) + if (!(machine_is_em_x270() || machine_is_exeda() + || machine_is_cm_x300())) return -ENODEV; em_x270_snd_device = platform_device_alloc("soc-audio", -1); @@ -95,5 +96,5 @@ module_exit(em_x270_exit); /* Module information */ MODULE_AUTHOR("Mike Rapoport"); -MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300"); MODULE_LICENSE("GPL"); -- cgit v1.1 From 7de0a0aee5cf24639c14b17ab4077f5dba0d7cb9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 May 2009 20:05:57 +0100 Subject: ASoC: Enforce symmetric rates for PXA2xx I2S There is a single I2S_SYNC pin on the chip. Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2f4b6e4..6014577 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -329,6 +329,7 @@ struct snd_soc_dai pxa_i2s_dai = { .rates = PXA2XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = &pxa_i2s_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(pxa_i2s_dai); -- cgit v1.1 From 97b8096dc92ae62b1d40e6bec7e7b257d2b30161 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Mon, 11 May 2009 20:36:08 +0900 Subject: ASoC: TWL4030: change DAPM for analog microphone selection The inputs of the twl4030 codec can be mixed, so we will use the mixer DAPM for the analog microphone registers(0x05, 0x06), but if we enable more than one input at the same time, the input impedance of the input amplifier will be reduced. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 52 +++++++++++++++++----------------------------- 1 file changed, 19 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd392c6..eaf91ab 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -422,36 +422,18 @@ static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control = SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); /* Left analog microphone selection */ -static const char *twl4030_analoglmic_texts[] = - {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; - -static const unsigned int twl4030_analoglmic_values[] = - {0x0, 0x1, 0x2, 0x4, 0x8}; - -static const struct soc_enum twl4030_analoglmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf, - ARRAY_SIZE(twl4030_analoglmic_texts), - twl4030_analoglmic_texts, - twl4030_analoglmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { + SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0), + SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0), +}; /* Right analog microphone selection */ -static const char *twl4030_analogrmic_texts[] = - {"Off", "Sub mic", "AUXR"}; - -static const unsigned int twl4030_analogrmic_values[] = - {0x0, 0x1, 0x4}; - -static const struct soc_enum twl4030_analogrmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5, - ARRAY_SIZE(twl4030_analogrmic_texts), - twl4030_analogrmic_texts, - twl4030_analogrmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { + SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), + SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0), +}; /* TX1 L/R Analog/Digital microphone selection */ static const char *twl4030_micpathtx1_texts[] = @@ -1138,11 +1120,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), - /* Analog input muxes with switch for the capture amplifiers */ - SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", - TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control), - SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", - TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control), + /* Analog input mixers for the capture amplifiers */ + SND_SOC_DAPM_MIXER("Analog Left Capture Route", + TWL4030_REG_ANAMICL, 4, 0, + &twl4030_dapm_analoglmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analoglmic_controls)), + SND_SOC_DAPM_MIXER("Analog Right Capture Route", + TWL4030_REG_ANAMICR, 4, 0, + &twl4030_dapm_analogrmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analogrmic_controls)), SND_SOC_DAPM_PGA("ADC Physical Left", TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0), -- cgit v1.1 From 511b4c171a8054506e8898c40833e31270689b8b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 May 2009 11:51:46 +0200 Subject: ALSA: hdsp - Add a comment about external firmwares for hdsp When the hdsp driver is built in kernel, the corresponding firmware files have to be built into the kernel as well (otherwise the boot will hang up for fairly long time), but there is no way to control it in Kconfig yet, unfortunately. Instead, show a comment for user just to make sure. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93422e3..7478971 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -532,6 +532,9 @@ config SND_HDSP To compile this driver as a module, choose M here: the module will be called snd-hdsp. +comment "Don't forget to add built-in firmwares for HDSP driver" + depends on SND_HDSP=y + config SND_HDSPM tristate "RME Hammerfall DSP MADI" select SND_HWDEP -- cgit v1.1 From 22b530e0287724f00dde9a15a94d408c3334e86c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 May 2009 11:32:52 +0200 Subject: ALSA: hda - Disable fallback to model=acer for Acer laptops The model=acer for ALC883/889 doesn't work well for the recent Acer Aspire laptops. Since model=auto works better nowadays, it's safer to use the default fallback instead of the Acer specific one. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c14020a..51954ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8622,8 +8622,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_4930G), - /* default Acer */ - SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), + /* default Acer -- disabled as it causes more problems. + * model=auto should work fine now + */ + /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), -- cgit v1.1 From 8c10dc4f54d315ce801dc9ef4018aab8d0d75a7b Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Mon, 11 May 2009 23:49:41 +0200 Subject: ASoC: pxa2xx-i2s: Proper initialization Reset FIFO logic and registers, and make sure REC and RPL functions along with FIFO service are disabled at probe. Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6014577..fce8a28 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -347,6 +347,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) if (ret != 0) clk_put(clk_i2s); + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + /* Make sure RPL and REC are disabled */ + SACR1 = SACR1_DRPL | SACR1_DREC; + /* Along with FIFO servicing */ + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return ret; } -- cgit v1.1 From bb74a6e5c5535ddd977aa161c72bef738e566daa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 13 May 2009 17:23:54 +0100 Subject: ASoC: Point at kernel.org git The Wolfson git is not currently tracking bleeding edge ASoC so change to my kernel.org git which is doing so. Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 17c8ec1..00401d8 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5273,7 +5273,7 @@ P: Liam Girdwood M: lrg@slimlogic.co.uk P: Mark Brown M: broonie@opensource.wolfsonmicro.com -T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc +T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git L: alsa-devel@alsa-project.org (subscribers-only) W: http://alsa-project.org/main/index.php/ASoC S: Supported -- cgit v1.1 From 8cc72361481f00253f1e468ade5795427386d593 Mon Sep 17 00:00:00 2001 From: Wai Yew CHAY Date: Thu, 14 May 2009 08:05:58 +0200 Subject: ALSA: SB X-Fi driver merge MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Sound Blaster X-Fi driver supports Creative solutions based on 20K1 and 20K2 chipsets. Supported hardware : Creative Sound Blaster X-Fi Titanium Fatal1ty® Champion Series Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series Creative Sound Blaster X-Fi Titanium Professional Audio Creative Sound Blaster X-Fi Titanium Creative Sound Blaster X-Fi Elite Pro Creative Sound Blaster X-Fi Platinum Creative Sound Blaster X-Fi Fatal1ty Creative Sound Blaster X-Fi XtremeGamer Creative Sound Blaster X-Fi XtremeMusic Current release features: * ALSA PCM Playback * ALSA Record * ALSA Mixer Note: * External I/O modules detection not included. Signed-off-by: Wai Yew CHAY Singed-off-by: Ryan RICHARDS Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 19 + sound/pci/Kconfig | 10 + sound/pci/Makefile | 1 + sound/pci/ctxfi/Makefile | 5 + sound/pci/ctxfi/ct20k1reg.h | 634 +++++++ sound/pci/ctxfi/ct20k2reg.h | 85 + sound/pci/ctxfi/ctamixer.c | 488 +++++ sound/pci/ctxfi/ctamixer.h | 96 + sound/pci/ctxfi/ctatc.c | 1605 ++++++++++++++++ sound/pci/ctxfi/ctatc.h | 155 ++ sound/pci/ctxfi/ctdaio.c | 769 ++++++++ sound/pci/ctxfi/ctdaio.h | 122 ++ sound/pci/ctxfi/ctdrv.h | 30 + sound/pci/ctxfi/cthardware.c | 108 ++ sound/pci/ctxfi/cthardware.h | 160 ++ sound/pci/ctxfi/cthw20k1.c | 2230 +++++++++++++++++++++++ sound/pci/ctxfi/cthw20k1.h | 26 + sound/pci/ctxfi/cthw20k2.c | 2133 ++++++++++++++++++++++ sound/pci/ctxfi/cthw20k2.h | 26 + sound/pci/ctxfi/ctimap.c | 112 ++ sound/pci/ctxfi/ctimap.h | 40 + sound/pci/ctxfi/ctmixer.c | 1108 +++++++++++ sound/pci/ctxfi/ctmixer.h | 67 + sound/pci/ctxfi/ctpcm.c | 499 +++++ sound/pci/ctxfi/ctpcm.h | 27 + sound/pci/ctxfi/ctresource.c | 297 +++ sound/pci/ctxfi/ctresource.h | 72 + sound/pci/ctxfi/ctsrc.c | 886 +++++++++ sound/pci/ctxfi/ctsrc.h | 149 ++ sound/pci/ctxfi/ctvmem.c | 254 +++ sound/pci/ctxfi/ctvmem.h | 49 + sound/pci/ctxfi/xfi.c | 125 ++ 32 files changed, 12387 insertions(+) create mode 100644 sound/pci/ctxfi/Makefile create mode 100644 sound/pci/ctxfi/ct20k1reg.h create mode 100644 sound/pci/ctxfi/ct20k2reg.h create mode 100644 sound/pci/ctxfi/ctamixer.c create mode 100644 sound/pci/ctxfi/ctamixer.h create mode 100644 sound/pci/ctxfi/ctatc.c create mode 100644 sound/pci/ctxfi/ctatc.h create mode 100644 sound/pci/ctxfi/ctdaio.c create mode 100644 sound/pci/ctxfi/ctdaio.h create mode 100644 sound/pci/ctxfi/ctdrv.h create mode 100644 sound/pci/ctxfi/cthardware.c create mode 100644 sound/pci/ctxfi/cthardware.h create mode 100644 sound/pci/ctxfi/cthw20k1.c create mode 100644 sound/pci/ctxfi/cthw20k1.h create mode 100644 sound/pci/ctxfi/cthw20k2.c create mode 100644 sound/pci/ctxfi/cthw20k2.h create mode 100644 sound/pci/ctxfi/ctimap.c create mode 100644 sound/pci/ctxfi/ctimap.h create mode 100644 sound/pci/ctxfi/ctmixer.c create mode 100644 sound/pci/ctxfi/ctmixer.h create mode 100644 sound/pci/ctxfi/ctpcm.c create mode 100644 sound/pci/ctxfi/ctpcm.h create mode 100644 sound/pci/ctxfi/ctresource.c create mode 100644 sound/pci/ctxfi/ctresource.h create mode 100644 sound/pci/ctxfi/ctsrc.c create mode 100644 sound/pci/ctxfi/ctsrc.h create mode 100644 sound/pci/ctxfi/ctvmem.c create mode 100644 sound/pci/ctxfi/ctvmem.h create mode 100644 sound/pci/ctxfi/xfi.c diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..c903db8 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -460,6 +460,25 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. + Module snd-ctxfi + ---------------- + + Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) + * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series + * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series + * Creative Sound Blaster X-Fi Titanium Professional Audio + * Creative Sound Blaster X-Fi Titanium + * Creative Sound Blaster X-Fi Elite Pro + * Creative Sound Blaster X-Fi Platinum + * Creative Sound Blaster X-Fi Fatal1ty + * Creative Sound Blaster X-Fi XtremeGamer + * Creative Sound Blaster X-Fi XtremeMusic + + reference_rate - reference sample rate, 44100 or 48000 (default) + multiple - multiple to ref. sample rate, 1 or 2 (default) + + This module supports multiple cards. + Module snd-darla20 ------------------ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93422e3..3a7640f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -275,6 +275,16 @@ config SND_CS5535AUDIO To compile this driver as a module, choose M here: the module will be called snd-cs5535audio. +config SND_CTXFI + tristate "Creative Sound Blaster X-Fi" + select SND_PCM + help + If you want to use soundcards based on Creative Sound Blastr X-Fi + boards with 20k1 or 20k2 chips, say Y here. + + To compile this driver as a module, choose M here: the module + will be called snd-ctxfi. + config SND_DARLA20 tristate "(Echoaudio) Darla20" select FW_LOADER diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 65b25d2..6a1281e 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -59,6 +59,7 @@ obj-$(CONFIG_SND) += \ ali5451/ \ au88x0/ \ aw2/ \ + ctxfi/ \ ca0106/ \ cs46xx/ \ cs5535audio/ \ diff --git a/sound/pci/ctxfi/Makefile b/sound/pci/ctxfi/Makefile new file mode 100644 index 0000000..2904323 --- /dev/null +++ b/sound/pci/ctxfi/Makefile @@ -0,0 +1,5 @@ +snd-ctxfi-objs := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \ + ctsrc.o ctamixer.o ctdaio.o ctimap.o cthardware.o \ + cthw20k2.o cthw20k1.o + +obj-$(CONFIG_SND_CTXFI) += snd-ctxfi.o diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h new file mode 100644 index 0000000..c62e677 --- /dev/null +++ b/sound/pci/ctxfi/ct20k1reg.h @@ -0,0 +1,634 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + */ + +#ifndef CT20K1REG_H +#define CT20k1REG_H + +/* 20k1 registers */ +#define DSPXRAM_START 0x000000 +#define DSPXRAM_END 0x013FFC +#define DSPAXRAM_START 0x020000 +#define DSPAXRAM_END 0x023FFC +#define DSPYRAM_START 0x040000 +#define DSPYRAM_END 0x04FFFC +#define DSPAYRAM_START 0x020000 +#define DSPAYRAM_END 0x063FFC +#define DSPMICRO_START 0x080000 +#define DSPMICRO_END 0x0B3FFC +#define DSP0IO_START 0x100000 +#define DSP0IO_END 0x101FFC +#define AUDIORINGIPDSP0_START 0x100000 +#define AUDIORINGIPDSP0_END 0x1003FC +#define AUDIORINGOPDSP0_START 0x100400 +#define AUDIORINGOPDSP0_END 0x1007FC +#define AUDPARARINGIODSP0_START 0x100800 +#define AUDPARARINGIODSP0_END 0x100BFC +#define DSP0LOCALHWREG_START 0x100C00 +#define DSP0LOCALHWREG_END 0x100C3C +#define DSP0XYRAMAGINDEX_START 0x100C40 +#define DSP0XYRAMAGINDEX_END 0x100C5C +#define DSP0XYRAMAGMDFR_START 0x100C60 +#define DSP0XYRAMAGMDFR_END 0x100C7C +#define DSP0INTCONTLVEC_START 0x100C80 +#define DSP0INTCONTLVEC_END 0x100CD8 +#define INTCONTLGLOBALREG_START 0x100D1C +#define INTCONTLGLOBALREG_END 0x100D3C +#define HOSTINTFPORTADDRCONTDSP0 0x100D40 +#define HOSTINTFPORTDATADSP0 0x100D44 +#define TIME0PERENBDSP0 0x100D60 +#define TIME0COUNTERDSP0 0x100D64 +#define TIME1PERENBDSP0 0x100D68 +#define TIME1COUNTERDSP0 0x100D6C +#define TIME2PERENBDSP0 0x100D70 +#define TIME2COUNTERDSP0 0x100D74 +#define TIME3PERENBDSP0 0x100D78 +#define TIME3COUNTERDSP0 0x100D7C +#define XRAMINDOPERREFNOUP_STARTDSP0 0x100D80 +#define XRAMINDOPERREFNOUP_ENDDSP0 0x100D9C +#define XRAMINDOPERREFUP_STARTDSP0 0x100DA0 +#define XRAMINDOPERREFUP_ENDDSP0 0x100DBC +#define YRAMINDOPERREFNOUP_STARTDSP0 0x100DC0 +#define YRAMINDOPERREFNOUP_ENDDSP0 0x100DDC +#define YRAMINDOPERREFUP_STARTDSP0 0x100DE0 +#define YRAMINDOPERREFUP_ENDDSP0 0x100DFC +#define DSP0CONDCODE 0x100E00 +#define DSP0STACKFLAG 0x100E04 +#define DSP0PROGCOUNTSTACKPTREG 0x100E08 +#define DSP0PROGCOUNTSTACKDATAREG 0x100E0C +#define DSP0CURLOOPADDRREG 0x100E10 +#define DSP0CURLOOPCOUNT 0x100E14 +#define DSP0TOPLOOPCOUNTSTACK 0x100E18 +#define DSP0TOPLOOPADDRSTACK 0x100E1C +#define DSP0LOOPSTACKPTR 0x100E20 +#define DSP0STASSTACKDATAREG 0x100E24 +#define DSP0STASSTACKPTR 0x100E28 +#define DSP0PROGCOUNT 0x100E2C +#define GLOBDSPDEBGREG 0x100E30 +#define GLOBDSPBREPTRREG 0x100E30 +#define DSP0XYRAMBASE_START 0x100EA0 +#define DSP0XYRAMBASE_END 0x100EBC +#define DSP0XYRAMLENG_START 0x100EC0 +#define DSP0XYRAMLENG_END 0x100EDC +#define SEMAPHOREREGDSP0 0x100EE0 +#define DSP0INTCONTMASKREG 0x100EE4 +#define DSP0INTCONTPENDREG 0x100EE8 +#define DSP0INTCONTSERVINT 0x100EEC +#define DSPINTCONTEXTINTMODREG 0x100EEC +#define GPIODSP0 0x100EFC +#define DMADSPBASEADDRREG_STARTDSP0 0x100F00 +#define DMADSPBASEADDRREG_ENDDSP0 0x100F1C +#define DMAHOSTBASEADDRREG_STARTDSP0 0x100F20 +#define DMAHOSTBASEADDRREG_ENDDSP0 0x100F3C +#define DMADSPCURADDRREG_STARTDSP0 0x100F40 +#define DMADSPCURADDRREG_ENDDSP0 0x100F5C +#define DMAHOSTCURADDRREG_STARTDSP0 0x100F60 +#define DMAHOSTCURADDRREG_ENDDSP0 0x100F7C +#define DMATANXCOUNTREG_STARTDSP0 0x100F80 +#define DMATANXCOUNTREG_ENDDSP0 0x100F9C +#define DMATIMEBUGREG_STARTDSP0 0x100FA0 +#define DMATIMEBUGREG_ENDDSP0 0x100FAC +#define DMACNTLMODFREG_STARTDSP0 0x100FA0 +#define DMACNTLMODFREG_ENDDSP0 0x100FAC + +#define DMAGLOBSTATSREGDSP0 0x100FEC +#define DSP0XGPRAM_START 0x101000 +#define DSP0XGPRAM_END 0x1017FC +#define DSP0YGPRAM_START 0x101800 +#define DSP0YGPRAM_END 0x101FFC + + + + +#define AUDIORINGIPDSP1_START 0x102000 +#define AUDIORINGIPDSP1_END 0x1023FC +#define AUDIORINGOPDSP1_START 0x102400 +#define AUDIORINGOPDSP1_END 0x1027FC +#define AUDPARARINGIODSP1_START 0x102800 +#define AUDPARARINGIODSP1_END 0x102BFC +#define DSP1LOCALHWREG_START 0x102C00 +#define DSP1LOCALHWREG_END 0x102C3C +#define DSP1XYRAMAGINDEX_START 0x102C40 +#define DSP1XYRAMAGINDEX_END 0x102C5C +#define DSP1XYRAMAGMDFR_START 0x102C60 +#define DSP1XYRAMAGMDFR_END 0x102C7C +#define DSP1INTCONTLVEC_START 0x102C80 +#define DSP1INTCONTLVEC_END 0x102CD8 +#define HOSTINTFPORTADDRCONTDSP1 0x102D40 +#define HOSTINTFPORTDATADSP1 0x102D44 +#define TIME0PERENBDSP1 0x102D60 +#define TIME0COUNTERDSP1 0x102D64 +#define TIME1PERENBDSP1 0x102D68 +#define TIME1COUNTERDSP1 0x102D6C +#define TIME2PERENBDSP1 0x102D70 +#define TIME2COUNTERDSP1 0x102D74 +#define TIME3PERENBDSP1 0x102D78 +#define TIME3COUNTERDSP1 0x102D7C +#define XRAMINDOPERREFNOUP_STARTDSP1 0x102D80 +#define XRAMINDOPERREFNOUP_ENDDSP1 0x102D9C +#define XRAMINDOPERREFUP_STARTDSP1 0x102DA0 +#define XRAMINDOPERREFUP_ENDDSP1 0x102DBC +#define YRAMINDOPERREFNOUP_STARTDSP1 0x102DC0 +#define YRAMINDOPERREFNOUP_ENDDSP1 0x102DDC +#define YRAMINDOPERREFUP_STARTDSP1 0x102DE0 +#define YRAMINDOPERREFUP_ENDDSP1 0x102DFC + +#define DSP1CONDCODE 0x102E00 +#define DSP1STACKFLAG 0x102E04 +#define DSP1PROGCOUNTSTACKPTREG 0x102E08 +#define DSP1PROGCOUNTSTACKDATAREG 0x102E0C +#define DSP1CURLOOPADDRREG 0x102E10 +#define DSP1CURLOOPCOUNT 0x102E14 +#define DSP1TOPLOOPCOUNTSTACK 0x102E18 +#define DSP1TOPLOOPADDRSTACK 0x102E1C +#define DSP1LOOPSTACKPTR 0x102E20 +#define DSP1STASSTACKDATAREG 0x102E24 +#define DSP1STASSTACKPTR 0x102E28 +#define DSP1PROGCOUNT 0x102E2C +#define DSP1XYRAMBASE_START 0x102EA0 +#define DSP1XYRAMBASE_END 0x102EBC +#define DSP1XYRAMLENG_START 0x102EC0 +#define DSP1XYRAMLENG_END 0x102EDC +#define SEMAPHOREREGDSP1 0x102EE0 +#define DSP1INTCONTMASKREG 0x102EE4 +#define DSP1INTCONTPENDREG 0x102EE8 +#define DSP1INTCONTSERVINT 0x102EEC +#define GPIODSP1 0x102EFC +#define DMADSPBASEADDRREG_STARTDSP1 0x102F00 +#define DMADSPBASEADDRREG_ENDDSP1 0x102F1C +#define DMAHOSTBASEADDRREG_STARTDSP1 0x102F20 +#define DMAHOSTBASEADDRREG_ENDDSP1 0x102F3C +#define DMADSPCURADDRREG_STARTDSP1 0x102F40 +#define DMADSPCURADDRREG_ENDDSP1 0x102F5C +#define DMAHOSTCURADDRREG_STARTDSP1 0x102F60 +#define DMAHOSTCURADDRREG_ENDDSP1 0x102F7C +#define DMATANXCOUNTREG_STARTDSP1 0x102F80 +#define DMATANXCOUNTREG_ENDDSP1 0x102F9C +#define DMATIMEBUGREG_STARTDSP1 0x102FA0 +#define DMATIMEBUGREG_ENDDSP1 0x102FAC +#define DMACNTLMODFREG_STARTDSP1 0x102FA0 +#define DMACNTLMODFREG_ENDDSP1 0x102FAC + +#define DMAGLOBSTATSREGDSP1 0x102FEC +#define DSP1XGPRAM_START 0x103000 +#define DSP1XGPRAM_END 0x1033FC +#define DSP1YGPRAM_START 0x103400 +#define DSP1YGPRAM_END 0x1037FC + + + +#define AUDIORINGIPDSP2_START 0x104000 +#define AUDIORINGIPDSP2_END 0x1043FC +#define AUDIORINGOPDSP2_START 0x104400 +#define AUDIORINGOPDSP2_END 0x1047FC +#define AUDPARARINGIODSP2_START 0x104800 +#define AUDPARARINGIODSP2_END 0x104BFC +#define DSP2LOCALHWREG_START 0x104C00 +#define DSP2LOCALHWREG_END 0x104C3C +#define DSP2XYRAMAGINDEX_START 0x104C40 +#define DSP2XYRAMAGINDEX_END 0x104C5C +#define DSP2XYRAMAGMDFR_START 0x104C60 +#define DSP2XYRAMAGMDFR_END 0x104C7C +#define DSP2INTCONTLVEC_START 0x104C80 +#define DSP2INTCONTLVEC_END 0x104CD8 +#define HOSTINTFPORTADDRCONTDSP2 0x104D40 +#define HOSTINTFPORTDATADSP2 0x104D44 +#define TIME0PERENBDSP2 0x104D60 +#define TIME0COUNTERDSP2 0x104D64 +#define TIME1PERENBDSP2 0x104D68 +#define TIME1COUNTERDSP2 0x104D6C +#define TIME2PERENBDSP2 0x104D70 +#define TIME2COUNTERDSP2 0x104D74 +#define TIME3PERENBDSP2 0x104D78 +#define TIME3COUNTERDSP2 0x104D7C +#define XRAMINDOPERREFNOUP_STARTDSP2 0x104D80 +#define XRAMINDOPERREFNOUP_ENDDSP2 0x104D9C +#define XRAMINDOPERREFUP_STARTDSP2 0x104DA0 +#define XRAMINDOPERREFUP_ENDDSP2 0x104DBC +#define YRAMINDOPERREFNOUP_STARTDSP2 0x104DC0 +#define YRAMINDOPERREFNOUP_ENDDSP2 0x104DDC +#define YRAMINDOPERREFUP_STARTDSP2 0x104DE0 +#define YRAMINDOPERREFUP_ENDDSP2 0x104DFC +#define DSP2CONDCODE 0x104E00 +#define DSP2STACKFLAG 0x104E04 +#define DSP2PROGCOUNTSTACKPTREG 0x104E08 +#define DSP2PROGCOUNTSTACKDATAREG 0x104E0C +#define DSP2CURLOOPADDRREG 0x104E10 +#define DSP2CURLOOPCOUNT 0x104E14 +#define DSP2TOPLOOPCOUNTSTACK 0x104E18 +#define DSP2TOPLOOPADDRSTACK 0x104E1C +#define DSP2LOOPSTACKPTR 0x104E20 +#define DSP2STASSTACKDATAREG 0x104E24 +#define DSP2STASSTACKPTR 0x104E28 +#define DSP2PROGCOUNT 0x104E2C +#define DSP2XYRAMBASE_START 0x104EA0 +#define DSP2XYRAMBASE_END 0x104EBC +#define DSP2XYRAMLENG_START 0x104EC0 +#define DSP2XYRAMLENG_END 0x104EDC +#define SEMAPHOREREGDSP2 0x104EE0 +#define DSP2INTCONTMASKREG 0x104EE4 +#define DSP2INTCONTPENDREG 0x104EE8 +#define DSP2INTCONTSERVINT 0x104EEC +#define GPIODSP2 0x104EFC +#define DMADSPBASEADDRREG_STARTDSP2 0x104F00 +#define DMADSPBASEADDRREG_ENDDSP2 0x104F1C +#define DMAHOSTBASEADDRREG_STARTDSP2 0x104F20 +#define DMAHOSTBASEADDRREG_ENDDSP2 0x104F3C +#define DMADSPCURADDRREG_STARTDSP2 0x104F40 +#define DMADSPCURADDRREG_ENDDSP2 0x104F5C +#define DMAHOSTCURADDRREG_STARTDSP2 0x104F60 +#define DMAHOSTCURADDRREG_ENDDSP2 0x104F7C +#define DMATANXCOUNTREG_STARTDSP2 0x104F80 +#define DMATANXCOUNTREG_ENDDSP2 0x104F9C +#define DMATIMEBUGREG_STARTDSP2 0x104FA0 +#define DMATIMEBUGREG_ENDDSP2 0x104FAC +#define DMACNTLMODFREG_STARTDSP2 0x104FA0 +#define DMACNTLMODFREG_ENDDSP2 0x104FAC + +#define DMAGLOBSTATSREGDSP2 0x104FEC +#define DSP2XGPRAM_START 0x105000 +#define DSP2XGPRAM_END 0x1051FC +#define DSP2YGPRAM_START 0x105800 +#define DSP2YGPRAM_END 0x1059FC + + + +#define AUDIORINGIPDSP3_START 0x106000 +#define AUDIORINGIPDSP3_END 0x1063FC +#define AUDIORINGOPDSP3_START 0x106400 +#define AUDIORINGOPDSP3_END 0x1067FC +#define AUDPARARINGIODSP3_START 0x106800 +#define AUDPARARINGIODSP3_END 0x106BFC +#define DSP3LOCALHWREG_START 0x106C00 +#define DSP3LOCALHWREG_END 0x106C3C +#define DSP3XYRAMAGINDEX_START 0x106C40 +#define DSP3XYRAMAGINDEX_END 0x106C5C +#define DSP3XYRAMAGMDFR_START 0x106C60 +#define DSP3XYRAMAGMDFR_END 0x106C7C +#define DSP3INTCONTLVEC_START 0x106C80 +#define DSP3INTCONTLVEC_END 0x106CD8 +#define HOSTINTFPORTADDRCONTDSP3 0x106D40 +#define HOSTINTFPORTDATADSP3 0x106D44 +#define TIME0PERENBDSP3 0x106D60 +#define TIME0COUNTERDSP3 0x106D64 +#define TIME1PERENBDSP3 0x106D68 +#define TIME1COUNTERDSP3 0x106D6C +#define TIME2PERENBDSP3 0x106D70 +#define TIME2COUNTERDSP3 0x106D74 +#define TIME3PERENBDSP3 0x106D78 +#define TIME3COUNTERDSP3 0x106D7C +#define XRAMINDOPERREFNOUP_STARTDSP3 0x106D80 +#define XRAMINDOPERREFNOUP_ENDDSP3 0x106D9C +#define XRAMINDOPERREFUP_STARTDSP3 0x106DA0 +#define XRAMINDOPERREFUP_ENDDSP3 0x106DBC +#define YRAMINDOPERREFNOUP_STARTDSP3 0x106DC0 +#define YRAMINDOPERREFNOUP_ENDDSP3 0x106DDC +#define YRAMINDOPERREFUP_STARTDSP3 0x106DE0 +#define YRAMINDOPERREFUP_ENDDSP3 0x100DFC + +#define DSP3CONDCODE 0x106E00 +#define DSP3STACKFLAG 0x106E04 +#define DSP3PROGCOUNTSTACKPTREG 0x106E08 +#define DSP3PROGCOUNTSTACKDATAREG 0x106E0C +#define DSP3CURLOOPADDRREG 0x106E10 +#define DSP3CURLOOPCOUNT 0x106E14 +#define DSP3TOPLOOPCOUNTSTACK 0x106E18 +#define DSP3TOPLOOPADDRSTACK 0x106E1C +#define DSP3LOOPSTACKPTR 0x106E20 +#define DSP3STASSTACKDATAREG 0x106E24 +#define DSP3STASSTACKPTR 0x106E28 +#define DSP3PROGCOUNT 0x106E2C +#define DSP3XYRAMBASE_START 0x106EA0 +#define DSP3XYRAMBASE_END 0x106EBC +#define DSP3XYRAMLENG_START 0x106EC0 +#define DSP3XYRAMLENG_END 0x106EDC +#define SEMAPHOREREGDSP3 0x106EE0 +#define DSP3INTCONTMASKREG 0x106EE4 +#define DSP3INTCONTPENDREG 0x106EE8 +#define DSP3INTCONTSERVINT 0x106EEC +#define GPIODSP3 0x106EFC +#define DMADSPBASEADDRREG_STARTDSP3 0x106F00 +#define DMADSPBASEADDRREG_ENDDSP3 0x106F1C +#define DMAHOSTBASEADDRREG_STARTDSP3 0x106F20 +#define DMAHOSTBASEADDRREG_ENDDSP3 0x106F3C +#define DMADSPCURADDRREG_STARTDSP3 0x106F40 +#define DMADSPCURADDRREG_ENDDSP3 0x106F5C +#define DMAHOSTCURADDRREG_STARTDSP3 0x106F60 +#define DMAHOSTCURADDRREG_ENDDSP3 0x106F7C +#define DMATANXCOUNTREG_STARTDSP3 0x106F80 +#define DMATANXCOUNTREG_ENDDSP3 0x106F9C +#define DMATIMEBUGREG_STARTDSP3 0x106FA0 +#define DMATIMEBUGREG_ENDDSP3 0x106FAC +#define DMACNTLMODFREG_STARTDSP3 0x106FA0 +#define DMACNTLMODFREG_ENDDSP3 0x106FAC + +#define DMAGLOBSTATSREGDSP3 0x106FEC +#define DSP3XGPRAM_START 0x107000 +#define DSP3XGPRAM_END 0x1071FC +#define DSP3YGPRAM_START 0x107800 +#define DSP3YGPRAM_END 0x1079FC + +/* end of DSP reg definitions */ + +#define DSPAIMAP_START 0x108000 +#define DSPAIMAP_END 0x1083FC +#define DSPPIMAP_START 0x108400 +#define DSPPIMAP_END 0x1087FC +#define DSPPOMAP_START 0x108800 +#define DSPPOMAP_END 0x108BFC +#define DSPPOCTL 0x108C00 +#define TKCTL_START 0x110000 +#define TKCTL_END 0x110FFC +#define TKCC_START 0x111000 +#define TKCC_END 0x111FFC +#define TKIMAP_START 0x112000 +#define TKIMAP_END 0x112FFC +#define TKDCTR16 0x113000 +#define TKPB16 0x113004 +#define TKBS16 0x113008 +#define TKDCTR32 0x11300C +#define TKPB32 0x113010 +#define TKBS32 0x113014 +#define ICDCTR16 0x113018 +#define ITBS16 0x11301C +#define ICDCTR32 0x113020 +#define ITBS32 0x113024 +#define ITSTART 0x113028 +#define TKSQ 0x11302C + +#define TKSCCTL_START 0x114000 +#define TKSCCTL_END 0x11403C +#define TKSCADR_START 0x114100 +#define TKSCADR_END 0x11413C +#define TKSCDATAX_START 0x114800 +#define TKSCDATAX_END 0x1149FC +#define TKPCDATAX_START 0x120000 +#define TKPCDATAX_END 0x12FFFC + +#define MALSA 0x130000 +#define MAPPHA 0x130004 +#define MAPPLA 0x130008 +#define MALSB 0x130010 +#define MAPPHB 0x130014 +#define MAPPLB 0x130018 + +#define TANSPORTMAPABREGS_START 0x130020 +#define TANSPORTMAPABREGS_END 0x13A2FC + +#define PTPAHX 0x13B000 +#define PTPALX 0x13B004 + +#define TANSPPAGETABLEPHYADDR015_START 0x13B008 +#define TANSPPAGETABLEPHYADDR015_END 0x13B07C +#define TRNQADRX_START 0x13B100 +#define TRNQADRX_END 0x13B13C +#define TRNQTIMX_START 0x13B200 +#define TRNQTIMX_END 0x13B23C +#define TRNQAPARMX_START 0x13B300 +#define TRNQAPARMX_END 0x13B33C + +#define TRNQCNT 0x13B400 +#define TRNCTL 0x13B404 +#define TRNIS 0x13B408 +#define TRNCURTS 0x13B40C + +#define AMOP_START 0x140000 +#define AMOPLO 0x140000 +#define AMOPHI 0x140004 +#define AMOP_END 0x147FFC +#define PMOP_START 0x148000 +#define PMOPLO 0x148000 +#define PMOPHI 0x148004 +#define PMOP_END 0x14FFFC +#define PCURR_START 0x150000 +#define PCURR_END 0x153FFC +#define PTRAG_START 0x154000 +#define PTRAG_END 0x157FFC +#define PSR_START 0x158000 +#define PSR_END 0x15BFFC + +#define PFSTAT4SEG_START 0x160000 +#define PFSTAT4SEG_END 0x160BFC +#define PFSTAT2SEG_START 0x160C00 +#define PFSTAT2SEG_END 0x1617FC +#define PFTARG4SEG_START 0x164000 +#define PFTARG4SEG_END 0x164BFC +#define PFTARG2SEG_START 0x164C00 +#define PFTARG2SEG_END 0x1657FC +#define PFSR4SEG_START 0x168000 +#define PFSR4SEG_END 0x168BFC +#define PFSR2SEG_START 0x168C00 +#define PFSR2SEG_END 0x1697FC +#define PCURRMS4SEG_START 0x16C000 +#define PCURRMS4SEG_END 0x16CCFC +#define PCURRMS2SEG_START 0x16CC00 +#define PCURRMS2SEG_END 0x16D7FC +#define PTARGMS4SEG_START 0x170000 +#define PTARGMS4SEG_END 0x172FFC +#define PTARGMS2SEG_START 0x173000 +#define PTARGMS2SEG_END 0x1747FC +#define PSRMS4SEG_START 0x170000 +#define PSRMS4SEG_END 0x172FFC +#define PSRMS2SEG_START 0x173000 +#define PSRMS2SEG_END 0x1747FC + +#define PRING_LO_START 0x190000 +#define PRING_LO_END 0x193FFC +#define PRING_HI_START 0x194000 +#define PRING_HI_END 0x197FFC +#define PRING_LO_HI_START 0x198000 +#define PRING_LO_HI 0x198000 +#define PRING_LO_HI_END 0x19BFFC + +#define PINTFIFO 0x1A0000 +#define SRCCTL 0x1B0000 +#define SRCCCR 0x1B0004 +#define SRCIMAP 0x1B0008 +#define SRCODDC 0x1B000C +#define SRCCA 0x1B0010 +#define SRCCF 0x1B0014 +#define SRCSA 0x1B0018 +#define SRCLA 0x1B001C +#define SRCCTLSWR 0x1B0020 + +/* SRC HERE */ +#define SRCALBA 0x1B002C +#define SRCMCTL 0x1B012C +#define SRCCERR 0x1B022C +#define SRCITB 0x1B032C +#define SRCIPM 0x1B082C +#define SRCIP 0x1B102C +#define SRCENBSTAT 0x1B202C +#define SRCENBLO 0x1B212C +#define SRCENBHI 0x1B222C +#define SRCENBS 0x1B232C +#define SRCENB 0x1B282C +#define SRCENB07 0x1B282C +#define SRCENBS07 0x1B302C + +#define SRCDN0Z 0x1B0030 +#define SRCDN0Z0 0x1B0030 +#define SRCDN0Z1 0x1B0034 +#define SRCDN0Z2 0x1B0038 +#define SRCDN0Z3 0x1B003C +#define SRCDN1Z 0x1B0040 +#define SRCDN1Z0 0x1B0040 +#define SRCDN1Z1 0x1B0044 +#define SRCDN1Z2 0x1B0048 +#define SRCDN1Z3 0x1B004C +#define SRCDN1Z4 0x1B0050 +#define SRCDN1Z5 0x1B0054 +#define SRCDN1Z6 0x1B0058 +#define SRCDN1Z7 0x1B005C +#define SRCUPZ 0x1B0060 +#define SRCUPZ0 0x1B0060 +#define SRCUPZ1 0x1B0064 +#define SRCUPZ2 0x1B0068 +#define SRCUPZ3 0x1B006C +#define SRCUPZ4 0x1B0070 +#define SRCUPZ5 0x1B0074 +#define SRCUPZ6 0x1B0078 +#define SRCUPZ7 0x1B007C +#define SRCCD0 0x1B0080 +#define SRCCD1 0x1B0084 +#define SRCCD2 0x1B0088 +#define SRCCD3 0x1B008C +#define SRCCD4 0x1B0090 +#define SRCCD5 0x1B0094 +#define SRCCD6 0x1B0098 +#define SRCCD7 0x1B009C +#define SRCCD8 0x1B00A0 +#define SRCCD9 0x1B00A4 +#define SRCCDA 0x1B00A8 +#define SRCCDB 0x1B00AC +#define SRCCDC 0x1B00B0 +#define SRCCDD 0x1B00B4 +#define SRCCDE 0x1B00B8 +#define SRCCDF 0x1B00BC +#define SRCCD10 0x1B00C0 +#define SRCCD11 0x1B00C4 +#define SRCCD12 0x1B00C8 +#define SRCCD13 0x1B00CC +#define SRCCD14 0x1B00D0 +#define SRCCD15 0x1B00D4 +#define SRCCD16 0x1B00D8 +#define SRCCD17 0x1B00DC +#define SRCCD18 0x1B00E0 +#define SRCCD19 0x1B00E4 +#define SRCCD1A 0x1B00E8 +#define SRCCD1B 0x1B00EC +#define SRCCD1C 0x1B00F0 +#define SRCCD1D 0x1B00F4 +#define SRCCD1E 0x1B00F8 +#define SRCCD1F 0x1B00FC + +#define SRCCONTRBLOCK_START 0x1B0100 +#define SRCCONTRBLOCK_END 0x1BFFFC +#define FILTOP_START 0x1C0000 +#define FILTOP_END 0x1C05FC +#define FILTIMAP_START 0x1C0800 +#define FILTIMAP_END 0x1C0DFC +#define FILTZ1_START 0x1C1000 +#define FILTZ1_END 0x1C15FC +#define FILTZ2_START 0x1C1800 +#define FILTZ2_END 0x1C1DFC +#define DAOIMAP_START 0x1C5000 +#define DAOIMAP 0x1C5000 +#define DAOIMAP_END 0x1C5124 + +#define AC97D 0x1C5400 +#define AC97A 0x1C5404 +#define AC97CTL 0x1C5408 +#define I2SCTL 0x1C5420 + +#define SPOS 0x1C5440 +#define SPOSA 0x1C5440 +#define SPOSB 0x1C5444 +#define SPOSC 0x1C5448 +#define SPOSD 0x1C544C + +#define SPISA 0x1C5450 +#define SPISB 0x1C5454 +#define SPISC 0x1C5458 +#define SPISD 0x1C545C + +#define SPFSCTL 0x1C5460 + +#define SPFS0 0x1C5468 +#define SPFS1 0x1C546C +#define SPFS2 0x1C5470 +#define SPFS3 0x1C5474 +#define SPFS4 0x1C5478 +#define SPFS5 0x1C547C + +#define SPOCTL 0x1C5480 +#define SPICTL 0x1C5484 +#define SPISTS 0x1C5488 +#define SPINTP 0x1C548C +#define SPINTE 0x1C5490 +#define SPUTCTLAB 0x1C5494 +#define SPUTCTLCD 0x1C5498 + +#define SRTSPA 0x1C54C0 +#define SRTSPB 0x1C54C4 +#define SRTSPC 0x1C54C8 +#define SRTSPD 0x1C54CC + +#define SRTSCTL 0x1C54D0 +#define SRTSCTLA 0x1C54D0 +#define SRTSCTLB 0x1C54D4 +#define SRTSCTLC 0x1C54D8 +#define SRTSCTLD 0x1C54DC + +#define SRTI2S 0x1C54E0 +#define SRTICTL 0x1C54F0 + +#define WC 0x1C6000 +#define TIMR 0x1C6004 + +#define GIP 0x1C6010 +#define GIE 0x1C6014 +#define DIE 0x1C6018 +#define DIC 0x1C601C +#define GPIO 0x1C6020 +#define GPIOCTL 0x1C6024 +#define GPIP 0x1C6028 +#define GPIE 0x1C602C +#define DSPINT0 0x1C6030 +#define DSPEIOC 0x1C6034 +#define MUADAT 0x1C6040 +#define MUACMD 0x1C6044 +#define MUASTAT 0x1C6044 +#define MUBDAT 0x1C6048 +#define MUBCMD 0x1C604C +#define MUBSTAT 0x1C604C +#define UARTCMA 0x1C6050 +#define UARTCMB 0x1C6054 +#define UARTIP 0x1C6058 +#define UARTIE 0x1C605C +#define PLLCTL 0x1C6060 +#define PLLDCD 0x1C6064 +#define GCTL 0x1C6070 +#define ID0 0x1C6080 +#define ID1 0x1C6084 +#define ID2 0x1C6088 +#define ID3 0x1C608C +#define SDRCTL 0x1C7000 + + +#define I2SA_L 0x0L +#define I2SA_R 0x1L +#define I2SB_L 0x8L +#define I2SB_R 0x9L +#define I2SC_L 0x10L +#define I2SC_R 0x11L +#define I2SD_L 0x18L +#define I2SD_R 0x19L + +#endif /* CT20K1REG_H */ + + diff --git a/sound/pci/ctxfi/ct20k2reg.h b/sound/pci/ctxfi/ct20k2reg.h new file mode 100644 index 0000000..2d07986 --- /dev/null +++ b/sound/pci/ctxfi/ct20k2reg.h @@ -0,0 +1,85 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + */ + +#ifndef _20K2REGISTERS_H_ +#define _20K2REGISTERS_H_ + + +/* Timer Registers */ +#define TIMER_TIMR 0x1B7004 +#define INTERRUPT_GIP 0x1B7010 +#define INTERRUPT_GIE 0x1B7014 + +/* I2C Registers */ +#define I2C_IF_ADDRESS 0x1B9000 +#define I2C_IF_WDATA 0x1B9004 +#define I2C_IF_RDATA 0x1B9008 +#define I2C_IF_STATUS 0x1B900C +#define I2C_IF_WLOCK 0x1B9010 + +/* Global Control Registers */ +#define GLOBAL_CNTL_GCTL 0x1B7090 + +/* PLL Registers */ +#define PLL_CTL 0x1B7080 +#define PLL_STAT 0x1B7084 +#define PLL_ENB 0x1B7088 + +/* SRC Registers */ +#define SRC_CTL 0x1A0000 /* 0x1A0000 + (256 * Chn) */ +#define SRC_CCR 0x1A0004 /* 0x1A0004 + (256 * Chn) */ +#define SRC_IMAP 0x1A0008 /* 0x1A0008 + (256 * Chn) */ +#define SRC_CA 0x1A0010 /* 0x1A0010 + (256 * Chn) */ +#define SRC_CF 0x1A0014 /* 0x1A0014 + (256 * Chn) */ +#define SRC_SA 0x1A0018 /* 0x1A0018 + (256 * Chn) */ +#define SRC_LA 0x1A001C /* 0x1A001C + (256 * Chn) */ +#define SRC_CTLSWR 0x1A0020 /* 0x1A0020 + (256 * Chn) */ +#define SRC_CD 0x1A0080 /* 0x1A0080 + (256 * Chn) + (4 * Regn) */ +#define SRC_MCTL 0x1A012C +#define SRC_IP 0x1A102C /* 0x1A102C + (256 * Regn) */ +#define SRC_ENB 0x1A282C /* 0x1A282C + (256 * Regn) */ +#define SRC_ENBSTAT 0x1A202C +#define SRC_ENBSA 0x1A232C +#define SRC_DN0Z 0x1A0030 +#define SRC_DN1Z 0x1A0040 +#define SRC_UPZ 0x1A0060 + +/* GPIO Registers */ +#define GPIO_DATA 0x1B7020 +#define GPIO_CTRL 0x1B7024 + +/* Virtual memory registers */ +#define VMEM_PTPAL 0x1C6300 /* 0x1C6300 + (16 * Chn) */ +#define VMEM_PTPAH 0x1C6304 /* 0x1C6304 + (16 * Chn) */ +#define VMEM_CTL 0x1C7000 + +/* Transport Registers */ +#define TRANSPORT_ENB 0x1B6000 +#define TRANSPORT_CTL 0x1B6004 +#define TRANSPORT_INT 0x1B6008 + +/* Audio IO */ +#define AUDIO_IO_AIM 0x1B5000 /* 0x1B5000 + (0x04 * Chn) */ +#define AUDIO_IO_TX_CTL 0x1B5400 /* 0x1B5400 + (0x40 * Chn) */ +#define AUDIO_IO_TX_CSTAT_L 0x1B5408 /* 0x1B5408 + (0x40 * Chn) */ +#define AUDIO_IO_TX_CSTAT_H 0x1B540C /* 0x1B540C + (0x40 * Chn) */ +#define AUDIO_IO_RX_CTL 0x1B5410 /* 0x1B5410 + (0x40 * Chn) */ +#define AUDIO_IO_RX_SRT_CTL 0x1B5420 /* 0x1B5420 + (0x40 * Chn) */ +#define AUDIO_IO_MCLK 0x1B5600 +#define AUDIO_IO_TX_BLRCLK 0x1B5604 +#define AUDIO_IO_RX_BLRCLK 0x1B5608 + +/* Mixer */ +#define MIXER_AMOPLO 0x130000 /* 0x130000 + (8 * Chn) [4095 : 0] */ +#define MIXER_AMOPHI 0x130004 /* 0x130004 + (8 * Chn) [4095 : 0] */ +#define MIXER_PRING_LO_HI 0x188000 /* 0x188000 + (4 * Chn) [4095 : 0] */ +#define MIXER_PMOPLO 0x138000 /* 0x138000 + (8 * Chn) [4095 : 0] */ +#define MIXER_PMOPHI 0x138004 /* 0x138004 + (8 * Chn) [4095 : 0] */ +#define MIXER_AR_ENABLE 0x19000C + +#endif diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c new file mode 100644 index 0000000..119791a --- /dev/null +++ b/sound/pci/ctxfi/ctamixer.c @@ -0,0 +1,488 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctamixer.c + * + * @Brief + * This file contains the implementation of the Audio Mixer + * resource management object. + * + * @Author Liu Chun + * @Date May 21 2008 + * + */ + +#include "ctamixer.h" +#include "cthardware.h" +#include + +#define AMIXER_RESOURCE_NUM 256 +#define SUM_RESOURCE_NUM 256 + +#define AMIXER_Y_IMMEDIATE 1 + +#define BLANK_SLOT 4094 + +static int amixer_master(struct rsc *rsc) +{ + rsc->conj = 0; + return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; +} + +static int amixer_next_conj(struct rsc *rsc) +{ + rsc->conj++; + return container_of(rsc, struct amixer, rsc)->idx[rsc->conj]; +} + +static int amixer_index(const struct rsc *rsc) +{ + return container_of(rsc, struct amixer, rsc)->idx[rsc->conj]; +} + +static int amixer_output_slot(const struct rsc *rsc) +{ + return (amixer_index(rsc) << 4) + 0x4; +} + +static struct rsc_ops amixer_basic_rsc_ops = { + .master = amixer_master, + .next_conj = amixer_next_conj, + .index = amixer_index, + .output_slot = amixer_output_slot, +}; + +static int amixer_set_input(struct amixer *amixer, struct rsc *rsc) +{ + struct hw *hw = NULL; + + hw = (struct hw *)amixer->rsc.hw; + hw->amixer_set_mode(amixer->rsc.ctrl_blk, AMIXER_Y_IMMEDIATE); + amixer->input = rsc; + if (NULL == rsc) + hw->amixer_set_x(amixer->rsc.ctrl_blk, BLANK_SLOT); + else + hw->amixer_set_x(amixer->rsc.ctrl_blk, + rsc->ops->output_slot(rsc)); + + return 0; +} + +/* y is a 14-bit immediate constant */ +static int amixer_set_y(struct amixer *amixer, unsigned int y) +{ + struct hw *hw = NULL; + + hw = (struct hw *)amixer->rsc.hw; + hw->amixer_set_y(amixer->rsc.ctrl_blk, y); + + return 0; +} + +static int amixer_set_invalid_squash(struct amixer *amixer, unsigned int iv) +{ + struct hw *hw = NULL; + + hw = (struct hw *)amixer->rsc.hw; + hw->amixer_set_iv(amixer->rsc.ctrl_blk, iv); + + return 0; +} + +static int amixer_set_sum(struct amixer *amixer, struct sum *sum) +{ + struct hw *hw = NULL; + + hw = (struct hw *)amixer->rsc.hw; + amixer->sum = sum; + if (NULL == sum) { + hw->amixer_set_se(amixer->rsc.ctrl_blk, 0); + } else { + hw->amixer_set_se(amixer->rsc.ctrl_blk, 1); + hw->amixer_set_sadr(amixer->rsc.ctrl_blk, + sum->rsc.ops->index(&sum->rsc)); + } + + return 0; +} + +static int amixer_commit_write(struct amixer *amixer) +{ + struct hw *hw = NULL; + unsigned int index = 0; + int i = 0; + struct rsc *input = NULL; + struct sum *sum = NULL; + + hw = (struct hw *)amixer->rsc.hw; + input = amixer->input; + sum = amixer->sum; + + /* Program master and conjugate resources */ + amixer->rsc.ops->master(&amixer->rsc); + if (NULL != input) + input->ops->master(input); + + if (NULL != sum) + sum->rsc.ops->master(&sum->rsc); + + for (i = 0; i < amixer->rsc.msr; i++) { + hw->amixer_set_dirty_all(amixer->rsc.ctrl_blk); + if (NULL != input) { + hw->amixer_set_x(amixer->rsc.ctrl_blk, + input->ops->output_slot(input)); + input->ops->next_conj(input); + } + if (NULL != sum) { + hw->amixer_set_sadr(amixer->rsc.ctrl_blk, + sum->rsc.ops->index(&sum->rsc)); + sum->rsc.ops->next_conj(&sum->rsc); + } + index = amixer->rsc.ops->output_slot(&amixer->rsc); + hw->amixer_commit_write(hw, index, amixer->rsc.ctrl_blk); + amixer->rsc.ops->next_conj(&amixer->rsc); + } + amixer->rsc.ops->master(&amixer->rsc); + if (NULL != input) + input->ops->master(input); + + if (NULL != sum) + sum->rsc.ops->master(&sum->rsc); + + return 0; +} + +static int amixer_commit_raw_write(struct amixer *amixer) +{ + struct hw *hw = NULL; + unsigned int index = 0; + + hw = (struct hw *)amixer->rsc.hw; + index = amixer->rsc.ops->output_slot(&amixer->rsc); + hw->amixer_commit_write(hw, index, amixer->rsc.ctrl_blk); + + return 0; +} + +static int amixer_get_y(struct amixer *amixer) +{ + struct hw *hw = NULL; + + hw = (struct hw *)amixer->rsc.hw; + return hw->amixer_get_y(amixer->rsc.ctrl_blk); +} + +static int amixer_setup(struct amixer *amixer, struct rsc *input, + unsigned int scale, struct sum *sum) +{ + amixer_set_input(amixer, input); + amixer_set_y(amixer, scale); + amixer_set_sum(amixer, sum); + amixer_commit_write(amixer); + return 0; +} + +static struct amixer_rsc_ops amixer_ops = { + .set_input = amixer_set_input, + .set_invalid_squash = amixer_set_invalid_squash, + .set_scale = amixer_set_y, + .set_sum = amixer_set_sum, + .commit_write = amixer_commit_write, + .commit_raw_write = amixer_commit_raw_write, + .setup = amixer_setup, + .get_scale = amixer_get_y, +}; + +static int amixer_rsc_init(struct amixer *amixer, + const struct amixer_desc *desc, + struct amixer_mgr *mgr) +{ + int err = 0; + + err = rsc_init(&amixer->rsc, amixer->idx[0], + AMIXER, desc->msr, mgr->mgr.hw); + if (err) + return err; + + /* Set amixer specific operations */ + amixer->rsc.ops = &amixer_basic_rsc_ops; + amixer->ops = &amixer_ops; + amixer->input = NULL; + amixer->sum = NULL; + + amixer_setup(amixer, NULL, 0, NULL); + + return 0; +} + +static int amixer_rsc_uninit(struct amixer *amixer) +{ + amixer_setup(amixer, NULL, 0, NULL); + rsc_uninit(&amixer->rsc); + amixer->ops = NULL; + amixer->input = NULL; + amixer->sum = NULL; + return 0; +} + +static int get_amixer_rsc(struct amixer_mgr *mgr, + const struct amixer_desc *desc, + struct amixer **ramixer) +{ + int err = 0, i = 0; + unsigned int idx = 0; + struct amixer *amixer = NULL; + unsigned long flags; + + *ramixer = NULL; + + /* Allocate mem for amixer resource */ + amixer = kzalloc(sizeof(*amixer), GFP_KERNEL); + if (NULL == amixer) { + err = -ENOMEM; + return err; + } + + /* Check whether there are sufficient + * amixer resources to meet request. */ + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i = 0; i < desc->msr; i++) { + err = mgr_get_resource(&mgr->mgr, 1, &idx); + if (err) + break; + + amixer->idx[i] = idx; + } + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + if (err) { + printk(KERN_ERR "Can't meet AMIXER resource request!\n"); + goto error; + } + + err = amixer_rsc_init(amixer, desc, mgr); + if (err) + goto error; + + *ramixer = amixer; + + return 0; + +error: + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i--; i >= 0; i--) + mgr_put_resource(&mgr->mgr, 1, amixer->idx[i]); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + kfree(amixer); + return err; +} + +static int put_amixer_rsc(struct amixer_mgr *mgr, struct amixer *amixer) +{ + unsigned long flags; + int i = 0; + + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i = 0; i < amixer->rsc.msr; i++) + mgr_put_resource(&mgr->mgr, 1, amixer->idx[i]); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + amixer_rsc_uninit(amixer); + kfree(amixer); + + return 0; +} + +int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr) +{ + int err = 0; + struct amixer_mgr *amixer_mgr; + + *ramixer_mgr = NULL; + amixer_mgr = kzalloc(sizeof(*amixer_mgr), GFP_KERNEL); + if (NULL == amixer_mgr) + return -ENOMEM; + + err = rsc_mgr_init(&amixer_mgr->mgr, AMIXER, AMIXER_RESOURCE_NUM, hw); + if (err) + goto error; + + spin_lock_init(&amixer_mgr->mgr_lock); + + amixer_mgr->get_amixer = get_amixer_rsc; + amixer_mgr->put_amixer = put_amixer_rsc; + + *ramixer_mgr = amixer_mgr; + + return 0; + +error: + kfree(amixer_mgr); + return err; +} + +int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr) +{ + rsc_mgr_uninit(&amixer_mgr->mgr); + kfree(amixer_mgr); + return 0; +} + +/* SUM resource management */ + +static int sum_master(struct rsc *rsc) +{ + rsc->conj = 0; + return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; +} + +static int sum_next_conj(struct rsc *rsc) +{ + rsc->conj++; + return container_of(rsc, struct sum, rsc)->idx[rsc->conj]; +} + +static int sum_index(const struct rsc *rsc) +{ + return container_of(rsc, struct sum, rsc)->idx[rsc->conj]; +} + +static int sum_output_slot(const struct rsc *rsc) +{ + return (sum_index(rsc) << 4) + 0xc; +} + +static struct rsc_ops sum_basic_rsc_ops = { + .master = sum_master, + .next_conj = sum_next_conj, + .index = sum_index, + .output_slot = sum_output_slot, +}; + +static int sum_rsc_init(struct sum *sum, + const struct sum_desc *desc, + struct sum_mgr *mgr) +{ + int err = 0; + + err = rsc_init(&sum->rsc, sum->idx[0], SUM, desc->msr, mgr->mgr.hw); + if (err) + return err; + + sum->rsc.ops = &sum_basic_rsc_ops; + + return 0; +} + +static int sum_rsc_uninit(struct sum *sum) +{ + rsc_uninit(&sum->rsc); + return 0; +} + +static int get_sum_rsc(struct sum_mgr *mgr, + const struct sum_desc *desc, + struct sum **rsum) +{ + int err = 0, i = 0; + unsigned int idx = 0; + struct sum *sum = NULL; + unsigned long flags; + + *rsum = NULL; + + /* Allocate mem for sum resource */ + sum = kzalloc(sizeof(*sum), GFP_KERNEL); + if (NULL == sum) { + err = -ENOMEM; + return err; + } + + /* Check whether there are sufficient sum resources to meet request. */ + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i = 0; i < desc->msr; i++) { + err = mgr_get_resource(&mgr->mgr, 1, &idx); + if (err) + break; + + sum->idx[i] = idx; + } + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + if (err) { + printk(KERN_ERR "Can't meet SUM resource request!\n"); + goto error; + } + + err = sum_rsc_init(sum, desc, mgr); + if (err) + goto error; + + *rsum = sum; + + return 0; + +error: + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i--; i >= 0; i--) + mgr_put_resource(&mgr->mgr, 1, sum->idx[i]); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + kfree(sum); + return err; +} + +static int put_sum_rsc(struct sum_mgr *mgr, struct sum *sum) +{ + unsigned long flags; + int i = 0; + + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i = 0; i < sum->rsc.msr; i++) + mgr_put_resource(&mgr->mgr, 1, sum->idx[i]); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + sum_rsc_uninit(sum); + kfree(sum); + + return 0; +} + +int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr) +{ + int err = 0; + struct sum_mgr *sum_mgr; + + *rsum_mgr = NULL; + sum_mgr = kzalloc(sizeof(*sum_mgr), GFP_KERNEL); + if (NULL == sum_mgr) + return -ENOMEM; + + err = rsc_mgr_init(&sum_mgr->mgr, SUM, SUM_RESOURCE_NUM, hw); + if (err) + goto error; + + spin_lock_init(&sum_mgr->mgr_lock); + + sum_mgr->get_sum = get_sum_rsc; + sum_mgr->put_sum = put_sum_rsc; + + *rsum_mgr = sum_mgr; + + return 0; + +error: + kfree(sum_mgr); + return err; +} + +int sum_mgr_destroy(struct sum_mgr *sum_mgr) +{ + rsc_mgr_uninit(&sum_mgr->mgr); + kfree(sum_mgr); + return 0; +} + diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h new file mode 100644 index 0000000..cc49e5a --- /dev/null +++ b/sound/pci/ctxfi/ctamixer.h @@ -0,0 +1,96 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctamixer.h + * + * @Brief + * This file contains the definition of the Audio Mixer + * resource management object. + * + * @Author Liu Chun + * @Date May 21 2008 + * + */ + +#ifndef CTAMIXER_H +#define CTAMIXER_H + +#include "ctresource.h" +#include + +/* Define the descriptor of a summation node resource */ +struct sum { + struct rsc rsc; /* Basic resource info */ + unsigned char idx[8]; +}; + +/* Define sum resource request description info */ +struct sum_desc { + unsigned int msr; +}; + +struct sum_mgr { + struct rsc_mgr mgr; /* Basic resource manager info */ + spinlock_t mgr_lock; + + /* request one sum resource */ + int (*get_sum)(struct sum_mgr *mgr, + const struct sum_desc *desc, struct sum **rsum); + /* return one sum resource */ + int (*put_sum)(struct sum_mgr *mgr, struct sum *sum); +}; + +/* Constructor and destructor of daio resource manager */ +int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr); +int sum_mgr_destroy(struct sum_mgr *sum_mgr); + +/* Define the descriptor of a amixer resource */ +struct amixer_rsc_ops; + +struct amixer { + struct rsc rsc; /* Basic resource info */ + unsigned char idx[8]; + struct rsc *input; /* pointer to a resource acting as source */ + struct sum *sum; /* Put amixer output to this summation node */ + struct amixer_rsc_ops *ops; /* AMixer specific operations */ +}; + +struct amixer_rsc_ops { + int (*set_input)(struct amixer *amixer, struct rsc *rsc); + int (*set_scale)(struct amixer *amixer, unsigned int scale); + int (*set_invalid_squash)(struct amixer *amixer, unsigned int iv); + int (*set_sum)(struct amixer *amixer, struct sum *sum); + int (*commit_write)(struct amixer *amixer); + /* Only for interleaved recording */ + int (*commit_raw_write)(struct amixer *amixer); + int (*setup)(struct amixer *amixer, struct rsc *input, + unsigned int scale, struct sum *sum); + int (*get_scale)(struct amixer *amixer); +}; + +/* Define amixer resource request description info */ +struct amixer_desc { + unsigned int msr; +}; + +struct amixer_mgr { + struct rsc_mgr mgr; /* Basic resource manager info */ + spinlock_t mgr_lock; + + /* request one amixer resource */ + int (*get_amixer)(struct amixer_mgr *mgr, + const struct amixer_desc *desc, + struct amixer **ramixer); + /* return one amixer resource */ + int (*put_amixer)(struct amixer_mgr *mgr, struct amixer *amixer); +}; + +/* Constructor and destructor of amixer resource manager */ +int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr); +int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr); + +#endif /* CTAMIXER_H */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c new file mode 100644 index 0000000..5f35374 --- /dev/null +++ b/sound/pci/ctxfi/ctatc.c @@ -0,0 +1,1605 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctatc.c + * + * @Brief + * This file contains the implementation of the device resource management + * object. + * + * @Author Liu Chun + * @Date Mar 28 2008 + */ + +#include "ctatc.h" +#include "ctpcm.h" +#include "ctmixer.h" +#include "ctdrv.h" +#include "cthardware.h" +#include "ctsrc.h" +#include "ctamixer.h" +#include "ctdaio.h" +#include +#include +#include +#include + +#define MONO_SUM_SCALE 0x19a8 /* 2^(-0.5) in 14-bit floating format */ +#define DAIONUM 7 +#define MAX_MULTI_CHN 8 + +#define IEC958_DEFAULT_CON ((IEC958_AES0_NONAUDIO \ + | IEC958_AES0_CON_NOT_COPYRIGHT) \ + | ((IEC958_AES1_CON_MIXER \ + | IEC958_AES1_CON_ORIGINAL) << 8) \ + | (0x10 << 16) \ + | ((IEC958_AES3_CON_FS_48000) << 24)) + +static const struct ct_atc_chip_sub_details atc_sub_details[NUM_CTCARDS] = { + [CTSB0760] = {.subsys = PCI_SUBSYS_CREATIVE_SB0760, + .nm_model = "SB076x"}, + [CTHENDRIX] = {.subsys = PCI_SUBSYS_CREATIVE_HENDRIX, + .nm_model = "Hendrix"}, + [CTSB08801] = {.subsys = PCI_SUBSYS_CREATIVE_SB08801, + .nm_model = "SB0880"}, + [CTSB08802] = {.subsys = PCI_SUBSYS_CREATIVE_SB08802, + .nm_model = "SB0880"}, + [CTSB08803] = {.subsys = PCI_SUBSYS_CREATIVE_SB08803, + .nm_model = "SB0880"} +}; + +static struct ct_atc_chip_details atc_chip_details[] = { + {.vendor = PCI_VENDOR_CREATIVE, .device = PCI_DEVICE_CREATIVE_20K1, + .sub_details = NULL, + .nm_card = "X-Fi 20k1"}, + {.vendor = PCI_VENDOR_CREATIVE, .device = PCI_DEVICE_CREATIVE_20K2, + .sub_details = atc_sub_details, + .nm_card = "X-Fi 20k2"}, + {} /* terminator */ +}; + +static struct { + int (*create)(struct ct_atc *atc, + enum CTALSADEVS device, const char *device_name); + int (*destroy)(void *alsa_dev); + const char *public_name; +} alsa_dev_funcs[NUM_CTALSADEVS] = { + [FRONT] = { .create = ct_alsa_pcm_create, + .destroy = NULL, + .public_name = "Front/WaveIn"}, + [REAR] = { .create = ct_alsa_pcm_create, + .destroy = NULL, + .public_name = "Rear"}, + [CLFE] = { .create = ct_alsa_pcm_create, + .destroy = NULL, + .public_name = "Center/LFE"}, + [SURROUND] = { .create = ct_alsa_pcm_create, + .destroy = NULL, + .public_name = "Surround"}, + [IEC958] = { .create = ct_alsa_pcm_create, + .destroy = NULL, + .public_name = "IEC958 Non-audio"}, + + [MIXER] = { .create = ct_alsa_mix_create, + .destroy = NULL, + .public_name = "Mixer"} +}; + +typedef int (*create_t)(void *, void **); +typedef int (*destroy_t)(void *); + +static struct { + int (*create)(void *hw, void **rmgr); + int (*destroy)(void *mgr); +} rsc_mgr_funcs[NUM_RSCTYP] = { + [SRC] = { .create = (create_t)src_mgr_create, + .destroy = (destroy_t)src_mgr_destroy }, + [SRCIMP] = { .create = (create_t)srcimp_mgr_create, + .destroy = (destroy_t)srcimp_mgr_destroy }, + [AMIXER] = { .create = (create_t)amixer_mgr_create, + .destroy = (destroy_t)amixer_mgr_destroy }, + [SUM] = { .create = (create_t)sum_mgr_create, + .destroy = (destroy_t)sum_mgr_destroy }, + [DAIO] = { .create = (create_t)daio_mgr_create, + .destroy = (destroy_t)daio_mgr_destroy } +}; + +static int +atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm); + +/* * + * Only mono and interleaved modes are supported now. + * Always allocates a contiguous channel block. + * */ + +static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + unsigned long flags; + struct snd_pcm_runtime *runtime; + struct ct_vm *vm; + + if (NULL == apcm->substream) + return 0; + + runtime = apcm->substream->runtime; + vm = atc->vm; + + spin_lock_irqsave(&atc->vm_lock, flags); + apcm->vm_block = vm->map(vm, runtime->dma_area, runtime->dma_bytes); + spin_unlock_irqrestore(&atc->vm_lock, flags); + + if (NULL == apcm->vm_block) + return -ENOENT; + + return 0; +} + +static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + unsigned long flags; + struct ct_vm *vm; + + if (NULL == apcm->vm_block) + return; + + vm = atc->vm; + + spin_lock_irqsave(&atc->vm_lock, flags); + vm->unmap(vm, apcm->vm_block); + spin_unlock_irqrestore(&atc->vm_lock, flags); + + apcm->vm_block = NULL; +} + +static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index) +{ + struct ct_vm *vm; + void *kvirt_addr; + unsigned long phys_addr; + unsigned long flags; + + spin_lock_irqsave(&atc->vm_lock, flags); + vm = atc->vm; + kvirt_addr = vm->get_ptp_virt(vm, index); + if (kvirt_addr == NULL) + phys_addr = (~0UL); + else + phys_addr = virt_to_phys(kvirt_addr); + + spin_unlock_irqrestore(&atc->vm_lock, flags); + + return phys_addr; +} + +static unsigned int convert_format(snd_pcm_format_t snd_format) +{ + switch (snd_format) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + return SRC_SF_U8; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_U16_LE: + return SRC_SF_S16; + case SNDRV_PCM_FORMAT_S24_3LE: + return SRC_SF_S24; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + return SRC_SF_S32; + default: + printk(KERN_ERR "not recognized snd format is %d \n", + snd_format); + return SRC_SF_S16; + } +} + +static unsigned int +atc_get_pitch(unsigned int input_rate, unsigned int output_rate) +{ + unsigned int pitch = 0; + int b = 0; + + /* get pitch and convert to fixed-point 8.24 format. */ + pitch = (input_rate / output_rate) << 24; + input_rate %= output_rate; + input_rate /= 100; + output_rate /= 100; + for (b = 31; ((b >= 0) && !(input_rate >> b)); ) + b--; + + if (b >= 0) { + input_rate <<= (31 - b); + input_rate /= output_rate; + b = 24 - (31 - b); + if (b >= 0) + input_rate <<= b; + else + input_rate >>= -b; + + pitch |= input_rate; + } + + return pitch; +} + +static int select_rom(unsigned int pitch) +{ + if ((pitch > 0x00428f5c) && (pitch < 0x01b851ec)) { + /* 0.26 <= pitch <= 1.72 */ + return 1; + } else if ((0x01d66666 == pitch) || (0x01d66667 == pitch)) { + /* pitch == 1.8375 */ + return 2; + } else if (0x02000000 == pitch) { + /* pitch == 2 */ + return 3; + } else if ((pitch >= 0x0) && (pitch <= 0x08000000)) { + /* 0 <= pitch <= 8 */ + return 0; + } else { + return -ENOENT; + } +} + +static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src_mgr *src_mgr = atc->rsc_mgrs[SRC]; + struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; + struct src_desc desc = {0}; + struct amixer_desc mix_dsc = {0}; + struct src *src = NULL; + struct amixer *amixer = NULL; + int err = 0; + int n_amixer = apcm->substream->runtime->channels, i = 0; + int device = apcm->substream->pcm->device; + unsigned int pitch = 0; + unsigned long flags; + + if (NULL != apcm->src) { + /* Prepared pcm playback */ + return 0; + } + + /* Get SRC resource */ + desc.multi = apcm->substream->runtime->channels; + desc.msr = atc->msr; + desc.mode = MEMRD; + err = src_mgr->get_src(src_mgr, &desc, (struct src **)&apcm->src); + if (err) + goto error1; + + pitch = atc_get_pitch(apcm->substream->runtime->rate, + (atc->rsr * atc->msr)); + src = apcm->src; + src->ops->set_pitch(src, pitch); + src->ops->set_rom(src, select_rom(pitch)); + src->ops->set_sf(src, convert_format(apcm->substream->runtime->format)); + src->ops->set_pm(src, (src->ops->next_interleave(src) != NULL)); + + /* Get AMIXER resource */ + n_amixer = (n_amixer < 2) ? 2 : n_amixer; + apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); + if (NULL == apcm->amixers) { + err = -ENOMEM; + goto error1; + } + mix_dsc.msr = atc->msr; + for (i = 0, apcm->n_amixer = 0; i < n_amixer; i++) { + err = amixer_mgr->get_amixer(amixer_mgr, &mix_dsc, + (struct amixer **)&apcm->amixers[i]); + if (err) + goto error1; + + apcm->n_amixer++; + } + + /* Set up device virtual mem map */ + err = ct_map_audio_buffer(atc, apcm); + if (err < 0) + goto error1; + + /* Connect resources */ + src = apcm->src; + for (i = 0; i < n_amixer; i++) { + amixer = apcm->amixers[i]; + spin_lock_irqsave(&atc->atc_lock, flags); + amixer->ops->setup(amixer, &src->rsc, + INIT_VOL, atc->pcm[i+device*2]); + spin_unlock_irqrestore(&atc->atc_lock, flags); + src = src->ops->next_interleave(src); + if (NULL == src) + src = apcm->src; + } + + return 0; + +error1: + atc_pcm_release_resources(atc, apcm); + return err; +} + +static int +atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src_mgr *src_mgr = atc->rsc_mgrs[SRC]; + struct srcimp_mgr *srcimp_mgr = atc->rsc_mgrs[SRCIMP]; + struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; + struct sum_mgr *sum_mgr = atc->rsc_mgrs[SUM]; + struct srcimp *srcimp = NULL; + int i = 0; + + if (NULL != apcm->srcimps) { + for (i = 0; i < apcm->n_srcimp; i++) { + srcimp = apcm->srcimps[i]; + srcimp->ops->unmap(srcimp); + srcimp_mgr->put_srcimp(srcimp_mgr, srcimp); + apcm->srcimps[i] = NULL; + } + kfree(apcm->srcimps); + apcm->srcimps = NULL; + } + + if (NULL != apcm->srccs) { + for (i = 0; i < apcm->n_srcc; i++) { + src_mgr->put_src(src_mgr, apcm->srccs[i]); + apcm->srccs[i] = NULL; + } + kfree(apcm->srccs); + apcm->srccs = NULL; + } + + if (NULL != apcm->amixers) { + for (i = 0; i < apcm->n_amixer; i++) { + amixer_mgr->put_amixer(amixer_mgr, apcm->amixers[i]); + apcm->amixers[i] = NULL; + } + kfree(apcm->amixers); + apcm->amixers = NULL; + } + + if (NULL != apcm->mono) { + sum_mgr->put_sum(sum_mgr, apcm->mono); + apcm->mono = NULL; + } + + if (NULL != apcm->src) { + src_mgr->put_src(src_mgr, apcm->src); + apcm->src = NULL; + } + + if (NULL != apcm->vm_block) { + /* Undo device virtual mem map */ + ct_unmap_audio_buffer(atc, apcm); + apcm->vm_block = NULL; + } + + return 0; +} + +static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + unsigned int max_cisz = 0; + struct src *src = apcm->src; + + max_cisz = src->multi * src->rsc.msr; + max_cisz = 0x80 * (max_cisz < 8 ? max_cisz : 8); + + src->ops->set_sa(src, apcm->vm_block->addr); + src->ops->set_la(src, apcm->vm_block->addr + apcm->vm_block->size); + src->ops->set_ca(src, apcm->vm_block->addr + max_cisz); + src->ops->set_cisz(src, max_cisz); + + src->ops->set_bm(src, 1); + src->ops->set_state(src, SRC_STATE_INIT); + src->ops->commit_write(src); + + return 0; +} + +static int atc_pcm_stop(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src *src = NULL; + int i = 0; + + src = apcm->src; + src->ops->set_bm(src, 0); + src->ops->set_state(src, SRC_STATE_OFF); + src->ops->commit_write(src); + + if (NULL != apcm->srccs) { + for (i = 0; i < apcm->n_srcc; i++) { + src = apcm->srccs[i]; + src->ops->set_bm(src, 0); + src->ops->set_state(src, SRC_STATE_OFF); + src->ops->commit_write(src); + } + } + + apcm->started = 0; + + return 0; +} + +static int +atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src *src = apcm->src; + u32 size = 0, max_cisz = 0; + int position = 0; + + position = src->ops->get_ca(src); + + size = apcm->vm_block->size; + max_cisz = src->multi * src->rsc.msr; + max_cisz = 128 * (max_cisz < 8 ? max_cisz : 8); + + return (position + size - max_cisz - apcm->vm_block->addr) % size; +} + +struct src_node_conf_t { + unsigned int pitch; + unsigned int msr:8; + unsigned int mix_msr:8; + unsigned int imp_msr:8; + unsigned int vo:1; +}; + +static void setup_src_node_conf(struct ct_atc *atc, struct ct_atc_pcm *apcm, + struct src_node_conf_t *conf, int *n_srcc) +{ + unsigned int pitch = 0; + + /* get pitch and convert to fixed-point 8.24 format. */ + pitch = atc_get_pitch((atc->rsr * atc->msr), + apcm->substream->runtime->rate); + *n_srcc = 0; + + if (1 == atc->msr) { + *n_srcc = apcm->substream->runtime->channels; + conf[0].pitch = pitch; + conf[0].mix_msr = conf[0].imp_msr = conf[0].msr = 1; + conf[0].vo = 1; + } else if (2 == atc->msr) { + if (0x8000000 < pitch) { + /* Need two-stage SRCs, SRCIMPs and + * AMIXERs for converting format */ + conf[0].pitch = (atc->msr << 24); + conf[0].msr = conf[0].mix_msr = 1; + conf[0].imp_msr = atc->msr; + conf[0].vo = 0; + conf[1].pitch = atc_get_pitch(atc->rsr, + apcm->substream->runtime->rate); + conf[1].msr = conf[1].mix_msr = conf[1].imp_msr = 1; + conf[1].vo = 1; + *n_srcc = apcm->substream->runtime->channels * 2; + } else if (0x1000000 < pitch) { + /* Need one-stage SRCs, SRCIMPs and + * AMIXERs for converting format */ + conf[0].pitch = pitch; + conf[0].msr = conf[0].mix_msr + = conf[0].imp_msr = atc->msr; + conf[0].vo = 1; + *n_srcc = apcm->substream->runtime->channels; + } + } +} + +static int +atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src_mgr *src_mgr = atc->rsc_mgrs[SRC]; + struct srcimp_mgr *srcimp_mgr = atc->rsc_mgrs[SRCIMP]; + struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; + struct sum_mgr *sum_mgr = atc->rsc_mgrs[SUM]; + struct src_desc src_dsc = {0}; + struct src *src = NULL; + struct srcimp_desc srcimp_dsc = {0}; + struct srcimp *srcimp = NULL; + struct amixer_desc mix_dsc = {0}; + struct sum_desc sum_dsc = {0}; + unsigned int pitch = 0; + int multi = 0, err = 0, i = 0; + int n_srcimp = 0, n_amixer = 0, n_srcc = 0, n_sum = 0; + struct src_node_conf_t src_node_conf[2] = {{0} }; + + /* The numbers of converting SRCs and SRCIMPs should be determined + * by pitch value. */ + + multi = apcm->substream->runtime->channels; + + /* get pitch and convert to fixed-point 8.24 format. */ + pitch = atc_get_pitch((atc->rsr * atc->msr), + apcm->substream->runtime->rate); + + setup_src_node_conf(atc, apcm, src_node_conf, &n_srcc); + n_sum = (1 == multi) ? 1 : 0; + n_amixer += n_sum * 2 + n_srcc; + n_srcimp += n_srcc; + if ((multi > 1) && (0x8000000 >= pitch)) { + /* Need extra AMIXERs and SRCIMPs for special treatment + * of interleaved recording of conjugate channels */ + n_amixer += multi * atc->msr; + n_srcimp += multi * atc->msr; + } else { + n_srcimp += multi; + } + + if (n_srcc) { + apcm->srccs = kzalloc(sizeof(void *)*n_srcc, GFP_KERNEL); + if (NULL == apcm->srccs) + return -ENOMEM; + } + if (n_amixer) { + apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); + if (NULL == apcm->amixers) { + err = -ENOMEM; + goto error1; + } + } + apcm->srcimps = kzalloc(sizeof(void *)*n_srcimp, GFP_KERNEL); + if (NULL == apcm->srcimps) { + err = -ENOMEM; + goto error1; + } + + /* Allocate SRCs for sample rate conversion if needed */ + src_dsc.multi = 1; + src_dsc.mode = ARCRW; + for (i = 0, apcm->n_srcc = 0; i < n_srcc; i++) { + src_dsc.msr = src_node_conf[i/multi].msr; + err = src_mgr->get_src(src_mgr, &src_dsc, + (struct src **)&apcm->srccs[i]); + if (err) + goto error1; + + src = apcm->srccs[i]; + pitch = src_node_conf[i/multi].pitch; + src->ops->set_pitch(src, pitch); + src->ops->set_rom(src, select_rom(pitch)); + src->ops->set_vo(src, src_node_conf[i/multi].vo); + + apcm->n_srcc++; + } + + /* Allocate AMIXERs for routing SRCs of conversion if needed */ + for (i = 0, apcm->n_amixer = 0; i < n_amixer; i++) { + if (i < (n_sum*2)) + mix_dsc.msr = atc->msr; + else if (i < (n_sum*2+n_srcc)) + mix_dsc.msr = src_node_conf[(i-n_sum*2)/multi].mix_msr; + else + mix_dsc.msr = 1; + + err = amixer_mgr->get_amixer(amixer_mgr, &mix_dsc, + (struct amixer **)&apcm->amixers[i]); + if (err) + goto error1; + + apcm->n_amixer++; + } + + /* Allocate a SUM resource to mix all input channels together */ + sum_dsc.msr = atc->msr; + err = sum_mgr->get_sum(sum_mgr, &sum_dsc, (struct sum **)&apcm->mono); + if (err) + goto error1; + + pitch = atc_get_pitch((atc->rsr * atc->msr), + apcm->substream->runtime->rate); + /* Allocate SRCIMP resources */ + for (i = 0, apcm->n_srcimp = 0; i < n_srcimp; i++) { + if (i < (n_srcc)) + srcimp_dsc.msr = src_node_conf[i/multi].imp_msr; + else if (1 == multi) + srcimp_dsc.msr = (pitch <= 0x8000000) ? atc->msr : 1; + else + srcimp_dsc.msr = 1; + + err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, &srcimp); + if (err) + goto error1; + + apcm->srcimps[i] = srcimp; + apcm->n_srcimp++; + } + + /* Allocate a SRC for writing data to host memory */ + src_dsc.multi = apcm->substream->runtime->channels; + src_dsc.msr = 1; + src_dsc.mode = MEMWR; + err = src_mgr->get_src(src_mgr, &src_dsc, (struct src **)&apcm->src); + if (err) + goto error1; + + src = apcm->src; + src->ops->set_pitch(src, pitch); + + /* Set up device virtual mem map */ + err = ct_map_audio_buffer(atc, apcm); + if (err < 0) + goto error1; + + return 0; + +error1: + atc_pcm_release_resources(atc, apcm); + return err; +} + +static int atc_pcm_capture_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src *src = NULL; + struct amixer *amixer = NULL; + struct srcimp *srcimp = NULL; + struct ct_mixer *mixer = atc->mixer; + struct sum *mono = NULL; + struct rsc *out_ports[8] = {NULL}; + int err = 0, i = 0, j = 0, n_sum = 0, multi = 0; + unsigned int pitch = 0; + int mix_base = 0, imp_base = 0; + + if (NULL != apcm->src) { + /* Prepared pcm capture */ + return 0; + } + + /* Get needed resources. */ + err = atc_pcm_capture_get_resources(atc, apcm); + if (err) + return err; + + /* Connect resources */ + mixer->get_output_ports(mixer, MIX_PCMO_FRONT, + &out_ports[0], &out_ports[1]); + + multi = apcm->substream->runtime->channels; + if (1 == multi) { + mono = apcm->mono; + for (i = 0; i < 2; i++) { + amixer = apcm->amixers[i]; + amixer->ops->setup(amixer, out_ports[i], + MONO_SUM_SCALE, mono); + } + out_ports[0] = &mono->rsc; + n_sum = 1; + mix_base = n_sum * 2; + } + + for (i = 0; i < apcm->n_srcc; i++) { + src = apcm->srccs[i]; + srcimp = apcm->srcimps[imp_base+i]; + amixer = apcm->amixers[mix_base+i]; + srcimp->ops->map(srcimp, src, out_ports[i%multi]); + amixer->ops->setup(amixer, &src->rsc, INIT_VOL, NULL); + out_ports[i%multi] = &amixer->rsc; + } + + pitch = atc_get_pitch((atc->rsr * atc->msr), + apcm->substream->runtime->rate); + + if ((multi > 1) && (pitch <= 0x8000000)) { + /* Special connection for interleaved + * recording with conjugate channels */ + for (i = 0; i < multi; i++) { + out_ports[i]->ops->master(out_ports[i]); + for (j = 0; j < atc->msr; j++) { + amixer = apcm->amixers[apcm->n_srcc+j*multi+i]; + amixer->ops->set_input(amixer, out_ports[i]); + amixer->ops->set_scale(amixer, INIT_VOL); + amixer->ops->set_sum(amixer, NULL); + amixer->ops->commit_raw_write(amixer); + out_ports[i]->ops->next_conj(out_ports[i]); + + srcimp = apcm->srcimps[apcm->n_srcc+j*multi+i]; + srcimp->ops->map(srcimp, apcm->src, + &amixer->rsc); + } + } + } else { + for (i = 0; i < multi; i++) { + srcimp = apcm->srcimps[apcm->n_srcc+i]; + srcimp->ops->map(srcimp, apcm->src, out_ports[i]); + } + } + + return 0; +} + +static int atc_pcm_capture_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src *src = NULL; + struct src_mgr *src_mgr = atc->rsc_mgrs[SRC]; + int i = 0, multi = 0; + + if (apcm->started) + return 0; + + apcm->started = 1; + multi = apcm->substream->runtime->channels; + /* Set up converting SRCs */ + for (i = 0; i < apcm->n_srcc; i++) { + src = apcm->srccs[i]; + src->ops->set_pm(src, ((i%multi) != (multi-1))); + src_mgr->src_disable(src_mgr, src); + } + + /* Set up recording SRC */ + src = apcm->src; + src->ops->set_sf(src, convert_format(apcm->substream->runtime->format)); + src->ops->set_sa(src, apcm->vm_block->addr); + src->ops->set_la(src, apcm->vm_block->addr + apcm->vm_block->size); + src->ops->set_ca(src, apcm->vm_block->addr); + src_mgr->src_disable(src_mgr, src); + + /* Disable relevant SRCs firstly */ + src_mgr->commit_write(src_mgr); + + /* Enable SRCs respectively */ + for (i = 0; i < apcm->n_srcc; i++) { + src = apcm->srccs[i]; + src->ops->set_state(src, SRC_STATE_RUN); + src->ops->commit_write(src); + src_mgr->src_enable_s(src_mgr, src); + } + src = apcm->src; + src->ops->set_bm(src, 1); + src->ops->set_state(src, SRC_STATE_RUN); + src->ops->commit_write(src); + src_mgr->src_enable_s(src_mgr, src); + + /* Enable relevant SRCs synchronously */ + src_mgr->commit_write(src_mgr); + + return 0; +} + +static int +atc_pcm_capture_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src *src = apcm->src; + + return src->ops->get_ca(src) - apcm->vm_block->addr; +} + +static int spdif_passthru_playback_get_resources(struct ct_atc *atc, + struct ct_atc_pcm *apcm) +{ + struct src_mgr *src_mgr = atc->rsc_mgrs[SRC]; + struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; + struct src_desc desc = {0}; + struct amixer_desc mix_dsc = {0}; + struct src *src = NULL; + int err = 0; + int n_amixer = apcm->substream->runtime->channels, i = 0; + unsigned int pitch = 0, rsr = atc->pll_rate; + + /* Get SRC resource */ + desc.multi = apcm->substream->runtime->channels; + desc.msr = 1; + while (apcm->substream->runtime->rate > (rsr * desc.msr)) + desc.msr <<= 1; + + desc.mode = MEMRD; + err = src_mgr->get_src(src_mgr, &desc, (struct src **)&apcm->src); + if (err) + goto error1; + + pitch = atc_get_pitch(apcm->substream->runtime->rate, (rsr * desc.msr)); + src = apcm->src; + src->ops->set_pitch(src, pitch); + src->ops->set_rom(src, select_rom(pitch)); + src->ops->set_sf(src, convert_format(apcm->substream->runtime->format)); + src->ops->set_pm(src, (src->ops->next_interleave(src) != NULL)); + src->ops->set_bp(src, 1); + + /* Get AMIXER resource */ + n_amixer = (n_amixer < 2) ? 2 : n_amixer; + apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); + if (NULL == apcm->amixers) { + err = -ENOMEM; + goto error1; + } + mix_dsc.msr = desc.msr; + for (i = 0, apcm->n_amixer = 0; i < n_amixer; i++) { + err = amixer_mgr->get_amixer(amixer_mgr, &mix_dsc, + (struct amixer **)&apcm->amixers[i]); + if (err) + goto error1; + + apcm->n_amixer++; + } + + /* Set up device virtual mem map */ + err = ct_map_audio_buffer(atc, apcm); + if (err < 0) + goto error1; + + return 0; + +error1: + atc_pcm_release_resources(atc, apcm); + return err; +} + +static int +spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct dao *dao = container_of(atc->daios[SPDIFOO], struct dao, daio); + unsigned long flags; + unsigned int rate = apcm->substream->runtime->rate; + unsigned int status = 0; + int err = 0; + unsigned char iec958_con_fs = 0; + + switch (rate) { + case 48000: + iec958_con_fs = IEC958_AES3_CON_FS_48000; + break; + case 44100: + iec958_con_fs = IEC958_AES3_CON_FS_44100; + break; + case 32000: + iec958_con_fs = IEC958_AES3_CON_FS_32000; + break; + default: + return -ENOENT; + } + + spin_lock_irqsave(&atc->atc_lock, flags); + dao->ops->get_spos(dao, &status); + if (((status >> 24) & IEC958_AES3_CON_FS) != iec958_con_fs) { + status &= ((~IEC958_AES3_CON_FS) << 24); + status |= (iec958_con_fs << 24); + dao->ops->set_spos(dao, status); + dao->ops->commit_write(dao); + } + if ((rate != atc->pll_rate) && (32000 != rate)) { + err = ((struct hw *)atc->hw)->pll_init(atc->hw, rate); + atc->pll_rate = err ? 0 : rate; + } + spin_unlock_irqrestore(&atc->atc_lock, flags); + + return err; +} + +static int +spdif_passthru_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) +{ + struct src *src = NULL; + struct amixer *amixer = NULL; + struct dao *dao = NULL; + int err = 0; + int i = 0; + unsigned long flags; + + if (NULL != apcm->src) + return 0; + + /* Configure SPDIFOO and PLL to passthrough mode; + * determine pll_rate. */ + err = spdif_passthru_playback_setup(atc, apcm); + if (err) + return err; + + /* Get needed resources. */ + err = spdif_passthru_playback_get_resources(atc, apcm); + if (err) + return err; + + /* Connect resources */ + src = apcm->src; + for (i = 0; i < apcm->n_amixer; i++) { + amixer = apcm->amixers[i]; + amixer->ops->setup(amixer, &src->rsc, INIT_VOL, NULL); + src = src->ops->next_interleave(src); + if (NULL == src) + src = apcm->src; + } + /* Connect to SPDIFOO */ + spin_lock_irqsave(&atc->atc_lock, flags); + dao = container_of(atc->daios[SPDIFOO], struct dao, daio); + amixer = apcm->amixers[0]; + dao->ops->set_left_input(dao, &amixer->rsc); + amixer = apcm->amixers[1]; + dao->ops->set_right_input(dao, &amixer->rsc); + spin_unlock_irqrestore(&atc->atc_lock, flags); + + return 0; +} + +static int atc_select_line_in(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + struct ct_mixer *mixer = atc->mixer; + struct src *src = NULL; + + if (hw->is_adc_source_selected(hw, ADC_LINEIN)) + return 0; + + mixer->set_input_left(mixer, MIX_MIC_IN, NULL); + mixer->set_input_right(mixer, MIX_MIC_IN, NULL); + + hw->select_adc_source(hw, ADC_LINEIN); + + src = atc->srcs[2]; + mixer->set_input_left(mixer, MIX_LINE_IN, &src->rsc); + src = atc->srcs[3]; + mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc); + + return 0; +} + +static int atc_select_mic_in(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + struct ct_mixer *mixer = atc->mixer; + struct src *src = NULL; + + if (hw->is_adc_source_selected(hw, ADC_MICIN)) + return 0; + + mixer->set_input_left(mixer, MIX_LINE_IN, NULL); + mixer->set_input_right(mixer, MIX_LINE_IN, NULL); + + hw->select_adc_source(hw, ADC_MICIN); + + src = atc->srcs[2]; + mixer->set_input_left(mixer, MIX_MIC_IN, &src->rsc); + src = atc->srcs[3]; + mixer->set_input_right(mixer, MIX_MIC_IN, &src->rsc); + + return 0; +} + +static int atc_have_digit_io_switch(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + + return hw->have_digit_io_switch(hw); +} + +static int atc_select_digit_io(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + + if (hw->is_adc_source_selected(hw, ADC_NONE)) + return 0; + + hw->select_adc_source(hw, ADC_NONE); + + return 0; +} + +static int atc_daio_unmute(struct ct_atc *atc, unsigned char state, int type) +{ + struct daio_mgr *daio_mgr = atc->rsc_mgrs[DAIO]; + + if (state) + daio_mgr->daio_enable(daio_mgr, atc->daios[type]); + else + daio_mgr->daio_disable(daio_mgr, atc->daios[type]); + + daio_mgr->commit_write(daio_mgr); + + return 0; +} + +static int +atc_dao_get_status(struct ct_atc *atc, unsigned int *status, int type) +{ + struct dao *dao = container_of(atc->daios[type], struct dao, daio); + return dao->ops->get_spos(dao, status); +} + +static int +atc_dao_set_status(struct ct_atc *atc, unsigned int status, int type) +{ + struct dao *dao = container_of(atc->daios[type], struct dao, daio); + + dao->ops->set_spos(dao, status); + dao->ops->commit_write(dao); + return 0; +} + +static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, LINEO1); +} + +static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, LINEO4); +} + +static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, LINEO3); +} + +static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, LINEO2); +} + +static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, LINEIM); +} + +static int atc_spdif_out_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, SPDIFOO); +} + +static int atc_spdif_in_unmute(struct ct_atc *atc, unsigned char state) +{ + return atc_daio_unmute(atc, state, SPDIFIO); +} + +static int atc_spdif_out_get_status(struct ct_atc *atc, unsigned int *status) +{ + return atc_dao_get_status(atc, status, SPDIFOO); +} + +static int atc_spdif_out_set_status(struct ct_atc *atc, unsigned int status) +{ + return atc_dao_set_status(atc, status, SPDIFOO); +} + +static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state) +{ + unsigned long flags; + struct dao_desc da_dsc = {0}; + struct dao *dao = NULL; + int err = 0; + struct ct_mixer *mixer = atc->mixer; + struct rsc *rscs[2] = {NULL}; + unsigned int spos = 0; + + spin_lock_irqsave(&atc->atc_lock, flags); + dao = container_of(atc->daios[SPDIFOO], struct dao, daio); + da_dsc.msr = state ? 1 : atc->msr; + da_dsc.passthru = state ? 1 : 0; + err = dao->ops->reinit(dao, &da_dsc); + if (state) { + spos = IEC958_DEFAULT_CON; + } else { + mixer->get_output_ports(mixer, MIX_SPDIF_OUT, + &rscs[0], &rscs[1]); + dao->ops->set_left_input(dao, rscs[0]); + dao->ops->set_right_input(dao, rscs[1]); + /* Restore PLL to atc->rsr if needed. */ + if (atc->pll_rate != atc->rsr) { + err = ((struct hw *)atc->hw)->pll_init(atc->hw, + atc->rsr); + atc->pll_rate = err ? 0 : atc->rsr; + } + } + dao->ops->set_spos(dao, spos); + dao->ops->commit_write(dao); + spin_unlock_irqrestore(&atc->atc_lock, flags); + + return err; +} + +static int ct_atc_destroy(struct ct_atc *atc) +{ + struct daio_mgr *daio_mgr = NULL; + struct dao *dao = NULL; + struct dai *dai = NULL; + struct daio *daio = NULL; + struct sum_mgr *sum_mgr = NULL; + struct src_mgr *src_mgr = NULL; + struct srcimp_mgr *srcimp_mgr = NULL; + struct srcimp *srcimp = NULL; + struct ct_mixer *mixer = NULL; + int i = 0; + + if (NULL == atc) + return 0; + + /* Stop hardware and disable all interrupts */ + if (NULL != atc->hw) + ((struct hw *)atc->hw)->card_stop(atc->hw); + + /* Destroy internal mixer objects */ + if (NULL != atc->mixer) { + mixer = atc->mixer; + mixer->set_input_left(mixer, MIX_LINE_IN, NULL); + mixer->set_input_right(mixer, MIX_LINE_IN, NULL); + mixer->set_input_left(mixer, MIX_MIC_IN, NULL); + mixer->set_input_right(mixer, MIX_MIC_IN, NULL); + mixer->set_input_left(mixer, MIX_SPDIF_IN, NULL); + mixer->set_input_right(mixer, MIX_SPDIF_IN, NULL); + ct_mixer_destroy(atc->mixer); + } + + if (NULL != atc->daios) { + daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; + for (i = 0; i < atc->n_daio; i++) { + daio = atc->daios[i]; + if (daio->type < LINEIM) { + dao = container_of(daio, struct dao, daio); + dao->ops->clear_left_input(dao); + dao->ops->clear_right_input(dao); + } else { + dai = container_of(daio, struct dai, daio); + /* some thing to do for dai ... */ + } + daio_mgr->put_daio(daio_mgr, daio); + } + kfree(atc->daios); + } + + if (NULL != atc->pcm) { + sum_mgr = atc->rsc_mgrs[SUM]; + for (i = 0; i < atc->n_pcm; i++) + sum_mgr->put_sum(sum_mgr, atc->pcm[i]); + + kfree(atc->pcm); + } + + if (NULL != atc->srcs) { + src_mgr = atc->rsc_mgrs[SRC]; + for (i = 0; i < atc->n_src; i++) + src_mgr->put_src(src_mgr, atc->srcs[i]); + + kfree(atc->srcs); + } + + if (NULL != atc->srcimps) { + srcimp_mgr = atc->rsc_mgrs[SRCIMP]; + for (i = 0; i < atc->n_srcimp; i++) { + srcimp = atc->srcimps[i]; + srcimp->ops->unmap(srcimp); + srcimp_mgr->put_srcimp(srcimp_mgr, atc->srcimps[i]); + } + kfree(atc->srcimps); + } + + for (i = 0; i < NUM_RSCTYP; i++) { + if ((NULL != rsc_mgr_funcs[i].destroy) && + (NULL != atc->rsc_mgrs[i])) + rsc_mgr_funcs[i].destroy(atc->rsc_mgrs[i]); + + } + + if (NULL != atc->hw) + destroy_hw_obj((struct hw *)atc->hw); + + /* Destroy device virtual memory manager object */ + if (NULL != atc->vm) { + ct_vm_destroy(atc->vm); + atc->vm = NULL; + } + + kfree(atc); + + return 0; +} + +static int atc_dev_free(struct snd_device *dev) +{ + struct ct_atc *atc = dev->device_data; + return ct_atc_destroy(atc); +} + +static int atc_identify_card(struct ct_atc *atc) +{ + u16 subsys = 0; + u8 revision = 0; + struct pci_dev *pci = atc->pci; + const struct ct_atc_chip_details *d; + enum CTCARDS i; + + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsys); + pci_read_config_byte(pci, PCI_REVISION_ID, &revision); + atc->chip_details = NULL; + atc->model = NUM_CTCARDS; + for (d = atc_chip_details; d->vendor; d++) { + if (d->vendor != pci->vendor || d->device != pci->device) + continue; + + if (NULL == d->sub_details) { + atc->chip_details = d; + break; + } + for (i = 0; i < NUM_CTCARDS; i++) { + if ((d->sub_details[i].subsys == subsys) || + (((subsys & 0x6000) == 0x6000) && + ((d->sub_details[i].subsys & 0x6000) == 0x6000))) { + atc->model = i; + break; + } + } + if (i >= NUM_CTCARDS) + continue; + + atc->chip_details = d; + break; + /* not take revision into consideration now */ + } + if (!d->vendor) + return -ENOENT; + + return 0; +} + +static int ct_create_alsa_devs(struct ct_atc *atc) +{ + enum CTALSADEVS i; + struct hw *hw = atc->hw; + int err; + + switch (hw->get_chip_type(hw)) { + case ATC20K1: + alsa_dev_funcs[MIXER].public_name = "20K1"; + break; + case ATC20K2: + alsa_dev_funcs[MIXER].public_name = "20K2"; + break; + default: + alsa_dev_funcs[MIXER].public_name = "Unknown"; + break; + } + + for (i = 0; i < NUM_CTALSADEVS; i++) { + if (NULL == alsa_dev_funcs[i].create) + continue; + + err = alsa_dev_funcs[i].create(atc, i, + alsa_dev_funcs[i].public_name); + if (err) { + printk(KERN_ERR "Creating alsa device %d failed!\n", i); + return err; + } + } + + return 0; +} + +static int atc_create_hw_devs(struct ct_atc *atc) +{ + struct hw *hw = NULL; + struct card_conf info = {0}; + int i = 0, err = 0; + + err = create_hw_obj(atc->pci, &hw); + if (err) { + printk(KERN_ERR "Failed to create hw obj!!!\n"); + return err; + } + atc->hw = hw; + + /* Initialize card hardware. */ + info.rsr = atc->rsr; + info.msr = atc->msr; + info.vm_pgt_phys = atc_get_ptp_phys(atc, 0); + err = hw->card_init(hw, &info); + if (err < 0) + return err; + + for (i = 0; i < NUM_RSCTYP; i++) { + if (NULL == rsc_mgr_funcs[i].create) + continue; + + err = rsc_mgr_funcs[i].create(atc->hw, &atc->rsc_mgrs[i]); + if (err) { + printk(KERN_ERR "Failed to create rsc_mgr %d!!!\n", i); + return err; + } + } + + return 0; +} + +static int atc_get_resources(struct ct_atc *atc) +{ + struct daio_desc da_desc = {0}; + struct daio_mgr *daio_mgr = NULL; + struct src_desc src_dsc = {0}; + struct src_mgr *src_mgr = NULL; + struct srcimp_desc srcimp_dsc = {0}; + struct srcimp_mgr *srcimp_mgr = NULL; + struct sum_desc sum_dsc = {0}; + struct sum_mgr *sum_mgr = NULL; + int err = 0, i = 0; + unsigned short subsys_id = 0; + + atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); + if (NULL == atc->daios) + return -ENOMEM; + + atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); + if (NULL == atc->srcs) + return -ENOMEM; + + atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); + if (NULL == atc->srcimps) + return -ENOMEM; + + atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); + if (NULL == atc->pcm) + return -ENOMEM; + + daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; + da_desc.msr = atc->msr; + for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) { + da_desc.type = i; + err = daio_mgr->get_daio(daio_mgr, &da_desc, + (struct daio **)&atc->daios[i]); + if (err) { + printk(KERN_ERR "Failed to get DAIO " + "resource %d!!!\n", i); + return err; + } + atc->n_daio++; + } + pci_read_config_word(atc->pci, PCI_SUBSYSTEM_ID, &subsys_id); + if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { + /* SB073x cards */ + da_desc.type = SPDIFI1; + } else { + da_desc.type = SPDIFIO; + } + err = daio_mgr->get_daio(daio_mgr, &da_desc, + (struct daio **)&atc->daios[i]); + if (err) { + printk(KERN_ERR "Failed to get S/PDIF-in resource!!!\n"); + return err; + } + atc->n_daio++; + + src_mgr = atc->rsc_mgrs[SRC]; + src_dsc.multi = 1; + src_dsc.msr = atc->msr; + src_dsc.mode = ARCRW; + for (i = 0, atc->n_src = 0; i < (2*2); i++) { + err = src_mgr->get_src(src_mgr, &src_dsc, + (struct src **)&atc->srcs[i]); + if (err) + return err; + + atc->n_src++; + } + + srcimp_mgr = atc->rsc_mgrs[SRCIMP]; + srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */ + for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) { + err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, + (struct srcimp **)&atc->srcimps[i]); + if (err) + return err; + + atc->n_srcimp++; + } + srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */ + for (i = 0; i < (2*1); i++) { + err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, + (struct srcimp **)&atc->srcimps[2*1+i]); + if (err) + return err; + + atc->n_srcimp++; + } + + sum_mgr = atc->rsc_mgrs[SUM]; + sum_dsc.msr = atc->msr; + for (i = 0, atc->n_pcm = 0; i < (2*4); i++) { + err = sum_mgr->get_sum(sum_mgr, &sum_dsc, + (struct sum **)&atc->pcm[i]); + if (err) + return err; + + atc->n_pcm++; + } + + err = ct_mixer_create(atc, (struct ct_mixer **)&atc->mixer); + if (err) { + printk(KERN_ERR "Failed to create mixer obj!!!\n"); + return err; + } + + return 0; +} + +static void +atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, + struct src **srcs, struct srcimp **srcimps) +{ + struct rsc *rscs[2] = {NULL}; + struct src *src = NULL; + struct srcimp *srcimp = NULL; + int i = 0; + + rscs[0] = &dai->daio.rscl; + rscs[1] = &dai->daio.rscr; + for (i = 0; i < 2; i++) { + src = srcs[i]; + srcimp = srcimps[i]; + srcimp->ops->map(srcimp, src, rscs[i]); + src_mgr->src_disable(src_mgr, src); + } + + src_mgr->commit_write(src_mgr); /* Actually disable SRCs */ + + src = srcs[0]; + src->ops->set_pm(src, 1); + for (i = 0; i < 2; i++) { + src = srcs[i]; + src->ops->set_state(src, SRC_STATE_RUN); + src->ops->commit_write(src); + src_mgr->src_enable_s(src_mgr, src); + } + + dai->ops->set_srt_srcl(dai, &(srcs[0]->rsc)); + dai->ops->set_srt_srcr(dai, &(srcs[1]->rsc)); + + dai->ops->set_enb_src(dai, 1); + dai->ops->set_enb_srt(dai, 1); + dai->ops->commit_write(dai); + + src_mgr->commit_write(src_mgr); /* Synchronously enable SRCs */ +} + +static void atc_connect_resources(struct ct_atc *atc) +{ + struct dai *dai = NULL; + struct dao *dao = NULL; + struct src *src = NULL; + struct sum *sum = NULL; + struct ct_mixer *mixer = NULL; + struct rsc *rscs[2] = {NULL}; + int i = 0, j = 0; + + mixer = atc->mixer; + + for (i = MIX_WAVE_FRONT, j = LINEO1; i <= MIX_SPDIF_OUT; i++, j++) { + mixer->get_output_ports(mixer, i, &rscs[0], &rscs[1]); + dao = container_of(atc->daios[j], struct dao, daio); + dao->ops->set_left_input(dao, rscs[0]); + dao->ops->set_right_input(dao, rscs[1]); + } + + dai = container_of(atc->daios[LINEIM], struct dai, daio); + atc_connect_dai(atc->rsc_mgrs[SRC], dai, + (struct src **)&atc->srcs[2], + (struct srcimp **)&atc->srcimps[2]); + src = atc->srcs[2]; + mixer->set_input_left(mixer, MIX_LINE_IN, &src->rsc); + src = atc->srcs[3]; + mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc); + + dai = container_of(atc->daios[SPDIFIO], struct dai, daio); + atc_connect_dai(atc->rsc_mgrs[SRC], dai, + (struct src **)&atc->srcs[0], + (struct srcimp **)&atc->srcimps[0]); + + src = atc->srcs[0]; + mixer->set_input_left(mixer, MIX_SPDIF_IN, &src->rsc); + src = atc->srcs[1]; + mixer->set_input_right(mixer, MIX_SPDIF_IN, &src->rsc); + + for (i = MIX_PCMI_FRONT, j = 0; i <= MIX_PCMI_SURROUND; i++, j += 2) { + sum = atc->pcm[j]; + mixer->set_input_left(mixer, i, &sum->rsc); + sum = atc->pcm[j+1]; + mixer->set_input_right(mixer, i, &sum->rsc); + } +} + +static void atc_set_ops(struct ct_atc *atc) +{ + /* Set operations */ + atc->map_audio_buffer = ct_map_audio_buffer; + atc->unmap_audio_buffer = ct_unmap_audio_buffer; + atc->pcm_playback_prepare = atc_pcm_playback_prepare; + atc->pcm_release_resources = atc_pcm_release_resources; + atc->pcm_playback_start = atc_pcm_playback_start; + atc->pcm_playback_stop = atc_pcm_stop; + atc->pcm_playback_position = atc_pcm_playback_position; + atc->pcm_capture_prepare = atc_pcm_capture_prepare; + atc->pcm_capture_start = atc_pcm_capture_start; + atc->pcm_capture_stop = atc_pcm_stop; + atc->pcm_capture_position = atc_pcm_capture_position; + atc->spdif_passthru_playback_prepare = spdif_passthru_playback_prepare; + atc->get_ptp_phys = atc_get_ptp_phys; + atc->select_line_in = atc_select_line_in; + atc->select_mic_in = atc_select_mic_in; + atc->select_digit_io = atc_select_digit_io; + atc->line_front_unmute = atc_line_front_unmute; + atc->line_surround_unmute = atc_line_surround_unmute; + atc->line_clfe_unmute = atc_line_clfe_unmute; + atc->line_rear_unmute = atc_line_rear_unmute; + atc->line_in_unmute = atc_line_in_unmute; + atc->spdif_out_unmute = atc_spdif_out_unmute; + atc->spdif_in_unmute = atc_spdif_in_unmute; + atc->spdif_out_get_status = atc_spdif_out_get_status; + atc->spdif_out_set_status = atc_spdif_out_set_status; + atc->spdif_out_passthru = atc_spdif_out_passthru; + atc->have_digit_io_switch = atc_have_digit_io_switch; +} + +/** + * ct_atc_create - create and initialize a hardware manager + * @card: corresponding alsa card object + * @pci: corresponding kernel pci device object + * @ratc: return created object address in it + * + * Creates and initializes a hardware manager. + * + * Creates kmallocated ct_atc structure. Initializes hardware. + * Returns 0 if suceeds, or negative error code if fails. + */ + +int ct_atc_create(struct snd_card *card, struct pci_dev *pci, + unsigned int rsr, unsigned int msr, struct ct_atc **ratc) +{ + struct ct_atc *atc = NULL; + static struct snd_device_ops ops = { + .dev_free = atc_dev_free, + }; + int err = 0; + + *ratc = NULL; + + atc = kzalloc(sizeof(*atc), GFP_KERNEL); + if (NULL == atc) + return -ENOMEM; + + atc->card = card; + atc->pci = pci; + atc->rsr = rsr; + atc->msr = msr; + + /* Set operations */ + atc_set_ops(atc); + + spin_lock_init(&atc->atc_lock); + spin_lock_init(&atc->vm_lock); + + /* Find card model */ + err = atc_identify_card(atc); + if (err < 0) { + printk(KERN_ERR "ctatc: Card not recognised\n"); + goto error1; + } + + /* Set up device virtual memory management object */ + err = ct_vm_create(&atc->vm); + if (err < 0) + goto error1; + + /* Create all atc hw devices */ + err = atc_create_hw_devs(atc); + if (err < 0) + goto error1; + + /* Get resources */ + err = atc_get_resources(atc); + if (err < 0) + goto error1; + + /* Build topology */ + atc_connect_resources(atc); + + atc->create_alsa_devs = ct_create_alsa_devs; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); + if (err < 0) + goto error1; + + snd_card_set_dev(card, &pci->dev); + + *ratc = atc; + return 0; + +error1: + ct_atc_destroy(atc); + printk(KERN_ERR "Something wrong!!!\n"); + return err; +} + diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h new file mode 100644 index 0000000..286c993 --- /dev/null +++ b/sound/pci/ctxfi/ctatc.h @@ -0,0 +1,155 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctatc.h + * + * @Brief + * This file contains the definition of the device resource management object. + * + * @Author Liu Chun + * @Date Mar 28 2008 + * + */ + +#ifndef CTATC_H +#define CTATC_H + +#include +#include +#include +#include +#include + +#include "ctvmem.h" +#include "ctresource.h" + +enum CTALSADEVS { /* Types of alsa devices */ + FRONT, + REAR, + CLFE, + SURROUND, + IEC958, + MIXER, + NUM_CTALSADEVS /* This should always be the last */ +}; + +enum CTCARDS { + CTSB0760, + CTHENDRIX, + CTSB08801, + CTSB08802, + CTSB08803, + NUM_CTCARDS /* This should always be the last */ +}; + +struct ct_atc_chip_sub_details { + u16 subsys; + const char *nm_model; +}; + +struct ct_atc_chip_details { + u16 vendor; + u16 device; + const struct ct_atc_chip_sub_details *sub_details; + const char *nm_card; +}; + +struct ct_atc; + +/* alsa pcm stream descriptor */ +struct ct_atc_pcm { + struct snd_pcm_substream *substream; + void (*interrupt)(struct ct_atc_pcm *apcm); + unsigned int started:1; + unsigned int stop_timer:1; + struct timer_list timer; + spinlock_t timer_lock; + unsigned int position; + + /* Only mono and interleaved modes are supported now. */ + struct ct_vm_block *vm_block; + void *src; /* SRC for interacting with host memory */ + void **srccs; /* SRCs for sample rate conversion */ + void **srcimps; /* SRC Input Mappers */ + void **amixers; /* AMIXERs for routing converted data */ + void *mono; /* A SUM resource for mixing chs to one */ + unsigned char n_srcc; /* Number of converting SRCs */ + unsigned char n_srcimp; /* Number of SRC Input Mappers */ + unsigned char n_amixer; /* Number of AMIXERs */ +}; + +/* Chip resource management object */ +struct ct_atc { + struct pci_dev *pci; + struct snd_card *card; + unsigned int rsr; /* reference sample rate in Hz */ + unsigned int msr; /* master sample rate in rsr */ + unsigned int pll_rate; /* current rate of Phase Lock Loop */ + + const struct ct_atc_chip_details *chip_details; + enum CTCARDS model; + /* Create all alsa devices */ + int (*create_alsa_devs)(struct ct_atc *atc); + + struct ct_vm *vm; /* device virtual memory manager for this card */ + int (*map_audio_buffer)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + void (*unmap_audio_buffer)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + unsigned long (*get_ptp_phys)(struct ct_atc *atc, int index); + + spinlock_t atc_lock; + spinlock_t vm_lock; + + int (*pcm_playback_prepare)(struct ct_atc *atc, + struct ct_atc_pcm *apcm); + int (*pcm_playback_start)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + int (*pcm_playback_stop)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + int (*pcm_playback_position)(struct ct_atc *atc, + struct ct_atc_pcm *apcm); + int (*spdif_passthru_playback_prepare)(struct ct_atc *atc, + struct ct_atc_pcm *apcm); + int (*pcm_capture_prepare)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + int (*pcm_capture_start)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + int (*pcm_capture_stop)(struct ct_atc *atc, struct ct_atc_pcm *apcm); + int (*pcm_capture_position)(struct ct_atc *atc, + struct ct_atc_pcm *apcm); + int (*pcm_release_resources)(struct ct_atc *atc, + struct ct_atc_pcm *apcm); + int (*select_line_in)(struct ct_atc *atc); + int (*select_mic_in)(struct ct_atc *atc); + int (*select_digit_io)(struct ct_atc *atc); + int (*line_front_unmute)(struct ct_atc *atc, unsigned char state); + int (*line_surround_unmute)(struct ct_atc *atc, unsigned char state); + int (*line_clfe_unmute)(struct ct_atc *atc, unsigned char state); + int (*line_rear_unmute)(struct ct_atc *atc, unsigned char state); + int (*line_in_unmute)(struct ct_atc *atc, unsigned char state); + int (*spdif_out_unmute)(struct ct_atc *atc, unsigned char state); + int (*spdif_in_unmute)(struct ct_atc *atc, unsigned char state); + int (*spdif_out_get_status)(struct ct_atc *atc, unsigned int *status); + int (*spdif_out_set_status)(struct ct_atc *atc, unsigned int status); + int (*spdif_out_passthru)(struct ct_atc *atc, unsigned char state); + int (*have_digit_io_switch)(struct ct_atc *atc); + + /* Don't touch! Used for internal object. */ + void *rsc_mgrs[NUM_RSCTYP]; /* chip resource managers */ + void *mixer; /* internal mixer object */ + void *hw; /* chip specific hardware access object */ + void **daios; /* digital audio io resources */ + void **pcm; /* SUMs for collecting all pcm stream */ + void **srcs; /* Sample Rate Converters for input signal */ + void **srcimps; /* input mappers for SRCs */ + unsigned char n_daio; + unsigned char n_src; + unsigned char n_srcimp; + unsigned char n_pcm; +}; + + +int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, + unsigned int rsr, unsigned int msr, + struct ct_atc **ratc); + +#endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c new file mode 100644 index 0000000..a2aea39 --- /dev/null +++ b/sound/pci/ctxfi/ctdaio.c @@ -0,0 +1,769 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctdaio.c + * + * @Brief + * This file contains the implementation of Digital Audio Input Output + * resource management object. + * + * @Author Liu Chun + * @Date May 23 2008 + * + */ + +#include "ctdaio.h" +#include "cthardware.h" +#include "ctimap.h" +#include +#include + +#define DAIO_RESOURCE_NUM NUM_DAIOTYP +#define DAIO_OUT_MAX SPDIFOO + +union daio_usage { + struct { + unsigned short lineo1:1; + unsigned short lineo2:1; + unsigned short lineo3:1; + unsigned short lineo4:1; + unsigned short spdifoo:1; + unsigned short lineim:1; + unsigned short spdifio:1; + unsigned short spdifi1:1; + } bf; + unsigned short data; +}; + +struct daio_rsc_idx { + unsigned short left; + unsigned short right; +}; + +struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { + [LINEO1] = {.left = 0x00, .right = 0x01}, + [LINEO2] = {.left = 0x18, .right = 0x19}, + [LINEO3] = {.left = 0x08, .right = 0x09}, + [LINEO4] = {.left = 0x10, .right = 0x11}, + [LINEIM] = {.left = 0x1b5, .right = 0x1bd}, + [SPDIFOO] = {.left = 0x20, .right = 0x21}, + [SPDIFIO] = {.left = 0x15, .right = 0x1d}, + [SPDIFI1] = {.left = 0x95, .right = 0x9d}, +}; + +struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { + [LINEO1] = {.left = 0x40, .right = 0x41}, + [LINEO2] = {.left = 0x70, .right = 0x71}, + [LINEO3] = {.left = 0x50, .right = 0x51}, + [LINEO4] = {.left = 0x60, .right = 0x61}, + [LINEIM] = {.left = 0x45, .right = 0xc5}, + [SPDIFOO] = {.left = 0x00, .right = 0x01}, + [SPDIFIO] = {.left = 0x05, .right = 0x85}, +}; + +static int daio_master(struct rsc *rsc) +{ + /* Actually, this is not the resource index of DAIO. + * For DAO, it is the input mapper index. And, for DAI, + * it is the output time-slot index. */ + return rsc->conj = rsc->idx; +} + +static int daio_index(const struct rsc *rsc) +{ + return rsc->conj; +} + +static int daio_out_next_conj(struct rsc *rsc) +{ + return rsc->conj += 2; +} + +static int daio_in_next_conj_20k1(struct rsc *rsc) +{ + return rsc->conj += 0x200; +} + +static int daio_in_next_conj_20k2(struct rsc *rsc) +{ + return rsc->conj += 0x100; +} + +static struct rsc_ops daio_out_rsc_ops = { + .master = daio_master, + .next_conj = daio_out_next_conj, + .index = daio_index, + .output_slot = NULL, +}; + +static struct rsc_ops daio_in_rsc_ops_20k1 = { + .master = daio_master, + .next_conj = daio_in_next_conj_20k1, + .index = NULL, + .output_slot = daio_index, +}; + +static struct rsc_ops daio_in_rsc_ops_20k2 = { + .master = daio_master, + .next_conj = daio_in_next_conj_20k2, + .index = NULL, + .output_slot = daio_index, +}; + +static unsigned int daio_device_index(enum DAIOTYP type, struct hw *hw) +{ + switch (hw->get_chip_type(hw)) { + case ATC20K1: + switch (type) { + case SPDIFOO: return 0; + case SPDIFIO: return 0; + case SPDIFI1: return 1; + case LINEO1: return 4; + case LINEO2: return 7; + case LINEO3: return 5; + case LINEO4: return 6; + case LINEIM: return 7; + default: return -EINVAL; + } + case ATC20K2: + switch (type) { + case SPDIFOO: return 0; + case SPDIFIO: return 0; + case LINEO1: return 4; + case LINEO2: return 7; + case LINEO3: return 5; + case LINEO4: return 6; + case LINEIM: return 4; + default: return -EINVAL; + } + default: + return -EINVAL; + } +} + +static int dao_rsc_reinit(struct dao *dao, const struct dao_desc *desc); + +static int dao_spdif_get_spos(struct dao *dao, unsigned int *spos) +{ + ((struct hw *)dao->hw)->dao_get_spos(dao->ctrl_blk, spos); + return 0; +} + +static int dao_spdif_set_spos(struct dao *dao, unsigned int spos) +{ + ((struct hw *)dao->hw)->dao_set_spos(dao->ctrl_blk, spos); + return 0; +} + +static int dao_commit_write(struct dao *dao) +{ + ((struct hw *)dao->hw)->dao_commit_write(dao->hw, + daio_device_index(dao->daio.type, dao->hw), dao->ctrl_blk); + return 0; +} + +static int dao_set_left_input(struct dao *dao, struct rsc *input) +{ + struct imapper *entry = NULL; + struct daio *daio = &dao->daio; + int i = 0; + + entry = kzalloc((sizeof(*entry) * daio->rscl.msr), GFP_KERNEL); + if (NULL == entry) + return -ENOMEM; + + /* Program master and conjugate resources */ + input->ops->master(input); + daio->rscl.ops->master(&daio->rscl); + for (i = 0; i < daio->rscl.msr; i++, entry++) { + entry->slot = input->ops->output_slot(input); + entry->user = entry->addr = daio->rscl.ops->index(&daio->rscl); + dao->mgr->imap_add(dao->mgr, entry); + dao->imappers[i] = entry; + + input->ops->next_conj(input); + daio->rscl.ops->next_conj(&daio->rscl); + } + input->ops->master(input); + daio->rscl.ops->master(&daio->rscl); + + return 0; +} + +static int dao_set_right_input(struct dao *dao, struct rsc *input) +{ + struct imapper *entry = NULL; + struct daio *daio = &dao->daio; + int i = 0; + + entry = kzalloc((sizeof(*entry) * daio->rscr.msr), GFP_KERNEL); + if (NULL == entry) + return -ENOMEM; + + /* Program master and conjugate resources */ + input->ops->master(input); + daio->rscr.ops->master(&daio->rscr); + for (i = 0; i < daio->rscr.msr; i++, entry++) { + entry->slot = input->ops->output_slot(input); + entry->user = entry->addr = daio->rscr.ops->index(&daio->rscr); + dao->mgr->imap_add(dao->mgr, entry); + dao->imappers[daio->rscl.msr + i] = entry; + + input->ops->next_conj(input); + daio->rscr.ops->next_conj(&daio->rscr); + } + input->ops->master(input); + daio->rscr.ops->master(&daio->rscr); + + return 0; +} + +static int dao_clear_left_input(struct dao *dao) +{ + struct imapper *entry = NULL; + struct daio *daio = &dao->daio; + int i = 0; + + if (NULL == dao->imappers[0]) + return 0; + + entry = dao->imappers[0]; + dao->mgr->imap_delete(dao->mgr, entry); + /* Program conjugate resources */ + for (i = 1; i < daio->rscl.msr; i++) { + entry = dao->imappers[i]; + dao->mgr->imap_delete(dao->mgr, entry); + dao->imappers[i] = NULL; + } + + kfree(dao->imappers[0]); + dao->imappers[0] = NULL; + + return 0; +} + +static int dao_clear_right_input(struct dao *dao) +{ + struct imapper *entry = NULL; + struct daio *daio = &dao->daio; + int i = 0; + + if (NULL == dao->imappers[daio->rscl.msr]) + return 0; + + entry = dao->imappers[daio->rscl.msr]; + dao->mgr->imap_delete(dao->mgr, entry); + /* Program conjugate resources */ + for (i = 1; i < daio->rscr.msr; i++) { + entry = dao->imappers[daio->rscl.msr + i]; + dao->mgr->imap_delete(dao->mgr, entry); + dao->imappers[daio->rscl.msr + i] = NULL; + } + + kfree(dao->imappers[daio->rscl.msr]); + dao->imappers[daio->rscl.msr] = NULL; + + return 0; +} + +static struct dao_rsc_ops dao_ops = { + .set_spos = dao_spdif_set_spos, + .commit_write = dao_commit_write, + .get_spos = dao_spdif_get_spos, + .reinit = dao_rsc_reinit, + .set_left_input = dao_set_left_input, + .set_right_input = dao_set_right_input, + .clear_left_input = dao_clear_left_input, + .clear_right_input = dao_clear_right_input, +}; + +static int dai_set_srt_srcl(struct dai *dai, struct rsc *src) +{ + src->ops->master(src); + ((struct hw *)dai->hw)->dai_srt_set_srcm(dai->ctrl_blk, + src->ops->index(src)); + return 0; +} + +static int dai_set_srt_srcr(struct dai *dai, struct rsc *src) +{ + src->ops->master(src); + ((struct hw *)dai->hw)->dai_srt_set_srco(dai->ctrl_blk, + src->ops->index(src)); + return 0; +} + +static int dai_set_srt_msr(struct dai *dai, unsigned int msr) +{ + unsigned int rsr = 0; + + for (rsr = 0; msr > 1; msr >>= 1) + rsr++; + + ((struct hw *)dai->hw)->dai_srt_set_rsr(dai->ctrl_blk, rsr); + return 0; +} + +static int dai_set_enb_src(struct dai *dai, unsigned int enb) +{ + ((struct hw *)dai->hw)->dai_srt_set_ec(dai->ctrl_blk, enb); + return 0; +} + +static int dai_set_enb_srt(struct dai *dai, unsigned int enb) +{ + ((struct hw *)dai->hw)->dai_srt_set_et(dai->ctrl_blk, enb); + return 0; +} + +static int dai_commit_write(struct dai *dai) +{ + ((struct hw *)dai->hw)->dai_commit_write(dai->hw, + daio_device_index(dai->daio.type, dai->hw), dai->ctrl_blk); + return 0; +} + +static struct dai_rsc_ops dai_ops = { + .set_srt_srcl = dai_set_srt_srcl, + .set_srt_srcr = dai_set_srt_srcr, + .set_srt_msr = dai_set_srt_msr, + .set_enb_src = dai_set_enb_src, + .set_enb_srt = dai_set_enb_srt, + .commit_write = dai_commit_write, +}; + +static int daio_rsc_init(struct daio *daio, + const struct daio_desc *desc, + void *hw) +{ + int err = 0; + unsigned int idx_l = 0, idx_r = 0; + + switch (((struct hw *)hw)->get_chip_type(hw)) { + case ATC20K1: + idx_l = idx_20k1[desc->type].left; + idx_r = idx_20k1[desc->type].right; + break; + case ATC20K2: + idx_l = idx_20k2[desc->type].left; + idx_r = idx_20k2[desc->type].right; + break; + default: + return -EINVAL; + } + err = rsc_init(&daio->rscl, idx_l, DAIO, desc->msr, hw); + if (err) + return err; + + err = rsc_init(&daio->rscr, idx_r, DAIO, desc->msr, hw); + if (err) + goto error1; + + /* Set daio->rscl/r->ops to daio specific ones */ + if (desc->type <= DAIO_OUT_MAX) { + daio->rscl.ops = daio->rscr.ops = &daio_out_rsc_ops; + } else { + switch (((struct hw *)hw)->get_chip_type(hw)) { + case ATC20K1: + daio->rscl.ops = daio->rscr.ops = &daio_in_rsc_ops_20k1; + break; + case ATC20K2: + daio->rscl.ops = daio->rscr.ops = &daio_in_rsc_ops_20k2; + break; + default: + break; + } + } + daio->type = desc->type; + + return 0; + +error1: + rsc_uninit(&daio->rscl); + return err; +} + +static int daio_rsc_uninit(struct daio *daio) +{ + rsc_uninit(&daio->rscl); + rsc_uninit(&daio->rscr); + + return 0; +} + +static int dao_rsc_init(struct dao *dao, + const struct daio_desc *desc, + struct daio_mgr *mgr) +{ + struct hw *hw = mgr->mgr.hw; + unsigned int conf = 0; + int err = 0; + + err = daio_rsc_init(&dao->daio, desc, mgr->mgr.hw); + if (err) + return err; + + dao->imappers = kzalloc(sizeof(void *)*desc->msr*2, GFP_KERNEL); + if (NULL == dao->imappers) { + err = -ENOMEM; + goto error1; + } + dao->ops = &dao_ops; + dao->mgr = mgr; + dao->hw = hw; + err = hw->dao_get_ctrl_blk(&dao->ctrl_blk); + if (err) + goto error2; + + hw->daio_mgr_dsb_dao(mgr->mgr.ctrl_blk, + daio_device_index(dao->daio.type, hw)); + hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk); + + conf |= (desc->msr & 0x7) | (desc->passthru << 3); + hw->daio_mgr_dao_init(mgr->mgr.ctrl_blk, + daio_device_index(dao->daio.type, hw), conf); + hw->daio_mgr_enb_dao(mgr->mgr.ctrl_blk, + daio_device_index(dao->daio.type, hw)); + hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk); + + return 0; + +error2: + kfree(dao->imappers); + dao->imappers = NULL; +error1: + daio_rsc_uninit(&dao->daio); + return err; +} + +static int dao_rsc_uninit(struct dao *dao) +{ + if (NULL != dao->imappers) { + if (NULL != dao->imappers[0]) + dao_clear_left_input(dao); + + if (NULL != dao->imappers[dao->daio.rscl.msr]) + dao_clear_right_input(dao); + + kfree(dao->imappers); + dao->imappers = NULL; + } + ((struct hw *)dao->hw)->dao_put_ctrl_blk(dao->ctrl_blk); + dao->hw = dao->ctrl_blk = NULL; + daio_rsc_uninit(&dao->daio); + + return 0; +} + +static int dao_rsc_reinit(struct dao *dao, const struct dao_desc *desc) +{ + struct daio_mgr *mgr = dao->mgr; + struct daio_desc dsc = {0}; + + dsc.type = dao->daio.type; + dsc.msr = desc->msr; + dsc.passthru = desc->passthru; + dao_rsc_uninit(dao); + return dao_rsc_init(dao, &dsc, mgr); +} + +static int dai_rsc_init(struct dai *dai, + const struct daio_desc *desc, + struct daio_mgr *mgr) +{ + int err = 0; + struct hw *hw = mgr->mgr.hw; + unsigned int rsr = 0, msr = 0; + + err = daio_rsc_init(&dai->daio, desc, mgr->mgr.hw); + if (err) + return err; + + dai->ops = &dai_ops; + dai->hw = mgr->mgr.hw; + err = hw->dai_get_ctrl_blk(&dai->ctrl_blk); + if (err) + goto error1; + + for (rsr = 0, msr = desc->msr; msr > 1; msr >>= 1) + rsr++; + + hw->dai_srt_set_rsr(dai->ctrl_blk, rsr); + hw->dai_srt_set_drat(dai->ctrl_blk, 0); + /* default to disabling control of a SRC */ + hw->dai_srt_set_ec(dai->ctrl_blk, 0); + hw->dai_srt_set_et(dai->ctrl_blk, 0); /* default to disabling SRT */ + hw->dai_commit_write(hw, + daio_device_index(dai->daio.type, dai->hw), dai->ctrl_blk); + + return 0; + +error1: + daio_rsc_uninit(&dai->daio); + return err; +} + +static int dai_rsc_uninit(struct dai *dai) +{ + ((struct hw *)dai->hw)->dai_put_ctrl_blk(dai->ctrl_blk); + dai->hw = dai->ctrl_blk = NULL; + daio_rsc_uninit(&dai->daio); + return 0; +} + +static int daio_mgr_get_rsc(struct rsc_mgr *mgr, enum DAIOTYP type) +{ + if (((union daio_usage *)mgr->rscs)->data & (0x1 << type)) + return -ENOENT; + + ((union daio_usage *)mgr->rscs)->data |= (0x1 << type); + + return 0; +} + +static int daio_mgr_put_rsc(struct rsc_mgr *mgr, enum DAIOTYP type) +{ + ((union daio_usage *)mgr->rscs)->data &= ~(0x1 << type); + + return 0; +} + +static int get_daio_rsc(struct daio_mgr *mgr, + const struct daio_desc *desc, + struct daio **rdaio) +{ + int err = 0; + struct dai *dai = NULL; + struct dao *dao = NULL; + unsigned long flags; + + *rdaio = NULL; + + /* Check whether there are sufficient daio resources to meet request. */ + spin_lock_irqsave(&mgr->mgr_lock, flags); + err = daio_mgr_get_rsc(&mgr->mgr, desc->type); + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + if (err) { + printk(KERN_ERR "Can't meet DAIO resource request!\n"); + return err; + } + + /* Allocate mem for daio resource */ + if (desc->type <= DAIO_OUT_MAX) { + dao = kzalloc(sizeof(*dao), GFP_KERNEL); + if (NULL == dao) { + err = -ENOMEM; + goto error; + } + err = dao_rsc_init(dao, desc, mgr); + if (err) + goto error; + + *rdaio = &dao->daio; + } else { + dai = kzalloc(sizeof(*dai), GFP_KERNEL); + if (NULL == dai) { + err = -ENOMEM; + goto error; + } + err = dai_rsc_init(dai, desc, mgr); + if (err) + goto error; + + *rdaio = &dai->daio; + } + + mgr->daio_enable(mgr, *rdaio); + mgr->commit_write(mgr); + + return 0; + +error: + if (NULL != dao) + kfree(dao); + else if (NULL != dai) + kfree(dai); + + spin_lock_irqsave(&mgr->mgr_lock, flags); + daio_mgr_put_rsc(&mgr->mgr, desc->type); + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + return err; +} + +static int put_daio_rsc(struct daio_mgr *mgr, struct daio *daio) +{ + unsigned long flags; + + mgr->daio_disable(mgr, daio); + mgr->commit_write(mgr); + + spin_lock_irqsave(&mgr->mgr_lock, flags); + daio_mgr_put_rsc(&mgr->mgr, daio->type); + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + + if (daio->type <= DAIO_OUT_MAX) { + dao_rsc_uninit(container_of(daio, struct dao, daio)); + kfree(container_of(daio, struct dao, daio)); + } else { + dai_rsc_uninit(container_of(daio, struct dai, daio)); + kfree(container_of(daio, struct dai, daio)); + } + + return 0; +} + +static int daio_mgr_enb_daio(struct daio_mgr *mgr, struct daio *daio) +{ + struct hw *hw = mgr->mgr.hw; + + if (DAIO_OUT_MAX >= daio->type) { + hw->daio_mgr_enb_dao(mgr->mgr.ctrl_blk, + daio_device_index(daio->type, hw)); + } else { + hw->daio_mgr_enb_dai(mgr->mgr.ctrl_blk, + daio_device_index(daio->type, hw)); + } + return 0; +} + +static int daio_mgr_dsb_daio(struct daio_mgr *mgr, struct daio *daio) +{ + struct hw *hw = mgr->mgr.hw; + + if (DAIO_OUT_MAX >= daio->type) { + hw->daio_mgr_dsb_dao(mgr->mgr.ctrl_blk, + daio_device_index(daio->type, hw)); + } else { + hw->daio_mgr_dsb_dai(mgr->mgr.ctrl_blk, + daio_device_index(daio->type, hw)); + } + return 0; +} + +static int daio_map_op(void *data, struct imapper *entry) +{ + struct rsc_mgr *mgr = &((struct daio_mgr *)data)->mgr; + struct hw *hw = mgr->hw; + + hw->daio_mgr_set_imaparc(mgr->ctrl_blk, entry->slot); + hw->daio_mgr_set_imapnxt(mgr->ctrl_blk, entry->next); + hw->daio_mgr_set_imapaddr(mgr->ctrl_blk, entry->addr); + hw->daio_mgr_commit_write(mgr->hw, mgr->ctrl_blk); + + return 0; +} + +static int daio_imap_add(struct daio_mgr *mgr, struct imapper *entry) +{ + unsigned long flags; + int err = 0; + + spin_lock_irqsave(&mgr->imap_lock, flags); + if ((0 == entry->addr) && (mgr->init_imap_added)) { + input_mapper_delete(&mgr->imappers, mgr->init_imap, + daio_map_op, mgr); + mgr->init_imap_added = 0; + } + err = input_mapper_add(&mgr->imappers, entry, daio_map_op, mgr); + spin_unlock_irqrestore(&mgr->imap_lock, flags); + + return err; +} + +static int daio_imap_delete(struct daio_mgr *mgr, struct imapper *entry) +{ + unsigned long flags; + int err = 0; + + spin_lock_irqsave(&mgr->imap_lock, flags); + err = input_mapper_delete(&mgr->imappers, entry, daio_map_op, mgr); + if (list_empty(&mgr->imappers)) { + input_mapper_add(&mgr->imappers, mgr->init_imap, + daio_map_op, mgr); + mgr->init_imap_added = 1; + } + spin_unlock_irqrestore(&mgr->imap_lock, flags); + + return err; +} + +static int daio_mgr_commit_write(struct daio_mgr *mgr) +{ + struct hw *hw = mgr->mgr.hw; + + hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk); + return 0; +} + +int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr) +{ + int err = 0, i = 0; + struct daio_mgr *daio_mgr; + struct imapper *entry; + + *rdaio_mgr = NULL; + daio_mgr = kzalloc(sizeof(*daio_mgr), GFP_KERNEL); + if (NULL == daio_mgr) + return -ENOMEM; + + err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw); + if (err) + goto error1; + + spin_lock_init(&daio_mgr->mgr_lock); + spin_lock_init(&daio_mgr->imap_lock); + INIT_LIST_HEAD(&daio_mgr->imappers); + entry = kzalloc(sizeof(*entry), GFP_KERNEL); + if (NULL == entry) { + err = -ENOMEM; + goto error2; + } + entry->slot = entry->addr = entry->next = entry->user = 0; + list_add(&entry->list, &daio_mgr->imappers); + daio_mgr->init_imap = entry; + daio_mgr->init_imap_added = 1; + + daio_mgr->get_daio = get_daio_rsc; + daio_mgr->put_daio = put_daio_rsc; + daio_mgr->daio_enable = daio_mgr_enb_daio; + daio_mgr->daio_disable = daio_mgr_dsb_daio; + daio_mgr->imap_add = daio_imap_add; + daio_mgr->imap_delete = daio_imap_delete; + daio_mgr->commit_write = daio_mgr_commit_write; + + for (i = 0; i < 8; i++) { + ((struct hw *)hw)->daio_mgr_dsb_dao(daio_mgr->mgr.ctrl_blk, i); + ((struct hw *)hw)->daio_mgr_dsb_dai(daio_mgr->mgr.ctrl_blk, i); + } + ((struct hw *)hw)->daio_mgr_commit_write(hw, daio_mgr->mgr.ctrl_blk); + + *rdaio_mgr = daio_mgr; + + return 0; + +error2: + rsc_mgr_uninit(&daio_mgr->mgr); +error1: + kfree(daio_mgr); + return err; +} + +int daio_mgr_destroy(struct daio_mgr *daio_mgr) +{ + unsigned long flags; + + /* free daio input mapper list */ + spin_lock_irqsave(&daio_mgr->imap_lock, flags); + free_input_mapper_list(&daio_mgr->imappers); + spin_unlock_irqrestore(&daio_mgr->imap_lock, flags); + + rsc_mgr_uninit(&daio_mgr->mgr); + kfree(daio_mgr); + + return 0; +} + diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h new file mode 100644 index 0000000..0f52ce5 --- /dev/null +++ b/sound/pci/ctxfi/ctdaio.h @@ -0,0 +1,122 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctdaio.h + * + * @Brief + * This file contains the definition of Digital Audio Input Output + * resource management object. + * + * @Author Liu Chun + * @Date May 23 2008 + * + */ + +#ifndef CTDAIO_H +#define CTDAIO_H + +#include "ctresource.h" +#include "ctimap.h" +#include +#include + +/* Define the descriptor of a daio resource */ +enum DAIOTYP { + LINEO1, + LINEO2, + LINEO3, + LINEO4, + SPDIFOO, /* S/PDIF Out (Flexijack/Optical) */ + LINEIM, + SPDIFIO, /* S/PDIF In (Flexijack/Optical) on the card */ + SPDIFI1, /* S/PDIF In on internal Drive Bay */ + NUM_DAIOTYP +}; + +struct dao_rsc_ops; +struct dai_rsc_ops; +struct daio_mgr; + +struct daio { + struct rsc rscl; /* Basic resource info for left TX/RX */ + struct rsc rscr; /* Basic resource info for right TX/RX */ + enum DAIOTYP type; +}; + +struct dao { + struct daio daio; + struct dao_rsc_ops *ops; /* DAO specific operations */ + struct imapper **imappers; + struct daio_mgr *mgr; + void *hw; + void *ctrl_blk; +}; + +struct dai { + struct daio daio; + struct dai_rsc_ops *ops; /* DAI specific operations */ + void *hw; + void *ctrl_blk; +}; + +struct dao_desc { + unsigned int msr:4; + unsigned int passthru:1; +}; + +struct dao_rsc_ops { + int (*set_spos)(struct dao *dao, unsigned int spos); + int (*commit_write)(struct dao *dao); + int (*get_spos)(struct dao *dao, unsigned int *spos); + int (*reinit)(struct dao *dao, const struct dao_desc *desc); + int (*set_left_input)(struct dao *dao, struct rsc *input); + int (*set_right_input)(struct dao *dao, struct rsc *input); + int (*clear_left_input)(struct dao *dao); + int (*clear_right_input)(struct dao *dao); +}; + +struct dai_rsc_ops { + int (*set_srt_srcl)(struct dai *dai, struct rsc *src); + int (*set_srt_srcr)(struct dai *dai, struct rsc *src); + int (*set_srt_msr)(struct dai *dai, unsigned int msr); + int (*set_enb_src)(struct dai *dai, unsigned int enb); + int (*set_enb_srt)(struct dai *dai, unsigned int enb); + int (*commit_write)(struct dai *dai); +}; + +/* Define daio resource request description info */ +struct daio_desc { + unsigned int type:4; + unsigned int msr:4; + unsigned int passthru:1; +}; + +struct daio_mgr { + struct rsc_mgr mgr; /* Basic resource manager info */ + spinlock_t mgr_lock; + spinlock_t imap_lock; + struct list_head imappers; + struct imapper *init_imap; + unsigned int init_imap_added; + + /* request one daio resource */ + int (*get_daio)(struct daio_mgr *mgr, + const struct daio_desc *desc, struct daio **rdaio); + /* return one daio resource */ + int (*put_daio)(struct daio_mgr *mgr, struct daio *daio); + int (*daio_enable)(struct daio_mgr *mgr, struct daio *daio); + int (*daio_disable)(struct daio_mgr *mgr, struct daio *daio); + int (*imap_add)(struct daio_mgr *mgr, struct imapper *entry); + int (*imap_delete)(struct daio_mgr *mgr, struct imapper *entry); + int (*commit_write)(struct daio_mgr *mgr); +}; + +/* Constructor and destructor of daio resource manager */ +int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr); +int daio_mgr_destroy(struct daio_mgr *daio_mgr); + +#endif /* CTDAIO_H */ diff --git a/sound/pci/ctxfi/ctdrv.h b/sound/pci/ctxfi/ctdrv.h new file mode 100644 index 0000000..f776a44 --- /dev/null +++ b/sound/pci/ctxfi/ctdrv.h @@ -0,0 +1,30 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @file ctdrv.h + * + * @breaf + * This file contains the definition of card IDs supported by this driver. + * + * @author Liu Chun + * + */ + +#ifndef CTDRV_H +#define CTDRV_H + +#define PCI_VENDOR_CREATIVE 0x1102 +#define PCI_DEVICE_CREATIVE_20K1 0x0005 +#define PCI_DEVICE_CREATIVE_20K2 0x000B +#define PCI_SUBVENDOR_CREATIVE 0x1102 +#define PCI_SUBSYS_CREATIVE_SB0760 0x0024 +#define PCI_SUBSYS_CREATIVE_SB08801 0x0041 +#define PCI_SUBSYS_CREATIVE_SB08802 0x0042 +#define PCI_SUBSYS_CREATIVE_SB08803 0x0043 +#define PCI_SUBSYS_CREATIVE_HENDRIX 0x6000 + +#endif /* CTDRV_H */ diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c new file mode 100644 index 0000000..8e58860 --- /dev/null +++ b/sound/pci/ctxfi/cthardware.c @@ -0,0 +1,108 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File cthardware.c + * + * @Brief + * This file contains the implementation of hardware access methord. + * + * @Author Liu Chun + * @Date Jun 26 2008 + * + */ + +#include "cthardware.h" +#include "cthw20k1.h" +#include "cthw20k2.h" +#include + +static enum CHIPTYP get_chip_type(struct hw *hw) +{ + enum CHIPTYP type = ATCNONE; + + switch (hw->pci->device) { + case 0x0005: /* 20k1 device */ + type = ATC20K1; + break; + case 0x000B: /* 20k2 device */ + type = ATC20K2; + break; + default: + type = ATCNONE; + break; + } + + return type; +} + +int create_hw_obj(struct pci_dev *pci, struct hw **rhw) +{ + int err = 0; + + switch (pci->device) { + case 0x0005: /* 20k1 device */ + err = create_20k1_hw_obj(rhw); + break; + case 0x000B: /* 20k2 device */ + err = create_20k2_hw_obj(rhw); + break; + default: + err = -ENODEV; + break; + } + if (err) + return err; + + (*rhw)->pci = pci; + (*rhw)->get_chip_type = get_chip_type; + + return 0; +} + +int destroy_hw_obj(struct hw *hw) +{ + int err = 0; + + switch (hw->pci->device) { + case 0x0005: /* 20k1 device */ + err = destroy_20k1_hw_obj(hw); + break; + case 0x000B: /* 20k2 device */ + err = destroy_20k2_hw_obj(hw); + break; + default: + err = -ENODEV; + break; + } + + return err; +} + +unsigned int get_field(unsigned int data, unsigned int field) +{ + int i; + + BUG_ON(!field); + /* @field should always be greater than 0 */ + for (i = 0; !(field & (1 << i)); ) + i++; + + return (data & field) >> i; +} + +void set_field(unsigned int *data, unsigned int field, unsigned int value) +{ + int i; + + BUG_ON(!field); + /* @field should always be greater than 0 */ + for (i = 0; !(field & (1 << i)); ) + i++; + + *data = (*data & (~field)) | ((value << i) & field); +} + diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h new file mode 100644 index 0000000..b0512df --- /dev/null +++ b/sound/pci/ctxfi/cthardware.h @@ -0,0 +1,160 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File cthardware.h + * + * @Brief + * This file contains the definition of hardware access methord. + * + * @Author Liu Chun + * @Date May 13 2008 + * + */ + +#ifndef CTHARDWARE_H +#define CTHARDWARE_H + +#include +#include + +enum CHIPTYP { + ATC20K1, + ATC20K2, + ATCNONE +}; + +/* Type of input source for ADC */ +enum ADCSRC{ + ADC_MICIN, + ADC_LINEIN, + ADC_VIDEO, + ADC_AUX, + ADC_NONE /* Switch to digital input */ +}; + +struct card_conf { + /* device virtual mem page table page physical addr + * (supporting one page table page now) */ + unsigned long vm_pgt_phys; + unsigned int rsr; /* reference sample rate in Hzs*/ + unsigned int msr; /* master sample rate in rsrs */ +}; + +struct hw { + int (*card_init)(struct hw *hw, struct card_conf *info); + int (*card_stop)(struct hw *hw); + int (*pll_init)(struct hw *hw, unsigned int rsr); + enum CHIPTYP (*get_chip_type)(struct hw *hw); + int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source); + int (*select_adc_source)(struct hw *hw, enum ADCSRC source); + int (*have_digit_io_switch)(struct hw *hw); + + /* SRC operations */ + int (*src_rsc_get_ctrl_blk)(void **rblk); + int (*src_rsc_put_ctrl_blk)(void *blk); + int (*src_set_state)(void *blk, unsigned int state); + int (*src_set_bm)(void *blk, unsigned int bm); + int (*src_set_rsr)(void *blk, unsigned int rsr); + int (*src_set_sf)(void *blk, unsigned int sf); + int (*src_set_wr)(void *blk, unsigned int wr); + int (*src_set_pm)(void *blk, unsigned int pm); + int (*src_set_rom)(void *blk, unsigned int rom); + int (*src_set_vo)(void *blk, unsigned int vo); + int (*src_set_st)(void *blk, unsigned int st); + int (*src_set_ie)(void *blk, unsigned int ie); + int (*src_set_ilsz)(void *blk, unsigned int ilsz); + int (*src_set_bp)(void *blk, unsigned int bp); + int (*src_set_cisz)(void *blk, unsigned int cisz); + int (*src_set_ca)(void *blk, unsigned int ca); + int (*src_set_sa)(void *blk, unsigned int sa); + int (*src_set_la)(void *blk, unsigned int la); + int (*src_set_pitch)(void *blk, unsigned int pitch); + int (*src_set_clear_zbufs)(void *blk, unsigned int clear); + int (*src_set_dirty)(void *blk, unsigned int flags); + int (*src_set_dirty_all)(void *blk); + int (*src_commit_write)(struct hw *hw, unsigned int idx, void *blk); + int (*src_get_ca)(struct hw *hw, unsigned int idx, void *blk); + unsigned int (*src_get_dirty)(void *blk); + unsigned int (*src_dirty_conj_mask)(void); + int (*src_mgr_get_ctrl_blk)(void **rblk); + int (*src_mgr_put_ctrl_blk)(void *blk); + /* syncly enable src @idx */ + int (*src_mgr_enbs_src)(void *blk, unsigned int idx); + /* enable src @idx */ + int (*src_mgr_enb_src)(void *blk, unsigned int idx); + /* disable src @idx */ + int (*src_mgr_dsb_src)(void *blk, unsigned int idx); + int (*src_mgr_commit_write)(struct hw *hw, void *blk); + + /* SRC Input Mapper operations */ + int (*srcimp_mgr_get_ctrl_blk)(void **rblk); + int (*srcimp_mgr_put_ctrl_blk)(void *blk); + int (*srcimp_mgr_set_imaparc)(void *blk, unsigned int slot); + int (*srcimp_mgr_set_imapuser)(void *blk, unsigned int user); + int (*srcimp_mgr_set_imapnxt)(void *blk, unsigned int next); + int (*srcimp_mgr_set_imapaddr)(void *blk, unsigned int addr); + int (*srcimp_mgr_commit_write)(struct hw *hw, void *blk); + + /* AMIXER operations */ + int (*amixer_rsc_get_ctrl_blk)(void **rblk); + int (*amixer_rsc_put_ctrl_blk)(void *blk); + int (*amixer_mgr_get_ctrl_blk)(void **rblk); + int (*amixer_mgr_put_ctrl_blk)(void *blk); + int (*amixer_set_mode)(void *blk, unsigned int mode); + int (*amixer_set_iv)(void *blk, unsigned int iv); + int (*amixer_set_x)(void *blk, unsigned int x); + int (*amixer_set_y)(void *blk, unsigned int y); + int (*amixer_set_sadr)(void *blk, unsigned int sadr); + int (*amixer_set_se)(void *blk, unsigned int se); + int (*amixer_set_dirty)(void *blk, unsigned int flags); + int (*amixer_set_dirty_all)(void *blk); + int (*amixer_commit_write)(struct hw *hw, unsigned int idx, void *blk); + int (*amixer_get_y)(void *blk); + unsigned int (*amixer_get_dirty)(void *blk); + + /* DAIO operations */ + int (*dai_get_ctrl_blk)(void **rblk); + int (*dai_put_ctrl_blk)(void *blk); + int (*dai_srt_set_srco)(void *blk, unsigned int src); + int (*dai_srt_set_srcm)(void *blk, unsigned int src); + int (*dai_srt_set_rsr)(void *blk, unsigned int rsr); + int (*dai_srt_set_drat)(void *blk, unsigned int drat); + int (*dai_srt_set_ec)(void *blk, unsigned int ec); + int (*dai_srt_set_et)(void *blk, unsigned int et); + int (*dai_commit_write)(struct hw *hw, unsigned int idx, void *blk); + int (*dao_get_ctrl_blk)(void **rblk); + int (*dao_put_ctrl_blk)(void *blk); + int (*dao_set_spos)(void *blk, unsigned int spos); + int (*dao_commit_write)(struct hw *hw, unsigned int idx, void *blk); + int (*dao_get_spos)(void *blk, unsigned int *spos); + + int (*daio_mgr_get_ctrl_blk)(struct hw *hw, void **rblk); + int (*daio_mgr_put_ctrl_blk)(void *blk); + int (*daio_mgr_enb_dai)(void *blk, unsigned int idx); + int (*daio_mgr_dsb_dai)(void *blk, unsigned int idx); + int (*daio_mgr_enb_dao)(void *blk, unsigned int idx); + int (*daio_mgr_dsb_dao)(void *blk, unsigned int idx); + int (*daio_mgr_dao_init)(void *blk, unsigned int idx, + unsigned int conf); + int (*daio_mgr_set_imaparc)(void *blk, unsigned int slot); + int (*daio_mgr_set_imapnxt)(void *blk, unsigned int next); + int (*daio_mgr_set_imapaddr)(void *blk, unsigned int addr); + int (*daio_mgr_commit_write)(struct hw *hw, void *blk); + + struct pci_dev *pci; /* the pci kernel structure of this card */ + int irq; + unsigned long io_base; + unsigned long mem_base; +}; + +int create_hw_obj(struct pci_dev *pci, struct hw **rhw); +int destroy_hw_obj(struct hw *hw); + +unsigned int get_field(unsigned int data, unsigned int field); +void set_field(unsigned int *data, unsigned int field, unsigned int value); + +#endif /* CTHARDWARE_H */ diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c new file mode 100644 index 0000000..53572d92 --- /dev/null +++ b/sound/pci/ctxfi/cthw20k1.c @@ -0,0 +1,2230 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File cthw20k1.c + * + * @Brief + * This file contains the implementation of hardware access methord for 20k1. + * + * @Author Liu Chun + * @Date Jun 24 2008 + * + */ + +#include "cthw20k1.h" +#include "ct20k1reg.h" +#include +#include +#include +#include +#include +#include +#include +#include + +#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bits */ + +struct hw20k1 { + struct hw hw; + spinlock_t reg_20k1_lock; + spinlock_t reg_pci_lock; +}; + +static u32 hw_read_20kx(struct hw *hw, u32 reg); +static void hw_write_20kx(struct hw *hw, u32 reg, u32 data); +static u32 hw_read_pci(struct hw *hw, u32 reg); +static void hw_write_pci(struct hw *hw, u32 reg, u32 data); + +/* + * Type definition block. + * The layout of control structures can be directly applied on 20k2 chip. + */ + +/* + * SRC control block definitions. + */ + +/* SRC resource control block */ +#define SRCCTL_STATE 0x00000007 +#define SRCCTL_BM 0x00000008 +#define SRCCTL_RSR 0x00000030 +#define SRCCTL_SF 0x000001C0 +#define SRCCTL_WR 0x00000200 +#define SRCCTL_PM 0x00000400 +#define SRCCTL_ROM 0x00001800 +#define SRCCTL_VO 0x00002000 +#define SRCCTL_ST 0x00004000 +#define SRCCTL_IE 0x00008000 +#define SRCCTL_ILSZ 0x000F0000 +#define SRCCTL_BP 0x00100000 + +#define SRCCCR_CISZ 0x000007FF +#define SRCCCR_CWA 0x001FF800 +#define SRCCCR_D 0x00200000 +#define SRCCCR_RS 0x01C00000 +#define SRCCCR_NAL 0x3E000000 +#define SRCCCR_RA 0xC0000000 + +#define SRCCA_CA 0x03FFFFFF +#define SRCCA_RS 0x1C000000 +#define SRCCA_NAL 0xE0000000 + +#define SRCSA_SA 0x03FFFFFF + +#define SRCLA_LA 0x03FFFFFF + +/* Mixer Parameter Ring ram Low and Hight register. + * Fixed-point value in 8.24 format for parameter channel */ +#define MPRLH_PITCH 0xFFFFFFFF + +/* SRC resource register dirty flags */ +union src_dirty { + struct { + u16 ctl:1; + u16 ccr:1; + u16 sa:1; + u16 la:1; + u16 ca:1; + u16 mpr:1; + u16 czbfs:1; /* Clear Z-Buffers */ + u16 rsv:9; + } bf; + u16 data; +}; + +struct src_rsc_ctrl_blk { + unsigned int ctl; + unsigned int ccr; + unsigned int ca; + unsigned int sa; + unsigned int la; + unsigned int mpr; + union src_dirty dirty; +}; + +/* SRC manager control block */ +union src_mgr_dirty { + struct { + u16 enb0:1; + u16 enb1:1; + u16 enb2:1; + u16 enb3:1; + u16 enb4:1; + u16 enb5:1; + u16 enb6:1; + u16 enb7:1; + u16 enbsa:1; + u16 rsv:7; + } bf; + u16 data; +}; + +struct src_mgr_ctrl_blk { + unsigned int enbsa; + unsigned int enb[8]; + union src_mgr_dirty dirty; +}; + +/* SRCIMP manager control block */ +#define SRCAIM_ARC 0x00000FFF +#define SRCAIM_NXT 0x00FF0000 +#define SRCAIM_SRC 0xFF000000 + +struct srcimap { + unsigned int srcaim; + unsigned int idx; +}; + +/* SRCIMP manager register dirty flags */ +union srcimp_mgr_dirty { + struct { + u16 srcimap:1; + u16 rsv:15; + } bf; + u16 data; +}; + +struct srcimp_mgr_ctrl_blk { + struct srcimap srcimap; + union srcimp_mgr_dirty dirty; +}; + +/* + * Function implementation block. + */ + +static int src_get_rsc_ctrl_blk(void **rblk) +{ + struct src_rsc_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int src_put_rsc_ctrl_blk(void *blk) +{ + kfree((struct src_rsc_ctrl_blk *)blk); + + return 0; +} + +static int src_set_state(void *blk, unsigned int state) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_STATE, state); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_bm(void *blk, unsigned int bm) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_BM, bm); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_rsr(void *blk, unsigned int rsr) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_RSR, rsr); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_sf(void *blk, unsigned int sf) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_SF, sf); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_wr(void *blk, unsigned int wr) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_WR, wr); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_pm(void *blk, unsigned int pm) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_PM, pm); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_rom(void *blk, unsigned int rom) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_ROM, rom); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_vo(void *blk, unsigned int vo) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_VO, vo); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_st(void *blk, unsigned int st) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_ST, st); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_ie(void *blk, unsigned int ie) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_IE, ie); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_ilsz(void *blk, unsigned int ilsz) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_ILSZ, ilsz); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_bp(void *blk, unsigned int bp) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_BP, bp); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_cisz(void *blk, unsigned int cisz) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ccr, SRCCCR_CISZ, cisz); + ctl->dirty.bf.ccr = 1; + return 0; +} + +static int src_set_ca(void *blk, unsigned int ca) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ca, SRCCA_CA, ca); + ctl->dirty.bf.ca = 1; + return 0; +} + +static int src_set_sa(void *blk, unsigned int sa) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->sa, SRCSA_SA, sa); + ctl->dirty.bf.sa = 1; + return 0; +} + +static int src_set_la(void *blk, unsigned int la) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->la, SRCLA_LA, la); + ctl->dirty.bf.la = 1; + return 0; +} + +static int src_set_pitch(void *blk, unsigned int pitch) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->mpr, MPRLH_PITCH, pitch); + ctl->dirty.bf.mpr = 1; + return 0; +} + +static int src_set_clear_zbufs(void *blk, unsigned int clear) +{ + ((struct src_rsc_ctrl_blk *)blk)->dirty.bf.czbfs = (clear ? 1 : 0); + return 0; +} + +static int src_set_dirty(void *blk, unsigned int flags) +{ + ((struct src_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff); + return 0; +} + +static int src_set_dirty_all(void *blk) +{ + ((struct src_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0); + return 0; +} + +#define AR_SLOT_SIZE 4096 +#define AR_SLOT_BLOCK_SIZE 16 +#define AR_PTS_PITCH 6 +#define AR_PARAM_SRC_OFFSET 0x60 + +static unsigned int src_param_pitch_mixer(unsigned int src_idx) +{ + return ((src_idx << 4) + AR_PTS_PITCH + AR_SLOT_SIZE + - AR_PARAM_SRC_OFFSET) % AR_SLOT_SIZE; + +} + +static int src_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct src_rsc_ctrl_blk *ctl = blk; + int i = 0; + + if (ctl->dirty.bf.czbfs) { + /* Clear Z-Buffer registers */ + for (i = 0; i < 8; i++) + hw_write_20kx(hw, SRCUPZ+idx*0x100+i*0x4, 0); + + for (i = 0; i < 4; i++) + hw_write_20kx(hw, SRCDN0Z+idx*0x100+i*0x4, 0); + + for (i = 0; i < 8; i++) + hw_write_20kx(hw, SRCDN1Z+idx*0x100+i*0x4, 0); + + ctl->dirty.bf.czbfs = 0; + } + if (ctl->dirty.bf.mpr) { + /* Take the parameter mixer resource in the same group as that + * the idx src is in for simplicity. Unlike src, all conjugate + * parameter mixer resources must be programmed for + * corresponding conjugate src resources. */ + unsigned int pm_idx = src_param_pitch_mixer(idx); + hw_write_20kx(hw, PRING_LO_HI+4*pm_idx, ctl->mpr); + hw_write_20kx(hw, PMOPLO+8*pm_idx, 0x3); + hw_write_20kx(hw, PMOPHI+8*pm_idx, 0x0); + ctl->dirty.bf.mpr = 0; + } + if (ctl->dirty.bf.sa) { + hw_write_20kx(hw, SRCSA+idx*0x100, ctl->sa); + ctl->dirty.bf.sa = 0; + } + if (ctl->dirty.bf.la) { + hw_write_20kx(hw, SRCLA+idx*0x100, ctl->la); + ctl->dirty.bf.la = 0; + } + if (ctl->dirty.bf.ca) { + hw_write_20kx(hw, SRCCA+idx*0x100, ctl->ca); + ctl->dirty.bf.ca = 0; + } + + /* Write srccf register */ + hw_write_20kx(hw, SRCCF+idx*0x100, 0x0); + + if (ctl->dirty.bf.ccr) { + hw_write_20kx(hw, SRCCCR+idx*0x100, ctl->ccr); + ctl->dirty.bf.ccr = 0; + } + if (ctl->dirty.bf.ctl) { + hw_write_20kx(hw, SRCCTL+idx*0x100, ctl->ctl); + ctl->dirty.bf.ctl = 0; + } + + return 0; +} + +static int src_get_ca(struct hw *hw, unsigned int idx, void *blk) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + ctl->ca = hw_read_20kx(hw, SRCCA+idx*0x100); + ctl->dirty.bf.ca = 0; + + return get_field(ctl->ca, SRCCA_CA); +} + +static unsigned int src_get_dirty(void *blk) +{ + return ((struct src_rsc_ctrl_blk *)blk)->dirty.data; +} + +static unsigned int src_dirty_conj_mask(void) +{ + return 0x20; +} + +static int src_mgr_enbs_src(void *blk, unsigned int idx) +{ + ((struct src_mgr_ctrl_blk *)blk)->enbsa = ~(0x0); + ((struct src_mgr_ctrl_blk *)blk)->dirty.bf.enbsa = 1; + ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32)); + return 0; +} + +static int src_mgr_enb_src(void *blk, unsigned int idx) +{ + ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32)); + ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32)); + return 0; +} + +static int src_mgr_dsb_src(void *blk, unsigned int idx) +{ + ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] &= ~(0x1 << (idx%32)); + ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32)); + return 0; +} + +static int src_mgr_commit_write(struct hw *hw, void *blk) +{ + struct src_mgr_ctrl_blk *ctl = blk; + int i = 0; + unsigned int ret = 0; + + if (ctl->dirty.bf.enbsa) { + do { + ret = hw_read_20kx(hw, SRCENBSTAT); + } while (ret & 0x1); + hw_write_20kx(hw, SRCENBS, ctl->enbsa); + ctl->dirty.bf.enbsa = 0; + } + for (i = 0; i < 8; i++) { + if ((ctl->dirty.data & (0x1 << i))) { + hw_write_20kx(hw, SRCENB+(i*0x100), ctl->enb[i]); + ctl->dirty.data &= ~(0x1 << i); + } + } + + return 0; +} + +static int src_mgr_get_ctrl_blk(void **rblk) +{ + struct src_mgr_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int src_mgr_put_ctrl_blk(void *blk) +{ + kfree((struct src_mgr_ctrl_blk *)blk); + + return 0; +} + +static int srcimp_mgr_get_ctrl_blk(void **rblk) +{ + struct srcimp_mgr_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int srcimp_mgr_put_ctrl_blk(void *blk) +{ + kfree((struct srcimp_mgr_ctrl_blk *)blk); + + return 0; +} + +static int srcimp_mgr_set_imaparc(void *blk, unsigned int slot) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->srcimap.srcaim, SRCAIM_ARC, slot); + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_set_imapuser(void *blk, unsigned int user) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->srcimap.srcaim, SRCAIM_SRC, user); + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_set_imapnxt(void *blk, unsigned int next) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->srcimap.srcaim, SRCAIM_NXT, next); + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_set_imapaddr(void *blk, unsigned int addr) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + ctl->srcimap.idx = addr; + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_commit_write(struct hw *hw, void *blk) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.srcimap) { + hw_write_20kx(hw, SRCIMAP+ctl->srcimap.idx*0x100, + ctl->srcimap.srcaim); + ctl->dirty.bf.srcimap = 0; + } + + return 0; +} + +/* + * AMIXER control block definitions. + */ + +#define AMOPLO_M 0x00000003 +#define AMOPLO_X 0x0003FFF0 +#define AMOPLO_Y 0xFFFC0000 + +#define AMOPHI_SADR 0x000000FF +#define AMOPHI_SE 0x80000000 + +/* AMIXER resource register dirty flags */ +union amixer_dirty { + struct { + u16 amoplo:1; + u16 amophi:1; + u16 rsv:14; + } bf; + u16 data; +}; + +/* AMIXER resource control block */ +struct amixer_rsc_ctrl_blk { + unsigned int amoplo; + unsigned int amophi; + union amixer_dirty dirty; +}; + +static int amixer_set_mode(void *blk, unsigned int mode) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_M, mode); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_iv(void *blk, unsigned int iv) +{ + /* 20k1 amixer does not have this field */ + return 0; +} + +static int amixer_set_x(void *blk, unsigned int x) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_X, x); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_y(void *blk, unsigned int y) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_Y, y); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_sadr(void *blk, unsigned int sadr) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amophi, AMOPHI_SADR, sadr); + ctl->dirty.bf.amophi = 1; + return 0; +} + +static int amixer_set_se(void *blk, unsigned int se) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amophi, AMOPHI_SE, se); + ctl->dirty.bf.amophi = 1; + return 0; +} + +static int amixer_set_dirty(void *blk, unsigned int flags) +{ + ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff); + return 0; +} + +static int amixer_set_dirty_all(void *blk) +{ + ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0); + return 0; +} + +static int amixer_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.amoplo || ctl->dirty.bf.amophi) { + hw_write_20kx(hw, AMOPLO+idx*8, ctl->amoplo); + ctl->dirty.bf.amoplo = 0; + hw_write_20kx(hw, AMOPHI+idx*8, ctl->amophi); + ctl->dirty.bf.amophi = 0; + } + + return 0; +} + +static int amixer_get_y(void *blk) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + return get_field(ctl->amoplo, AMOPLO_Y); +} + +static unsigned int amixer_get_dirty(void *blk) +{ + return ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data; +} + +static int amixer_rsc_get_ctrl_blk(void **rblk) +{ + struct amixer_rsc_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int amixer_rsc_put_ctrl_blk(void *blk) +{ + kfree((struct amixer_rsc_ctrl_blk *)blk); + + return 0; +} + +static int amixer_mgr_get_ctrl_blk(void **rblk) +{ + /*amixer_mgr_ctrl_blk_t *blk;*/ + + *rblk = NULL; + /*blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk;*/ + + return 0; +} + +static int amixer_mgr_put_ctrl_blk(void *blk) +{ + /*kfree((amixer_mgr_ctrl_blk_t *)blk);*/ + + return 0; +} + +/* + * DAIO control block definitions. + */ + +/* Receiver Sample Rate Tracker Control register */ +#define SRTCTL_SRCR 0x000000FF +#define SRTCTL_SRCL 0x0000FF00 +#define SRTCTL_RSR 0x00030000 +#define SRTCTL_DRAT 0x000C0000 +#define SRTCTL_RLE 0x10000000 +#define SRTCTL_RLP 0x20000000 +#define SRTCTL_EC 0x40000000 +#define SRTCTL_ET 0x80000000 + +/* DAIO Receiver register dirty flags */ +union dai_dirty { + struct { + u16 srtctl:1; + u16 rsv:15; + } bf; + u16 data; +}; + +/* DAIO Receiver control block */ +struct dai_ctrl_blk { + unsigned int srtctl; + union dai_dirty dirty; +}; + +/* S/PDIF Transmitter register dirty flags */ +union dao_dirty { + struct { + u16 spos:1; + u16 rsv:15; + } bf; + u16 data; +}; + +/* S/PDIF Transmitter control block */ +struct dao_ctrl_blk { + unsigned int spos; /* S/PDIF Output Channel Status Register */ + union dao_dirty dirty; +}; + +/* Audio Input Mapper RAM */ +#define AIM_ARC 0x00000FFF +#define AIM_NXT 0x007F0000 + +struct daoimap { + unsigned int aim; + unsigned int idx; +}; + +/* I2S Transmitter/Receiver Control register */ +#define I2SCTL_EA 0x00000004 +#define I2SCTL_EI 0x00000010 + +/* S/PDIF Transmitter Control register */ +#define SPOCTL_OE 0x00000001 +#define SPOCTL_OS 0x0000000E +#define SPOCTL_RIV 0x00000010 +#define SPOCTL_LIV 0x00000020 +#define SPOCTL_SR 0x000000C0 + +/* S/PDIF Receiver Control register */ +#define SPICTL_EN 0x00000001 +#define SPICTL_I24 0x00000002 +#define SPICTL_IB 0x00000004 +#define SPICTL_SM 0x00000008 +#define SPICTL_VM 0x00000010 + +/* DAIO manager register dirty flags */ +union daio_mgr_dirty { + struct { + u32 i2soctl:4; + u32 i2sictl:4; + u32 spoctl:4; + u32 spictl:4; + u32 daoimap:1; + u32 rsv:15; + } bf; + u32 data; +}; + +/* DAIO manager control block */ +struct daio_mgr_ctrl_blk { + unsigned int i2sctl; + unsigned int spoctl; + unsigned int spictl; + struct daoimap daoimap; + union daio_mgr_dirty dirty; +}; + +static int dai_srt_set_srcr(void *blk, unsigned int src) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srtctl, SRTCTL_SRCR, src); + ctl->dirty.bf.srtctl = 1; + return 0; +} + +static int dai_srt_set_srcl(void *blk, unsigned int src) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srtctl, SRTCTL_SRCL, src); + ctl->dirty.bf.srtctl = 1; + return 0; +} + +static int dai_srt_set_rsr(void *blk, unsigned int rsr) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srtctl, SRTCTL_RSR, rsr); + ctl->dirty.bf.srtctl = 1; + return 0; +} + +static int dai_srt_set_drat(void *blk, unsigned int drat) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srtctl, SRTCTL_DRAT, drat); + ctl->dirty.bf.srtctl = 1; + return 0; +} + +static int dai_srt_set_ec(void *blk, unsigned int ec) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srtctl, SRTCTL_EC, ec ? 1 : 0); + ctl->dirty.bf.srtctl = 1; + return 0; +} + +static int dai_srt_set_et(void *blk, unsigned int et) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srtctl, SRTCTL_ET, et ? 1 : 0); + ctl->dirty.bf.srtctl = 1; + return 0; +} + +static int dai_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct dai_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.srtctl) { + if (idx < 4) { + /* S/PDIF SRTs */ + hw_write_20kx(hw, SRTSCTL+0x4*idx, ctl->srtctl); + } else { + /* I2S SRT */ + hw_write_20kx(hw, SRTICTL, ctl->srtctl); + } + ctl->dirty.bf.srtctl = 0; + } + + return 0; +} + +static int dai_get_ctrl_blk(void **rblk) +{ + struct dai_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int dai_put_ctrl_blk(void *blk) +{ + kfree((struct dai_ctrl_blk *)blk); + + return 0; +} + +static int dao_set_spos(void *blk, unsigned int spos) +{ + ((struct dao_ctrl_blk *)blk)->spos = spos; + ((struct dao_ctrl_blk *)blk)->dirty.bf.spos = 1; + return 0; +} + +static int dao_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct dao_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.spos) { + if (idx < 4) { + /* S/PDIF SPOSx */ + hw_write_20kx(hw, SPOS+0x4*idx, ctl->spos); + } + ctl->dirty.bf.spos = 0; + } + + return 0; +} + +static int dao_get_spos(void *blk, unsigned int *spos) +{ + *spos = ((struct dao_ctrl_blk *)blk)->spos; + return 0; +} + +static int dao_get_ctrl_blk(void **rblk) +{ + struct dao_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int dao_put_ctrl_blk(void *blk) +{ + kfree((struct dao_ctrl_blk *)blk); + + return 0; +} + +static int daio_mgr_enb_dai(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + if (idx < 4) { + /* S/PDIF input */ + set_field(&ctl->spictl, SPICTL_EN << (idx*8), 1); + ctl->dirty.bf.spictl |= (0x1 << idx); + } else { + /* I2S input */ + idx %= 4; + set_field(&ctl->i2sctl, I2SCTL_EI << (idx*8), 1); + ctl->dirty.bf.i2sictl |= (0x1 << idx); + } + return 0; +} + +static int daio_mgr_dsb_dai(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + if (idx < 4) { + /* S/PDIF input */ + set_field(&ctl->spictl, SPICTL_EN << (idx*8), 0); + ctl->dirty.bf.spictl |= (0x1 << idx); + } else { + /* I2S input */ + idx %= 4; + set_field(&ctl->i2sctl, I2SCTL_EI << (idx*8), 0); + ctl->dirty.bf.i2sictl |= (0x1 << idx); + } + return 0; +} + +static int daio_mgr_enb_dao(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + if (idx < 4) { + /* S/PDIF output */ + set_field(&ctl->spoctl, SPOCTL_OE << (idx*8), 1); + ctl->dirty.bf.spoctl |= (0x1 << idx); + } else { + /* I2S output */ + idx %= 4; + set_field(&ctl->i2sctl, I2SCTL_EA << (idx*8), 1); + ctl->dirty.bf.i2soctl |= (0x1 << idx); + } + return 0; +} + +static int daio_mgr_dsb_dao(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + if (idx < 4) { + /* S/PDIF output */ + set_field(&ctl->spoctl, SPOCTL_OE << (idx*8), 0); + ctl->dirty.bf.spoctl |= (0x1 << idx); + } else { + /* I2S output */ + idx %= 4; + set_field(&ctl->i2sctl, I2SCTL_EA << (idx*8), 0); + ctl->dirty.bf.i2soctl |= (0x1 << idx); + } + return 0; +} + +static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + if (idx < 4) { + /* S/PDIF output */ + switch ((conf & 0x7)) { + case 0: + set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 3); + break; /* CDIF */ + case 1: + set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 0); + break; + case 2: + set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 1); + break; + case 4: + set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 2); + break; + default: + break; + } + set_field(&ctl->spoctl, SPOCTL_LIV << (idx*8), + (conf >> 4) & 0x1); /* Non-audio */ + set_field(&ctl->spoctl, SPOCTL_RIV << (idx*8), + (conf >> 4) & 0x1); /* Non-audio */ + set_field(&ctl->spoctl, SPOCTL_OS << (idx*8), + ((conf >> 3) & 0x1) ? 2 : 2); /* Raw */ + + ctl->dirty.bf.spoctl |= (0x1 << idx); + } else { + /* I2S output */ + /*idx %= 4; */ + } + return 0; +} + +static int daio_mgr_set_imaparc(void *blk, unsigned int slot) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->daoimap.aim, AIM_ARC, slot); + ctl->dirty.bf.daoimap = 1; + return 0; +} + +static int daio_mgr_set_imapnxt(void *blk, unsigned int next) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->daoimap.aim, AIM_NXT, next); + ctl->dirty.bf.daoimap = 1; + return 0; +} + +static int daio_mgr_set_imapaddr(void *blk, unsigned int addr) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + ctl->daoimap.idx = addr; + ctl->dirty.bf.daoimap = 1; + return 0; +} + +static int daio_mgr_commit_write(struct hw *hw, void *blk) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + int i = 0; + + if (ctl->dirty.bf.i2sictl || ctl->dirty.bf.i2soctl) { + for (i = 0; i < 4; i++) { + if ((ctl->dirty.bf.i2sictl & (0x1 << i))) + ctl->dirty.bf.i2sictl &= ~(0x1 << i); + + if ((ctl->dirty.bf.i2soctl & (0x1 << i))) + ctl->dirty.bf.i2soctl &= ~(0x1 << i); + } + hw_write_20kx(hw, I2SCTL, ctl->i2sctl); + mdelay(1); + } + if (ctl->dirty.bf.spoctl) { + for (i = 0; i < 4; i++) { + if ((ctl->dirty.bf.spoctl & (0x1 << i))) + ctl->dirty.bf.spoctl &= ~(0x1 << i); + } + hw_write_20kx(hw, SPOCTL, ctl->spoctl); + mdelay(1); + } + if (ctl->dirty.bf.spictl) { + for (i = 0; i < 4; i++) { + if ((ctl->dirty.bf.spictl & (0x1 << i))) + ctl->dirty.bf.spictl &= ~(0x1 << i); + } + hw_write_20kx(hw, SPICTL, ctl->spictl); + mdelay(1); + } + if (ctl->dirty.bf.daoimap) { + hw_write_20kx(hw, DAOIMAP+ctl->daoimap.idx*4, + ctl->daoimap.aim); + ctl->dirty.bf.daoimap = 0; + } + + return 0; +} + +static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) +{ + struct daio_mgr_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + blk->i2sctl = hw_read_20kx(hw, I2SCTL); + blk->spoctl = hw_read_20kx(hw, SPOCTL); + blk->spictl = hw_read_20kx(hw, SPICTL); + + *rblk = blk; + + return 0; +} + +static int daio_mgr_put_ctrl_blk(void *blk) +{ + kfree((struct daio_mgr_ctrl_blk *)blk); + + return 0; +} + +/* Card hardware initialization block */ +struct dac_conf { + unsigned int msr; /* master sample rate in rsrs */ +}; + +struct adc_conf { + unsigned int msr; /* master sample rate in rsrs */ + unsigned char input; /* the input source of ADC */ + unsigned char mic20db; /* boost mic by 20db if input is microphone */ +}; + +struct daio_conf { + unsigned int msr; /* master sample rate in rsrs */ +}; + +struct trn_conf { + unsigned long vm_pgt_phys; +}; + +static int hw_daio_init(struct hw *hw, const struct daio_conf *info) +{ + u32 i2sorg = 0; + u32 spdorg = 0; + + /* Read I2S CTL. Keep original value. */ + /*i2sorg = hw_read_20kx(hw, I2SCTL);*/ + i2sorg = 0x94040404; /* enable all audio out and I2S-D input */ + /* Program I2S with proper master sample rate and enable + * the correct I2S channel. */ + i2sorg &= 0xfffffffc; + + /* Enable S/PDIF-out-A in fixed 24-bit data + * format and default to 48kHz. */ + /* Disable all before doing any changes. */ + hw_write_20kx(hw, SPOCTL, 0x0); + spdorg = 0x05; + + switch (info->msr) { + case 1: + i2sorg |= 1; + spdorg |= (0x0 << 6); + break; + case 2: + i2sorg |= 2; + spdorg |= (0x1 << 6); + break; + case 4: + i2sorg |= 3; + spdorg |= (0x2 << 6); + break; + default: + i2sorg |= 1; + break; + } + + hw_write_20kx(hw, I2SCTL, i2sorg); + hw_write_20kx(hw, SPOCTL, spdorg); + + /* Enable S/PDIF-in-A in fixed 24-bit data format. */ + /* Disable all before doing any changes. */ + hw_write_20kx(hw, SPICTL, 0x0); + mdelay(1); + spdorg = 0x0a0a0a0a; + hw_write_20kx(hw, SPICTL, spdorg); + mdelay(1); + + return 0; +} + +/* TRANSPORT operations */ +static int hw_trn_init(struct hw *hw, const struct trn_conf *info) +{ + u32 trnctl = 0; + unsigned long ptp_phys_low = 0, ptp_phys_high = 0; + + /* Set up device page table */ + if ((~0UL) == info->vm_pgt_phys) { + printk(KERN_ERR "Wrong device page table page address!\n"); + return -1; + } + + trnctl = 0x13; /* 32-bit, 4k-size page */ +#if BITS_PER_LONG == 64 + ptp_phys_low = info->vm_pgt_phys & ((1UL<<32)-1); + ptp_phys_high = (info->vm_pgt_phys>>32) & ((1UL<<32)-1); + trnctl |= (1<<2); +#elif BITS_PER_LONG == 32 + ptp_phys_low = info->vm_pgt_phys & (~0UL); + ptp_phys_high = 0; +#else +# error "Unknown BITS_PER_LONG!" +#endif +#if PAGE_SIZE == 8192 + trnctl |= (1<<5); +#endif + hw_write_20kx(hw, PTPALX, ptp_phys_low); + hw_write_20kx(hw, PTPAHX, ptp_phys_high); + hw_write_20kx(hw, TRNCTL, trnctl); + hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */ + + return 0; +} + +/* Card initialization */ +#define GCTL_EAC 0x00000001 +#define GCTL_EAI 0x00000002 +#define GCTL_BEP 0x00000004 +#define GCTL_BES 0x00000008 +#define GCTL_DSP 0x00000010 +#define GCTL_DBP 0x00000020 +#define GCTL_ABP 0x00000040 +#define GCTL_TBP 0x00000080 +#define GCTL_SBP 0x00000100 +#define GCTL_FBP 0x00000200 +#define GCTL_XA 0x00000400 +#define GCTL_ET 0x00000800 +#define GCTL_PR 0x00001000 +#define GCTL_MRL 0x00002000 +#define GCTL_SDE 0x00004000 +#define GCTL_SDI 0x00008000 +#define GCTL_SM 0x00010000 +#define GCTL_SR 0x00020000 +#define GCTL_SD 0x00040000 +#define GCTL_SE 0x00080000 +#define GCTL_AID 0x00100000 + +static int hw_pll_init(struct hw *hw, unsigned int rsr) +{ + unsigned int pllctl; + int i = 0; + + pllctl = (48000 == rsr) ? 0x1480a001 : 0x1480a731; + for (i = 0; i < 3; i++) { + if (hw_read_20kx(hw, PLLCTL) == pllctl) + break; + + hw_write_20kx(hw, PLLCTL, pllctl); + mdelay(40); + } + if (i >= 3) { + printk(KERN_ALERT "PLL initialization failed!!!\n"); + return -EBUSY; + } + + return 0; +} + +static int hw_auto_init(struct hw *hw) +{ + unsigned int gctl; + int i; + + gctl = hw_read_20kx(hw, GCTL); + set_field(&gctl, GCTL_EAI, 0); + hw_write_20kx(hw, GCTL, gctl); + set_field(&gctl, GCTL_EAI, 1); + hw_write_20kx(hw, GCTL, gctl); + mdelay(10); + for (i = 0; i < 400000; i++) { + gctl = hw_read_20kx(hw, GCTL); + if (get_field(gctl, GCTL_AID)) + break; + } + if (!get_field(gctl, GCTL_AID)) { + printk(KERN_ALERT "Card Auto-init failed!!!\n"); + return -EBUSY; + } + + return 0; +} + +static int i2c_unlock(struct hw *hw) +{ + if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa) + return 0; + + hw_write_pci(hw, 0xcc, 0x8c); + hw_write_pci(hw, 0xcc, 0x0e); + if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa) + return 0; + + hw_write_pci(hw, 0xcc, 0xee); + hw_write_pci(hw, 0xcc, 0xaa); + if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa) + return 0; + + return -1; +} + +static void i2c_lock(struct hw *hw) +{ + if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa) + hw_write_pci(hw, 0xcc, 0x00); +} + +static void i2c_write(struct hw *hw, u32 device, u32 addr, u32 data) +{ + unsigned int ret = 0; + + do { + ret = hw_read_pci(hw, 0xEC); + } while (!(ret & 0x800000)); + hw_write_pci(hw, 0xE0, device); + hw_write_pci(hw, 0xE4, (data << 8) | (addr & 0xff)); +} + +/* DAC operations */ + +static int hw_reset_dac(struct hw *hw) +{ + u32 i = 0; + u16 gpioorg = 0; + unsigned int ret = 0; + + if (i2c_unlock(hw)) + return -1; + + do { + ret = hw_read_pci(hw, 0xEC); + } while (!(ret & 0x800000)); + hw_write_pci(hw, 0xEC, 0x05); /* write to i2c status control */ + + /* To be effective, need to reset the DAC twice. */ + for (i = 0; i < 2; i++) { + /* set gpio */ + mdelay(100); + gpioorg = (u16)hw_read_20kx(hw, GPIO); + gpioorg &= 0xfffd; + hw_write_20kx(hw, GPIO, gpioorg); + mdelay(1); + hw_write_20kx(hw, GPIO, gpioorg | 0x2); + } + + i2c_write(hw, 0x00180080, 0x01, 0x80); + i2c_write(hw, 0x00180080, 0x02, 0x10); + + i2c_lock(hw); + + return 0; +} + +static int hw_dac_init(struct hw *hw, const struct dac_conf *info) +{ + u32 data = 0; + u16 gpioorg = 0; + u16 subsys_id = 0; + unsigned int ret = 0; + + pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); + if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { + /* SB055x, unmute outputs */ + gpioorg = (u16)hw_read_20kx(hw, GPIO); + gpioorg &= 0xffbf; /* set GPIO6 to low */ + gpioorg |= 2; /* set GPIO1 to high */ + hw_write_20kx(hw, GPIO, gpioorg); + return 0; + } + + /* mute outputs */ + gpioorg = (u16)hw_read_20kx(hw, GPIO); + gpioorg &= 0xffbf; + hw_write_20kx(hw, GPIO, gpioorg); + + hw_reset_dac(hw); + + if (i2c_unlock(hw)) + return -1; + + hw_write_pci(hw, 0xEC, 0x05); /* write to i2c status control */ + do { + ret = hw_read_pci(hw, 0xEC); + } while (!(ret & 0x800000)); + + switch (info->msr) { + case 1: + data = 0x24; + break; + case 2: + data = 0x25; + break; + case 4: + data = 0x26; + break; + default: + data = 0x24; + break; + } + + i2c_write(hw, 0x00180080, 0x06, data); + i2c_write(hw, 0x00180080, 0x09, data); + i2c_write(hw, 0x00180080, 0x0c, data); + i2c_write(hw, 0x00180080, 0x0f, data); + + i2c_lock(hw); + + /* unmute outputs */ + gpioorg = (u16)hw_read_20kx(hw, GPIO); + gpioorg = gpioorg | 0x40; + hw_write_20kx(hw, GPIO, gpioorg); + + return 0; +} + +/* ADC operations */ + +static int is_adc_input_selected_SB055x(struct hw *hw, enum ADCSRC type) +{ + u32 data = 0; + return data; +} + +static int is_adc_input_selected_SBx(struct hw *hw, enum ADCSRC type) +{ + u32 data = 0; + + data = hw_read_20kx(hw, GPIO); + switch (type) { + case ADC_MICIN: + data = ((data & (0x1<<7)) && (data & (0x1<<8))); + break; + case ADC_LINEIN: + data = (!(data & (0x1<<7)) && (data & (0x1<<8))); + break; + case ADC_NONE: /* Digital I/O */ + data = (!(data & (0x1<<8))); + break; + default: + data = 0; + } + return data; +} + +static int is_adc_input_selected_hendrix(struct hw *hw, enum ADCSRC type) +{ + u32 data = 0; + + data = hw_read_20kx(hw, GPIO); + switch (type) { + case ADC_MICIN: + data = (data & (0x1 << 7)) ? 1 : 0; + break; + case ADC_LINEIN: + data = (data & (0x1 << 7)) ? 0 : 1; + break; + default: + data = 0; + } + return data; +} + +static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) +{ + u16 subsys_id = 0; + + pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); + if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { + /* SB055x cards */ + return is_adc_input_selected_SB055x(hw, type); + } else if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { + /* SB073x cards */ + return is_adc_input_selected_hendrix(hw, type); + } else if ((subsys_id & 0xf000) == 0x6000) { + /* Vista compatible cards */ + return is_adc_input_selected_hendrix(hw, type); + } else { + return is_adc_input_selected_SBx(hw, type); + } +} + +static int +adc_input_select_SB055x(struct hw *hw, enum ADCSRC type, unsigned char boost) +{ + u32 data = 0; + + /* + * check and set the following GPIO bits accordingly + * ADC_Gain = GPIO2 + * DRM_off = GPIO3 + * Mic_Pwr_on = GPIO7 + * Digital_IO_Sel = GPIO8 + * Mic_Sw = GPIO9 + * Aux/MicLine_Sw = GPIO12 + */ + data = hw_read_20kx(hw, GPIO); + data &= 0xec73; + switch (type) { + case ADC_MICIN: + data |= (0x1<<7) | (0x1<<8) | (0x1<<9) ; + data |= boost ? (0x1<<2) : 0; + break; + case ADC_LINEIN: + data |= (0x1<<8); + break; + case ADC_AUX: + data |= (0x1<<8) | (0x1<<12); + break; + case ADC_NONE: + data |= (0x1<<12); /* set to digital */ + break; + default: + return -1; + } + + hw_write_20kx(hw, GPIO, data); + + return 0; +} + + +static int +adc_input_select_SBx(struct hw *hw, enum ADCSRC type, unsigned char boost) +{ + u32 data = 0; + u32 i2c_data = 0; + unsigned int ret = 0; + + if (i2c_unlock(hw)) + return -1; + + do { + ret = hw_read_pci(hw, 0xEC); + } while (!(ret & 0x800000)); /* i2c ready poll */ + /* set i2c access mode as Direct Control */ + hw_write_pci(hw, 0xEC, 0x05); + + data = hw_read_20kx(hw, GPIO); + switch (type) { + case ADC_MICIN: + data |= ((0x1 << 7) | (0x1 << 8)); + i2c_data = 0x1; /* Mic-in */ + break; + case ADC_LINEIN: + data &= ~(0x1 << 7); + data |= (0x1 << 8); + i2c_data = 0x2; /* Line-in */ + break; + case ADC_NONE: + data &= ~(0x1 << 8); + i2c_data = 0x0; /* set to Digital */ + break; + default: + i2c_lock(hw); + return -1; + } + hw_write_20kx(hw, GPIO, data); + i2c_write(hw, 0x001a0080, 0x2a, i2c_data); + if (boost) { + i2c_write(hw, 0x001a0080, 0x1c, 0xe7); /* +12dB boost */ + i2c_write(hw, 0x001a0080, 0x1e, 0xe7); /* +12dB boost */ + } else { + i2c_write(hw, 0x001a0080, 0x1c, 0xcf); /* No boost */ + i2c_write(hw, 0x001a0080, 0x1e, 0xcf); /* No boost */ + } + + i2c_lock(hw); + + return 0; +} + +static int +adc_input_select_hendrix(struct hw *hw, enum ADCSRC type, unsigned char boost) +{ + u32 data = 0; + u32 i2c_data = 0; + unsigned int ret = 0; + + if (i2c_unlock(hw)) + return -1; + + do { + ret = hw_read_pci(hw, 0xEC); + } while (!(ret & 0x800000)); /* i2c ready poll */ + /* set i2c access mode as Direct Control */ + hw_write_pci(hw, 0xEC, 0x05); + + data = hw_read_20kx(hw, GPIO); + switch (type) { + case ADC_MICIN: + data |= (0x1 << 7); + i2c_data = 0x1; /* Mic-in */ + break; + case ADC_LINEIN: + data &= ~(0x1 << 7); + i2c_data = 0x2; /* Line-in */ + break; + default: + i2c_lock(hw); + return -1; + } + hw_write_20kx(hw, GPIO, data); + i2c_write(hw, 0x001a0080, 0x2a, i2c_data); + if (boost) { + i2c_write(hw, 0x001a0080, 0x1c, 0xe7); /* +12dB boost */ + i2c_write(hw, 0x001a0080, 0x1e, 0xe7); /* +12dB boost */ + } else { + i2c_write(hw, 0x001a0080, 0x1c, 0xcf); /* No boost */ + i2c_write(hw, 0x001a0080, 0x1e, 0xcf); /* No boost */ + } + + i2c_lock(hw); + + return 0; +} + +static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) +{ + u16 subsys_id = 0; + + pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); + if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { + /* SB055x cards */ + return adc_input_select_SB055x(hw, type, (ADC_MICIN == type)); + } else if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { + /* SB073x cards */ + return adc_input_select_hendrix(hw, type, (ADC_MICIN == type)); + } else if ((subsys_id & 0xf000) == 0x6000) { + /* Vista compatible cards */ + return adc_input_select_hendrix(hw, type, (ADC_MICIN == type)); + } else { + return adc_input_select_SBx(hw, type, (ADC_MICIN == type)); + } +} + +static int adc_init_SB055x(struct hw *hw, int input, int mic20db) +{ + return adc_input_select_SB055x(hw, input, mic20db); +} + +static int adc_init_SBx(struct hw *hw, int input, int mic20db) +{ + u16 gpioorg; + u16 input_source; + u32 adcdata = 0; + unsigned int ret = 0; + + input_source = 0x100; /* default to analog */ + switch (input) { + case ADC_MICIN: + adcdata = 0x1; + input_source = 0x180; /* set GPIO7 to select Mic */ + break; + case ADC_LINEIN: + adcdata = 0x2; + break; + case ADC_VIDEO: + adcdata = 0x4; + break; + case ADC_AUX: + adcdata = 0x8; + break; + case ADC_NONE: + adcdata = 0x0; + input_source = 0x0; /* set to Digital */ + break; + default: + break; + } + + if (i2c_unlock(hw)) + return -1; + + do { + ret = hw_read_pci(hw, 0xEC); + } while (!(ret & 0x800000)); /* i2c ready poll */ + hw_write_pci(hw, 0xEC, 0x05); /* write to i2c status control */ + + i2c_write(hw, 0x001a0080, 0x0e, 0x08); + i2c_write(hw, 0x001a0080, 0x18, 0x0a); + i2c_write(hw, 0x001a0080, 0x28, 0x86); + i2c_write(hw, 0x001a0080, 0x2a, adcdata); + + if (mic20db) { + i2c_write(hw, 0x001a0080, 0x1c, 0xf7); + i2c_write(hw, 0x001a0080, 0x1e, 0xf7); + } else { + i2c_write(hw, 0x001a0080, 0x1c, 0xcf); + i2c_write(hw, 0x001a0080, 0x1e, 0xcf); + } + + if (!(hw_read_20kx(hw, ID0) & 0x100)) + i2c_write(hw, 0x001a0080, 0x16, 0x26); + + i2c_lock(hw); + + gpioorg = (u16)hw_read_20kx(hw, GPIO); + gpioorg &= 0xfe7f; + gpioorg |= input_source; + hw_write_20kx(hw, GPIO, gpioorg); + + return 0; +} + +static int hw_adc_init(struct hw *hw, const struct adc_conf *info) +{ + int err = 0; + u16 subsys_id = 0; + + pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); + if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { + /* Sb055x card */ + err = adc_init_SB055x(hw, info->input, info->mic20db); + } else { + err = adc_init_SBx(hw, info->input, info->mic20db); + } + + return err; +} + +static int hw_have_digit_io_switch(struct hw *hw) +{ + u16 subsys_id = 0; + + pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); + /* SB073x and Vista compatible cards have no digit IO switch */ + return !((subsys_id == 0x0029) || (subsys_id == 0x0031) + || ((subsys_id & 0xf000) == 0x6000)); +} + +#define UAA_CFG_PWRSTATUS 0x44 +#define UAA_CFG_SPACE_FLAG 0xA0 +#define UAA_CORE_CHANGE 0x3FFC +static int uaa_to_xfi(struct pci_dev *pci) +{ + unsigned int bar0, bar1, bar2, bar3, bar4, bar5; + unsigned int cmd, irq, cl_size, l_timer, pwr; + unsigned int CTLA, CTLZ, CTLL, CTLX, CTL_, CTLF, CTLi; + unsigned int is_uaa = 0; + unsigned int data[4] = {0}; + unsigned int io_base; + void *mem_base; + int i = 0; + + /* By default, Hendrix card UAA Bar0 should be using memory... */ + io_base = pci_resource_start(pci, 0); + mem_base = ioremap(io_base, pci_resource_len(pci, 0)); + if (NULL == mem_base) + return -ENOENT; + + CTLX = ___constant_swab32(*((unsigned int *)"CTLX")); + CTL_ = ___constant_swab32(*((unsigned int *)"CTL-")); + CTLF = ___constant_swab32(*((unsigned int *)"CTLF")); + CTLi = ___constant_swab32(*((unsigned int *)"CTLi")); + CTLA = ___constant_swab32(*((unsigned int *)"CTLA")); + CTLZ = ___constant_swab32(*((unsigned int *)"CTLZ")); + CTLL = ___constant_swab32(*((unsigned int *)"CTLL")); + + /* Read current mode from Mode Change Register */ + for (i = 0; i < 4; i++) + data[i] = readl(mem_base + UAA_CORE_CHANGE); + + /* Determine current mode... */ + if (data[0] == CTLA) { + is_uaa = ((data[1] == CTLZ && data[2] == CTLL + && data[3] == CTLA) || (data[1] == CTLA + && data[2] == CTLZ && data[3] == CTLL)); + } else if (data[0] == CTLZ) { + is_uaa = (data[1] == CTLL + && data[2] == CTLA && data[3] == CTLA); + } else if (data[0] == CTLL) { + is_uaa = (data[1] == CTLA + && data[2] == CTLA && data[3] == CTLZ); + } else { + is_uaa = 0; + } + + if (!is_uaa) { + /* Not in UAA mode currently. Return directly. */ + iounmap(mem_base); + return 0; + } + + pci_read_config_dword(pci, PCI_BASE_ADDRESS_0, &bar0); + pci_read_config_dword(pci, PCI_BASE_ADDRESS_1, &bar1); + pci_read_config_dword(pci, PCI_BASE_ADDRESS_2, &bar2); + pci_read_config_dword(pci, PCI_BASE_ADDRESS_3, &bar3); + pci_read_config_dword(pci, PCI_BASE_ADDRESS_4, &bar4); + pci_read_config_dword(pci, PCI_BASE_ADDRESS_5, &bar5); + pci_read_config_dword(pci, PCI_INTERRUPT_LINE, &irq); + pci_read_config_dword(pci, PCI_CACHE_LINE_SIZE, &cl_size); + pci_read_config_dword(pci, PCI_LATENCY_TIMER, &l_timer); + pci_read_config_dword(pci, UAA_CFG_PWRSTATUS, &pwr); + pci_read_config_dword(pci, PCI_COMMAND, &cmd); + + /* Set up X-Fi core PCI configuration space. */ + /* Switch to X-Fi config space with BAR0 exposed. */ + pci_write_config_dword(pci, UAA_CFG_SPACE_FLAG, 0x87654321); + /* Copy UAA's BAR5 into X-Fi BAR0 */ + pci_write_config_dword(pci, PCI_BASE_ADDRESS_0, bar5); + /* Switch to X-Fi config space without BAR0 exposed. */ + pci_write_config_dword(pci, UAA_CFG_SPACE_FLAG, 0x12345678); + pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, bar1); + pci_write_config_dword(pci, PCI_BASE_ADDRESS_2, bar2); + pci_write_config_dword(pci, PCI_BASE_ADDRESS_3, bar3); + pci_write_config_dword(pci, PCI_BASE_ADDRESS_4, bar4); + pci_write_config_dword(pci, PCI_INTERRUPT_LINE, irq); + pci_write_config_dword(pci, PCI_CACHE_LINE_SIZE, cl_size); + pci_write_config_dword(pci, PCI_LATENCY_TIMER, l_timer); + pci_write_config_dword(pci, UAA_CFG_PWRSTATUS, pwr); + pci_write_config_dword(pci, PCI_COMMAND, cmd); + + /* Switch to X-Fi mode */ + writel(CTLX, (mem_base + UAA_CORE_CHANGE)); + writel(CTL_, (mem_base + UAA_CORE_CHANGE)); + writel(CTLF, (mem_base + UAA_CORE_CHANGE)); + writel(CTLi, (mem_base + UAA_CORE_CHANGE)); + + iounmap(mem_base); + + return 0; +} + +static int hw_card_start(struct hw *hw) +{ + int err = 0; + struct pci_dev *pci = hw->pci; + u16 subsys_id = 0; + unsigned int dma_mask = 0; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + /* Set DMA transfer mask */ + dma_mask = CT_XFI_DMA_MASK; + if (pci_set_dma_mask(pci, dma_mask) < 0 || + pci_set_consistent_dma_mask(pci, dma_mask) < 0) { + printk(KERN_ERR "architecture does not support PCI " + "busmaster DMA with mask 0x%x\n", dma_mask); + err = -ENXIO; + goto error1; + } + + err = pci_request_regions(pci, "XFi"); + if (err < 0) + goto error1; + + /* Switch to X-Fi mode from UAA mode if neeeded */ + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsys_id); + if ((0x5 == pci->device) && (0x6000 == (subsys_id & 0x6000))) { + err = uaa_to_xfi(pci); + if (err) + goto error2; + + hw->io_base = pci_resource_start(pci, 5); + } else { + hw->io_base = pci_resource_start(pci, 0); + } + + /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED, + atc->chip_details->nm_card, hw))) { + goto error2; + } + hw->irq = pci->irq; + */ + + pci_set_master(pci); + + return 0; + +error2: + pci_release_regions(pci); + hw->io_base = 0; +error1: + pci_disable_device(pci); + return err; +} + +static int hw_card_stop(struct hw *hw) +{ + /* TODO: Disable interrupt and so on... */ + return 0; +} + +static int hw_card_shutdown(struct hw *hw) +{ + if (hw->irq >= 0) + free_irq(hw->irq, hw); + + hw->irq = -1; + + if (NULL != ((void *)hw->mem_base)) + iounmap((void *)hw->mem_base); + + hw->mem_base = (unsigned long)NULL; + + if (hw->io_base) + pci_release_regions(hw->pci); + + hw->io_base = 0; + + pci_disable_device(hw->pci); + + return 0; +} + +static int hw_card_init(struct hw *hw, struct card_conf *info) +{ + int err; + unsigned int gctl; + u16 subsys_id = 0; + u32 data = 0; + struct dac_conf dac_info = {0}; + struct adc_conf adc_info = {0}; + struct daio_conf daio_info = {0}; + struct trn_conf trn_info = {0}; + + /* Get PCI io port base address and do Hendrix switch if needed. */ + if (!hw->io_base) { + err = hw_card_start(hw); + if (err) + return err; + } + + /* PLL init */ + err = hw_pll_init(hw, info->rsr); + if (err < 0) + return err; + + /* kick off auto-init */ + err = hw_auto_init(hw); + if (err < 0) + return err; + + /* Enable audio ring */ + gctl = hw_read_20kx(hw, GCTL); + set_field(&gctl, GCTL_EAC, 1); + set_field(&gctl, GCTL_DBP, 1); + set_field(&gctl, GCTL_TBP, 1); + set_field(&gctl, GCTL_FBP, 1); + set_field(&gctl, GCTL_ET, 1); + hw_write_20kx(hw, GCTL, gctl); + mdelay(10); + + /* Reset all global pending interrupts */ + hw_write_20kx(hw, GIE, 0); + /* Reset all SRC pending interrupts */ + hw_write_20kx(hw, SRCIP, 0); + mdelay(30); + + pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); + /* Detect the card ID and configure GPIO accordingly. */ + if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { + /* SB055x cards */ + hw_write_20kx(hw, GPIOCTL, 0x13fe); + } else if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { + /* SB073x cards */ + hw_write_20kx(hw, GPIOCTL, 0x00e6); + } else if ((subsys_id & 0xf000) == 0x6000) { + /* Vista compatible cards */ + hw_write_20kx(hw, GPIOCTL, 0x00c2); + } else { + hw_write_20kx(hw, GPIOCTL, 0x01e6); + } + + trn_info.vm_pgt_phys = info->vm_pgt_phys; + err = hw_trn_init(hw, &trn_info); + if (err < 0) + return err; + + daio_info.msr = info->msr; + err = hw_daio_init(hw, &daio_info); + if (err < 0) + return err; + + dac_info.msr = info->msr; + err = hw_dac_init(hw, &dac_info); + if (err < 0) + return err; + + adc_info.msr = info->msr; + adc_info.input = ADC_LINEIN; + adc_info.mic20db = 0; + err = hw_adc_init(hw, &adc_info); + if (err < 0) + return err; + + data = hw_read_20kx(hw, SRCMCTL); + data |= 0x1; /* Enables input from the audio ring */ + hw_write_20kx(hw, SRCMCTL, data); + + return 0; +} + +static u32 hw_read_20kx(struct hw *hw, u32 reg) +{ + u32 value; + unsigned long flags; + + spin_lock_irqsave( + &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags); + outl(reg, hw->io_base + 0x0); + value = inl(hw->io_base + 0x4); + spin_unlock_irqrestore( + &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags); + + return value; +} + +static void hw_write_20kx(struct hw *hw, u32 reg, u32 data) +{ + unsigned long flags; + + spin_lock_irqsave( + &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags); + outl(reg, hw->io_base + 0x0); + outl(data, hw->io_base + 0x4); + spin_unlock_irqrestore( + &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags); + +} + +static u32 hw_read_pci(struct hw *hw, u32 reg) +{ + u32 value; + unsigned long flags; + + spin_lock_irqsave( + &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); + outl(reg, hw->io_base + 0x10); + value = inl(hw->io_base + 0x14); + spin_unlock_irqrestore( + &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); + + return value; +} + +static void hw_write_pci(struct hw *hw, u32 reg, u32 data) +{ + unsigned long flags; + + spin_lock_irqsave( + &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); + outl(reg, hw->io_base + 0x10); + outl(data, hw->io_base + 0x14); + spin_unlock_irqrestore( + &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); +} + +int create_20k1_hw_obj(struct hw **rhw) +{ + struct hw *hw; + struct hw20k1 *hw20k1; + + *rhw = NULL; + hw20k1 = kzalloc(sizeof(*hw20k1), GFP_KERNEL); + if (NULL == hw20k1) + return -ENOMEM; + + spin_lock_init(&hw20k1->reg_20k1_lock); + spin_lock_init(&hw20k1->reg_pci_lock); + + hw = &hw20k1->hw; + + hw->io_base = 0; + hw->mem_base = (unsigned long)NULL; + hw->irq = -1; + + hw->card_init = hw_card_init; + hw->card_stop = hw_card_stop; + hw->pll_init = hw_pll_init; + hw->is_adc_source_selected = hw_is_adc_input_selected; + hw->select_adc_source = hw_adc_input_select; + hw->have_digit_io_switch = hw_have_digit_io_switch; + + hw->src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk; + hw->src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk; + hw->src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk; + hw->src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk; + hw->src_set_state = src_set_state; + hw->src_set_bm = src_set_bm; + hw->src_set_rsr = src_set_rsr; + hw->src_set_sf = src_set_sf; + hw->src_set_wr = src_set_wr; + hw->src_set_pm = src_set_pm; + hw->src_set_rom = src_set_rom; + hw->src_set_vo = src_set_vo; + hw->src_set_st = src_set_st; + hw->src_set_ie = src_set_ie; + hw->src_set_ilsz = src_set_ilsz; + hw->src_set_bp = src_set_bp; + hw->src_set_cisz = src_set_cisz; + hw->src_set_ca = src_set_ca; + hw->src_set_sa = src_set_sa; + hw->src_set_la = src_set_la; + hw->src_set_pitch = src_set_pitch; + hw->src_set_dirty = src_set_dirty; + hw->src_set_clear_zbufs = src_set_clear_zbufs; + hw->src_set_dirty_all = src_set_dirty_all; + hw->src_commit_write = src_commit_write; + hw->src_get_ca = src_get_ca; + hw->src_get_dirty = src_get_dirty; + hw->src_dirty_conj_mask = src_dirty_conj_mask; + hw->src_mgr_enbs_src = src_mgr_enbs_src; + hw->src_mgr_enb_src = src_mgr_enb_src; + hw->src_mgr_dsb_src = src_mgr_dsb_src; + hw->src_mgr_commit_write = src_mgr_commit_write; + + hw->srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk; + hw->srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk; + hw->srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc; + hw->srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser; + hw->srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt; + hw->srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr; + hw->srcimp_mgr_commit_write = srcimp_mgr_commit_write; + + hw->amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk; + hw->amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk; + hw->amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk; + hw->amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk; + hw->amixer_set_mode = amixer_set_mode; + hw->amixer_set_iv = amixer_set_iv; + hw->amixer_set_x = amixer_set_x; + hw->amixer_set_y = amixer_set_y; + hw->amixer_set_sadr = amixer_set_sadr; + hw->amixer_set_se = amixer_set_se; + hw->amixer_set_dirty = amixer_set_dirty; + hw->amixer_set_dirty_all = amixer_set_dirty_all; + hw->amixer_commit_write = amixer_commit_write; + hw->amixer_get_y = amixer_get_y; + hw->amixer_get_dirty = amixer_get_dirty; + + hw->dai_get_ctrl_blk = dai_get_ctrl_blk; + hw->dai_put_ctrl_blk = dai_put_ctrl_blk; + hw->dai_srt_set_srco = dai_srt_set_srcr; + hw->dai_srt_set_srcm = dai_srt_set_srcl; + hw->dai_srt_set_rsr = dai_srt_set_rsr; + hw->dai_srt_set_drat = dai_srt_set_drat; + hw->dai_srt_set_ec = dai_srt_set_ec; + hw->dai_srt_set_et = dai_srt_set_et; + hw->dai_commit_write = dai_commit_write; + + hw->dao_get_ctrl_blk = dao_get_ctrl_blk; + hw->dao_put_ctrl_blk = dao_put_ctrl_blk; + hw->dao_set_spos = dao_set_spos; + hw->dao_commit_write = dao_commit_write; + hw->dao_get_spos = dao_get_spos; + + hw->daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk; + hw->daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk; + hw->daio_mgr_enb_dai = daio_mgr_enb_dai; + hw->daio_mgr_dsb_dai = daio_mgr_dsb_dai; + hw->daio_mgr_enb_dao = daio_mgr_enb_dao; + hw->daio_mgr_dsb_dao = daio_mgr_dsb_dao; + hw->daio_mgr_dao_init = daio_mgr_dao_init; + hw->daio_mgr_set_imaparc = daio_mgr_set_imaparc; + hw->daio_mgr_set_imapnxt = daio_mgr_set_imapnxt; + hw->daio_mgr_set_imapaddr = daio_mgr_set_imapaddr; + hw->daio_mgr_commit_write = daio_mgr_commit_write; + + *rhw = hw; + + return 0; +} + +int destroy_20k1_hw_obj(struct hw *hw) +{ + if (hw->io_base) + hw_card_shutdown(hw); + + kfree(container_of(hw, struct hw20k1, hw)); + return 0; +} diff --git a/sound/pci/ctxfi/cthw20k1.h b/sound/pci/ctxfi/cthw20k1.h new file mode 100644 index 0000000..02f72fb --- /dev/null +++ b/sound/pci/ctxfi/cthw20k1.h @@ -0,0 +1,26 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File cthw20k1.h + * + * @Brief + * This file contains the definition of hardware access methord. + * + * @Author Liu Chun + * @Date May 13 2008 + * + */ + +#ifndef CTHW20K1_H +#define CTHW20K1_H + +#include "cthardware.h" + +int create_20k1_hw_obj(struct hw **rhw); +int destroy_20k1_hw_obj(struct hw *hw); + +#endif /* CTHW20K1_H */ diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c new file mode 100644 index 0000000..cdcb75c --- /dev/null +++ b/sound/pci/ctxfi/cthw20k2.c @@ -0,0 +1,2133 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File cthw20k2.c + * + * @Brief + * This file contains the implementation of hardware access methord for 20k2. + * + * @Author Liu Chun + * @Date May 14 2008 + * + */ + +#include "cthw20k2.h" +#include "ct20k2reg.h" +#include +#include +#include +#include +#include +#include +#include + +#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bits */ + +static u32 hw_read_20kx(struct hw *hw, u32 reg); +static void hw_write_20kx(struct hw *hw, u32 reg, u32 data); + +/* + * Type definition block. + * The layout of control structures can be directly applied on 20k2 chip. + */ + +/* + * SRC control block definitions. + */ + +/* SRC resource control block */ +#define SRCCTL_STATE 0x00000007 +#define SRCCTL_BM 0x00000008 +#define SRCCTL_RSR 0x00000030 +#define SRCCTL_SF 0x000001C0 +#define SRCCTL_WR 0x00000200 +#define SRCCTL_PM 0x00000400 +#define SRCCTL_ROM 0x00001800 +#define SRCCTL_VO 0x00002000 +#define SRCCTL_ST 0x00004000 +#define SRCCTL_IE 0x00008000 +#define SRCCTL_ILSZ 0x000F0000 +#define SRCCTL_BP 0x00100000 + +#define SRCCCR_CISZ 0x000007FF +#define SRCCCR_CWA 0x001FF800 +#define SRCCCR_D 0x00200000 +#define SRCCCR_RS 0x01C00000 +#define SRCCCR_NAL 0x3E000000 +#define SRCCCR_RA 0xC0000000 + +#define SRCCA_CA 0x0FFFFFFF +#define SRCCA_RS 0xE0000000 + +#define SRCSA_SA 0x0FFFFFFF + +#define SRCLA_LA 0x0FFFFFFF + +/* Mixer Parameter Ring ram Low and Hight register. + * Fixed-point value in 8.24 format for parameter channel */ +#define MPRLH_PITCH 0xFFFFFFFF + +/* SRC resource register dirty flags */ +union src_dirty { + struct { + u16 ctl:1; + u16 ccr:1; + u16 sa:1; + u16 la:1; + u16 ca:1; + u16 mpr:1; + u16 czbfs:1; /* Clear Z-Buffers */ + u16 rsv:9; + } bf; + u16 data; +}; + +struct src_rsc_ctrl_blk { + unsigned int ctl; + unsigned int ccr; + unsigned int ca; + unsigned int sa; + unsigned int la; + unsigned int mpr; + union src_dirty dirty; +}; + +/* SRC manager control block */ +union src_mgr_dirty { + struct { + u16 enb0:1; + u16 enb1:1; + u16 enb2:1; + u16 enb3:1; + u16 enb4:1; + u16 enb5:1; + u16 enb6:1; + u16 enb7:1; + u16 enbsa:1; + u16 rsv:7; + } bf; + u16 data; +}; + +struct src_mgr_ctrl_blk { + unsigned int enbsa; + unsigned int enb[8]; + union src_mgr_dirty dirty; +}; + +/* SRCIMP manager control block */ +#define SRCAIM_ARC 0x00000FFF +#define SRCAIM_NXT 0x00FF0000 +#define SRCAIM_SRC 0xFF000000 + +struct srcimap { + unsigned int srcaim; + unsigned int idx; +}; + +/* SRCIMP manager register dirty flags */ +union srcimp_mgr_dirty { + struct { + u16 srcimap:1; + u16 rsv:15; + } bf; + u16 data; +}; + +struct srcimp_mgr_ctrl_blk { + struct srcimap srcimap; + union srcimp_mgr_dirty dirty; +}; + +/* + * Function implementation block. + */ + +static int src_get_rsc_ctrl_blk(void **rblk) +{ + struct src_rsc_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int src_put_rsc_ctrl_blk(void *blk) +{ + kfree((struct src_rsc_ctrl_blk *)blk); + + return 0; +} + +static int src_set_state(void *blk, unsigned int state) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_STATE, state); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_bm(void *blk, unsigned int bm) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_BM, bm); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_rsr(void *blk, unsigned int rsr) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_RSR, rsr); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_sf(void *blk, unsigned int sf) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_SF, sf); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_wr(void *blk, unsigned int wr) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_WR, wr); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_pm(void *blk, unsigned int pm) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_PM, pm); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_rom(void *blk, unsigned int rom) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_ROM, rom); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_vo(void *blk, unsigned int vo) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_VO, vo); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_st(void *blk, unsigned int st) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_ST, st); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_ie(void *blk, unsigned int ie) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_IE, ie); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_ilsz(void *blk, unsigned int ilsz) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_ILSZ, ilsz); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_bp(void *blk, unsigned int bp) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ctl, SRCCTL_BP, bp); + ctl->dirty.bf.ctl = 1; + return 0; +} + +static int src_set_cisz(void *blk, unsigned int cisz) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ccr, SRCCCR_CISZ, cisz); + ctl->dirty.bf.ccr = 1; + return 0; +} + +static int src_set_ca(void *blk, unsigned int ca) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->ca, SRCCA_CA, ca); + ctl->dirty.bf.ca = 1; + return 0; +} + +static int src_set_sa(void *blk, unsigned int sa) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->sa, SRCSA_SA, sa); + ctl->dirty.bf.sa = 1; + return 0; +} + +static int src_set_la(void *blk, unsigned int la) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->la, SRCLA_LA, la); + ctl->dirty.bf.la = 1; + return 0; +} + +static int src_set_pitch(void *blk, unsigned int pitch) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->mpr, MPRLH_PITCH, pitch); + ctl->dirty.bf.mpr = 1; + return 0; +} + +static int src_set_clear_zbufs(void *blk, unsigned int clear) +{ + ((struct src_rsc_ctrl_blk *)blk)->dirty.bf.czbfs = (clear ? 1 : 0); + return 0; +} + +static int src_set_dirty(void *blk, unsigned int flags) +{ + ((struct src_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff); + return 0; +} + +static int src_set_dirty_all(void *blk) +{ + ((struct src_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0); + return 0; +} + +#define AR_SLOT_SIZE 4096 +#define AR_SLOT_BLOCK_SIZE 16 +#define AR_PTS_PITCH 6 +#define AR_PARAM_SRC_OFFSET 0x60 + +static unsigned int src_param_pitch_mixer(unsigned int src_idx) +{ + return ((src_idx << 4) + AR_PTS_PITCH + AR_SLOT_SIZE + - AR_PARAM_SRC_OFFSET) % AR_SLOT_SIZE; + +} + +static int src_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct src_rsc_ctrl_blk *ctl = blk; + int i = 0; + + if (ctl->dirty.bf.czbfs) { + /* Clear Z-Buffer registers */ + for (i = 0; i < 8; i++) + hw_write_20kx(hw, SRC_UPZ+idx*0x100+i*0x4, 0); + + for (i = 0; i < 4; i++) + hw_write_20kx(hw, SRC_DN0Z+idx*0x100+i*0x4, 0); + + for (i = 0; i < 8; i++) + hw_write_20kx(hw, SRC_DN1Z+idx*0x100+i*0x4, 0); + + ctl->dirty.bf.czbfs = 0; + } + if (ctl->dirty.bf.mpr) { + /* Take the parameter mixer resource in the same group as that + * the idx src is in for simplicity. Unlike src, all conjugate + * parameter mixer resources must be programmed for + * corresponding conjugate src resources. */ + unsigned int pm_idx = src_param_pitch_mixer(idx); + hw_write_20kx(hw, MIXER_PRING_LO_HI+4*pm_idx, ctl->mpr); + hw_write_20kx(hw, MIXER_PMOPLO+8*pm_idx, 0x3); + hw_write_20kx(hw, MIXER_PMOPHI+8*pm_idx, 0x0); + ctl->dirty.bf.mpr = 0; + } + if (ctl->dirty.bf.sa) { + hw_write_20kx(hw, SRC_SA+idx*0x100, ctl->sa); + ctl->dirty.bf.sa = 0; + } + if (ctl->dirty.bf.la) { + hw_write_20kx(hw, SRC_LA+idx*0x100, ctl->la); + ctl->dirty.bf.la = 0; + } + if (ctl->dirty.bf.ca) { + hw_write_20kx(hw, SRC_CA+idx*0x100, ctl->ca); + ctl->dirty.bf.ca = 0; + } + + /* Write srccf register */ + hw_write_20kx(hw, SRC_CF+idx*0x100, 0x0); + + if (ctl->dirty.bf.ccr) { + hw_write_20kx(hw, SRC_CCR+idx*0x100, ctl->ccr); + ctl->dirty.bf.ccr = 0; + } + if (ctl->dirty.bf.ctl) { + hw_write_20kx(hw, SRC_CTL+idx*0x100, ctl->ctl); + ctl->dirty.bf.ctl = 0; + } + + return 0; +} + +static int src_get_ca(struct hw *hw, unsigned int idx, void *blk) +{ + struct src_rsc_ctrl_blk *ctl = blk; + + ctl->ca = hw_read_20kx(hw, SRC_CA+idx*0x100); + ctl->dirty.bf.ca = 0; + + return get_field(ctl->ca, SRCCA_CA); +} + +static unsigned int src_get_dirty(void *blk) +{ + return ((struct src_rsc_ctrl_blk *)blk)->dirty.data; +} + +static unsigned int src_dirty_conj_mask(void) +{ + return 0x20; +} + +static int src_mgr_enbs_src(void *blk, unsigned int idx) +{ + ((struct src_mgr_ctrl_blk *)blk)->enbsa |= (0x1 << ((idx%128)/4)); + ((struct src_mgr_ctrl_blk *)blk)->dirty.bf.enbsa = 1; + ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32)); + return 0; +} + +static int src_mgr_enb_src(void *blk, unsigned int idx) +{ + ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32)); + ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32)); + return 0; +} + +static int src_mgr_dsb_src(void *blk, unsigned int idx) +{ + ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] &= ~(0x1 << (idx%32)); + ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32)); + return 0; +} + +static int src_mgr_commit_write(struct hw *hw, void *blk) +{ + struct src_mgr_ctrl_blk *ctl = blk; + int i = 0; + unsigned int ret = 0; + + if (ctl->dirty.bf.enbsa) { + do { + ret = hw_read_20kx(hw, SRC_ENBSTAT); + } while (ret & 0x1); + hw_write_20kx(hw, SRC_ENBSA, ctl->enbsa); + ctl->dirty.bf.enbsa = 0; + } + for (i = 0; i < 8; i++) { + if ((ctl->dirty.data & (0x1 << i))) { + hw_write_20kx(hw, SRC_ENB+(i*0x100), ctl->enb[i]); + ctl->dirty.data &= ~(0x1 << i); + } + } + + return 0; +} + +static int src_mgr_get_ctrl_blk(void **rblk) +{ + struct src_mgr_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int src_mgr_put_ctrl_blk(void *blk) +{ + kfree((struct src_mgr_ctrl_blk *)blk); + + return 0; +} + +static int srcimp_mgr_get_ctrl_blk(void **rblk) +{ + struct srcimp_mgr_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int srcimp_mgr_put_ctrl_blk(void *blk) +{ + kfree((struct srcimp_mgr_ctrl_blk *)blk); + + return 0; +} + +static int srcimp_mgr_set_imaparc(void *blk, unsigned int slot) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->srcimap.srcaim, SRCAIM_ARC, slot); + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_set_imapuser(void *blk, unsigned int user) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->srcimap.srcaim, SRCAIM_SRC, user); + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_set_imapnxt(void *blk, unsigned int next) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->srcimap.srcaim, SRCAIM_NXT, next); + ctl->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_set_imapaddr(void *blk, unsigned int addr) +{ + ((struct srcimp_mgr_ctrl_blk *)blk)->srcimap.idx = addr; + ((struct srcimp_mgr_ctrl_blk *)blk)->dirty.bf.srcimap = 1; + return 0; +} + +static int srcimp_mgr_commit_write(struct hw *hw, void *blk) +{ + struct srcimp_mgr_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.srcimap) { + hw_write_20kx(hw, SRC_IMAP+ctl->srcimap.idx*0x100, + ctl->srcimap.srcaim); + ctl->dirty.bf.srcimap = 0; + } + + return 0; +} + +/* + * AMIXER control block definitions. + */ + +#define AMOPLO_M 0x00000003 +#define AMOPLO_IV 0x00000004 +#define AMOPLO_X 0x0003FFF0 +#define AMOPLO_Y 0xFFFC0000 + +#define AMOPHI_SADR 0x000000FF +#define AMOPHI_SE 0x80000000 + +/* AMIXER resource register dirty flags */ +union amixer_dirty { + struct { + u16 amoplo:1; + u16 amophi:1; + u16 rsv:14; + } bf; + u16 data; +}; + +/* AMIXER resource control block */ +struct amixer_rsc_ctrl_blk { + unsigned int amoplo; + unsigned int amophi; + union amixer_dirty dirty; +}; + +static int amixer_set_mode(void *blk, unsigned int mode) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_M, mode); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_iv(void *blk, unsigned int iv) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_IV, iv); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_x(void *blk, unsigned int x) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_X, x); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_y(void *blk, unsigned int y) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amoplo, AMOPLO_Y, y); + ctl->dirty.bf.amoplo = 1; + return 0; +} + +static int amixer_set_sadr(void *blk, unsigned int sadr) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amophi, AMOPHI_SADR, sadr); + ctl->dirty.bf.amophi = 1; + return 0; +} + +static int amixer_set_se(void *blk, unsigned int se) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + set_field(&ctl->amophi, AMOPHI_SE, se); + ctl->dirty.bf.amophi = 1; + return 0; +} + +static int amixer_set_dirty(void *blk, unsigned int flags) +{ + ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff); + return 0; +} + +static int amixer_set_dirty_all(void *blk) +{ + ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0); + return 0; +} + +static int amixer_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.amoplo || ctl->dirty.bf.amophi) { + hw_write_20kx(hw, MIXER_AMOPLO+idx*8, ctl->amoplo); + ctl->dirty.bf.amoplo = 0; + hw_write_20kx(hw, MIXER_AMOPHI+idx*8, ctl->amophi); + ctl->dirty.bf.amophi = 0; + } + + return 0; +} + +static int amixer_get_y(void *blk) +{ + struct amixer_rsc_ctrl_blk *ctl = blk; + + return get_field(ctl->amoplo, AMOPLO_Y); +} + +static unsigned int amixer_get_dirty(void *blk) +{ + return ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data; +} + +static int amixer_rsc_get_ctrl_blk(void **rblk) +{ + struct amixer_rsc_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int amixer_rsc_put_ctrl_blk(void *blk) +{ + kfree((struct amixer_rsc_ctrl_blk *)blk); + + return 0; +} + +static int amixer_mgr_get_ctrl_blk(void **rblk) +{ + *rblk = NULL; + + return 0; +} + +static int amixer_mgr_put_ctrl_blk(void *blk) +{ + return 0; +} + +/* + * DAIO control block definitions. + */ + +/* Receiver Sample Rate Tracker Control register */ +#define SRTCTL_SRCO 0x000000FF +#define SRTCTL_SRCM 0x0000FF00 +#define SRTCTL_RSR 0x00030000 +#define SRTCTL_DRAT 0x00300000 +#define SRTCTL_EC 0x01000000 +#define SRTCTL_ET 0x10000000 + +/* DAIO Receiver register dirty flags */ +union dai_dirty { + struct { + u16 srt:1; + u16 rsv:15; + } bf; + u16 data; +}; + +/* DAIO Receiver control block */ +struct dai_ctrl_blk { + unsigned int srt; + union dai_dirty dirty; +}; + +/* Audio Input Mapper RAM */ +#define AIM_ARC 0x00000FFF +#define AIM_NXT 0x007F0000 + +struct daoimap { + unsigned int aim; + unsigned int idx; +}; + +/* Audio Transmitter Control and Status register */ +#define ATXCTL_EN 0x00000001 +#define ATXCTL_MODE 0x00000010 +#define ATXCTL_CD 0x00000020 +#define ATXCTL_RAW 0x00000100 +#define ATXCTL_MT 0x00000200 +#define ATXCTL_NUC 0x00003000 +#define ATXCTL_BEN 0x00010000 +#define ATXCTL_BMUX 0x00700000 +#define ATXCTL_B24 0x01000000 +#define ATXCTL_CPF 0x02000000 +#define ATXCTL_RIV 0x10000000 +#define ATXCTL_LIV 0x20000000 +#define ATXCTL_RSAT 0x40000000 +#define ATXCTL_LSAT 0x80000000 + +/* XDIF Transmitter register dirty flags */ +union dao_dirty { + struct { + u16 atxcsl:1; + u16 rsv:15; + } bf; + u16 data; +}; + +/* XDIF Transmitter control block */ +struct dao_ctrl_blk { + /* XDIF Transmitter Channel Status Low Register */ + unsigned int atxcsl; + union dao_dirty dirty; +}; + +/* Audio Receiver Control register */ +#define ARXCTL_EN 0x00000001 + +/* DAIO manager register dirty flags */ +union daio_mgr_dirty { + struct { + u32 atxctl:8; + u32 arxctl:8; + u32 daoimap:1; + u32 rsv:15; + } bf; + u32 data; +}; + +/* DAIO manager control block */ +struct daio_mgr_ctrl_blk { + struct daoimap daoimap; + unsigned int txctl[8]; + unsigned int rxctl[8]; + union daio_mgr_dirty dirty; +}; + +static int dai_srt_set_srco(void *blk, unsigned int src) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srt, SRTCTL_SRCO, src); + ctl->dirty.bf.srt = 1; + return 0; +} + +static int dai_srt_set_srcm(void *blk, unsigned int src) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srt, SRTCTL_SRCM, src); + ctl->dirty.bf.srt = 1; + return 0; +} + +static int dai_srt_set_rsr(void *blk, unsigned int rsr) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srt, SRTCTL_RSR, rsr); + ctl->dirty.bf.srt = 1; + return 0; +} + +static int dai_srt_set_drat(void *blk, unsigned int drat) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srt, SRTCTL_DRAT, drat); + ctl->dirty.bf.srt = 1; + return 0; +} + +static int dai_srt_set_ec(void *blk, unsigned int ec) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srt, SRTCTL_EC, ec ? 1 : 0); + ctl->dirty.bf.srt = 1; + return 0; +} + +static int dai_srt_set_et(void *blk, unsigned int et) +{ + struct dai_ctrl_blk *ctl = blk; + + set_field(&ctl->srt, SRTCTL_ET, et ? 1 : 0); + ctl->dirty.bf.srt = 1; + return 0; +} + +static int dai_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct dai_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.srt) { + hw_write_20kx(hw, AUDIO_IO_RX_SRT_CTL+0x40*idx, ctl->srt); + ctl->dirty.bf.srt = 0; + } + + return 0; +} + +static int dai_get_ctrl_blk(void **rblk) +{ + struct dai_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int dai_put_ctrl_blk(void *blk) +{ + kfree((struct dai_ctrl_blk *)blk); + + return 0; +} + +static int dao_set_spos(void *blk, unsigned int spos) +{ + ((struct dao_ctrl_blk *)blk)->atxcsl = spos; + ((struct dao_ctrl_blk *)blk)->dirty.bf.atxcsl = 1; + return 0; +} + +static int dao_commit_write(struct hw *hw, unsigned int idx, void *blk) +{ + struct dao_ctrl_blk *ctl = blk; + + if (ctl->dirty.bf.atxcsl) { + if (idx < 4) { + /* S/PDIF SPOSx */ + hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_L+0x40*idx, + ctl->atxcsl); + } + ctl->dirty.bf.atxcsl = 0; + } + + return 0; +} + +static int dao_get_spos(void *blk, unsigned int *spos) +{ + *spos = ((struct dao_ctrl_blk *)blk)->atxcsl; + return 0; +} + +static int dao_get_ctrl_blk(void **rblk) +{ + struct dao_ctrl_blk *blk; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + *rblk = blk; + + return 0; +} + +static int dao_put_ctrl_blk(void *blk) +{ + kfree((struct dao_ctrl_blk *)blk); + + return 0; +} + +static int daio_mgr_enb_dai(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->rxctl[idx], ARXCTL_EN, 1); + ctl->dirty.bf.arxctl |= (0x1 << idx); + return 0; +} + +static int daio_mgr_dsb_dai(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->rxctl[idx], ARXCTL_EN, 0); + + ctl->dirty.bf.arxctl |= (0x1 << idx); + return 0; +} + +static int daio_mgr_enb_dao(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->txctl[idx], ATXCTL_EN, 1); + ctl->dirty.bf.atxctl |= (0x1 << idx); + return 0; +} + +static int daio_mgr_dsb_dao(void *blk, unsigned int idx) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->txctl[idx], ATXCTL_EN, 0); + ctl->dirty.bf.atxctl |= (0x1 << idx); + return 0; +} + +static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + if (idx < 4) { + /* S/PDIF output */ + switch ((conf & 0x7)) { + case 1: + set_field(&ctl->txctl[idx], ATXCTL_NUC, 0); + break; + case 2: + set_field(&ctl->txctl[idx], ATXCTL_NUC, 1); + break; + case 4: + set_field(&ctl->txctl[idx], ATXCTL_NUC, 2); + break; + case 8: + set_field(&ctl->txctl[idx], ATXCTL_NUC, 3); + break; + default: + break; + } + /* CDIF */ + set_field(&ctl->txctl[idx], ATXCTL_CD, (!(conf & 0x7))); + /* Non-audio */ + set_field(&ctl->txctl[idx], ATXCTL_LIV, (conf >> 4) & 0x1); + /* Non-audio */ + set_field(&ctl->txctl[idx], ATXCTL_RIV, (conf >> 4) & 0x1); + set_field(&ctl->txctl[idx], ATXCTL_RAW, + ((conf >> 3) & 0x1) ? 0 : 0); + ctl->dirty.bf.atxctl |= (0x1 << idx); + } else { + /* I2S output */ + /*idx %= 4; */ + } + return 0; +} + +static int daio_mgr_set_imaparc(void *blk, unsigned int slot) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->daoimap.aim, AIM_ARC, slot); + ctl->dirty.bf.daoimap = 1; + return 0; +} + +static int daio_mgr_set_imapnxt(void *blk, unsigned int next) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + + set_field(&ctl->daoimap.aim, AIM_NXT, next); + ctl->dirty.bf.daoimap = 1; + return 0; +} + +static int daio_mgr_set_imapaddr(void *blk, unsigned int addr) +{ + ((struct daio_mgr_ctrl_blk *)blk)->daoimap.idx = addr; + ((struct daio_mgr_ctrl_blk *)blk)->dirty.bf.daoimap = 1; + return 0; +} + +static int daio_mgr_commit_write(struct hw *hw, void *blk) +{ + struct daio_mgr_ctrl_blk *ctl = blk; + unsigned int data = 0; + int i = 0; + + for (i = 0; i < 8; i++) { + if ((ctl->dirty.bf.atxctl & (0x1 << i))) { + data = ctl->txctl[i]; + hw_write_20kx(hw, (AUDIO_IO_TX_CTL+(0x40*i)), data); + ctl->dirty.bf.atxctl &= ~(0x1 << i); + mdelay(1); + } + if ((ctl->dirty.bf.arxctl & (0x1 << i))) { + data = ctl->rxctl[i]; + hw_write_20kx(hw, (AUDIO_IO_RX_CTL+(0x40*i)), data); + ctl->dirty.bf.arxctl &= ~(0x1 << i); + mdelay(1); + } + } + if (ctl->dirty.bf.daoimap) { + hw_write_20kx(hw, AUDIO_IO_AIM+ctl->daoimap.idx*4, + ctl->daoimap.aim); + ctl->dirty.bf.daoimap = 0; + } + + return 0; +} + +static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) +{ + struct daio_mgr_ctrl_blk *blk; + int i = 0; + + *rblk = NULL; + blk = kzalloc(sizeof(*blk), GFP_KERNEL); + if (NULL == blk) + return -ENOMEM; + + for (i = 0; i < 8; i++) { + blk->txctl[i] = hw_read_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i)); + blk->rxctl[i] = hw_read_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i)); + } + + *rblk = blk; + + return 0; +} + +static int daio_mgr_put_ctrl_blk(void *blk) +{ + kfree((struct daio_mgr_ctrl_blk *)blk); + + return 0; +} + +/* Card hardware initialization block */ +struct dac_conf { + unsigned int msr; /* master sample rate in rsrs */ +}; + +struct adc_conf { + unsigned int msr; /* master sample rate in rsrs */ + unsigned char input; /* the input source of ADC */ + unsigned char mic20db; /* boost mic by 20db if input is microphone */ +}; + +struct daio_conf { + unsigned int msr; /* master sample rate in rsrs */ +}; + +struct trn_conf { + unsigned long vm_pgt_phys; +}; + +static int hw_daio_init(struct hw *hw, const struct daio_conf *info) +{ + u32 dwData = 0; + int i; + + /* Program I2S with proper sample rate and enable the correct I2S + * channel. ED(0/8/16/24): Enable all I2S/I2X master clock output */ + if (1 == info->msr) { + hw_write_20kx(hw, AUDIO_IO_MCLK, 0x01010101); + hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x01010101); + hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0); + } else if (2 == info->msr) { + hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111); + /* Specify all playing 96khz + * EA [0] - Enabled + * RTA [4:5] - 96kHz + * EB [8] - Enabled + * RTB [12:13] - 96kHz + * EC [16] - Enabled + * RTC [20:21] - 96kHz + * ED [24] - Enabled + * RTD [28:29] - 96kHz */ + hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x11111111); + hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0); + } else { + printk(KERN_ALERT "ERROR!!! Invalid sampling rate!!!\n"); + return -EINVAL; + } + + for (i = 0; i < 8; i++) { + if (i <= 3) { + /* 1st 3 channels are SPDIFs (SB0960) */ + if (i == 3) + dwData = 0x1001001; + else + dwData = 0x1000001; + + hw_write_20kx(hw, (AUDIO_IO_TX_CTL+(0x40*i)), dwData); + hw_write_20kx(hw, (AUDIO_IO_RX_CTL+(0x40*i)), dwData); + + /* Initialize the SPDIF Out Channel status registers. + * The value specified here is based on the typical + * values provided in the specification, namely: Clock + * Accuracy of 1000ppm, Sample Rate of 48KHz, + * unspecified source number, Generation status = 1, + * Category code = 0x12 (Digital Signal Mixer), + * Mode = 0, Emph = 0, Copy Permitted, AN = 0 + * (indicating that we're transmitting digital audio, + * and the Professional Use bit is 0. */ + + hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_L+(0x40*i), + 0x02109204); /* Default to 48kHz */ + + hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_H+(0x40*i), 0x0B); + } else { + /* Next 5 channels are I2S (SB0960) */ + dwData = 0x11; + hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), dwData); + if (2 == info->msr) { + /* Four channels per sample period */ + dwData |= 0x1000; + } + hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), dwData); + } + } + + return 0; +} + +/* TRANSPORT operations */ +static int hw_trn_init(struct hw *hw, const struct trn_conf *info) +{ + u32 vmctl = 0, data = 0; + unsigned long ptp_phys_low = 0, ptp_phys_high = 0; + int i = 0; + + /* Set up device page table */ + if ((~0UL) == info->vm_pgt_phys) { + printk(KERN_ALERT "Wrong device page table page address!!!\n"); + return -1; + } + + vmctl = 0x80000C0F; /* 32-bit, 4k-size page */ +#if BITS_PER_LONG == 64 + ptp_phys_low = info->vm_pgt_phys & ((1UL<<32)-1); + ptp_phys_high = (info->vm_pgt_phys>>32) & ((1UL<<32)-1); + vmctl |= (3<<8); +#elif BITS_PER_LONG == 32 + ptp_phys_low = info->vm_pgt_phys & (~0UL); + ptp_phys_high = 0; +#else +# error "Unknown BITS_PER_LONG!" +#endif +#if PAGE_SIZE == 8192 +# error "Don't support 8k-page!" +#endif + /* Write page table physical address to all PTPAL registers */ + for (i = 0; i < 64; i++) { + hw_write_20kx(hw, VMEM_PTPAL+(16*i), ptp_phys_low); + hw_write_20kx(hw, VMEM_PTPAH+(16*i), ptp_phys_high); + } + /* Enable virtual memory transfer */ + hw_write_20kx(hw, VMEM_CTL, vmctl); + /* Enable transport bus master and queueing of request */ + hw_write_20kx(hw, TRANSPORT_CTL, 0x03); + hw_write_20kx(hw, TRANSPORT_INT, 0x200c01); + /* Enable transport ring */ + data = hw_read_20kx(hw, TRANSPORT_ENB); + hw_write_20kx(hw, TRANSPORT_ENB, (data | 0x03)); + + return 0; +} + +/* Card initialization */ +#define GCTL_AIE 0x00000001 +#define GCTL_UAA 0x00000002 +#define GCTL_DPC 0x00000004 +#define GCTL_DBP 0x00000008 +#define GCTL_ABP 0x00000010 +#define GCTL_TBP 0x00000020 +#define GCTL_SBP 0x00000040 +#define GCTL_FBP 0x00000080 +#define GCTL_ME 0x00000100 +#define GCTL_AID 0x00001000 + +#define PLLCTL_SRC 0x00000007 +#define PLLCTL_SPE 0x00000008 +#define PLLCTL_RD 0x000000F0 +#define PLLCTL_FD 0x0001FF00 +#define PLLCTL_OD 0x00060000 +#define PLLCTL_B 0x00080000 +#define PLLCTL_AS 0x00100000 +#define PLLCTL_LF 0x03E00000 +#define PLLCTL_SPS 0x1C000000 +#define PLLCTL_AD 0x60000000 + +#define PLLSTAT_CCS 0x00000007 +#define PLLSTAT_SPL 0x00000008 +#define PLLSTAT_CRD 0x000000F0 +#define PLLSTAT_CFD 0x0001FF00 +#define PLLSTAT_SL 0x00020000 +#define PLLSTAT_FAS 0x00040000 +#define PLLSTAT_B 0x00080000 +#define PLLSTAT_PD 0x00100000 +#define PLLSTAT_OCA 0x00200000 +#define PLLSTAT_NCA 0x00400000 + +static int hw_pll_init(struct hw *hw, unsigned int rsr) +{ + unsigned int pllenb; + unsigned int pllctl; + unsigned int pllstat; + int i; + + pllenb = 0xB; + hw_write_20kx(hw, PLL_ENB, pllenb); + pllctl = 0x20D00000; + set_field(&pllctl, PLLCTL_FD, 16 - 4); + hw_write_20kx(hw, PLL_CTL, pllctl); + mdelay(40); + pllctl = hw_read_20kx(hw, PLL_CTL); + set_field(&pllctl, PLLCTL_B, 0); + if (48000 == rsr) { + set_field(&pllctl, PLLCTL_FD, 16 - 2); + set_field(&pllctl, PLLCTL_RD, 1 - 1); + } else { /* 44100 */ + set_field(&pllctl, PLLCTL_FD, 147 - 2); + set_field(&pllctl, PLLCTL_RD, 10 - 1); + } + hw_write_20kx(hw, PLL_CTL, pllctl); + mdelay(40); + for (i = 0; i < 1000; i++) { + pllstat = hw_read_20kx(hw, PLL_STAT); + if (get_field(pllstat, PLLSTAT_PD)) + continue; + + if (get_field(pllstat, PLLSTAT_B) != + get_field(pllctl, PLLCTL_B)) + continue; + + if (get_field(pllstat, PLLSTAT_CCS) != + get_field(pllctl, PLLCTL_SRC)) + continue; + + if (get_field(pllstat, PLLSTAT_CRD) != + get_field(pllctl, PLLCTL_RD)) + continue; + + if (get_field(pllstat, PLLSTAT_CFD) != + get_field(pllctl, PLLCTL_FD)) + continue; + + break; + } + if (i >= 1000) { + printk(KERN_ALERT "PLL initialization failed!!!\n"); + return -EBUSY; + } + + return 0; +} + +static int hw_auto_init(struct hw *hw) +{ + unsigned int gctl; + int i; + + gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); + set_field(&gctl, GCTL_AIE, 0); + hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); + set_field(&gctl, GCTL_AIE, 1); + hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); + mdelay(10); + for (i = 0; i < 400000; i++) { + gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); + if (get_field(gctl, GCTL_AID)) + break; + } + if (!get_field(gctl, GCTL_AID)) { + printk(KERN_ALERT "Card Auto-init failed!!!\n"); + return -EBUSY; + } + + return 0; +} + +/* DAC operations */ + +#define CS4382_MC1 0x1 +#define CS4382_MC2 0x2 +#define CS4382_MC3 0x3 +#define CS4382_FC 0x4 +#define CS4382_IC 0x5 +#define CS4382_XC1 0x6 +#define CS4382_VCA1 0x7 +#define CS4382_VCB1 0x8 +#define CS4382_XC2 0x9 +#define CS4382_VCA2 0xA +#define CS4382_VCB2 0xB +#define CS4382_XC3 0xC +#define CS4382_VCA3 0xD +#define CS4382_VCB3 0xE +#define CS4382_XC4 0xF +#define CS4382_VCA4 0x10 +#define CS4382_VCB4 0x11 +#define CS4382_CREV 0x12 + +/* I2C status */ +#define STATE_LOCKED 0x00 +#define STATE_UNLOCKED 0xAA +#define DATA_READY 0x800000 /* Used with I2C_IF_STATUS */ +#define DATA_ABORT 0x10000 /* Used with I2C_IF_STATUS */ + +#define I2C_STATUS_DCM 0x00000001 +#define I2C_STATUS_BC 0x00000006 +#define I2C_STATUS_APD 0x00000008 +#define I2C_STATUS_AB 0x00010000 +#define I2C_STATUS_DR 0x00800000 + +#define I2C_ADDRESS_PTAD 0x0000FFFF +#define I2C_ADDRESS_SLAD 0x007F0000 + +struct REGS_CS4382 { + u32 dwModeControl_1; + u32 dwModeControl_2; + u32 dwModeControl_3; + + u32 dwFilterControl; + u32 dwInvertControl; + + u32 dwMixControl_P1; + u32 dwVolControl_A1; + u32 dwVolControl_B1; + + u32 dwMixControl_P2; + u32 dwVolControl_A2; + u32 dwVolControl_B2; + + u32 dwMixControl_P3; + u32 dwVolControl_A3; + u32 dwVolControl_B3; + + u32 dwMixControl_P4; + u32 dwVolControl_A4; + u32 dwVolControl_B4; +}; + +static u8 m_bAddressSize, m_bDataSize, m_bDeviceID; + +static int I2CUnlockFullAccess(struct hw *hw) +{ + u8 UnlockKeySequence_FLASH_FULLACCESS_MODE[2] = {0xB3, 0xD4}; + + /* Send keys for forced BIOS mode */ + hw_write_20kx(hw, I2C_IF_WLOCK, + UnlockKeySequence_FLASH_FULLACCESS_MODE[0]); + hw_write_20kx(hw, I2C_IF_WLOCK, + UnlockKeySequence_FLASH_FULLACCESS_MODE[1]); + /* Check whether the chip is unlocked */ + if (hw_read_20kx(hw, I2C_IF_WLOCK) == STATE_UNLOCKED) + return 0; + + return -1; +} + +static int I2CLockChip(struct hw *hw) +{ + /* Write twice */ + hw_write_20kx(hw, I2C_IF_WLOCK, STATE_LOCKED); + hw_write_20kx(hw, I2C_IF_WLOCK, STATE_LOCKED); + if (hw_read_20kx(hw, I2C_IF_WLOCK) == STATE_LOCKED) + return 0; + + return -1; +} + +static int I2CInit(struct hw *hw, u8 bDeviceID, u8 bAddressSize, u8 bDataSize) +{ + int err = 0; + unsigned int RegI2CStatus; + unsigned int RegI2CAddress; + + err = I2CUnlockFullAccess(hw); + if (err < 0) + return err; + + m_bAddressSize = bAddressSize; + m_bDataSize = bDataSize; + m_bDeviceID = bDeviceID; + + RegI2CAddress = 0; + set_field(&RegI2CAddress, I2C_ADDRESS_SLAD, bDeviceID); + + hw_write_20kx(hw, I2C_IF_ADDRESS, RegI2CAddress); + + RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + + set_field(&RegI2CStatus, I2C_STATUS_DCM, 1); /* Direct control mode */ + + hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); + + return 0; +} + +static int I2CUninit(struct hw *hw) +{ + unsigned int RegI2CStatus; + unsigned int RegI2CAddress; + + RegI2CAddress = 0; + set_field(&RegI2CAddress, I2C_ADDRESS_SLAD, 0x57); /* I2C id */ + + hw_write_20kx(hw, I2C_IF_ADDRESS, RegI2CAddress); + + RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + + set_field(&RegI2CStatus, I2C_STATUS_DCM, 0); /* I2C mode */ + + hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); + + return I2CLockChip(hw); +} + +static int I2CWaitDataReady(struct hw *hw) +{ + int i = 0x400000; + unsigned int ret = 0; + + do { + ret = hw_read_20kx(hw, I2C_IF_STATUS); + } while ((!(ret & DATA_READY)) && --i); + + return i; +} + +static int I2CRead(struct hw *hw, u16 wAddress, u32 *pdwData) +{ + unsigned int RegI2CStatus; + + RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + set_field(&RegI2CStatus, I2C_STATUS_BC, + (4 == m_bAddressSize) ? 0 : m_bAddressSize); + hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); + if (!I2CWaitDataReady(hw)) + return -1; + + hw_write_20kx(hw, I2C_IF_WDATA, (u32)wAddress); + if (!I2CWaitDataReady(hw)) + return -1; + + /* Force a read operation */ + hw_write_20kx(hw, I2C_IF_RDATA, 0); + if (!I2CWaitDataReady(hw)) + return -1; + + *pdwData = hw_read_20kx(hw, I2C_IF_RDATA); + + return 0; +} + +static int I2CWrite(struct hw *hw, u16 wAddress, u32 dwData) +{ + unsigned int dwI2CData = (dwData << (m_bAddressSize * 8)) | wAddress; + unsigned int RegI2CStatus; + + RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + + set_field(&RegI2CStatus, I2C_STATUS_BC, + (4 == (m_bAddressSize + m_bDataSize)) ? + 0 : (m_bAddressSize + m_bDataSize)); + + hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); + I2CWaitDataReady(hw); + /* Dummy write to trigger the write oprtation */ + hw_write_20kx(hw, I2C_IF_WDATA, 0); + I2CWaitDataReady(hw); + + /* This is the real data */ + hw_write_20kx(hw, I2C_IF_WDATA, dwI2CData); + I2CWaitDataReady(hw); + + return 0; +} + +static int hw_dac_init(struct hw *hw, const struct dac_conf *info) +{ + int err = 0; + u32 dwData = 0; + int i = 0; + struct REGS_CS4382 cs4382_Read = {0}; + struct REGS_CS4382 cs4382_Def = { + 0x00000001, /* Mode Control 1 */ + 0x00000000, /* Mode Control 2 */ + 0x00000084, /* Mode Control 3 */ + 0x00000000, /* Filter Control */ + 0x00000000, /* Invert Control */ + 0x00000024, /* Mixing Control Pair 1 */ + 0x00000000, /* Vol Control A1 */ + 0x00000000, /* Vol Control B1 */ + 0x00000024, /* Mixing Control Pair 2 */ + 0x00000000, /* Vol Control A2 */ + 0x00000000, /* Vol Control B2 */ + 0x00000024, /* Mixing Control Pair 3 */ + 0x00000000, /* Vol Control A3 */ + 0x00000000, /* Vol Control B3 */ + 0x00000024, /* Mixing Control Pair 4 */ + 0x00000000, /* Vol Control A4 */ + 0x00000000 /* Vol Control B4 */ + }; + + /* Set DAC reset bit as output */ + dwData = hw_read_20kx(hw, GPIO_CTRL); + dwData |= 0x02; + hw_write_20kx(hw, GPIO_CTRL, dwData); + + err = I2CInit(hw, 0x18, 1, 1); + if (err < 0) + goto End; + + for (i = 0; i < 2; i++) { + /* Reset DAC twice just in-case the chip + * didn't initialized properly */ + dwData = hw_read_20kx(hw, GPIO_DATA); + /* GPIO data bit 1 */ + dwData &= 0xFFFFFFFD; + hw_write_20kx(hw, GPIO_DATA, dwData); + mdelay(10); + dwData |= 0x2; + hw_write_20kx(hw, GPIO_DATA, dwData); + mdelay(50); + + /* Reset the 2nd time */ + dwData &= 0xFFFFFFFD; + hw_write_20kx(hw, GPIO_DATA, dwData); + mdelay(10); + dwData |= 0x2; + hw_write_20kx(hw, GPIO_DATA, dwData); + mdelay(50); + + if (I2CRead(hw, CS4382_MC1, &cs4382_Read.dwModeControl_1)) + continue; + + if (I2CRead(hw, CS4382_MC2, &cs4382_Read.dwModeControl_2)) + continue; + + if (I2CRead(hw, CS4382_MC3, &cs4382_Read.dwModeControl_3)) + continue; + + if (I2CRead(hw, CS4382_FC, &cs4382_Read.dwFilterControl)) + continue; + + if (I2CRead(hw, CS4382_IC, &cs4382_Read.dwInvertControl)) + continue; + + if (I2CRead(hw, CS4382_XC1, &cs4382_Read.dwMixControl_P1)) + continue; + + if (I2CRead(hw, CS4382_VCA1, &cs4382_Read.dwVolControl_A1)) + continue; + + if (I2CRead(hw, CS4382_VCB1, &cs4382_Read.dwVolControl_B1)) + continue; + + if (I2CRead(hw, CS4382_XC2, &cs4382_Read.dwMixControl_P2)) + continue; + + if (I2CRead(hw, CS4382_VCA2, &cs4382_Read.dwVolControl_A2)) + continue; + + if (I2CRead(hw, CS4382_VCB2, &cs4382_Read.dwVolControl_B2)) + continue; + + if (I2CRead(hw, CS4382_XC3, &cs4382_Read.dwMixControl_P3)) + continue; + + if (I2CRead(hw, CS4382_VCA3, &cs4382_Read.dwVolControl_A3)) + continue; + + if (I2CRead(hw, CS4382_VCB3, &cs4382_Read.dwVolControl_B3)) + continue; + + if (I2CRead(hw, CS4382_XC4, &cs4382_Read.dwMixControl_P4)) + continue; + + if (I2CRead(hw, CS4382_VCA4, &cs4382_Read.dwVolControl_A4)) + continue; + + if (I2CRead(hw, CS4382_VCB4, &cs4382_Read.dwVolControl_B4)) + continue; + + if (memcmp(&cs4382_Read, &cs4382_Def, + sizeof(struct REGS_CS4382))) + continue; + else + break; + } + + if (i >= 2) + goto End; + + /* Note: Every I2C write must have some delay. + * This is not a requirement but the delay works here... */ + I2CWrite(hw, CS4382_MC1, 0x80); + I2CWrite(hw, CS4382_MC2, 0x10); + if (1 == info->msr) { + I2CWrite(hw, CS4382_XC1, 0x24); + I2CWrite(hw, CS4382_XC2, 0x24); + I2CWrite(hw, CS4382_XC3, 0x24); + I2CWrite(hw, CS4382_XC4, 0x24); + } else if (2 == info->msr) { + I2CWrite(hw, CS4382_XC1, 0x25); + I2CWrite(hw, CS4382_XC2, 0x25); + I2CWrite(hw, CS4382_XC3, 0x25); + I2CWrite(hw, CS4382_XC4, 0x25); + } else { + I2CWrite(hw, CS4382_XC1, 0x26); + I2CWrite(hw, CS4382_XC2, 0x26); + I2CWrite(hw, CS4382_XC3, 0x26); + I2CWrite(hw, CS4382_XC4, 0x26); + } + + return 0; +End: + + I2CUninit(hw); + return -1; +} + +/* ADC operations */ +#define MAKE_WM8775_ADDR(addr, data) (u32)(((addr<<1)&0xFE)|((data>>8)&0x1)) +#define MAKE_WM8775_DATA(data) (u32)(data&0xFF) + +#define WM8775_IC 0x0B +#define WM8775_MMC 0x0C +#define WM8775_AADCL 0x0E +#define WM8775_AADCR 0x0F +#define WM8775_ADCMC 0x15 +#define WM8775_RESET 0x17 + +static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) +{ + u32 data = 0; + + data = hw_read_20kx(hw, GPIO_DATA); + switch (type) { + case ADC_MICIN: + data = (data & (0x1 << 14)) ? 1 : 0; + break; + case ADC_LINEIN: + data = (data & (0x1 << 14)) ? 0 : 1; + break; + default: + data = 0; + } + return data; +} + +static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) +{ + u32 data = 0; + + data = hw_read_20kx(hw, GPIO_DATA); + switch (type) { + case ADC_MICIN: + data |= (0x1 << 14); + hw_write_20kx(hw, GPIO_DATA, data); + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), + MAKE_WM8775_DATA(0x101)); /* Mic-in */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), + MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), + MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ + break; + case ADC_LINEIN: + data &= ~(0x1 << 14); + hw_write_20kx(hw, GPIO_DATA, data); + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102), + MAKE_WM8775_DATA(0x102)); /* Line-in */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF), + MAKE_WM8775_DATA(0xCF)); /* No boost */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), + MAKE_WM8775_DATA(0xCF)); /* No boost */ + break; + default: + break; + } + + return 0; +} + +static int hw_adc_init(struct hw *hw, const struct adc_conf *info) +{ + int err = 0; + u32 dwMux = 2, dwData = 0, dwCtl = 0; + + /* Set ADC reset bit as output */ + dwData = hw_read_20kx(hw, GPIO_CTRL); + dwData |= (0x1 << 15); + hw_write_20kx(hw, GPIO_CTRL, dwData); + + /* Initialize I2C */ + err = I2CInit(hw, 0x1A, 1, 1); + if (err < 0) { + printk(KERN_ALERT "Failure to acquire I2C!!!\n"); + goto error; + } + + /* Make ADC in normal operation */ + dwData = hw_read_20kx(hw, GPIO_DATA); + dwData &= ~(0x1 << 15); + mdelay(10); + dwData |= (0x1 << 15); + hw_write_20kx(hw, GPIO_DATA, dwData); + mdelay(50); + + /* Set the master mode (256fs) */ + if (1 == info->msr) { + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02), + MAKE_WM8775_DATA(0x02)); + } else if (2 == info->msr) { + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A), + MAKE_WM8775_DATA(0x0A)); + } else { + printk(KERN_ALERT "Invalid master sampling " + "rate (msr %d)!!!\n", info->msr); + err = -EINVAL; + goto error; + } + + /* Configure GPIO bit 14 change to line-in/mic-in */ + dwCtl = hw_read_20kx(hw, GPIO_CTRL); + dwCtl |= 0x1<<14; + hw_write_20kx(hw, GPIO_CTRL, dwCtl); + + /* Check using Mic-in or Line-in */ + dwData = hw_read_20kx(hw, GPIO_DATA); + + if (dwMux == 1) { + /* Configures GPIO data to select Mic-in */ + dwData |= 0x1<<14; + hw_write_20kx(hw, GPIO_DATA, dwData); + + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), + MAKE_WM8775_DATA(0x101)); /* Mic-in */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), + MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), + MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ + } else if (dwMux == 2) { + /* Configures GPIO data to select Line-in */ + dwData &= ~(0x1<<14); + hw_write_20kx(hw, GPIO_DATA, dwData); + + /* Setup ADC */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102), + MAKE_WM8775_DATA(0x102)); /* Line-in */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF), + MAKE_WM8775_DATA(0xCF)); /* No boost */ + I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), + MAKE_WM8775_DATA(0xCF)); /* No boost */ + } else { + printk(KERN_ALERT "ERROR!!! Invalid input mux!!!\n"); + err = -EINVAL; + goto error; + } + + return 0; + +error: + I2CUninit(hw); + return err; +} + +static int hw_have_digit_io_switch(struct hw *hw) +{ + return 0; +} + +static int hw_card_start(struct hw *hw) +{ + int err = 0; + struct pci_dev *pci = hw->pci; + unsigned int gctl; + unsigned int dma_mask = 0; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + /* Set DMA transfer mask */ + dma_mask = CT_XFI_DMA_MASK; + if (pci_set_dma_mask(pci, dma_mask) < 0 || + pci_set_consistent_dma_mask(pci, dma_mask) < 0) { + printk(KERN_ERR "architecture does not support PCI " + "busmaster DMA with mask 0x%x\n", dma_mask); + err = -ENXIO; + goto error1; + } + + err = pci_request_regions(pci, "XFi"); + if (err < 0) + goto error1; + + hw->io_base = pci_resource_start(hw->pci, 2); + hw->mem_base = (unsigned long)ioremap(hw->io_base, + pci_resource_len(hw->pci, 2)); + if (NULL == (void *)hw->mem_base) { + err = -ENOENT; + goto error2; + } + + /* Switch to 20k2 mode from UAA mode. */ + gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); + set_field(&gctl, GCTL_UAA, 0); + hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); + + /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED, + atc->chip_details->nm_card, hw))) { + goto error3; + } + hw->irq = pci->irq; + */ + + pci_set_master(pci); + + return 0; + +/*error3: + iounmap((void *)hw->mem_base); + hw->mem_base = (unsigned long)NULL;*/ +error2: + pci_release_regions(pci); + hw->io_base = 0; +error1: + pci_disable_device(pci); + return err; +} + +static int hw_card_stop(struct hw *hw) +{ + /* TODO: Disable interrupt and so on... */ + return 0; +} + +static int hw_card_shutdown(struct hw *hw) +{ + if (hw->irq >= 0) + free_irq(hw->irq, hw); + + hw->irq = -1; + + if (NULL != ((void *)hw->mem_base)) + iounmap((void *)hw->mem_base); + + hw->mem_base = (unsigned long)NULL; + + if (hw->io_base) + pci_release_regions(hw->pci); + + hw->io_base = 0; + + pci_disable_device(hw->pci); + + return 0; +} + +static int hw_card_init(struct hw *hw, struct card_conf *info) +{ + int err; + unsigned int gctl; + u32 data = 0; + struct dac_conf dac_info = {0}; + struct adc_conf adc_info = {0}; + struct daio_conf daio_info = {0}; + struct trn_conf trn_info = {0}; + + /* Get PCI io port/memory base address and + * do 20kx core switch if needed. */ + if (!hw->io_base) { + err = hw_card_start(hw); + if (err) + return err; + } + + /* PLL init */ + err = hw_pll_init(hw, info->rsr); + if (err < 0) + return err; + + /* kick off auto-init */ + err = hw_auto_init(hw); + if (err < 0) + return err; + + gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); + set_field(&gctl, GCTL_DBP, 1); + set_field(&gctl, GCTL_TBP, 1); + set_field(&gctl, GCTL_FBP, 1); + set_field(&gctl, GCTL_DPC, 0); + hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); + + /* Reset all global pending interrupts */ + hw_write_20kx(hw, INTERRUPT_GIE, 0); + /* Reset all SRC pending interrupts */ + hw_write_20kx(hw, SRC_IP, 0); + + /* TODO: detect the card ID and configure GPIO accordingly. */ + /* Configures GPIO (0xD802 0x98028) */ + /*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/ + /* Configures GPIO (SB0880) */ + /*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/ + hw_write_20kx(hw, GPIO_CTRL, 0xD802); + + /* Enable audio ring */ + hw_write_20kx(hw, MIXER_AR_ENABLE, 0x01); + + trn_info.vm_pgt_phys = info->vm_pgt_phys; + err = hw_trn_init(hw, &trn_info); + if (err < 0) + return err; + + daio_info.msr = info->msr; + err = hw_daio_init(hw, &daio_info); + if (err < 0) + return err; + + dac_info.msr = info->msr; + err = hw_dac_init(hw, &dac_info); + if (err < 0) + return err; + + adc_info.msr = info->msr; + adc_info.input = ADC_LINEIN; + adc_info.mic20db = 0; + err = hw_adc_init(hw, &adc_info); + if (err < 0) + return err; + + data = hw_read_20kx(hw, SRC_MCTL); + data |= 0x1; /* Enables input from the audio ring */ + hw_write_20kx(hw, SRC_MCTL, data); + + return 0; +} + +static u32 hw_read_20kx(struct hw *hw, u32 reg) +{ + return readl((void *)(hw->mem_base + reg)); +} + +static void hw_write_20kx(struct hw *hw, u32 reg, u32 data) +{ + writel(data, (void *)(hw->mem_base + reg)); +} + +int create_20k2_hw_obj(struct hw **rhw) +{ + struct hw *hw; + + *rhw = NULL; + hw = kzalloc(sizeof(*hw), GFP_KERNEL); + if (NULL == hw) + return -ENOMEM; + + hw->io_base = 0; + hw->mem_base = (unsigned long)NULL; + hw->irq = -1; + + hw->card_init = hw_card_init; + hw->card_stop = hw_card_stop; + hw->pll_init = hw_pll_init; + hw->is_adc_source_selected = hw_is_adc_input_selected; + hw->select_adc_source = hw_adc_input_select; + hw->have_digit_io_switch = hw_have_digit_io_switch; + + hw->src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk; + hw->src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk; + hw->src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk; + hw->src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk; + hw->src_set_state = src_set_state; + hw->src_set_bm = src_set_bm; + hw->src_set_rsr = src_set_rsr; + hw->src_set_sf = src_set_sf; + hw->src_set_wr = src_set_wr; + hw->src_set_pm = src_set_pm; + hw->src_set_rom = src_set_rom; + hw->src_set_vo = src_set_vo; + hw->src_set_st = src_set_st; + hw->src_set_ie = src_set_ie; + hw->src_set_ilsz = src_set_ilsz; + hw->src_set_bp = src_set_bp; + hw->src_set_cisz = src_set_cisz; + hw->src_set_ca = src_set_ca; + hw->src_set_sa = src_set_sa; + hw->src_set_la = src_set_la; + hw->src_set_pitch = src_set_pitch; + hw->src_set_dirty = src_set_dirty; + hw->src_set_clear_zbufs = src_set_clear_zbufs; + hw->src_set_dirty_all = src_set_dirty_all; + hw->src_commit_write = src_commit_write; + hw->src_get_ca = src_get_ca; + hw->src_get_dirty = src_get_dirty; + hw->src_dirty_conj_mask = src_dirty_conj_mask; + hw->src_mgr_enbs_src = src_mgr_enbs_src; + hw->src_mgr_enb_src = src_mgr_enb_src; + hw->src_mgr_dsb_src = src_mgr_dsb_src; + hw->src_mgr_commit_write = src_mgr_commit_write; + + hw->srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk; + hw->srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk; + hw->srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc; + hw->srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser; + hw->srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt; + hw->srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr; + hw->srcimp_mgr_commit_write = srcimp_mgr_commit_write; + + hw->amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk; + hw->amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk; + hw->amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk; + hw->amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk; + hw->amixer_set_mode = amixer_set_mode; + hw->amixer_set_iv = amixer_set_iv; + hw->amixer_set_x = amixer_set_x; + hw->amixer_set_y = amixer_set_y; + hw->amixer_set_sadr = amixer_set_sadr; + hw->amixer_set_se = amixer_set_se; + hw->amixer_set_dirty = amixer_set_dirty; + hw->amixer_set_dirty_all = amixer_set_dirty_all; + hw->amixer_commit_write = amixer_commit_write; + hw->amixer_get_y = amixer_get_y; + hw->amixer_get_dirty = amixer_get_dirty; + + hw->dai_get_ctrl_blk = dai_get_ctrl_blk; + hw->dai_put_ctrl_blk = dai_put_ctrl_blk; + hw->dai_srt_set_srco = dai_srt_set_srco; + hw->dai_srt_set_srcm = dai_srt_set_srcm; + hw->dai_srt_set_rsr = dai_srt_set_rsr; + hw->dai_srt_set_drat = dai_srt_set_drat; + hw->dai_srt_set_ec = dai_srt_set_ec; + hw->dai_srt_set_et = dai_srt_set_et; + hw->dai_commit_write = dai_commit_write; + + hw->dao_get_ctrl_blk = dao_get_ctrl_blk; + hw->dao_put_ctrl_blk = dao_put_ctrl_blk; + hw->dao_set_spos = dao_set_spos; + hw->dao_commit_write = dao_commit_write; + hw->dao_get_spos = dao_get_spos; + + hw->daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk; + hw->daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk; + hw->daio_mgr_enb_dai = daio_mgr_enb_dai; + hw->daio_mgr_dsb_dai = daio_mgr_dsb_dai; + hw->daio_mgr_enb_dao = daio_mgr_enb_dao; + hw->daio_mgr_dsb_dao = daio_mgr_dsb_dao; + hw->daio_mgr_dao_init = daio_mgr_dao_init; + hw->daio_mgr_set_imaparc = daio_mgr_set_imaparc; + hw->daio_mgr_set_imapnxt = daio_mgr_set_imapnxt; + hw->daio_mgr_set_imapaddr = daio_mgr_set_imapaddr; + hw->daio_mgr_commit_write = daio_mgr_commit_write; + + *rhw = hw; + + return 0; +} + +int destroy_20k2_hw_obj(struct hw *hw) +{ + if (hw->io_base) + hw_card_shutdown(hw); + + kfree(hw); + return 0; +} diff --git a/sound/pci/ctxfi/cthw20k2.h b/sound/pci/ctxfi/cthw20k2.h new file mode 100644 index 0000000..d2b7daa --- /dev/null +++ b/sound/pci/ctxfi/cthw20k2.h @@ -0,0 +1,26 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File cthw20k2.h + * + * @Brief + * This file contains the definition of hardware access methord. + * + * @Author Liu Chun + * @Date May 13 2008 + * + */ + +#ifndef CTHW20K2_H +#define CTHW20K2_H + +#include "cthardware.h" + +int create_20k2_hw_obj(struct hw **rhw); +int destroy_20k2_hw_obj(struct hw *hw); + +#endif /* CTHW20K2_H */ diff --git a/sound/pci/ctxfi/ctimap.c b/sound/pci/ctxfi/ctimap.c new file mode 100644 index 0000000..d34eacd --- /dev/null +++ b/sound/pci/ctxfi/ctimap.c @@ -0,0 +1,112 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctimap.c + * + * @Brief + * This file contains the implementation of generic input mapper operations + * for input mapper management. + * + * @Author Liu Chun + * @Date May 23 2008 + * + */ + +#include "ctimap.h" +#include + +int input_mapper_add(struct list_head *mappers, struct imapper *entry, + int (*map_op)(void *, struct imapper *), void *data) +{ + struct list_head *pos, *pre, *head; + struct imapper *pre_ent, *pos_ent; + + head = mappers; + + if (list_empty(head)) { + entry->next = entry->addr; + map_op(data, entry); + list_add(&entry->list, head); + return 0; + } + + list_for_each(pos, head) { + pos_ent = list_entry(pos, struct imapper, list); + if (pos_ent->slot > entry->slot) { + /* found a position in list */ + break; + } + } + + if (pos != head) { + pre = pos->prev; + if (pre == head) + pre = head->prev; + + __list_add(&entry->list, pos->prev, pos); + } else { + pre = head->prev; + pos = head->next; + list_add_tail(&entry->list, head); + } + + pre_ent = list_entry(pre, struct imapper, list); + pos_ent = list_entry(pos, struct imapper, list); + + entry->next = pos_ent->addr; + map_op(data, entry); + pre_ent->next = entry->addr; + map_op(data, pre_ent); + + return 0; +} + +int input_mapper_delete(struct list_head *mappers, struct imapper *entry, + int (*map_op)(void *, struct imapper *), void *data) +{ + struct list_head *next, *pre, *head; + struct imapper *pre_ent, *next_ent; + + head = mappers; + + if (list_empty(head)) + return 0; + + pre = (entry->list.prev == head) ? head->prev : entry->list.prev; + next = (entry->list.next == head) ? head->next : entry->list.next; + + if (pre == &entry->list) { + /* entry is the only one node in mappers list */ + entry->next = entry->addr = entry->user = entry->slot = 0; + map_op(data, entry); + list_del(&entry->list); + return 0; + } + + pre_ent = list_entry(pre, struct imapper, list); + next_ent = list_entry(next, struct imapper, list); + + pre_ent->next = next_ent->addr; + map_op(data, pre_ent); + list_del(&entry->list); + + return 0; +} + +void free_input_mapper_list(struct list_head *head) +{ + struct imapper *entry = NULL; + struct list_head *pos = NULL; + + while (!list_empty(head)) { + pos = head->next; + list_del(pos); + entry = list_entry(pos, struct imapper, list); + kfree(entry); + } +} + diff --git a/sound/pci/ctxfi/ctimap.h b/sound/pci/ctxfi/ctimap.h new file mode 100644 index 0000000..53ccf9b --- /dev/null +++ b/sound/pci/ctxfi/ctimap.h @@ -0,0 +1,40 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctimap.h + * + * @Brief + * This file contains the definition of generic input mapper operations + * for input mapper management. + * + * @Author Liu Chun + * @Date May 23 2008 + * + */ + +#ifndef CTIMAP_H +#define CTIMAP_H + +#include + +struct imapper { + unsigned short slot; /* the id of the slot containing input data */ + unsigned short user; /* the id of the user resource consuming data */ + unsigned short addr; /* the input mapper ram id */ + unsigned short next; /* the next input mapper ram id */ + struct list_head list; +}; + +int input_mapper_add(struct list_head *mappers, struct imapper *entry, + int (*map_op)(void *, struct imapper *), void *data); + +int input_mapper_delete(struct list_head *mappers, struct imapper *entry, + int (*map_op)(void *, struct imapper *), void *data); + +void free_input_mapper_list(struct list_head *mappers); + +#endif /* CTIMAP_H */ diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c new file mode 100644 index 0000000..c80d692 --- /dev/null +++ b/sound/pci/ctxfi/ctmixer.c @@ -0,0 +1,1108 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctmixer.c + * + * @Brief + * This file contains the implementation of alsa mixer device functions. + * + * @Author Liu Chun + * @Date May 28 2008 + * + */ + + +#include "ctmixer.h" +#include "ctamixer.h" +#include +#include +#include +#include +#include + +enum CT_SUM_CTL { + SUM_IN_F, + SUM_IN_R, + SUM_IN_C, + SUM_IN_S, + SUM_IN_F_C, + + NUM_CT_SUMS +}; + +enum CT_AMIXER_CTL { + /* volume control mixers */ + AMIXER_MASTER_F, + AMIXER_MASTER_R, + AMIXER_MASTER_C, + AMIXER_MASTER_S, + AMIXER_PCM_F, + AMIXER_PCM_R, + AMIXER_PCM_C, + AMIXER_PCM_S, + AMIXER_SPDIFI, + AMIXER_LINEIN, + AMIXER_MIC, + AMIXER_SPDIFO, + AMIXER_WAVE_F, + AMIXER_WAVE_R, + AMIXER_WAVE_C, + AMIXER_WAVE_S, + AMIXER_MASTER_F_C, + AMIXER_PCM_F_C, + AMIXER_SPDIFI_C, + AMIXER_LINEIN_C, + AMIXER_MIC_C, + + /* this should always be the last one */ + NUM_CT_AMIXERS +}; + +enum CTALSA_MIXER_CTL { + /* volume control mixers */ + MIXER_MASTER_P, + MIXER_PCM_P, + MIXER_LINEIN_P, + MIXER_MIC_P, + MIXER_SPDIFI_P, + MIXER_SPDIFO_P, + MIXER_WAVEF_P, + MIXER_WAVER_P, + MIXER_WAVEC_P, + MIXER_WAVES_P, + MIXER_MASTER_C, + MIXER_PCM_C, + MIXER_LINEIN_C, + MIXER_MIC_C, + MIXER_SPDIFI_C, + + /* switch control mixers */ + MIXER_PCM_C_S, + MIXER_LINEIN_C_S, + MIXER_MIC_C_S, + MIXER_SPDIFI_C_S, + MIXER_LINEIN_P_S, + MIXER_SPDIFO_P_S, + MIXER_SPDIFI_P_S, + MIXER_WAVEF_P_S, + MIXER_WAVER_P_S, + MIXER_WAVEC_P_S, + MIXER_WAVES_P_S, + MIXER_DIGITAL_IO_S, + MIXER_IEC958_MASK, + MIXER_IEC958_DEFAULT, + MIXER_IEC958_STREAM, + + /* this should always be the last one */ + NUM_CTALSA_MIXERS +}; + +#define VOL_MIXER_START MIXER_MASTER_P +#define VOL_MIXER_END MIXER_SPDIFI_C +#define VOL_MIXER_NUM (VOL_MIXER_END - VOL_MIXER_START + 1) +#define SWH_MIXER_START MIXER_PCM_C_S +#define SWH_MIXER_END MIXER_DIGITAL_IO_S +#define SWH_CAPTURE_START MIXER_PCM_C_S +#define SWH_CAPTURE_END MIXER_SPDIFI_C_S + +#define CHN_NUM 2 + +struct ct_kcontrol_init { + unsigned char ctl; + char *name; +}; + +static struct ct_kcontrol_init +ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = { + [MIXER_MASTER_P] = { + .ctl = 1, + .name = "Master Playback Volume", + }, + [MIXER_MASTER_C] = { + .ctl = 1, + .name = "Master Capture Volume", + }, + [MIXER_PCM_P] = { + .ctl = 1, + .name = "PCM Playback Volume", + }, + [MIXER_PCM_C] = { + .ctl = 1, + .name = "PCM Capture Volume", + }, + [MIXER_LINEIN_P] = { + .ctl = 1, + .name = "Line-in Playback Volume", + }, + [MIXER_LINEIN_C] = { + .ctl = 1, + .name = "Line-in Capture Volume", + }, + [MIXER_MIC_P] = { + .ctl = 1, + .name = "Mic Playback Volume", + }, + [MIXER_MIC_C] = { + .ctl = 1, + .name = "Mic Capture Volume", + }, + [MIXER_SPDIFI_P] = { + .ctl = 1, + .name = "S/PDIF-in Playback Volume", + }, + [MIXER_SPDIFI_C] = { + .ctl = 1, + .name = "S/PDIF-in Capture Volume", + }, + [MIXER_SPDIFO_P] = { + .ctl = 1, + .name = "S/PDIF-out Playback Volume", + }, + [MIXER_WAVEF_P] = { + .ctl = 1, + .name = "Front Playback Volume", + }, + [MIXER_WAVES_P] = { + .ctl = 1, + .name = "Surround Playback Volume", + }, + [MIXER_WAVEC_P] = { + .ctl = 1, + .name = "Center/LFE Playback Volume", + }, + [MIXER_WAVER_P] = { + .ctl = 1, + .name = "Rear Playback Volume", + }, + + [MIXER_PCM_C_S] = { + .ctl = 1, + .name = "PCM Capture Switch", + }, + [MIXER_LINEIN_C_S] = { + .ctl = 1, + .name = "Line-in Capture Switch", + }, + [MIXER_MIC_C_S] = { + .ctl = 1, + .name = "Mic Capture Switch", + }, + [MIXER_SPDIFI_C_S] = { + .ctl = 1, + .name = "S/PDIF-in Capture Switch", + }, + [MIXER_LINEIN_P_S] = { + .ctl = 1, + .name = "Line-in Playback Switch", + }, + [MIXER_SPDIFO_P_S] = { + .ctl = 1, + .name = "S/PDIF-out Playback Switch", + }, + [MIXER_SPDIFI_P_S] = { + .ctl = 1, + .name = "S/PDIF-in Playback Switch", + }, + [MIXER_WAVEF_P_S] = { + .ctl = 1, + .name = "Front Playback Switch", + }, + [MIXER_WAVES_P_S] = { + .ctl = 1, + .name = "Surround Playback Switch", + }, + [MIXER_WAVEC_P_S] = { + .ctl = 1, + .name = "Center/LFE Playback Switch", + }, + [MIXER_WAVER_P_S] = { + .ctl = 1, + .name = "Rear Playback Switch", + }, + [MIXER_DIGITAL_IO_S] = { + .ctl = 0, + .name = "Digit-IO Playback Switch", + }, +}; + +static void +ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type); + +static void +ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type); + +static struct snd_kcontrol *kctls[2] = {NULL}; + +static enum CT_AMIXER_CTL get_amixer_index(enum CTALSA_MIXER_CTL alsa_index) +{ + switch (alsa_index) { + case MIXER_MASTER_P: return AMIXER_MASTER_F; + case MIXER_MASTER_C: return AMIXER_MASTER_F_C; + case MIXER_PCM_P: return AMIXER_PCM_F; + case MIXER_PCM_C: + case MIXER_PCM_C_S: return AMIXER_PCM_F_C; + case MIXER_LINEIN_P: return AMIXER_LINEIN; + case MIXER_LINEIN_C: + case MIXER_LINEIN_C_S: return AMIXER_LINEIN_C; + case MIXER_MIC_P: return AMIXER_MIC; + case MIXER_MIC_C: + case MIXER_MIC_C_S: return AMIXER_MIC_C; + case MIXER_SPDIFI_P: return AMIXER_SPDIFI; + case MIXER_SPDIFI_C: + case MIXER_SPDIFI_C_S: return AMIXER_SPDIFI_C; + case MIXER_SPDIFO_P: return AMIXER_SPDIFO; + case MIXER_WAVEF_P: return AMIXER_WAVE_F; + case MIXER_WAVES_P: return AMIXER_WAVE_S; + case MIXER_WAVEC_P: return AMIXER_WAVE_C; + case MIXER_WAVER_P: return AMIXER_WAVE_R; + default: return NUM_CT_AMIXERS; + } +} + +static enum CT_AMIXER_CTL get_recording_amixer(enum CT_AMIXER_CTL index) +{ + switch (index) { + case AMIXER_MASTER_F: return AMIXER_MASTER_F_C; + case AMIXER_PCM_F: return AMIXER_PCM_F_C; + case AMIXER_SPDIFI: return AMIXER_SPDIFI_C; + case AMIXER_LINEIN: return AMIXER_LINEIN_C; + case AMIXER_MIC: return AMIXER_MIC_C; + default: return NUM_CT_AMIXERS; + } +} + +static unsigned char +get_switch_state(struct ct_mixer *mixer, enum CTALSA_MIXER_CTL type) +{ + return (mixer->switch_state & (0x1 << (type - SWH_MIXER_START))) + ? 1 : 0; +} + +static void +set_switch_state(struct ct_mixer *mixer, + enum CTALSA_MIXER_CTL type, unsigned char state) +{ + if (state) + mixer->switch_state |= (0x1 << (type - SWH_MIXER_START)); + else + mixer->switch_state &= ~(0x1 << (type - SWH_MIXER_START)); +} + +/* Map integer value ranging from 0 to 65535 to 14-bit float value ranging + * from 2^-6 to (1+1023/1024) */ +static unsigned int uint16_to_float14(unsigned int x) +{ + unsigned int i = 0; + + if (x < 17) + return 0; + + x *= 2031; + x /= 65535; + x += 16; + + /* i <= 6 */ + for (i = 0; !(x & 0x400); i++) + x <<= 1; + + x = (((7 - i) & 0x7) << 10) | (x & 0x3ff); + + return x; +} + +static unsigned int float14_to_uint16(unsigned int x) +{ + unsigned int e = 0; + + if (!x) + return x; + + e = (x >> 10) & 0x7; + x &= 0x3ff; + x += 1024; + x >>= (7 - e); + x -= 16; + x *= 65535; + x /= 2031; + + return x; +} + +static int ct_alsa_mix_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 43690; + uinfo->value.integer.step = 128; + + return 0; +} + +static int ct_alsa_mix_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ct_atc *atc = snd_kcontrol_chip(kcontrol); + enum CT_AMIXER_CTL type = get_amixer_index(kcontrol->private_value); + struct amixer *amixer = NULL; + int i = 0; + + for (i = 0; i < 2; i++) { + amixer = ((struct ct_mixer *)atc->mixer)-> + amixers[type*CHN_NUM+i]; + /* Convert 14-bit float-point scale to 16-bit integer volume */ + ucontrol->value.integer.value[i] = + (float14_to_uint16(amixer->ops->get_scale(amixer)) & 0xffff); + } + + return 0; +} + +static int ct_alsa_mix_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ct_atc *atc = snd_kcontrol_chip(kcontrol); + struct ct_mixer *mixer = atc->mixer; + enum CT_AMIXER_CTL type = get_amixer_index(kcontrol->private_value); + struct amixer *amixer = NULL; + int i = 0, j = 0, change = 0, val = 0; + + for (i = 0; i < 2; i++) { + /* Convert 16-bit integer volume to 14-bit float-point scale */ + val = (ucontrol->value.integer.value[i] & 0xffff); + amixer = mixer->amixers[type*CHN_NUM+i]; + if ((float14_to_uint16(amixer->ops->get_scale(amixer)) & 0xff80) + != (val & 0xff80)) { + val = uint16_to_float14(val); + amixer->ops->set_scale(amixer, val); + amixer->ops->commit_write(amixer); + change = 1; + /* Synchronize Master/PCM playback AMIXERs. */ + if (AMIXER_MASTER_F == type || AMIXER_PCM_F == type) { + for (j = 1; j < 4; j++) { + amixer = mixer-> + amixers[(type+j)*CHN_NUM+i]; + amixer->ops->set_scale(amixer, val); + amixer->ops->commit_write(amixer); + } + } + } + } + + return change; +} + +static struct snd_kcontrol_new vol_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = ct_alsa_mix_volume_info, + .get = ct_alsa_mix_volume_get, + .put = ct_alsa_mix_volume_put +}; + +static void +do_line_mic_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type) +{ + + if (MIXER_LINEIN_C_S == type) { + atc->select_line_in(atc); + set_switch_state(atc->mixer, MIXER_MIC_C_S, 0); + snd_ctl_notify(atc->card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctls[1]->id); + } else if (MIXER_MIC_C_S == type) { + atc->select_mic_in(atc); + set_switch_state(atc->mixer, MIXER_LINEIN_C_S, 0); + snd_ctl_notify(atc->card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctls[0]->id); + } +} + +static void +do_digit_io_switch(struct ct_atc *atc, int state) +{ + struct ct_mixer *mixer = atc->mixer; + + if (state) { + atc->select_digit_io(atc); + atc->spdif_out_unmute(atc, + get_switch_state(mixer, MIXER_SPDIFO_P_S)); + atc->spdif_in_unmute(atc, 1); + atc->line_in_unmute(atc, 0); + return; + } + + if (get_switch_state(mixer, MIXER_LINEIN_C_S)) + atc->select_line_in(atc); + else if (get_switch_state(mixer, MIXER_MIC_C_S)) + atc->select_mic_in(atc); + + atc->spdif_out_unmute(atc, 0); + atc->spdif_in_unmute(atc, 0); + atc->line_in_unmute(atc, 1); + return; +} + +static int ct_alsa_mix_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ct_alsa_mix_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ct_mixer *mixer = + ((struct ct_atc *)snd_kcontrol_chip(kcontrol))->mixer; + enum CTALSA_MIXER_CTL type = kcontrol->private_value; + + ucontrol->value.integer.value[0] = get_switch_state(mixer, type); + return 0; +} + +static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ct_atc *atc = snd_kcontrol_chip(kcontrol); + struct ct_mixer *mixer = atc->mixer; + enum CTALSA_MIXER_CTL type = kcontrol->private_value; + int state = 0; + + state = ucontrol->value.integer.value[0]; + if (get_switch_state(mixer, type) == state) + return 0; + + set_switch_state(mixer, type, state); + /* Do changes in mixer. */ + if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) { + if (state) { + ct_mixer_recording_select(mixer, + get_amixer_index(type)); + } else { + ct_mixer_recording_unselect(mixer, + get_amixer_index(type)); + } + } + /* Do changes out of mixer. */ + if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type)) + do_line_mic_switch(atc, type); + else if (MIXER_WAVEF_P_S == type) + atc->line_front_unmute(atc, state); + else if (MIXER_WAVES_P_S == type) + atc->line_surround_unmute(atc, state); + else if (MIXER_WAVEC_P_S == type) + atc->line_clfe_unmute(atc, state); + else if (MIXER_WAVER_P_S == type) + atc->line_rear_unmute(atc, state); + else if (MIXER_LINEIN_P_S == type) + atc->line_in_unmute(atc, state); + else if (MIXER_SPDIFO_P_S == type) + atc->spdif_out_unmute(atc, state); + else if (MIXER_SPDIFI_P_S == type) + atc->spdif_in_unmute(atc, state); + else if (MIXER_DIGITAL_IO_S == type) + do_digit_io_switch(atc, state); + + return 1; +} + +static struct snd_kcontrol_new swh_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = ct_alsa_mix_switch_info, + .get = ct_alsa_mix_switch_get, + .put = ct_alsa_mix_switch_put +}; + +static int ct_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int ct_spdif_get_mask(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0xff; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0xff; + ucontrol->value.iec958.status[3] = 0xff; + return 0; +} + +static int ct_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int status = SNDRV_PCM_DEFAULT_CON_SPDIF; + + ucontrol->value.iec958.status[0] = (status >> 0) & 0xff; + ucontrol->value.iec958.status[1] = (status >> 8) & 0xff; + ucontrol->value.iec958.status[2] = (status >> 16) & 0xff; + ucontrol->value.iec958.status[3] = (status >> 24) & 0xff; + + return 0; +} + +static int ct_spdif_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ct_atc *atc = snd_kcontrol_chip(kcontrol); + unsigned int status = 0; + + atc->spdif_out_get_status(atc, &status); + ucontrol->value.iec958.status[0] = (status >> 0) & 0xff; + ucontrol->value.iec958.status[1] = (status >> 8) & 0xff; + ucontrol->value.iec958.status[2] = (status >> 16) & 0xff; + ucontrol->value.iec958.status[3] = (status >> 24) & 0xff; + + return 0; +} + +static int ct_spdif_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ct_atc *atc = snd_kcontrol_chip(kcontrol); + int change = 1; + unsigned int status = 0, old_status = 0; + + status = (ucontrol->value.iec958.status[0] << 0) | + (ucontrol->value.iec958.status[1] << 8) | + (ucontrol->value.iec958.status[2] << 16) | + (ucontrol->value.iec958.status[3] << 24); + + atc->spdif_out_get_status(atc, &old_status); + change = (old_status != status); + if (change) + atc->spdif_out_set_status(atc, status); + + return change; +} + +static struct snd_kcontrol_new iec958_mask_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .count = 1, + .info = ct_spdif_info, + .get = ct_spdif_get_mask, + .private_value = MIXER_IEC958_MASK +}; + +static struct snd_kcontrol_new iec958_default_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .count = 1, + .info = ct_spdif_info, + .get = ct_spdif_default_get, + .put = ct_spdif_put, + .private_value = MIXER_IEC958_DEFAULT +}; + +static struct snd_kcontrol_new iec958_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM), + .count = 1, + .info = ct_spdif_info, + .get = ct_spdif_get, + .put = ct_spdif_put, + .private_value = MIXER_IEC958_STREAM +}; + +#define NUM_IEC958_CTL 3 + +static int +ct_mixer_kcontrol_new(struct ct_mixer *mixer, struct snd_kcontrol_new *new) +{ + struct snd_kcontrol *kctl = NULL; + int err = 0; + + kctl = snd_ctl_new1(new, mixer->atc); + if (NULL == kctl) + return -ENOMEM; + + if (SNDRV_CTL_ELEM_IFACE_PCM == kctl->id.iface) + kctl->id.device = IEC958; + + err = snd_ctl_add(mixer->atc->card, kctl); + if (err) + return err; + + switch (new->private_value) { + case MIXER_LINEIN_C_S: + kctls[0] = kctl; break; + case MIXER_MIC_C_S: + kctls[1] = kctl; break; + default: + break; + } + + return 0; +} + +static int ct_mixer_kcontrols_create(struct ct_mixer *mixer) +{ + enum CTALSA_MIXER_CTL type = 0; + struct ct_atc *atc = mixer->atc; + int err = 0; + + /* Create snd kcontrol instances on demand */ + for (type = VOL_MIXER_START; type <= VOL_MIXER_END; type++) { + if (ct_kcontrol_init_table[type].ctl) { + vol_ctl.name = ct_kcontrol_init_table[type].name; + vol_ctl.private_value = (unsigned long)type; + err = ct_mixer_kcontrol_new(mixer, &vol_ctl); + if (err) + return err; + } + } + + ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl = + atc->have_digit_io_switch(atc); + for (type = SWH_MIXER_START; type <= SWH_MIXER_END; type++) { + if (ct_kcontrol_init_table[type].ctl) { + swh_ctl.name = ct_kcontrol_init_table[type].name; + swh_ctl.private_value = (unsigned long)type; + err = ct_mixer_kcontrol_new(mixer, &swh_ctl); + if (err) + return err; + } + } + + err = ct_mixer_kcontrol_new(mixer, &iec958_mask_ctl); + if (err) + return err; + + err = ct_mixer_kcontrol_new(mixer, &iec958_default_ctl); + if (err) + return err; + + err = ct_mixer_kcontrol_new(mixer, &iec958_ctl); + if (err) + return err; + + atc->line_front_unmute(atc, 1); + set_switch_state(mixer, MIXER_WAVEF_P_S, 1); + atc->line_surround_unmute(atc, 0); + set_switch_state(mixer, MIXER_WAVES_P_S, 0); + atc->line_clfe_unmute(atc, 0); + set_switch_state(mixer, MIXER_WAVEC_P_S, 0); + atc->line_rear_unmute(atc, 0); + set_switch_state(mixer, MIXER_WAVER_P_S, 0); + atc->spdif_out_unmute(atc, 0); + set_switch_state(mixer, MIXER_SPDIFO_P_S, 0); + atc->line_in_unmute(atc, 0); + set_switch_state(mixer, MIXER_LINEIN_P_S, 0); + atc->spdif_in_unmute(atc, 0); + set_switch_state(mixer, MIXER_SPDIFI_P_S, 0); + + set_switch_state(mixer, MIXER_PCM_C_S, 1); + set_switch_state(mixer, MIXER_LINEIN_C_S, 1); + set_switch_state(mixer, MIXER_SPDIFI_C_S, 1); + + return 0; +} + +static void +ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type) +{ + struct amixer *amix_d = NULL; + struct sum *sum_c = NULL; + int i = 0; + + for (i = 0; i < 2; i++) { + amix_d = mixer->amixers[type*CHN_NUM+i]; + sum_c = mixer->sums[SUM_IN_F_C*CHN_NUM+i]; + amix_d->ops->set_sum(amix_d, sum_c); + amix_d->ops->commit_write(amix_d); + } +} + +static void +ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type) +{ + struct amixer *amix_d = NULL; + int i = 0; + + for (i = 0; i < 2; i++) { + amix_d = mixer->amixers[type*CHN_NUM+i]; + amix_d->ops->set_sum(amix_d, NULL); + amix_d->ops->commit_write(amix_d); + } +} + +static int ct_mixer_get_resources(struct ct_mixer *mixer) +{ + struct sum_mgr *sum_mgr = NULL; + struct sum *sum = NULL; + struct sum_desc sum_desc = {0}; + struct amixer_mgr *amixer_mgr = NULL; + struct amixer *amixer = NULL; + struct amixer_desc am_desc = {0}; + int err = 0; + int i = 0; + + /* Allocate sum resources for mixer obj */ + sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM]; + sum_desc.msr = mixer->atc->msr; + for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) { + err = sum_mgr->get_sum(sum_mgr, &sum_desc, &sum); + if (err) { + printk(KERN_ERR "Failed to get sum resources for " + "front output!\n"); + break; + } + mixer->sums[i] = sum; + } + if (err) + goto error1; + + /* Allocate amixer resources for mixer obj */ + amixer_mgr = (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER]; + am_desc.msr = mixer->atc->msr; + for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { + err = amixer_mgr->get_amixer(amixer_mgr, &am_desc, &amixer); + if (err) { + printk(KERN_ERR "Failed to get amixer resources for " + "mixer obj!\n"); + break; + } + mixer->amixers[i] = amixer; + } + if (err) + goto error2; + + return 0; + +error2: + for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { + if (NULL != mixer->amixers[i]) { + amixer = mixer->amixers[i]; + amixer_mgr->put_amixer(amixer_mgr, amixer); + mixer->amixers[i] = NULL; + } + } +error1: + for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) { + if (NULL != mixer->sums[i]) { + sum_mgr->put_sum(sum_mgr, (struct sum *)mixer->sums[i]); + mixer->sums[i] = NULL; + } + } + + return err; +} + +static int ct_mixer_get_mem(struct ct_mixer **rmixer) +{ + struct ct_mixer *mixer = NULL; + int err = 0; + + *rmixer = NULL; + /* Allocate mem for mixer obj */ + mixer = kzalloc(sizeof(*mixer), GFP_KERNEL); + if (NULL == mixer) + return -ENOMEM; + + mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM), + GFP_KERNEL); + if (NULL == mixer->amixers) { + err = -ENOMEM; + goto error1; + } + mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL); + if (NULL == mixer->sums) { + err = -ENOMEM; + goto error2; + } + + *rmixer = mixer; + return 0; + +error2: + kfree(mixer->amixers); +error1: + kfree(mixer); + return err; +} + +static int ct_mixer_topology_build(struct ct_mixer *mixer) +{ + struct sum *sum = NULL; + struct amixer *amix_d = NULL, *amix_s = NULL; + enum CT_AMIXER_CTL i = 0, j = 0; + + /* Build topology from destination to source */ + + /* Set up Master mixer */ + for (i = AMIXER_MASTER_F, j = SUM_IN_F; + i <= AMIXER_MASTER_S; i++, j++) { + amix_d = mixer->amixers[i*CHN_NUM]; + sum = mixer->sums[j*CHN_NUM]; + amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); + amix_d = mixer->amixers[i*CHN_NUM+1]; + sum = mixer->sums[j*CHN_NUM+1]; + amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); + } + + /* Set up Wave-out mixer */ + for (i = AMIXER_WAVE_F, j = AMIXER_MASTER_F; + i <= AMIXER_WAVE_S; i++, j++) { + amix_d = mixer->amixers[i*CHN_NUM]; + amix_s = mixer->amixers[j*CHN_NUM]; + amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); + amix_d = mixer->amixers[i*CHN_NUM+1]; + amix_s = mixer->amixers[j*CHN_NUM+1]; + amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); + } + + /* Set up S/PDIF-out mixer */ + amix_d = mixer->amixers[AMIXER_SPDIFO*CHN_NUM]; + amix_s = mixer->amixers[AMIXER_MASTER_F*CHN_NUM]; + amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); + amix_d = mixer->amixers[AMIXER_SPDIFO*CHN_NUM+1]; + amix_s = mixer->amixers[AMIXER_MASTER_F*CHN_NUM+1]; + amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); + + /* Set up PCM-in mixer */ + for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) { + amix_d = mixer->amixers[i*CHN_NUM]; + sum = mixer->sums[j*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[i*CHN_NUM+1]; + sum = mixer->sums[j*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + } + + /* Set up Line-in mixer */ + amix_d = mixer->amixers[AMIXER_LINEIN*CHN_NUM]; + sum = mixer->sums[SUM_IN_F*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_LINEIN*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + /* Set up Mic-in mixer */ + amix_d = mixer->amixers[AMIXER_MIC*CHN_NUM]; + sum = mixer->sums[SUM_IN_F*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_MIC*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + /* Set up S/PDIF-in mixer */ + amix_d = mixer->amixers[AMIXER_SPDIFI*CHN_NUM]; + sum = mixer->sums[SUM_IN_F*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_SPDIFI*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + /* Set up Master recording mixer */ + amix_d = mixer->amixers[AMIXER_MASTER_F_C*CHN_NUM]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM]; + amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); + amix_d = mixer->amixers[AMIXER_MASTER_F_C*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1]; + amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); + + /* Set up PCM-in recording mixer */ + amix_d = mixer->amixers[AMIXER_PCM_F_C*CHN_NUM]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_PCM_F_C*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + /* Set up Line-in recording mixer */ + amix_d = mixer->amixers[AMIXER_LINEIN_C*CHN_NUM]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_LINEIN_C*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + /* Set up Mic-in recording mixer */ + amix_d = mixer->amixers[AMIXER_MIC_C*CHN_NUM]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_MIC_C*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + /* Set up S/PDIF-in recording mixer */ + amix_d = mixer->amixers[AMIXER_SPDIFI_C*CHN_NUM]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + amix_d = mixer->amixers[AMIXER_SPDIFI_C*CHN_NUM+1]; + sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1]; + amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); + + return 0; +} + +static int mixer_set_input_port(struct amixer *amixer, struct rsc *rsc) +{ + amixer->ops->set_input(amixer, rsc); + amixer->ops->commit_write(amixer); + + return 0; +} + +static enum CT_AMIXER_CTL port_to_amixer(enum MIXER_PORT_T type) +{ + switch (type) { + case MIX_WAVE_FRONT: return AMIXER_WAVE_F; + case MIX_WAVE_SURROUND: return AMIXER_WAVE_S; + case MIX_WAVE_CENTLFE: return AMIXER_WAVE_C; + case MIX_WAVE_REAR: return AMIXER_WAVE_R; + case MIX_PCMO_FRONT: return AMIXER_MASTER_F_C; + case MIX_SPDIF_OUT: return AMIXER_SPDIFO; + case MIX_LINE_IN: return AMIXER_LINEIN; + case MIX_MIC_IN: return AMIXER_MIC; + case MIX_SPDIF_IN: return AMIXER_SPDIFI; + case MIX_PCMI_FRONT: return AMIXER_PCM_F; + case MIX_PCMI_SURROUND: return AMIXER_PCM_S; + case MIX_PCMI_CENTLFE: return AMIXER_PCM_C; + case MIX_PCMI_REAR: return AMIXER_PCM_R; + default: return 0; + } +} + +static int mixer_get_output_ports(struct ct_mixer *mixer, + enum MIXER_PORT_T type, + struct rsc **rleft, struct rsc **rright) +{ + enum CT_AMIXER_CTL amix = port_to_amixer(type); + + if (NULL != rleft) + *rleft = &((struct amixer *)mixer->amixers[amix*CHN_NUM])->rsc; + + if (NULL != rright) + *rright = + &((struct amixer *)mixer->amixers[amix*CHN_NUM+1])->rsc; + + return 0; +} + +static int mixer_set_input_left(struct ct_mixer *mixer, + enum MIXER_PORT_T type, struct rsc *rsc) +{ + enum CT_AMIXER_CTL amix = port_to_amixer(type); + + mixer_set_input_port(mixer->amixers[amix*CHN_NUM], rsc); + amix = get_recording_amixer(amix); + if (amix < NUM_CT_AMIXERS) + mixer_set_input_port(mixer->amixers[amix*CHN_NUM], rsc); + + return 0; +} + +static int +mixer_set_input_right(struct ct_mixer *mixer, + enum MIXER_PORT_T type, struct rsc *rsc) +{ + enum CT_AMIXER_CTL amix = port_to_amixer(type); + + mixer_set_input_port(mixer->amixers[amix*CHN_NUM+1], rsc); + amix = get_recording_amixer(amix); + if (amix < NUM_CT_AMIXERS) + mixer_set_input_port(mixer->amixers[amix*CHN_NUM+1], rsc); + + return 0; +} + +int ct_mixer_destroy(struct ct_mixer *mixer) +{ + struct sum_mgr *sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM]; + struct amixer_mgr *amixer_mgr = + (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER]; + struct amixer *amixer = NULL; + int i = 0; + + /* Release amixer resources */ + for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { + if (NULL != mixer->amixers[i]) { + amixer = mixer->amixers[i]; + amixer_mgr->put_amixer(amixer_mgr, amixer); + } + } + + /* Release sum resources */ + for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) { + if (NULL != mixer->sums[i]) + sum_mgr->put_sum(sum_mgr, (struct sum *)mixer->sums[i]); + } + + /* Release mem assigned to mixer object */ + kfree(mixer->sums); + kfree(mixer->amixers); + kfree(mixer); + + return 0; +} + +int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer) +{ + struct ct_mixer *mixer = NULL; + int err = 0; + + *rmixer = NULL; + + /* Allocate mem for mixer obj */ + err = ct_mixer_get_mem(&mixer); + if (err) + return err; + + mixer->switch_state = 0; + mixer->atc = atc; + /* Set operations */ + mixer->get_output_ports = mixer_get_output_ports; + mixer->set_input_left = mixer_set_input_left; + mixer->set_input_right = mixer_set_input_right; + + /* Allocate chip resources for mixer obj */ + err = ct_mixer_get_resources(mixer); + if (err) + goto error; + + /* Build internal mixer topology */ + ct_mixer_topology_build(mixer); + + *rmixer = mixer; + + return 0; + +error: + ct_mixer_destroy(mixer); + return err; +} + +int ct_alsa_mix_create(struct ct_atc *atc, + enum CTALSADEVS device, + const char *device_name) +{ + int err = 0; + + /* Create snd kcontrol instances on demand */ + vol_ctl.device = swh_ctl.device = device; + err = ct_mixer_kcontrols_create((struct ct_mixer *)atc->mixer); + if (err) + return err; + + strcpy(atc->card->mixername, device_name); + + return 0; +} diff --git a/sound/pci/ctxfi/ctmixer.h b/sound/pci/ctxfi/ctmixer.h new file mode 100644 index 0000000..e2d96eb --- /dev/null +++ b/sound/pci/ctxfi/ctmixer.h @@ -0,0 +1,67 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctmixer.h + * + * @Brief + * This file contains the definition of the mixer device functions. + * + * @Author Liu Chun + * @Date Mar 28 2008 + * + */ + +#ifndef CTMIXER_H +#define CTMIXER_H + +#include "ctatc.h" +#include "ctresource.h" + +#define INIT_VOL 0x1c00 + +enum MIXER_PORT_T { + MIX_WAVE_FRONT, + MIX_WAVE_REAR, + MIX_WAVE_CENTLFE, + MIX_WAVE_SURROUND, + MIX_SPDIF_OUT, + MIX_PCMO_FRONT, + MIX_MIC_IN, + MIX_LINE_IN, + MIX_SPDIF_IN, + MIX_PCMI_FRONT, + MIX_PCMI_REAR, + MIX_PCMI_CENTLFE, + MIX_PCMI_SURROUND, + + NUM_MIX_PORTS +}; + +/* alsa mixer descriptor */ +struct ct_mixer { + struct ct_atc *atc; + + void **amixers; /* amixer resources for volume control */ + void **sums; /* sum resources for signal collection */ + unsigned int switch_state; /* A bit-map to indicate state of switches */ + + int (*get_output_ports)(struct ct_mixer *mixer, enum MIXER_PORT_T type, + struct rsc **rleft, struct rsc **rright); + + int (*set_input_left)(struct ct_mixer *mixer, + enum MIXER_PORT_T type, struct rsc *rsc); + int (*set_input_right)(struct ct_mixer *mixer, + enum MIXER_PORT_T type, struct rsc *rsc); +}; + +int ct_alsa_mix_create(struct ct_atc *atc, + enum CTALSADEVS device, + const char *device_name); +int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer); +int ct_mixer_destroy(struct ct_mixer *mixer); + +#endif /* CTMIXER_H */ diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c new file mode 100644 index 0000000..73d4fdb --- /dev/null +++ b/sound/pci/ctxfi/ctpcm.c @@ -0,0 +1,499 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctpcm.c + * + * @Brief + * This file contains the definition of the pcm device functions. + * + * @Author Liu Chun + * @Date Apr 2 2008 + * + */ + +#include "ctpcm.h" +#include + +/* Hardware descriptions for playback */ +static struct snd_pcm_hardware ct_pcm_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_192000), + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = (64), + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware ct_spdif_passthru_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE), + .rates = (SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_32000), + .rate_min = 32000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = (64), + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +/* Hardware descriptions for capture */ +static struct snd_pcm_hardware ct_pcm_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_96000), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = (384), + .period_bytes_max = (64*1024), + .periods_min = 2, + .periods_max = 1024, + .fifo_size = 0, +}; + +static void ct_atc_pcm_interrupt(struct ct_atc_pcm *atc_pcm) +{ + struct ct_atc_pcm *apcm = atc_pcm; + + if (NULL == apcm->substream) + return; + + snd_pcm_period_elapsed(apcm->substream); +} + +static void ct_atc_pcm_free_substream(struct snd_pcm_runtime *runtime) +{ + struct ct_atc_pcm *apcm = runtime->private_data; + struct ct_atc *atc = snd_pcm_substream_chip(apcm->substream); + + atc->pcm_release_resources(atc, apcm); + kfree(apcm); + runtime->private_data = NULL; +} + +/* pcm playback operations */ +static int ct_pcm_playback_open(struct snd_pcm_substream *substream) +{ + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm; + int err; + + apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); + if (NULL == apcm) + return -ENOMEM; + + spin_lock_init(&apcm->timer_lock); + apcm->stop_timer = 0; + apcm->substream = substream; + apcm->interrupt = ct_atc_pcm_interrupt; + runtime->private_data = apcm; + runtime->private_free = ct_atc_pcm_free_substream; + if (IEC958 == substream->pcm->device) { + runtime->hw = ct_spdif_passthru_playback_hw; + atc->spdif_out_passthru(atc, 1); + } else { + runtime->hw = ct_pcm_playback_hw; + if (FRONT == substream->pcm->device) + runtime->hw.channels_max = 8; + } + + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + kfree(apcm); + return err; + } + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + 1024, UINT_MAX); + if (err < 0) { + kfree(apcm); + return err; + } + + return 0; +} + +static int ct_pcm_playback_close(struct snd_pcm_substream *substream) +{ + struct ct_atc *atc = snd_pcm_substream_chip(substream); + + /* TODO: Notify mixer inactive. */ + if (IEC958 == substream->pcm->device) + atc->spdif_out_passthru(atc, 0); + + /* The ct_atc_pcm object will be freed by runtime->private_free */ + + return 0; +} + +static int ct_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int ct_pcm_hw_free(struct snd_pcm_substream *substream) +{ + /* Free snd-allocated pages */ + return snd_pcm_lib_free_pages(substream); +} + +static void ct_pcm_timer_callback(unsigned long data) +{ + struct ct_atc_pcm *apcm = (struct ct_atc_pcm *)data; + struct snd_pcm_substream *substream = apcm->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int period_size = runtime->period_size; + unsigned int buffer_size = runtime->buffer_size; + unsigned long flags; + unsigned int position = 0, dist = 0, interval = 0; + + position = substream->ops->pointer(substream); + dist = (position + buffer_size - apcm->position) % buffer_size; + if ((dist >= period_size) || + (position/period_size != apcm->position/period_size)) { + apcm->interrupt(apcm); + apcm->position = position; + } + /* Add extra HZ*5/1000 to avoid overrun issue when recording + * at 8kHz in 8-bit format or at 88kHz in 24-bit format. */ + interval = ((period_size - (position % period_size)) + * HZ + (runtime->rate - 1)) / runtime->rate + HZ * 5 / 1000; + spin_lock_irqsave(&apcm->timer_lock, flags); + apcm->timer.expires = jiffies + interval; + if (!apcm->stop_timer) + add_timer(&apcm->timer); + + spin_unlock_irqrestore(&apcm->timer_lock, flags); +} + +static int ct_pcm_timer_prepare(struct ct_atc_pcm *apcm) +{ + unsigned long flags; + + spin_lock_irqsave(&apcm->timer_lock, flags); + if (timer_pending(&apcm->timer)) { + /* The timer has already been started. */ + spin_unlock_irqrestore(&apcm->timer_lock, flags); + return 0; + } + + init_timer(&apcm->timer); + apcm->timer.data = (unsigned long)apcm; + apcm->timer.function = ct_pcm_timer_callback; + spin_unlock_irqrestore(&apcm->timer_lock, flags); + apcm->position = 0; + + return 0; +} + +static int ct_pcm_timer_start(struct ct_atc_pcm *apcm) +{ + struct snd_pcm_runtime *runtime = apcm->substream->runtime; + unsigned long flags; + + spin_lock_irqsave(&apcm->timer_lock, flags); + if (timer_pending(&apcm->timer)) { + /* The timer has already been started. */ + spin_unlock_irqrestore(&apcm->timer_lock, flags); + return 0; + } + + apcm->timer.expires = jiffies + (runtime->period_size * HZ + + (runtime->rate - 1)) / runtime->rate; + apcm->stop_timer = 0; + add_timer(&apcm->timer); + spin_unlock_irqrestore(&apcm->timer_lock, flags); + + return 0; +} + +static int ct_pcm_timer_stop(struct ct_atc_pcm *apcm) +{ + unsigned long flags; + + spin_lock_irqsave(&apcm->timer_lock, flags); + apcm->stop_timer = 1; + del_timer(&apcm->timer); + spin_unlock_irqrestore(&apcm->timer_lock, flags); + + try_to_del_timer_sync(&apcm->timer); + + return 0; +} + +static int ct_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + int err; + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = runtime->private_data; + + if (IEC958 == substream->pcm->device) + err = atc->spdif_passthru_playback_prepare(atc, apcm); + else + err = atc->pcm_playback_prepare(atc, apcm); + + if (err < 0) { + printk(KERN_ERR "Preparing pcm playback failed!!!\n"); + return err; + } + + ct_pcm_timer_prepare(apcm); + + return 0; +} + +static int +ct_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + atc->pcm_playback_start(atc, apcm); + ct_pcm_timer_start(apcm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ct_pcm_timer_stop(apcm); + atc->pcm_playback_stop(atc, apcm); + break; + default: + break; + } + + return 0; +} + +static snd_pcm_uframes_t +ct_pcm_playback_pointer(struct snd_pcm_substream *substream) +{ + unsigned long position; + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = runtime->private_data; + + /* Read out playback position */ + position = atc->pcm_playback_position(atc, apcm); + position = bytes_to_frames(runtime, position); + return position; +} + +/* pcm capture operations */ +static int ct_pcm_capture_open(struct snd_pcm_substream *substream) +{ + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm; + int err; + + apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); + if (NULL == apcm) + return -ENOMEM; + + spin_lock_init(&apcm->timer_lock); + apcm->started = 0; + apcm->stop_timer = 0; + apcm->substream = substream; + apcm->interrupt = ct_atc_pcm_interrupt; + runtime->private_data = apcm; + runtime->private_free = ct_atc_pcm_free_substream; + runtime->hw = ct_pcm_capture_hw; + runtime->hw.rate_max = atc->rsr * atc->msr; + + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + kfree(apcm); + return err; + } + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + 1024, UINT_MAX); + if (err < 0) { + kfree(apcm); + return err; + } + + return 0; +} + +static int ct_pcm_capture_close(struct snd_pcm_substream *substream) +{ + /* The ct_atc_pcm object will be freed by runtime->private_free */ + /* TODO: Notify mixer inactive. */ + return 0; +} + +static int ct_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + int err; + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = runtime->private_data; + + err = atc->pcm_capture_prepare(atc, apcm); + if (err < 0) { + printk(KERN_ERR "Preparing pcm capture failed!!!\n"); + return err; + } + + ct_pcm_timer_prepare(apcm); + + return 0; +} + +static int +ct_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + atc->pcm_capture_start(atc, apcm); + ct_pcm_timer_start(apcm); + break; + case SNDRV_PCM_TRIGGER_STOP: + ct_pcm_timer_stop(apcm); + atc->pcm_capture_stop(atc, apcm); + break; + default: + ct_pcm_timer_stop(apcm); + atc->pcm_capture_stop(atc, apcm); + break; + } + + return 0; +} + +static snd_pcm_uframes_t +ct_pcm_capture_pointer(struct snd_pcm_substream *substream) +{ + unsigned long position; + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = runtime->private_data; + + /* Read out playback position */ + position = atc->pcm_capture_position(atc, apcm); + position = bytes_to_frames(runtime, position); + return position; +} + +/* PCM operators for playback */ +static struct snd_pcm_ops ct_pcm_playback_ops = { + .open = ct_pcm_playback_open, + .close = ct_pcm_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ct_pcm_hw_params, + .hw_free = ct_pcm_hw_free, + .prepare = ct_pcm_playback_prepare, + .trigger = ct_pcm_playback_trigger, + .pointer = ct_pcm_playback_pointer, +}; + +/* PCM operators for capture */ +static struct snd_pcm_ops ct_pcm_capture_ops = { + .open = ct_pcm_capture_open, + .close = ct_pcm_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ct_pcm_hw_params, + .hw_free = ct_pcm_hw_free, + .prepare = ct_pcm_capture_prepare, + .trigger = ct_pcm_capture_trigger, + .pointer = ct_pcm_capture_pointer, +}; + +/* Create ALSA pcm device */ +int ct_alsa_pcm_create(struct ct_atc *atc, + enum CTALSADEVS device, + const char *device_name) +{ + struct snd_pcm *pcm; + int err; + int playback_count, capture_count; + char name[128]; + + strncpy(name, device_name, sizeof(name)); + playback_count = (IEC958 == device) ? 1 : 8; + capture_count = (FRONT == device) ? 1 : 0; + err = snd_pcm_new(atc->card, name, device, + playback_count, capture_count, &pcm); + if (err < 0) { + printk(KERN_ERR "snd_pcm_new failed!! Err=%d\n", err); + return err; + } + + pcm->private_data = atc; + pcm->info_flags = 0; + pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; + strcpy(pcm->name, device_name); + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ct_pcm_playback_ops); + + if (FRONT == device) + snd_pcm_set_ops(pcm, + SNDRV_PCM_STREAM_CAPTURE, &ct_pcm_capture_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(atc->pci), 128*1024, 128*1024); + + return 0; +} diff --git a/sound/pci/ctxfi/ctpcm.h b/sound/pci/ctxfi/ctpcm.h new file mode 100644 index 0000000..178da0d --- /dev/null +++ b/sound/pci/ctxfi/ctpcm.h @@ -0,0 +1,27 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctpcm.h + * + * @Brief + * This file contains the definition of the pcm device functions. + * + * @Author Liu Chun + * @Date Mar 28 2008 + * + */ + +#ifndef CTPCM_H +#define CTPCM_H + +#include "ctatc.h" + +int ct_alsa_pcm_create(struct ct_atc *atc, + enum CTALSADEVS device, + const char *device_name); + +#endif /* CTPCM_H */ diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c new file mode 100644 index 0000000..414fc23 --- /dev/null +++ b/sound/pci/ctxfi/ctresource.c @@ -0,0 +1,297 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctresource.c + * + * @Brief + * This file contains the implementation of some generic helper functions. + * + * @Author Liu Chun + * @Date May 15 2008 + * + */ + +#include "ctresource.h" +#include "cthardware.h" +#include +#include + +#define AUDIO_SLOT_BLOCK_NUM 256 + +/* Resource allocation based on bit-map management mechanism */ +static int +get_resource(u8 *rscs, unsigned int amount, + unsigned int multi, unsigned int *ridx) +{ + int i = 0, j = 0, k = 0, n = 0; + + /* Check whether there are sufficient resources to meet request. */ + for (i = 0, n = multi; i < amount; i++) { + j = i / 8; + k = i % 8; + if (rscs[j] & ((u8)1 << k)) { + n = multi; + continue; + } + if (!(--n)) + break; /* found sufficient contiguous resources */ + } + + if (i >= amount) { + /* Can not find sufficient contiguous resources */ + return -ENOENT; + } + + /* Mark the contiguous bits in resource bit-map as used */ + for (n = multi; n > 0; n--) { + j = i / 8; + k = i % 8; + rscs[j] |= ((u8)1 << k); + i--; + } + + *ridx = i + 1; + + return 0; +} + +static int put_resource(u8 *rscs, unsigned int multi, unsigned int idx) +{ + unsigned int i = 0, j = 0, k = 0, n = 0; + + /* Mark the contiguous bits in resource bit-map as used */ + for (n = multi, i = idx; n > 0; n--) { + j = i / 8; + k = i % 8; + rscs[j] &= ~((u8)1 << k); + i++; + } + + return 0; +} + +int mgr_get_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int *ridx) +{ + int err = 0; + + if (n > mgr->avail) + return -ENOENT; + + err = get_resource(mgr->rscs, mgr->amount, n, ridx); + if (!err) + mgr->avail -= n; + + return err; +} + +int mgr_put_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int idx) +{ + put_resource(mgr->rscs, n, idx); + mgr->avail += n; + + return 0; +} + +static unsigned char offset_in_audio_slot_block[NUM_RSCTYP] = { + /* SRC channel is at Audio Ring slot 1 every 16 slots. */ + [SRC] = 0x1, + [AMIXER] = 0x4, + [SUM] = 0xc, +}; + +static int rsc_index(const struct rsc *rsc) +{ + return rsc->conj; +} + +static int audio_ring_slot(const struct rsc *rsc) +{ + return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type]; +} + +static int rsc_next_conj(struct rsc *rsc) +{ + unsigned int i; + for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); ) + i++; + rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i); + return rsc->conj; +} + +static int rsc_master(struct rsc *rsc) +{ + return rsc->conj = rsc->idx; +} + +static struct rsc_ops rsc_generic_ops = { + .index = rsc_index, + .output_slot = audio_ring_slot, + .master = rsc_master, + .next_conj = rsc_next_conj, +}; + +int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw) +{ + int err = 0; + + rsc->idx = idx; + rsc->conj = idx; + rsc->type = type; + rsc->msr = msr; + rsc->hw = hw; + rsc->ops = &rsc_generic_ops; + if (NULL == hw) { + rsc->ctrl_blk = NULL; + return 0; + } + + switch (type) { + case SRC: + err = ((struct hw *)hw)->src_rsc_get_ctrl_blk(&rsc->ctrl_blk); + break; + case AMIXER: + err = ((struct hw *)hw)-> + amixer_rsc_get_ctrl_blk(&rsc->ctrl_blk); + break; + case SRCIMP: + case SUM: + case DAIO: + break; + default: + printk(KERN_ERR "Invalid resource type value %d!\n", type); + return -EINVAL; + } + + if (err) { + printk(KERN_ERR "Failed to get resource control block!\n"); + return err; + } + + return 0; +} + +int rsc_uninit(struct rsc *rsc) +{ + if ((NULL != rsc->hw) && (NULL != rsc->ctrl_blk)) { + switch (rsc->type) { + case SRC: + ((struct hw *)rsc->hw)-> + src_rsc_put_ctrl_blk(rsc->ctrl_blk); + break; + case AMIXER: + ((struct hw *)rsc->hw)-> + amixer_rsc_put_ctrl_blk(rsc->ctrl_blk); + break; + case SUM: + case DAIO: + break; + default: + printk(KERN_ERR "Invalid resource type value %d!\n", + rsc->type); + break; + } + + rsc->hw = rsc->ctrl_blk = NULL; + } + + rsc->idx = rsc->conj = 0; + rsc->type = NUM_RSCTYP; + rsc->msr = 0; + + return 0; +} + +int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, + unsigned int amount, void *hw_obj) +{ + int err = 0; + struct hw *hw = hw_obj; + + mgr->type = NUM_RSCTYP; + + mgr->rscs = kzalloc(((amount + 8 - 1) / 8), GFP_KERNEL); + if (NULL == mgr->rscs) + return -ENOMEM; + + switch (type) { + case SRC: + err = hw->src_mgr_get_ctrl_blk(&mgr->ctrl_blk); + break; + case SRCIMP: + err = hw->srcimp_mgr_get_ctrl_blk(&mgr->ctrl_blk); + break; + case AMIXER: + err = hw->amixer_mgr_get_ctrl_blk(&mgr->ctrl_blk); + break; + case DAIO: + err = hw->daio_mgr_get_ctrl_blk(hw, &mgr->ctrl_blk); + break; + case SUM: + break; + default: + printk(KERN_ERR "Invalid resource type value %d!\n", type); + err = -EINVAL; + goto error; + } + + if (err) { + printk(KERN_ERR "Failed to get manager control block!\n"); + goto error; + } + + mgr->type = type; + mgr->avail = mgr->amount = amount; + mgr->hw = hw; + + return 0; + +error: + kfree(mgr->rscs); + return err; +} + +int rsc_mgr_uninit(struct rsc_mgr *mgr) +{ + if (NULL != mgr->rscs) { + kfree(mgr->rscs); + mgr->rscs = NULL; + } + + if ((NULL != mgr->hw) && (NULL != mgr->ctrl_blk)) { + switch (mgr->type) { + case SRC: + ((struct hw *)mgr->hw)-> + src_mgr_put_ctrl_blk(mgr->ctrl_blk); + break; + case SRCIMP: + ((struct hw *)mgr->hw)-> + srcimp_mgr_put_ctrl_blk(mgr->ctrl_blk); + break; + case AMIXER: + ((struct hw *)mgr->hw)-> + amixer_mgr_put_ctrl_blk(mgr->ctrl_blk); + break; + case DAIO: + ((struct hw *)mgr->hw)-> + daio_mgr_put_ctrl_blk(mgr->ctrl_blk); + break; + case SUM: + break; + default: + printk(KERN_ERR "Invalid resource type value %d!\n", + mgr->type); + break; + } + + mgr->hw = mgr->ctrl_blk = NULL; + } + + mgr->type = NUM_RSCTYP; + mgr->avail = mgr->amount = 0; + + return 0; +} diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h new file mode 100644 index 0000000..0838c2e --- /dev/null +++ b/sound/pci/ctxfi/ctresource.h @@ -0,0 +1,72 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctresource.h + * + * @Brief + * This file contains the definition of generic hardware resources for + * resource management. + * + * @Author Liu Chun + * @Date May 13 2008 + * + */ + +#ifndef CTRESOURCE_H +#define CTRESOURCE_H + +#include + +enum RSCTYP { + SRC, + SRCIMP, + AMIXER, + SUM, + DAIO, + NUM_RSCTYP /* This must be the last one and less than 16 */ +}; + +struct rsc_ops; + +struct rsc { + u32 idx:12; /* The index of a resource */ + u32 type:4; /* The type (RSCTYP) of a resource */ + u32 conj:12; /* Current conjugate index */ + u32 msr:4; /* The Master Sample Rate a resource working on */ + void *ctrl_blk; /* Chip specific control info block for a resource */ + void *hw; /* Chip specific object for hardware access means */ + struct rsc_ops *ops; /* Generic resource operations */ +}; + +struct rsc_ops { + int (*master)(struct rsc *rsc); /* Move to master resource */ + int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ + int (*index)(const struct rsc *rsc); /* Return the index of resource */ + /* Return the output slot number */ + int (*output_slot)(const struct rsc *rsc); +}; + +int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw); +int rsc_uninit(struct rsc *rsc); + +struct rsc_mgr { + enum RSCTYP type; /* The type (RSCTYP) of resource to manage */ + unsigned int amount; /* The total amount of a kind of resource */ + unsigned int avail; /* The amount of currently available resources */ + unsigned char *rscs; /* The bit-map for resource allocation */ + void *ctrl_blk; /* Chip specific control info block */ + void *hw; /* Chip specific object for hardware access */ +}; + +/* Resource management is based on bit-map mechanism */ +int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, + unsigned int amount, void *hw); +int rsc_mgr_uninit(struct rsc_mgr *mgr); +int mgr_get_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int *ridx); +int mgr_put_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int idx); + +#endif /* CTRESOURCE_H */ diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c new file mode 100644 index 0000000..d3e0ad5 --- /dev/null +++ b/sound/pci/ctxfi/ctsrc.c @@ -0,0 +1,886 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctsrc.c + * + * @Brief + * This file contains the implementation of the Sample Rate Convertor + * resource management object. + * + * @Author Liu Chun + * @Date May 13 2008 + * + */ + +#include "ctsrc.h" +#include "cthardware.h" +#include + +#define SRC_RESOURCE_NUM 64 +#define SRCIMP_RESOURCE_NUM 256 + +static unsigned int conj_mask; + +static int src_default_config_memrd(struct src *src); +static int src_default_config_memwr(struct src *src); +static int src_default_config_arcrw(struct src *src); + +static int (*src_default_config[3])(struct src *) = { + [MEMRD] = src_default_config_memrd, + [MEMWR] = src_default_config_memwr, + [ARCRW] = src_default_config_arcrw +}; + +static int src_set_state(struct src *src, unsigned int state) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_state(src->rsc.ctrl_blk, state); + + return 0; +} + +static int src_set_bm(struct src *src, unsigned int bm) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_bm(src->rsc.ctrl_blk, bm); + + return 0; +} + +static int src_set_sf(struct src *src, unsigned int sf) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_sf(src->rsc.ctrl_blk, sf); + + return 0; +} + +static int src_set_pm(struct src *src, unsigned int pm) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_pm(src->rsc.ctrl_blk, pm); + + return 0; +} + +static int src_set_rom(struct src *src, unsigned int rom) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_rom(src->rsc.ctrl_blk, rom); + + return 0; +} + +static int src_set_vo(struct src *src, unsigned int vo) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_vo(src->rsc.ctrl_blk, vo); + + return 0; +} + +static int src_set_st(struct src *src, unsigned int st) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_st(src->rsc.ctrl_blk, st); + + return 0; +} + +static int src_set_bp(struct src *src, unsigned int bp) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_bp(src->rsc.ctrl_blk, bp); + + return 0; +} + +static int src_set_cisz(struct src *src, unsigned int cisz) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_cisz(src->rsc.ctrl_blk, cisz); + + return 0; +} + +static int src_set_ca(struct src *src, unsigned int ca) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_ca(src->rsc.ctrl_blk, ca); + + return 0; +} + +static int src_set_sa(struct src *src, unsigned int sa) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_sa(src->rsc.ctrl_blk, sa); + + return 0; +} + +static int src_set_la(struct src *src, unsigned int la) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_la(src->rsc.ctrl_blk, la); + + return 0; +} + +static int src_set_pitch(struct src *src, unsigned int pitch) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_pitch(src->rsc.ctrl_blk, pitch); + + return 0; +} + +static int src_set_clear_zbufs(struct src *src) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1); + + return 0; +} + +static int src_commit_write(struct src *src) +{ + struct hw *hw = NULL; + int i = 0; + unsigned int dirty = 0; + + hw = (struct hw *)src->rsc.hw; + src->rsc.ops->master(&src->rsc); + if (src->rsc.msr > 1) { + /* Save dirty flags for conjugate resource programming */ + dirty = hw->src_get_dirty(src->rsc.ctrl_blk) & conj_mask; + } + hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); + + /* Program conjugate parameter mixer resources */ + if (MEMWR == src->mode) + return 0; + + for (i = 1; i < src->rsc.msr; i++) { + src->rsc.ops->next_conj(&src->rsc); + hw->src_set_dirty(src->rsc.ctrl_blk, dirty); + hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); + } + src->rsc.ops->master(&src->rsc); + + return 0; +} + +static int src_get_ca(struct src *src) +{ + struct hw *hw = NULL; + + hw = (struct hw *)src->rsc.hw; + return hw->src_get_ca(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); +} + +static int src_init(struct src *src) +{ + src_default_config[src->mode](src); + + return 0; +} + +static struct src *src_next_interleave(struct src *src) +{ + return src->intlv; +} + +static int src_default_config_memrd(struct src *src) +{ + struct hw *hw = src->rsc.hw; + unsigned int rsr = 0, msr = 0; + + hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF); + hw->src_set_bm(src->rsc.ctrl_blk, 1); + for (rsr = 0, msr = src->rsc.msr; msr > 1; msr >>= 1) + rsr++; + + hw->src_set_rsr(src->rsc.ctrl_blk, rsr); + hw->src_set_sf(src->rsc.ctrl_blk, SRC_SF_S16); + hw->src_set_wr(src->rsc.ctrl_blk, 0); + hw->src_set_pm(src->rsc.ctrl_blk, 0); + hw->src_set_rom(src->rsc.ctrl_blk, 0); + hw->src_set_vo(src->rsc.ctrl_blk, 0); + hw->src_set_st(src->rsc.ctrl_blk, 0); + hw->src_set_ilsz(src->rsc.ctrl_blk, src->multi - 1); + hw->src_set_cisz(src->rsc.ctrl_blk, 0x80); + hw->src_set_sa(src->rsc.ctrl_blk, 0x0); + hw->src_set_la(src->rsc.ctrl_blk, 0x1000); + hw->src_set_ca(src->rsc.ctrl_blk, 0x80); + hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000); + hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1); + + src->rsc.ops->master(&src->rsc); + hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); + + for (msr = 1; msr < src->rsc.msr; msr++) { + src->rsc.ops->next_conj(&src->rsc); + hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000); + hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); + } + src->rsc.ops->master(&src->rsc); + + return 0; +} + +static int src_default_config_memwr(struct src *src) +{ + struct hw *hw = src->rsc.hw; + + hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF); + hw->src_set_bm(src->rsc.ctrl_blk, 1); + hw->src_set_rsr(src->rsc.ctrl_blk, 0); + hw->src_set_sf(src->rsc.ctrl_blk, SRC_SF_S16); + hw->src_set_wr(src->rsc.ctrl_blk, 1); + hw->src_set_pm(src->rsc.ctrl_blk, 0); + hw->src_set_rom(src->rsc.ctrl_blk, 0); + hw->src_set_vo(src->rsc.ctrl_blk, 0); + hw->src_set_st(src->rsc.ctrl_blk, 0); + hw->src_set_ilsz(src->rsc.ctrl_blk, 0); + hw->src_set_cisz(src->rsc.ctrl_blk, 0x80); + hw->src_set_sa(src->rsc.ctrl_blk, 0x0); + hw->src_set_la(src->rsc.ctrl_blk, 0x1000); + hw->src_set_ca(src->rsc.ctrl_blk, 0x80); + hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000); + hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1); + + src->rsc.ops->master(&src->rsc); + hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); + + return 0; +} + +static int src_default_config_arcrw(struct src *src) +{ + struct hw *hw = src->rsc.hw; + unsigned int rsr = 0, msr = 0; + unsigned int dirty; + + hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF); + hw->src_set_bm(src->rsc.ctrl_blk, 0); + for (rsr = 0, msr = src->rsc.msr; msr > 1; msr >>= 1) + rsr++; + + hw->src_set_rsr(src->rsc.ctrl_blk, rsr); + hw->src_set_sf(src->rsc.ctrl_blk, SRC_SF_F32); + hw->src_set_wr(src->rsc.ctrl_blk, 0); + hw->src_set_pm(src->rsc.ctrl_blk, 0); + hw->src_set_rom(src->rsc.ctrl_blk, 0); + hw->src_set_vo(src->rsc.ctrl_blk, 0); + hw->src_set_st(src->rsc.ctrl_blk, 0); + hw->src_set_ilsz(src->rsc.ctrl_blk, 0); + hw->src_set_cisz(src->rsc.ctrl_blk, 0x80); + hw->src_set_sa(src->rsc.ctrl_blk, 0x0); + /*hw->src_set_sa(src->rsc.ctrl_blk, 0x100);*/ + hw->src_set_la(src->rsc.ctrl_blk, 0x1000); + /*hw->src_set_la(src->rsc.ctrl_blk, 0x03ffffe0);*/ + hw->src_set_ca(src->rsc.ctrl_blk, 0x80); + hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000); + hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1); + + dirty = hw->src_get_dirty(src->rsc.ctrl_blk); + src->rsc.ops->master(&src->rsc); + for (msr = 0; msr < src->rsc.msr; msr++) { + hw->src_set_dirty(src->rsc.ctrl_blk, dirty); + hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc), + src->rsc.ctrl_blk); + src->rsc.ops->next_conj(&src->rsc); + } + src->rsc.ops->master(&src->rsc); + + return 0; +} + +static struct src_rsc_ops src_rsc_ops = { + .set_state = src_set_state, + .set_bm = src_set_bm, + .set_sf = src_set_sf, + .set_pm = src_set_pm, + .set_rom = src_set_rom, + .set_vo = src_set_vo, + .set_st = src_set_st, + .set_bp = src_set_bp, + .set_cisz = src_set_cisz, + .set_ca = src_set_ca, + .set_sa = src_set_sa, + .set_la = src_set_la, + .set_pitch = src_set_pitch, + .set_clr_zbufs = src_set_clear_zbufs, + .commit_write = src_commit_write, + .get_ca = src_get_ca, + .init = src_init, + .next_interleave = src_next_interleave, +}; + +static int +src_rsc_init(struct src *src, u32 idx, + const struct src_desc *desc, struct src_mgr *mgr) +{ + int err = 0; + int i = 0, n = 0; + struct src *p; + + n = (MEMRD == desc->mode) ? desc->multi : 1; + for (i = 0, p = src; i < n; i++, p++) { + err = rsc_init(&p->rsc, idx + i, SRC, desc->msr, mgr->mgr.hw); + if (err) + goto error1; + + /* Initialize src specific rsc operations */ + p->ops = &src_rsc_ops; + p->multi = (0 == i) ? desc->multi : 1; + p->mode = desc->mode; + src_default_config[desc->mode](p); + mgr->src_enable(mgr, p); + p->intlv = p + 1; + } + (--p)->intlv = NULL; /* Set @intlv of the last SRC to NULL */ + + mgr->commit_write(mgr); + + return 0; + +error1: + for (i--, p--; i >= 0; i--, p--) { + mgr->src_disable(mgr, p); + rsc_uninit(&p->rsc); + } + mgr->commit_write(mgr); + return err; +} + +static int src_rsc_uninit(struct src *src, struct src_mgr *mgr) +{ + int i = 0, n = 0; + struct src *p; + + n = (MEMRD == src->mode) ? src->multi : 1; + for (i = 0, p = src; i < n; i++, p++) { + mgr->src_disable(mgr, p); + rsc_uninit(&p->rsc); + p->multi = 0; + p->ops = NULL; + p->mode = NUM_SRCMODES; + p->intlv = NULL; + } + mgr->commit_write(mgr); + + return 0; +} + +static int +get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc) +{ + unsigned int idx = SRC_RESOURCE_NUM; + int err = 0; + struct src *src = NULL; + unsigned long flags; + + *rsrc = NULL; + + /* Check whether there are sufficient src resources to meet request. */ + spin_lock_irqsave(&mgr->mgr_lock, flags); + if (MEMRD == desc->mode) + err = mgr_get_resource(&mgr->mgr, desc->multi, &idx); + else + err = mgr_get_resource(&mgr->mgr, 1, &idx); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + if (err) { + printk(KERN_ERR "Can't meet SRC resource request!\n"); + return err; + } + + /* Allocate mem for master src resource */ + if (MEMRD == desc->mode) + src = kzalloc(sizeof(*src)*desc->multi, GFP_KERNEL); + else + src = kzalloc(sizeof(*src), GFP_KERNEL); + + if (NULL == src) { + err = -ENOMEM; + goto error1; + } + + err = src_rsc_init(src, idx, desc, mgr); + if (err) + goto error2; + + *rsrc = src; + + return 0; + +error2: + kfree(src); +error1: + spin_lock_irqsave(&mgr->mgr_lock, flags); + if (MEMRD == desc->mode) + mgr_put_resource(&mgr->mgr, desc->multi, idx); + else + mgr_put_resource(&mgr->mgr, 1, idx); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + return err; +} + +static int put_src_rsc(struct src_mgr *mgr, struct src *src) +{ + unsigned long flags; + + spin_lock_irqsave(&mgr->mgr_lock, flags); + src->rsc.ops->master(&src->rsc); + if (MEMRD == src->mode) + mgr_put_resource(&mgr->mgr, src->multi, + src->rsc.ops->index(&src->rsc)); + else + mgr_put_resource(&mgr->mgr, 1, src->rsc.ops->index(&src->rsc)); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + src_rsc_uninit(src, mgr); + kfree(src); + + return 0; +} + +static int src_enable_s(struct src_mgr *mgr, struct src *src) +{ + struct hw *hw = mgr->mgr.hw; + int i = 0; + + src->rsc.ops->master(&src->rsc); + for (i = 0; i < src->rsc.msr; i++) { + hw->src_mgr_enbs_src(mgr->mgr.ctrl_blk, + src->rsc.ops->index(&src->rsc)); + src->rsc.ops->next_conj(&src->rsc); + } + src->rsc.ops->master(&src->rsc); + + return 0; +} + +static int src_enable(struct src_mgr *mgr, struct src *src) +{ + struct hw *hw = mgr->mgr.hw; + int i = 0; + + src->rsc.ops->master(&src->rsc); + for (i = 0; i < src->rsc.msr; i++) { + hw->src_mgr_enb_src(mgr->mgr.ctrl_blk, + src->rsc.ops->index(&src->rsc)); + src->rsc.ops->next_conj(&src->rsc); + } + src->rsc.ops->master(&src->rsc); + + return 0; +} + +static int src_disable(struct src_mgr *mgr, struct src *src) +{ + struct hw *hw = mgr->mgr.hw; + int i = 0; + + src->rsc.ops->master(&src->rsc); + for (i = 0; i < src->rsc.msr; i++) { + hw->src_mgr_dsb_src(mgr->mgr.ctrl_blk, + src->rsc.ops->index(&src->rsc)); + src->rsc.ops->next_conj(&src->rsc); + } + src->rsc.ops->master(&src->rsc); + + return 0; +} + +static int src_mgr_commit_write(struct src_mgr *mgr) +{ + struct hw *hw = mgr->mgr.hw; + + hw->src_mgr_commit_write(hw, mgr->mgr.ctrl_blk); + + return 0; +} + +int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr) +{ + int err = 0, i = 0; + struct src_mgr *src_mgr; + + *rsrc_mgr = NULL; + src_mgr = kzalloc(sizeof(*src_mgr), GFP_KERNEL); + if (NULL == src_mgr) + return -ENOMEM; + + err = rsc_mgr_init(&src_mgr->mgr, SRC, SRC_RESOURCE_NUM, hw); + if (err) + goto error1; + + spin_lock_init(&src_mgr->mgr_lock); + conj_mask = ((struct hw *)hw)->src_dirty_conj_mask(); + + src_mgr->get_src = get_src_rsc; + src_mgr->put_src = put_src_rsc; + src_mgr->src_enable_s = src_enable_s; + src_mgr->src_enable = src_enable; + src_mgr->src_disable = src_disable; + src_mgr->commit_write = src_mgr_commit_write; + + /* Disable all SRC resources. */ + for (i = 0; i < 256; i++) + ((struct hw *)hw)->src_mgr_dsb_src(src_mgr->mgr.ctrl_blk, i); + + ((struct hw *)hw)->src_mgr_commit_write(hw, src_mgr->mgr.ctrl_blk); + + *rsrc_mgr = src_mgr; + + return 0; + +error1: + kfree(src_mgr); + return err; +} + +int src_mgr_destroy(struct src_mgr *src_mgr) +{ + rsc_mgr_uninit(&src_mgr->mgr); + kfree(src_mgr); + + return 0; +} + +/* SRCIMP resource manager operations */ + +static int srcimp_master(struct rsc *rsc) +{ + rsc->conj = 0; + return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; +} + +static int srcimp_next_conj(struct rsc *rsc) +{ + rsc->conj++; + return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj]; +} + +static int srcimp_index(const struct rsc *rsc) +{ + return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj]; +} + +static struct rsc_ops srcimp_basic_rsc_ops = { + .master = srcimp_master, + .next_conj = srcimp_next_conj, + .index = srcimp_index, + .output_slot = NULL, +}; + +static int srcimp_map(struct srcimp *srcimp, struct src *src, struct rsc *input) +{ + struct imapper *entry = NULL; + int i = 0; + + srcimp->rsc.ops->master(&srcimp->rsc); + src->rsc.ops->master(&src->rsc); + input->ops->master(input); + + /* Program master and conjugate resources */ + for (i = 0; i < srcimp->rsc.msr; i++) { + entry = &srcimp->imappers[i]; + entry->slot = input->ops->output_slot(input); + entry->user = src->rsc.ops->index(&src->rsc); + entry->addr = srcimp->rsc.ops->index(&srcimp->rsc); + srcimp->mgr->imap_add(srcimp->mgr, entry); + srcimp->mapped |= (0x1 << i); + + srcimp->rsc.ops->next_conj(&srcimp->rsc); + input->ops->next_conj(input); + } + + srcimp->rsc.ops->master(&srcimp->rsc); + input->ops->master(input); + + return 0; +} + +static int srcimp_unmap(struct srcimp *srcimp) +{ + int i = 0; + + /* Program master and conjugate resources */ + for (i = 0; i < srcimp->rsc.msr; i++) { + if (srcimp->mapped & (0x1 << i)) { + srcimp->mgr->imap_delete(srcimp->mgr, + &srcimp->imappers[i]); + srcimp->mapped &= ~(0x1 << i); + } + } + + return 0; +} + +static struct srcimp_rsc_ops srcimp_ops = { + .map = srcimp_map, + .unmap = srcimp_unmap +}; + +static int srcimp_rsc_init(struct srcimp *srcimp, + const struct srcimp_desc *desc, + struct srcimp_mgr *mgr) +{ + int err = 0; + + err = rsc_init(&srcimp->rsc, srcimp->idx[0], + SRCIMP, desc->msr, mgr->mgr.hw); + if (err) + return err; + + /* Reserve memory for imapper nodes */ + srcimp->imappers = kzalloc(sizeof(struct imapper)*desc->msr, + GFP_KERNEL); + if (NULL == srcimp->imappers) { + err = -ENOMEM; + goto error1; + } + + /* Set srcimp specific operations */ + srcimp->rsc.ops = &srcimp_basic_rsc_ops; + srcimp->ops = &srcimp_ops; + srcimp->mgr = mgr; + + srcimp->rsc.ops->master(&srcimp->rsc); + + return 0; + +error1: + rsc_uninit(&srcimp->rsc); + return err; +} + +static int srcimp_rsc_uninit(struct srcimp *srcimp) +{ + if (NULL != srcimp->imappers) { + kfree(srcimp->imappers); + srcimp->imappers = NULL; + } + srcimp->ops = NULL; + srcimp->mgr = NULL; + rsc_uninit(&srcimp->rsc); + + return 0; +} + +static int get_srcimp_rsc(struct srcimp_mgr *mgr, + const struct srcimp_desc *desc, + struct srcimp **rsrcimp) +{ + int err = 0, i = 0; + unsigned int idx = 0; + struct srcimp *srcimp = NULL; + unsigned long flags; + + *rsrcimp = NULL; + + /* Allocate mem for SRCIMP resource */ + srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL); + if (NULL == srcimp) { + err = -ENOMEM; + return err; + } + + /* Check whether there are sufficient SRCIMP resources. */ + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i = 0; i < desc->msr; i++) { + err = mgr_get_resource(&mgr->mgr, 1, &idx); + if (err) + break; + + srcimp->idx[i] = idx; + } + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + if (err) { + printk(KERN_ERR "Can't meet SRCIMP resource request!\n"); + goto error1; + } + + err = srcimp_rsc_init(srcimp, desc, mgr); + if (err) + goto error1; + + *rsrcimp = srcimp; + + return 0; + +error1: + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i--; i >= 0; i--) + mgr_put_resource(&mgr->mgr, 1, srcimp->idx[i]); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + kfree(srcimp); + return err; +} + +static int put_srcimp_rsc(struct srcimp_mgr *mgr, struct srcimp *srcimp) +{ + unsigned long flags; + int i = 0; + + spin_lock_irqsave(&mgr->mgr_lock, flags); + for (i = 0; i < srcimp->rsc.msr; i++) + mgr_put_resource(&mgr->mgr, 1, srcimp->idx[i]); + + spin_unlock_irqrestore(&mgr->mgr_lock, flags); + srcimp_rsc_uninit(srcimp); + kfree(srcimp); + + return 0; +} + +static int srcimp_map_op(void *data, struct imapper *entry) +{ + struct rsc_mgr *mgr = &((struct srcimp_mgr *)data)->mgr; + struct hw *hw = mgr->hw; + + hw->srcimp_mgr_set_imaparc(mgr->ctrl_blk, entry->slot); + hw->srcimp_mgr_set_imapuser(mgr->ctrl_blk, entry->user); + hw->srcimp_mgr_set_imapnxt(mgr->ctrl_blk, entry->next); + hw->srcimp_mgr_set_imapaddr(mgr->ctrl_blk, entry->addr); + hw->srcimp_mgr_commit_write(mgr->hw, mgr->ctrl_blk); + + return 0; +} + +static int srcimp_imap_add(struct srcimp_mgr *mgr, struct imapper *entry) +{ + unsigned long flags; + int err = 0; + + spin_lock_irqsave(&mgr->imap_lock, flags); + if ((0 == entry->addr) && (mgr->init_imap_added)) { + input_mapper_delete(&mgr->imappers, + mgr->init_imap, srcimp_map_op, mgr); + mgr->init_imap_added = 0; + } + err = input_mapper_add(&mgr->imappers, entry, srcimp_map_op, mgr); + spin_unlock_irqrestore(&mgr->imap_lock, flags); + + return err; +} + +static int srcimp_imap_delete(struct srcimp_mgr *mgr, struct imapper *entry) +{ + unsigned long flags; + int err = 0; + + spin_lock_irqsave(&mgr->imap_lock, flags); + err = input_mapper_delete(&mgr->imappers, entry, srcimp_map_op, mgr); + if (list_empty(&mgr->imappers)) { + input_mapper_add(&mgr->imappers, mgr->init_imap, + srcimp_map_op, mgr); + mgr->init_imap_added = 1; + } + spin_unlock_irqrestore(&mgr->imap_lock, flags); + + return err; +} + +int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr) +{ + int err = 0; + struct srcimp_mgr *srcimp_mgr; + struct imapper *entry; + + *rsrcimp_mgr = NULL; + srcimp_mgr = kzalloc(sizeof(*srcimp_mgr), GFP_KERNEL); + if (NULL == srcimp_mgr) + return -ENOMEM; + + err = rsc_mgr_init(&srcimp_mgr->mgr, SRCIMP, SRCIMP_RESOURCE_NUM, hw); + if (err) + goto error1; + + spin_lock_init(&srcimp_mgr->mgr_lock); + spin_lock_init(&srcimp_mgr->imap_lock); + INIT_LIST_HEAD(&srcimp_mgr->imappers); + entry = kzalloc(sizeof(*entry), GFP_KERNEL); + if (NULL == entry) { + err = -ENOMEM; + goto error2; + } + entry->slot = entry->addr = entry->next = entry->user = 0; + list_add(&entry->list, &srcimp_mgr->imappers); + srcimp_mgr->init_imap = entry; + srcimp_mgr->init_imap_added = 1; + + srcimp_mgr->get_srcimp = get_srcimp_rsc; + srcimp_mgr->put_srcimp = put_srcimp_rsc; + srcimp_mgr->imap_add = srcimp_imap_add; + srcimp_mgr->imap_delete = srcimp_imap_delete; + + *rsrcimp_mgr = srcimp_mgr; + + return 0; + +error2: + rsc_mgr_uninit(&srcimp_mgr->mgr); +error1: + kfree(srcimp_mgr); + return err; +} + +int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr) +{ + unsigned long flags; + + /* free src input mapper list */ + spin_lock_irqsave(&srcimp_mgr->imap_lock, flags); + free_input_mapper_list(&srcimp_mgr->imappers); + spin_unlock_irqrestore(&srcimp_mgr->imap_lock, flags); + + rsc_mgr_uninit(&srcimp_mgr->mgr); + kfree(srcimp_mgr); + + return 0; +} diff --git a/sound/pci/ctxfi/ctsrc.h b/sound/pci/ctxfi/ctsrc.h new file mode 100644 index 0000000..259366a --- /dev/null +++ b/sound/pci/ctxfi/ctsrc.h @@ -0,0 +1,149 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctsrc.h + * + * @Brief + * This file contains the definition of the Sample Rate Convertor + * resource management object. + * + * @Author Liu Chun + * @Date May 13 2008 + * + */ + +#ifndef CTSRC_H +#define CTSRC_H + +#include "ctresource.h" +#include "ctimap.h" +#include +#include + +#define SRC_STATE_OFF 0x0 +#define SRC_STATE_INIT 0x4 +#define SRC_STATE_RUN 0x5 + +#define SRC_SF_U8 0x0 +#define SRC_SF_S16 0x1 +#define SRC_SF_S24 0x2 +#define SRC_SF_S32 0x3 +#define SRC_SF_F32 0x4 + +/* Define the descriptor of a src resource */ +enum SRCMODE { + MEMRD, /* Read data from host memory */ + MEMWR, /* Write data to host memory */ + ARCRW, /* Read from and write to audio ring channel */ + NUM_SRCMODES +}; + +struct src_rsc_ops; + +struct src { + struct rsc rsc; /* Basic resource info */ + struct src *intlv; /* Pointer to next interleaved SRC in a series */ + struct src_rsc_ops *ops; /* SRC specific operations */ + /* Number of contiguous srcs for interleaved usage */ + unsigned char multi; + unsigned char mode; /* Working mode of this SRC resource */ +}; + +struct src_rsc_ops { + int (*set_state)(struct src *src, unsigned int state); + int (*set_bm)(struct src *src, unsigned int bm); + int (*set_sf)(struct src *src, unsigned int sf); + int (*set_pm)(struct src *src, unsigned int pm); + int (*set_rom)(struct src *src, unsigned int rom); + int (*set_vo)(struct src *src, unsigned int vo); + int (*set_st)(struct src *src, unsigned int st); + int (*set_bp)(struct src *src, unsigned int bp); + int (*set_cisz)(struct src *src, unsigned int cisz); + int (*set_ca)(struct src *src, unsigned int ca); + int (*set_sa)(struct src *src, unsigned int sa); + int (*set_la)(struct src *src, unsigned int la); + int (*set_pitch)(struct src *src, unsigned int pitch); + int (*set_clr_zbufs)(struct src *src); + int (*commit_write)(struct src *src); + int (*get_ca)(struct src *src); + int (*init)(struct src *src); + struct src* (*next_interleave)(struct src *src); +}; + +/* Define src resource request description info */ +struct src_desc { + /* Number of contiguous master srcs for interleaved usage */ + unsigned char multi; + unsigned char msr; + unsigned char mode; /* Working mode of the requested srcs */ +}; + +/* Define src manager object */ +struct src_mgr { + struct rsc_mgr mgr; /* Basic resource manager info */ + spinlock_t mgr_lock; + + /* request src resource */ + int (*get_src)(struct src_mgr *mgr, + const struct src_desc *desc, struct src **rsrc); + /* return src resource */ + int (*put_src)(struct src_mgr *mgr, struct src *src); + int (*src_enable_s)(struct src_mgr *mgr, struct src *src); + int (*src_enable)(struct src_mgr *mgr, struct src *src); + int (*src_disable)(struct src_mgr *mgr, struct src *src); + int (*commit_write)(struct src_mgr *mgr); +}; + +/* Define the descriptor of a SRC Input Mapper resource */ +struct srcimp_mgr; +struct srcimp_rsc_ops; + +struct srcimp { + struct rsc rsc; + unsigned char idx[8]; + struct imapper *imappers; + unsigned int mapped; /* A bit-map indicating which conj rsc is mapped */ + struct srcimp_mgr *mgr; + struct srcimp_rsc_ops *ops; +}; + +struct srcimp_rsc_ops { + int (*map)(struct srcimp *srcimp, struct src *user, struct rsc *input); + int (*unmap)(struct srcimp *srcimp); +}; + +/* Define SRCIMP resource request description info */ +struct srcimp_desc { + unsigned int msr; +}; + +struct srcimp_mgr { + struct rsc_mgr mgr; /* Basic resource manager info */ + spinlock_t mgr_lock; + spinlock_t imap_lock; + struct list_head imappers; + struct imapper *init_imap; + unsigned int init_imap_added; + + /* request srcimp resource */ + int (*get_srcimp)(struct srcimp_mgr *mgr, + const struct srcimp_desc *desc, + struct srcimp **rsrcimp); + /* return srcimp resource */ + int (*put_srcimp)(struct srcimp_mgr *mgr, struct srcimp *srcimp); + int (*imap_add)(struct srcimp_mgr *mgr, struct imapper *entry); + int (*imap_delete)(struct srcimp_mgr *mgr, struct imapper *entry); +}; + +/* Constructor and destructor of SRC resource manager */ +int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr); +int src_mgr_destroy(struct src_mgr *src_mgr); +/* Constructor and destructor of SRCIMP resource manager */ +int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrc_mgr); +int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr); + +#endif /* CTSRC_H */ diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c new file mode 100644 index 0000000..46ca04c --- /dev/null +++ b/sound/pci/ctxfi/ctvmem.c @@ -0,0 +1,254 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctvmem.c + * + * @Brief + * This file contains the implementation of virtual memory management object + * for card device. + * + * @Author Liu Chun + * @Date Apr 1 2008 + */ + +#include "ctvmem.h" +#include +#include +#include /* for PAGE_SIZE macro definition */ +#include +#include + +#define CT_PTES_PER_PAGE (PAGE_SIZE / sizeof(void *)) +#define CT_ADDRS_PER_PAGE (CT_PTES_PER_PAGE * PAGE_SIZE) + +/* * + * Find or create vm block based on requested @size. + * @size must be page aligned. + * */ +static struct ct_vm_block * +get_vm_block(struct ct_vm *vm, unsigned int size) +{ + struct ct_vm_block *block = NULL, *entry = NULL; + struct list_head *pos = NULL; + + list_for_each(pos, &vm->unused) { + entry = list_entry(pos, struct ct_vm_block, list); + if (entry->size >= size) + break; /* found a block that is big enough */ + } + if (pos == &vm->unused) + return NULL; + + if (entry->size == size) { + /* Move the vm node from unused list to used list directly */ + list_del(&entry->list); + list_add(&entry->list, &vm->used); + vm->size -= size; + return entry; + } + + block = kzalloc(sizeof(*block), GFP_KERNEL); + if (NULL == block) + return NULL; + + block->addr = entry->addr; + block->size = size; + list_add(&block->list, &vm->used); + entry->addr += size; + entry->size -= size; + vm->size -= size; + + return block; +} + +static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block) +{ + struct ct_vm_block *entry = NULL, *pre_ent = NULL; + struct list_head *pos = NULL, *pre = NULL; + + list_del(&block->list); + vm->size += block->size; + + list_for_each(pos, &vm->unused) { + entry = list_entry(pos, struct ct_vm_block, list); + if (entry->addr >= (block->addr + block->size)) + break; /* found a position */ + } + if (pos == &vm->unused) { + list_add_tail(&block->list, &vm->unused); + entry = block; + } else { + if ((block->addr + block->size) == entry->addr) { + entry->addr = block->addr; + entry->size += block->size; + kfree(block); + } else { + __list_add(&block->list, pos->prev, pos); + entry = block; + } + } + + pos = &entry->list; + pre = pos->prev; + while (pre != &vm->unused) { + entry = list_entry(pos, struct ct_vm_block, list); + pre_ent = list_entry(pre, struct ct_vm_block, list); + if ((pre_ent->addr + pre_ent->size) > entry->addr) + break; + + pre_ent->size += entry->size; + list_del(pos); + kfree(entry); + pos = pre; + pre = pos->prev; + } +} + +/* Map host addr (kmalloced/vmalloced) to device logical addr. */ +static struct ct_vm_block * +ct_vm_map(struct ct_vm *vm, void *host_addr, int size) +{ + struct ct_vm_block *block = NULL; + unsigned long pte_start; + unsigned long i; + unsigned long pages; + unsigned long start_phys; + unsigned long *ptp; + + /* do mapping */ + if ((unsigned long)host_addr >= VMALLOC_START) { + printk(KERN_ERR "Fail! Not support vmalloced addr now!\n"); + return NULL; + } + + if (size > vm->size) { + printk(KERN_ERR "Fail! No sufficient device virtural " + "memory space available!\n"); + return NULL; + } + + start_phys = (virt_to_phys(host_addr) & PAGE_MASK); + pages = (PAGE_ALIGN(virt_to_phys(host_addr) + size) + - start_phys) >> PAGE_SHIFT; + + ptp = vm->ptp[0]; + + block = get_vm_block(vm, (pages << PAGE_SHIFT)); + if (block == NULL) { + printk(KERN_ERR "No virtual memory block that is big " + "enough to allocate!\n"); + return NULL; + } + + pte_start = (block->addr >> PAGE_SHIFT); + for (i = 0; i < pages; i++) + ptp[pte_start+i] = start_phys + (i << PAGE_SHIFT); + + block->addr += (virt_to_phys(host_addr) & (~PAGE_MASK)); + block->size = size; + + return block; +} + +static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block) +{ + /* do unmapping */ + block->size = ((block->addr + block->size + PAGE_SIZE - 1) + & PAGE_MASK) - (block->addr & PAGE_MASK); + block->addr &= PAGE_MASK; + put_vm_block(vm, block); +} + +/* * + * return the host (kmalloced) addr of the @index-th device + * page talbe page on success, or NULL on failure. + * The first returned NULL indicates the termination. + * */ +static void * +ct_get_ptp_virt(struct ct_vm *vm, int index) +{ + void *addr; + + addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index]; + + return addr; +} + +int ct_vm_create(struct ct_vm **rvm) +{ + struct ct_vm *vm; + struct ct_vm_block *block; + int i; + + *rvm = NULL; + + vm = kzalloc(sizeof(*vm), GFP_KERNEL); + if (NULL == vm) + return -ENOMEM; + + /* Allocate page table pages */ + for (i = 0; i < CT_PTP_NUM; i++) { + vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (NULL == vm->ptp[i]) + break; + } + if (!i) { + /* no page table pages are allocated */ + kfree(vm); + return -ENOMEM; + } + vm->size = CT_ADDRS_PER_PAGE * i; + /* Initialise remaining ptps */ + for (; i < CT_PTP_NUM; i++) + vm->ptp[i] = NULL; + + vm->map = ct_vm_map; + vm->unmap = ct_vm_unmap; + vm->get_ptp_virt = ct_get_ptp_virt; + INIT_LIST_HEAD(&vm->unused); + INIT_LIST_HEAD(&vm->used); + block = kzalloc(sizeof(*block), GFP_KERNEL); + if (NULL != block) { + block->addr = 0; + block->size = vm->size; + list_add(&block->list, &vm->unused); + } + + *rvm = vm; + return 0; +} + +/* The caller must ensure no mapping pages are being used + * by hardware before calling this function */ +void ct_vm_destroy(struct ct_vm *vm) +{ + int i; + struct list_head *pos = NULL; + struct ct_vm_block *entry = NULL; + + /* free used and unused list nodes */ + while (!list_empty(&vm->used)) { + pos = vm->used.next; + list_del(pos); + entry = list_entry(pos, struct ct_vm_block, list); + kfree(entry); + } + while (!list_empty(&vm->unused)) { + pos = vm->unused.next; + list_del(pos); + entry = list_entry(pos, struct ct_vm_block, list); + kfree(entry); + } + + /* free allocated page table pages */ + for (i = 0; i < CT_PTP_NUM; i++) + kfree(vm->ptp[i]); + + vm->size = 0; + + kfree(vm); +} diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h new file mode 100644 index 0000000..4eb5bdd --- /dev/null +++ b/sound/pci/ctxfi/ctvmem.h @@ -0,0 +1,49 @@ +/** + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + * + * @File ctvmem.h + * + * @Brief + * This file contains the definition of virtual memory management object + * for card device. + * + * @Author Liu Chun + * @Date Mar 28 2008 + */ + +#ifndef CTVMEM_H +#define CTVMEM_H + +#define CT_PTP_NUM 1 /* num of device page table pages */ + +#include +#include + +struct ct_vm_block { + unsigned int addr; /* starting logical addr of this block */ + unsigned int size; /* size of this device virtual mem block */ + struct list_head list; +}; + +/* Virtual memory management object for card device */ +struct ct_vm { + void *ptp[CT_PTP_NUM]; /* Device page table pages */ + unsigned int size; /* Available addr space in bytes */ + struct list_head unused; /* List of unused blocks */ + struct list_head used; /* List of used blocks */ + + /* Map host addr (kmalloced/vmalloced) to device logical addr. */ + struct ct_vm_block *(*map)(struct ct_vm *, void *host_addr, int size); + /* Unmap device logical addr area. */ + void (*unmap)(struct ct_vm *, struct ct_vm_block *block); + void *(*get_ptp_virt)(struct ct_vm *vm, int index); +}; + +int ct_vm_create(struct ct_vm **rvm); +void ct_vm_destroy(struct ct_vm *vm); + +#endif /* CTVMEM_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c new file mode 100644 index 0000000..f911440 --- /dev/null +++ b/sound/pci/ctxfi/xfi.c @@ -0,0 +1,125 @@ +/* + * xfi linux driver. + * + * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + */ + +#include +#include +#include +#include +#include +#include "ctatc.h" +#include "ctdrv.h" + +MODULE_AUTHOR("Creative Technology Ltd"); +MODULE_DESCRIPTION("X-Fi driver version 1.03"); +MODULE_LICENSE("GPLv2"); +MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); + +static unsigned int reference_rate = 48000; +static unsigned int multiple = 2; +module_param(reference_rate, uint, S_IRUGO); +module_param(multiple, uint, S_IRUGO); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static struct pci_device_id ct_pci_dev_ids[] = { + /* only X-Fi is supported, so... */ + { PCI_DEVICE(PCI_VENDOR_CREATIVE, PCI_DEVICE_CREATIVE_20K1) }, + { PCI_DEVICE(PCI_VENDOR_CREATIVE, PCI_DEVICE_CREATIVE_20K2) }, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, ct_pci_dev_ids); + +static int __devinit +ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct ct_atc *atc; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + + if (!enable[dev]) { + dev++; + return -ENOENT; + } + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err) + return err; + if ((reference_rate != 48000) && (reference_rate != 44100)) { + printk(KERN_ERR "Invalid reference_rate value %u!!!\n" + "The valid values for reference_rate " + "are 48000 and 44100.\nValue 48000 is " + "assumed.\n", reference_rate); + reference_rate = 48000; + } + if ((multiple != 1) && (multiple != 2)) { + printk(KERN_ERR "Invalid multiple value %u!!!\nThe valid " + "values for multiple are 1 and 2.\nValue 2 " + "is assumed.\n", multiple); + multiple = 2; + } + err = ct_atc_create(card, pci, reference_rate, multiple, &atc); + if (err < 0) + goto error; + + card->private_data = atc; + + /* Create alsa devices supported by this card */ + err = atc->create_alsa_devs(atc); + if (err < 0) + goto error; + + strcpy(card->driver, "SB-XFi"); + strcpy(card->shortname, "Creative X-Fi"); + strcpy(card->longname, "Creative ALSA Driver X-Fi"); + + err = snd_card_register(card); + if (err < 0) + goto error; + + pci_set_drvdata(pci, card); + dev++; + + return 0; + +error: + snd_card_free(card); + return err; +} + +static void __devexit ct_card_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver ct_driver = { + .name = "SB-XFi", + .id_table = ct_pci_dev_ids, + .probe = ct_card_probe, + .remove = __devexit_p(ct_card_remove), +}; + +static int __init ct_card_init(void) +{ + return pci_register_driver(&ct_driver); +} + +static void __exit ct_card_exit(void) +{ + pci_unregister_driver(&ct_driver); +} + +module_init(ct_card_init) +module_exit(ct_card_exit) -- cgit v1.1 From d0da727e025da8b443a4a614dbb7a031b89857d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 10:56:04 +0200 Subject: ALSA: ctxfi - Add missing inclusion of linux/delay.h Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k1.c | 1 + sound/pci/ctxfi/cthw20k2.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 53572d92..44283bd 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -25,6 +25,7 @@ #include #include #include +#include #define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bits */ diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index cdcb75c..9c2d38b 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -24,6 +24,7 @@ #include #include #include +#include #define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bits */ -- cgit v1.1 From 14610ce711a363028ffffad98947d57f21fa5372 Mon Sep 17 00:00:00 2001 From: Anuj Aggarwal Date: Thu, 14 May 2009 13:59:19 +0530 Subject: ASoC: Added OMAP3 EVM support in ASoC. Resending the patch after fixing the minor issues. Signed-off-by: Anuj Aggarwal Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 8 +++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3evm.c | 147 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 157 insertions(+) create mode 100644 sound/soc/omap/omap3evm.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 675732e..b771238 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -39,6 +39,14 @@ config SND_OMAP_SOC_OMAP2EVM help Say Y if you want to add support for SoC audio on the omap2evm board. +config SND_OMAP_SOC_OMAP3EVM + tristate "SoC Audio support for OMAP3EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap3evm board. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c9e4ac..a37f498 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,6 +10,7 @@ snd-soc-n810-objs := n810.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o +snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o @@ -18,6 +19,7 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c new file mode 100644 index 0000000..9114c26 --- /dev/null +++ b/sound/soc/omap/omap3evm.c @@ -0,0 +1,147 @@ +/* + * omap3evm.c -- ALSA SoC support for OMAP3 EVM + * + * Author: Anuj Aggarwal + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2008 Texas Instruments, Incorporated + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "Can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "Can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3evm_ops = { + .hw_params = omap3evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &omap3evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3evm = { + .name = "omap3evm", + .platform = &omap_soc_platform, + .dai_link = &omap3evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3evm_snd_devdata = { + .card = &snd_soc_omap3evm, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3evm_snd_device; + +static int __init omap3evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3evm()) { + pr_err("Not OMAP3 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3 EVM SoC init\n"); + + omap3evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata); + omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev; + *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1; + + ret = platform_device_add(omap3evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3evm_snd_device); + + return ret; +} + +static void __exit omap3evm_soc_exit(void) +{ + platform_device_unregister(omap3evm_snd_device); +} + +module_init(omap3evm_soc_init); +module_exit(omap3evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal "); +MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); +MODULE_LICENSE("GPLv2"); -- cgit v1.1 From d34c43078236b41146877c49af341ab85b5fb8db Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Wed, 13 May 2009 21:59:14 -0400 Subject: ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 format Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 496dc30..352d7ee 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -79,7 +79,8 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_OUT 1 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_S32_BE) struct snd_soc_dai_ops; struct snd_soc_dai; -- cgit v1.1 From 903dba1eae49270a8c2275636071c3a238c8104d Mon Sep 17 00:00:00 2001 From: Jean Delvare Date: Thu, 14 May 2009 14:37:21 +0200 Subject: ALSA: keywest: Get rid of useless i2c_device_name() macro The i2c_device_name() macro is used only once and doesn't have much value, it hurts redability more than it helps. Get rid of it. Signed-off-by: Jean Delvare Cc: Benjamin Herrenschmidt Signed-off-by: Takashi Iwai --- sound/ppc/keywest.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index a5afb26..b2e2aac 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -33,10 +33,6 @@ static struct pmac_keywest *keywest_ctx; -#ifndef i2c_device_name -#define i2c_device_name(x) ((x)->name) -#endif - static int keywest_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -56,7 +52,7 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) if (! keywest_ctx) return -EINVAL; - if (strncmp(i2c_device_name(adapter), "mac-io", 6)) + if (strncmp(adapter->name, "mac-io", 6)) return 0; /* ignored */ memset(&info, 0, sizeof(struct i2c_board_info)); -- cgit v1.1 From 9fc20f030ba457d20eb994e1def7e2ce7d5ae1a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 15:14:18 +0200 Subject: ALSA: ctxfi - Move PCI ID definitions to linux/pci_ids.h Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 7 +++++++ sound/pci/ctxfi/ctatc.c | 17 +++++++++-------- sound/pci/ctxfi/ctdrv.h | 30 ------------------------------ sound/pci/ctxfi/xfi.c | 6 +++--- 4 files changed, 19 insertions(+), 41 deletions(-) delete mode 100644 sound/pci/ctxfi/ctdrv.h diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 06ba90c..6191531 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -1314,6 +1314,13 @@ #define PCI_VENDOR_ID_CREATIVE 0x1102 /* duplicate: ECTIVA */ #define PCI_DEVICE_ID_CREATIVE_EMU10K1 0x0002 +#define PCI_DEVICE_ID_CREATIVE_20K1 0x0005 +#define PCI_DEVICE_ID_CREATIVE_20K2 0x000b +#define PCI_SUBDEVICE_ID_CREATIVE_SB0760 0x0024 +#define PCI_SUBDEVICE_ID_CREATIVE_SB08801 0x0041 +#define PCI_SUBDEVICE_ID_CREATIVE_SB08802 0x0042 +#define PCI_SUBDEVICE_ID_CREATIVE_SB08803 0x0043 +#define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX 0x6000 #define PCI_VENDOR_ID_ECTIVA 0x1102 /* duplicate: CREATIVE */ #define PCI_DEVICE_ID_ECTIVA_EV1938 0x8938 diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 5f35374..073fe2a 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -18,7 +18,6 @@ #include "ctatc.h" #include "ctpcm.h" #include "ctmixer.h" -#include "ctdrv.h" #include "cthardware.h" #include "ctsrc.h" #include "ctamixer.h" @@ -40,23 +39,25 @@ | ((IEC958_AES3_CON_FS_48000) << 24)) static const struct ct_atc_chip_sub_details atc_sub_details[NUM_CTCARDS] = { - [CTSB0760] = {.subsys = PCI_SUBSYS_CREATIVE_SB0760, + [CTSB0760] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB0760, .nm_model = "SB076x"}, - [CTHENDRIX] = {.subsys = PCI_SUBSYS_CREATIVE_HENDRIX, + [CTHENDRIX] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, .nm_model = "Hendrix"}, - [CTSB08801] = {.subsys = PCI_SUBSYS_CREATIVE_SB08801, + [CTSB08801] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB08801, .nm_model = "SB0880"}, - [CTSB08802] = {.subsys = PCI_SUBSYS_CREATIVE_SB08802, + [CTSB08802] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB08802, .nm_model = "SB0880"}, - [CTSB08803] = {.subsys = PCI_SUBSYS_CREATIVE_SB08803, + [CTSB08803] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB08803, .nm_model = "SB0880"} }; static struct ct_atc_chip_details atc_chip_details[] = { - {.vendor = PCI_VENDOR_CREATIVE, .device = PCI_DEVICE_CREATIVE_20K1, + {.vendor = PCI_VENDOR_ID_CREATIVE, + .device = PCI_DEVICE_ID_CREATIVE_20K1, .sub_details = NULL, .nm_card = "X-Fi 20k1"}, - {.vendor = PCI_VENDOR_CREATIVE, .device = PCI_DEVICE_CREATIVE_20K2, + {.vendor = PCI_VENDOR_ID_CREATIVE, + .device = PCI_DEVICE_ID_CREATIVE_20K2, .sub_details = atc_sub_details, .nm_card = "X-Fi 20k2"}, {} /* terminator */ diff --git a/sound/pci/ctxfi/ctdrv.h b/sound/pci/ctxfi/ctdrv.h deleted file mode 100644 index f776a44..0000000 --- a/sound/pci/ctxfi/ctdrv.h +++ /dev/null @@ -1,30 +0,0 @@ -/** - * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved. - * - * This source file is released under GPL v2 license (no other versions). - * See the COPYING file included in the main directory of this source - * distribution for the license terms and conditions. - * - * @file ctdrv.h - * - * @breaf - * This file contains the definition of card IDs supported by this driver. - * - * @author Liu Chun - * - */ - -#ifndef CTDRV_H -#define CTDRV_H - -#define PCI_VENDOR_CREATIVE 0x1102 -#define PCI_DEVICE_CREATIVE_20K1 0x0005 -#define PCI_DEVICE_CREATIVE_20K2 0x000B -#define PCI_SUBVENDOR_CREATIVE 0x1102 -#define PCI_SUBSYS_CREATIVE_SB0760 0x0024 -#define PCI_SUBSYS_CREATIVE_SB08801 0x0041 -#define PCI_SUBSYS_CREATIVE_SB08802 0x0042 -#define PCI_SUBSYS_CREATIVE_SB08803 0x0043 -#define PCI_SUBSYS_CREATIVE_HENDRIX 0x6000 - -#endif /* CTDRV_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index f911440..b7368fd 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -11,10 +11,10 @@ #include #include #include +#include #include #include #include "ctatc.h" -#include "ctdrv.h" MODULE_AUTHOR("Creative Technology Ltd"); MODULE_DESCRIPTION("X-Fi driver version 1.03"); @@ -32,8 +32,8 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ - { PCI_DEVICE(PCI_VENDOR_CREATIVE, PCI_DEVICE_CREATIVE_20K1) }, - { PCI_DEVICE(PCI_VENDOR_CREATIVE, PCI_DEVICE_CREATIVE_20K2) }, + { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1) }, + { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K2) }, { 0, } }; MODULE_DEVICE_TABLE(pci, ct_pci_dev_ids); -- cgit v1.1 From 35b053becb64eba13f3ea5c8c51023997169ff34 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 15:17:16 +0200 Subject: ALSA: ctxfi - Avoid unneeded pci_read_config_*() calls Use struct pci subsystem_device and revision fields instead of unneeded calls of pci_read_config_*(). Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 073fe2a..f523450 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1190,14 +1190,14 @@ static int atc_dev_free(struct snd_device *dev) static int atc_identify_card(struct ct_atc *atc) { - u16 subsys = 0; - u8 revision = 0; + u16 subsys; + u8 revision; struct pci_dev *pci = atc->pci; const struct ct_atc_chip_details *d; enum CTCARDS i; - pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsys); - pci_read_config_byte(pci, PCI_REVISION_ID, &revision); + subsys = pci->subsystem_device; + revision = pci->revision; atc->chip_details = NULL; atc->model = NUM_CTCARDS; for (d = atc_chip_details; d->vendor; d++) { @@ -1308,7 +1308,7 @@ static int atc_get_resources(struct ct_atc *atc) struct sum_desc sum_dsc = {0}; struct sum_mgr *sum_mgr = NULL; int err = 0, i = 0; - unsigned short subsys_id = 0; + unsigned short subsys_id; atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); if (NULL == atc->daios) @@ -1339,7 +1339,7 @@ static int atc_get_resources(struct ct_atc *atc) } atc->n_daio++; } - pci_read_config_word(atc->pci, PCI_SUBSYSTEM_ID, &subsys_id); + subsys_id = atc->pci->subsystem_device; if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { /* SB073x cards */ da_desc.type = SPDIFI1; -- cgit v1.1 From b3e0afe61e8271a8d082478752a67e5c279c8f23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 15:19:30 +0200 Subject: ALSA: ctxfi - Add prefix to debug prints Added ctxfi: prefix to each debug print. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctamixer.c | 4 ++-- sound/pci/ctxfi/ctatc.c | 16 +++++++++------- sound/pci/ctxfi/cthw20k2.c | 17 +++++++++-------- sound/pci/ctxfi/ctmixer.c | 6 +++--- sound/pci/ctxfi/ctpcm.c | 6 +++--- sound/pci/ctxfi/ctresource.c | 20 ++++++++++++-------- sound/pci/ctxfi/ctsrc.c | 4 ++-- sound/pci/ctxfi/ctvmem.c | 7 ++++--- sound/pci/ctxfi/xfi.c | 15 ++++++++------- 9 files changed, 52 insertions(+), 43 deletions(-) diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index 119791a..859e996 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -259,7 +259,7 @@ static int get_amixer_rsc(struct amixer_mgr *mgr, } spin_unlock_irqrestore(&mgr->mgr_lock, flags); if (err) { - printk(KERN_ERR "Can't meet AMIXER resource request!\n"); + printk(KERN_ERR "ctxfi: Can't meet AMIXER resource request!\n"); goto error; } @@ -413,7 +413,7 @@ static int get_sum_rsc(struct sum_mgr *mgr, } spin_unlock_irqrestore(&mgr->mgr_lock, flags); if (err) { - printk(KERN_ERR "Can't meet SUM resource request!\n"); + printk(KERN_ERR "ctxfi: Can't meet SUM resource request!\n"); goto error; } diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index f523450..ead104e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -191,7 +191,7 @@ static unsigned int convert_format(snd_pcm_format_t snd_format) case SNDRV_PCM_FORMAT_S32_LE: return SRC_SF_S32; default: - printk(KERN_ERR "not recognized snd format is %d \n", + printk(KERN_ERR "ctxfi: not recognized snd format is %d \n", snd_format); return SRC_SF_S16; } @@ -1254,7 +1254,8 @@ static int ct_create_alsa_devs(struct ct_atc *atc) err = alsa_dev_funcs[i].create(atc, i, alsa_dev_funcs[i].public_name); if (err) { - printk(KERN_ERR "Creating alsa device %d failed!\n", i); + printk(KERN_ERR "ctxfi: " + "Creating alsa device %d failed!\n", i); return err; } } @@ -1289,7 +1290,8 @@ static int atc_create_hw_devs(struct ct_atc *atc) err = rsc_mgr_funcs[i].create(atc->hw, &atc->rsc_mgrs[i]); if (err) { - printk(KERN_ERR "Failed to create rsc_mgr %d!!!\n", i); + printk(KERN_ERR "ctxfi: " + "Failed to create rsc_mgr %d!!!\n", i); return err; } } @@ -1333,7 +1335,7 @@ static int atc_get_resources(struct ct_atc *atc) err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); if (err) { - printk(KERN_ERR "Failed to get DAIO " + printk(KERN_ERR "ctxfi: Failed to get DAIO " "resource %d!!!\n", i); return err; } @@ -1349,7 +1351,7 @@ static int atc_get_resources(struct ct_atc *atc) err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); if (err) { - printk(KERN_ERR "Failed to get S/PDIF-in resource!!!\n"); + printk(KERN_ERR "ctxfi: Failed to get S/PDIF-in resource!!!\n"); return err; } atc->n_daio++; @@ -1400,7 +1402,7 @@ static int atc_get_resources(struct ct_atc *atc) err = ct_mixer_create(atc, (struct ct_mixer **)&atc->mixer); if (err) { - printk(KERN_ERR "Failed to create mixer obj!!!\n"); + printk(KERN_ERR "ctxfi: Failed to create mixer obj!!!\n"); return err; } @@ -1600,7 +1602,7 @@ int ct_atc_create(struct snd_card *card, struct pci_dev *pci, error1: ct_atc_destroy(atc); - printk(KERN_ERR "Something wrong!!!\n"); + printk(KERN_ERR "ctxfi: Something wrong!!!\n"); return err; } diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 9c2d38b..7c24c2c 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1144,7 +1144,7 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info) hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x11111111); hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0); } else { - printk(KERN_ALERT "ERROR!!! Invalid sampling rate!!!\n"); + printk(KERN_ALERT "ctxfi: ERROR!!! Invalid sampling rate!!!\n"); return -EINVAL; } @@ -1197,7 +1197,8 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) /* Set up device page table */ if ((~0UL) == info->vm_pgt_phys) { - printk(KERN_ALERT "Wrong device page table page address!!!\n"); + printk(KERN_ALERT "ctxfi: " + "Wrong device page table page address!!!\n"); return -1; } @@ -1314,7 +1315,7 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) break; } if (i >= 1000) { - printk(KERN_ALERT "PLL initialization failed!!!\n"); + printk(KERN_ALERT "ctxfi: PLL initialization failed!!!\n"); return -EBUSY; } @@ -1338,7 +1339,7 @@ static int hw_auto_init(struct hw *hw) break; } if (!get_field(gctl, GCTL_AID)) { - printk(KERN_ALERT "Card Auto-init failed!!!\n"); + printk(KERN_ALERT "ctxfi: Card Auto-init failed!!!\n"); return -EBUSY; } @@ -1762,7 +1763,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) /* Initialize I2C */ err = I2CInit(hw, 0x1A, 1, 1); if (err < 0) { - printk(KERN_ALERT "Failure to acquire I2C!!!\n"); + printk(KERN_ALERT "ctxfi: Failure to acquire I2C!!!\n"); goto error; } @@ -1782,7 +1783,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A), MAKE_WM8775_DATA(0x0A)); } else { - printk(KERN_ALERT "Invalid master sampling " + printk(KERN_ALERT "ctxfi: Invalid master sampling " "rate (msr %d)!!!\n", info->msr); err = -EINVAL; goto error; @@ -1820,7 +1821,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), MAKE_WM8775_DATA(0xCF)); /* No boost */ } else { - printk(KERN_ALERT "ERROR!!! Invalid input mux!!!\n"); + printk(KERN_ALERT "ctxfi: ERROR!!! Invalid input mux!!!\n"); err = -EINVAL; goto error; } @@ -1852,7 +1853,7 @@ static int hw_card_start(struct hw *hw) dma_mask = CT_XFI_DMA_MASK; if (pci_set_dma_mask(pci, dma_mask) < 0 || pci_set_consistent_dma_mask(pci, dma_mask) < 0) { - printk(KERN_ERR "architecture does not support PCI " + printk(KERN_ERR "ctxfi: architecture does not support PCI " "busmaster DMA with mask 0x%x\n", dma_mask); err = -ENXIO; goto error1; diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index c80d692..b795076 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -760,7 +760,7 @@ static int ct_mixer_get_resources(struct ct_mixer *mixer) for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) { err = sum_mgr->get_sum(sum_mgr, &sum_desc, &sum); if (err) { - printk(KERN_ERR "Failed to get sum resources for " + printk(KERN_ERR "ctxfi:Failed to get sum resources for " "front output!\n"); break; } @@ -775,8 +775,8 @@ static int ct_mixer_get_resources(struct ct_mixer *mixer) for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { err = amixer_mgr->get_amixer(amixer_mgr, &am_desc, &amixer); if (err) { - printk(KERN_ERR "Failed to get amixer resources for " - "mixer obj!\n"); + printk(KERN_ERR "ctxfi:Failed to get amixer resources " + "for mixer obj!\n"); break; } mixer->amixers[i] = amixer; diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 73d4fdb..a64cb0e 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -284,7 +284,7 @@ static int ct_pcm_playback_prepare(struct snd_pcm_substream *substream) err = atc->pcm_playback_prepare(atc, apcm); if (err < 0) { - printk(KERN_ERR "Preparing pcm playback failed!!!\n"); + printk(KERN_ERR "ctxfi: Preparing pcm playback failed!!!\n"); return err; } @@ -389,7 +389,7 @@ static int ct_pcm_capture_prepare(struct snd_pcm_substream *substream) err = atc->pcm_capture_prepare(atc, apcm); if (err < 0) { - printk(KERN_ERR "Preparing pcm capture failed!!!\n"); + printk(KERN_ERR "ctxfi: Preparing pcm capture failed!!!\n"); return err; } @@ -477,7 +477,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, err = snd_pcm_new(atc->card, name, device, playback_count, capture_count, &pcm); if (err < 0) { - printk(KERN_ERR "snd_pcm_new failed!! Err=%d\n", err); + printk(KERN_ERR "ctxfi: snd_pcm_new failed!! Err=%d\n", err); return err; } diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 414fc23..da21a71 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -162,12 +162,14 @@ int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw) case DAIO: break; default: - printk(KERN_ERR "Invalid resource type value %d!\n", type); + printk(KERN_ERR + "ctxfi: Invalid resource type value %d!\n", type); return -EINVAL; } if (err) { - printk(KERN_ERR "Failed to get resource control block!\n"); + printk(KERN_ERR + "ctxfi: Failed to get resource control block!\n"); return err; } @@ -190,8 +192,8 @@ int rsc_uninit(struct rsc *rsc) case DAIO: break; default: - printk(KERN_ERR "Invalid resource type value %d!\n", - rsc->type); + printk(KERN_ERR "ctxfi: " + "Invalid resource type value %d!\n", rsc->type); break; } @@ -233,13 +235,15 @@ int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, case SUM: break; default: - printk(KERN_ERR "Invalid resource type value %d!\n", type); + printk(KERN_ERR + "ctxfi: Invalid resource type value %d!\n", type); err = -EINVAL; goto error; } if (err) { - printk(KERN_ERR "Failed to get manager control block!\n"); + printk(KERN_ERR + "ctxfi: Failed to get manager control block!\n"); goto error; } @@ -282,8 +286,8 @@ int rsc_mgr_uninit(struct rsc_mgr *mgr) case SUM: break; default: - printk(KERN_ERR "Invalid resource type value %d!\n", - mgr->type); + printk(KERN_ERR "ctxfi: " + "Invalid resource type value %d!\n", mgr->type); break; } diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index d3e0ad5..77e118c 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -431,7 +431,7 @@ get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc) spin_unlock_irqrestore(&mgr->mgr_lock, flags); if (err) { - printk(KERN_ERR "Can't meet SRC resource request!\n"); + printk(KERN_ERR "ctxfi: Can't meet SRC resource request!\n"); return err; } @@ -740,7 +740,7 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr, } spin_unlock_irqrestore(&mgr->mgr_lock, flags); if (err) { - printk(KERN_ERR "Can't meet SRCIMP resource request!\n"); + printk(KERN_ERR "ctxfi: Can't meet SRCIMP resource request!\n"); goto error1; } diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 46ca04c..cecf77e 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -121,12 +121,13 @@ ct_vm_map(struct ct_vm *vm, void *host_addr, int size) /* do mapping */ if ((unsigned long)host_addr >= VMALLOC_START) { - printk(KERN_ERR "Fail! Not support vmalloced addr now!\n"); + printk(KERN_ERR "ctxfi: " + "Fail! Not support vmalloced addr now!\n"); return NULL; } if (size > vm->size) { - printk(KERN_ERR "Fail! No sufficient device virtural " + printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " "memory space available!\n"); return NULL; } @@ -139,7 +140,7 @@ ct_vm_map(struct ct_vm *vm, void *host_addr, int size) block = get_vm_block(vm, (pages << PAGE_SHIFT)); if (block == NULL) { - printk(KERN_ERR "No virtual memory block that is big " + printk(KERN_ERR "ctxfi: No virtual memory block that is big " "enough to allocate!\n"); return NULL; } diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index b7368fd..19310ed 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -57,16 +57,17 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) if (err) return err; if ((reference_rate != 48000) && (reference_rate != 44100)) { - printk(KERN_ERR "Invalid reference_rate value %u!!!\n" - "The valid values for reference_rate " - "are 48000 and 44100.\nValue 48000 is " - "assumed.\n", reference_rate); + printk(KERN_ERR "ctxfi: Invalid reference_rate value %u!!!\n", + reference_rate); + printk(KERN_ERR "ctxfi: The valid values for reference_rate " + "are 48000 and 44100, Value 48000 is assumed.\n"); reference_rate = 48000; } if ((multiple != 1) && (multiple != 2)) { - printk(KERN_ERR "Invalid multiple value %u!!!\nThe valid " - "values for multiple are 1 and 2.\nValue 2 " - "is assumed.\n", multiple); + printk(KERN_ERR "ctxfi: Invalid multiple value %u!!!\n", + multiple); + printk(KERN_ERR "ctxfi: The valid values for multiple are " + "1 and 2, Value 2 is assumed.\n"); multiple = 2; } err = ct_atc_create(card, pci, reference_rate, multiple, &atc); -- cgit v1.1 From 7442f9dadb8626bd9e7e01445560499560b99f5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 18:20:33 +0200 Subject: ALSA: hda - Add a quirk entry for Macbook Pro 5,1 Added the codec SSID for MacBook Pro 5,1 as compatible as MP51. However, the headphone auto-muting function doesn't work. So, this is just a tentative solution. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 51954ba..1bf9d05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7293,6 +7293,10 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MBP3; break; case 0x106b3f00: /* Macbook 5,1 */ + case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense + * seems not working, so apparently + * no perfect solution yet + */ board_config = ALC885_MB5; break; default: -- cgit v1.1 From b243b77c708665d7af8c5e42611c27c89f918788 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Thu, 14 May 2009 10:25:42 +0200 Subject: ASoC: pxa2xx-i2s: Proper hw initialization Make sure we are in a know good state at end of probe : Reset FIFO logic and registers, and make sure REC and RPL functions along with FIFO service are disabled (SACR0_RST enables REC and RPL). Resetting loses current settings so remove reset from stream startup. Now reset occurs only at probe. Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6014577..bb8630b 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -106,10 +106,8 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); - if (!cpu_dai->active) { - SACR0 |= SACR0_RST; + if (!cpu_dai->active) SACR0 = 0; - } return 0; } @@ -347,6 +345,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) if (ret != 0) clk_put(clk_i2s); + /* + * PXA Developer's Manual: + * If SACR0[ENB] is toggled in the middle of a normal operation, + * the SACR0[RST] bit must also be set and cleared to reset all + * I2S controller registers. + */ + SACR0 = SACR0_RST; + SACR0 = 0; + /* Make sure RPL and REC are disabled */ + SACR1 = SACR1_DRPL | SACR1_DREC; + /* Along with FIFO servicing */ + SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + return ret; } -- cgit v1.1 From 34555c1077ac8f4854e0db9ad11b989a6908d210 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Wed, 13 May 2009 22:16:46 +0200 Subject: ASoC: pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separately - hw_params enables both RPL and REC functions each time : Enable the appropriate function in pxa2xx_i2s_trigger. - pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is off : Turn it off only when both functions are off. Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index bb8630b..115b471 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -176,9 +176,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { - SACR0 = 0; - SACR1 = 0; if (pxa_i2s.master) SACR0 |= SACR0_BCKD; @@ -224,6 +222,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + SACR1 &= ~SACR1_DRPL; + else + SACR1 &= ~SACR1_DREC; SACR0 |= SACR0_ENB; break; case SNDRV_PCM_TRIGGER_RESUME: @@ -250,7 +252,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, SAIMR &= ~SAIMR_RFS; } - if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { + if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); clk_disable(clk_i2s); -- cgit v1.1 From 9bc04fd1677a956fdd7c5645a09de34ca9a8f1a6 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Wed, 13 May 2009 22:16:52 +0200 Subject: ASoC: pxa2xx-i2s: Fix inappropriate release of i2s clock i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown. Now we 'get' i2s_clk at probe and 'put' it at driver removal or when probe fails. Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 115b471..bc12a09 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -257,8 +257,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, pxa_i2s_wait(); clk_disable(clk_i2s); } - - clk_put(clk_i2s); } #ifdef CONFIG_PM -- cgit v1.1 From 916465a841937a60baac6144ae3f41b0d1560f3b Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Wed, 13 May 2009 22:16:59 +0200 Subject: ASoC: pxa2xx-i2s: Fix suspend/resume pxa2xx_i2s_resume is : - unconditionnaly setting SACR0_ENB - unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0 fix these. In pxa2xx_i2s_{resume,suspend}, save/restore registers even when !dai->active. Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index bc12a09..4743e26 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -262,9 +262,6 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - /* store registers */ pxa_i2s.sacr0 = SACR0; pxa_i2s.sacr1 = SACR1; @@ -279,16 +276,14 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai) static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) { - if (!dai->active) - return 0; - pxa_i2s_wait(); - SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; + SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB; SACR1 = pxa_i2s.sacr1; SAIMR = pxa_i2s.saimr; SADIV = pxa_i2s.sadiv; - SACR0 |= SACR0_ENB; + + SACR0 = pxa_i2s.sacr0; return 0; } -- cgit v1.1 From 63c26baa2aa624b023892d66ed696525fc787560 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Thu, 14 May 2009 20:52:46 +0100 Subject: ASoC: Support AC97 link off by default on WM9712 The WM9712 can be configured by resistor strapping GPIO4 to behave like the WM9713 and default to leaving the AC97 link disabled after cold reset until a warm reset occurs. In this configuration we need to issue a warm reset after cold to bring the link up so do so. The warm reset will be harmless on systems that don't need it. [Changelog rewritten to document the reasoning. -- broonie] Signed-off-by: Marek Vasut Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 550c903..1fd4e88 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -585,6 +585,8 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) } soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; -- cgit v1.1 From 2baaec28068d07db3d4ae6ba885fa07255b2ad79 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 May 2009 12:18:47 +0200 Subject: ASoC: Add missing __devexit in wm8940.c Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8940.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index a66dacc..b8e17d6 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -907,7 +907,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, return wm8940_register(wm8940); } -static int wm8940_i2c_remove(struct i2c_client *client) +static int __devexit wm8940_i2c_remove(struct i2c_client *client) { struct wm8940_priv *wm8940 = i2c_get_clientdata(client); -- cgit v1.1 From 2bf2778e0fb38255e55ab5e10022132b0a72420e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 May 2009 12:20:52 +0200 Subject: ASoC: Optimize switch/case in magician.c Use default to optimize the switch/case in magicial_playback_hw_params(), which also fixes the compile warnings below: sound/soc/pxa/magician.c:89: warning: 'acds' may be used uninitialized in this function sound/soc/pxa/magician.c:89: warning: 'acps' may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/soc/pxa/magician.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 0625c34..c89a3cd 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -106,7 +106,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ acds = PXA_SSP_CLK_AUDIO_DIV_16; break; - case 32: + default: /* 32 */ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ acds = PXA_SSP_CLK_AUDIO_DIV_8; } @@ -118,7 +118,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_4; break; - case 32: + default: /* 32 */ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; } @@ -130,7 +130,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; - case 32: + default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } @@ -142,7 +142,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; - case 32: + default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } @@ -154,19 +154,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; - case 32: + default: /* 32 */ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; case 96000: + default: acps = 12235000; switch (width) { case 16: /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; break; - case 32: + default: /* 32 */ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; div4 = PXA_SSP_CLK_SCDB_1; -- cgit v1.1 From 2fc998907947f92b1b9113faad531d7f5a857987 Mon Sep 17 00:00:00 2001 From: Nickolas Lloyd Date: Fri, 15 May 2009 15:33:30 +0200 Subject: ALSA: hda - add controls to toggle DC bias on mic ports This patch adds a mixer control for the STAC92XX boards to control the DC bias of mic ports, allowing recording from both powered and non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack Mode" to switch between Mic, Line In, and Line Out. Signed-off-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 111 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 107 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ecf53f7..0295098 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -634,6 +634,96 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } +static unsigned int stac92xx_vref_set(struct hda_codec *codec, + hda_nid_t nid, unsigned int new_vref) +{ + unsigned int error; + unsigned int pincfg; + pincfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + pincfg &= 0xff; + pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + pincfg |= new_vref; + + if (new_vref == AC_PINCTL_VREF_HIZ) + pincfg |= AC_PINCTL_OUT_EN; + else + pincfg |= AC_PINCTL_IN_EN; + + error = snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg); + if (error < 0) + return error; + else + return 1; +} + +static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int vref; + vref = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + vref &= AC_PINCTL_VREFEN; + return vref; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref; + unsigned int error; + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = AC_PINCTL_VREF_80; + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else + new_vref = AC_PINCTL_VREF_HIZ; + + if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { + error = stac92xx_vref_set(codec, + kcontrol->private_value, new_vref); + return error; + } + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); + if (vref == AC_PINCTL_VREF_80) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 3; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -995,6 +1085,17 @@ static struct hda_verb stac9205_core_init[] = { .private_value = verb_read | (verb_write << 16), \ } +#define DC_BIAS(xname, idx, nid) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = idx, \ + .info = stac92xx_dc_bias_info, \ + .get = stac92xx_dc_bias_get, \ + .put = stac92xx_dc_bias_put, \ + .private_value = nid, \ + } + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -2702,7 +2803,8 @@ enum { STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, - STAC_CTL_WIDGET_CLFE_SWITCH + STAC_CTL_WIDGET_CLFE_SWITCH, + STAC_CTL_WIDGET_DC_BIAS }; static struct snd_kcontrol_new stac92xx_control_templates[] = { @@ -2714,6 +2816,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), + DC_BIAS(NULL, 0, 0), }; /* add dynamic controls */ @@ -3165,9 +3268,9 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Mic as Output Switch", - (spec->mic_switch << 8) | 1); + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, + "Mic Jack Mode", + spec->mic_switch); if (err < 0) return err; } -- cgit v1.1 From b04b4f7862de8d6e8b536853aaf66a6c1bb05cbd Mon Sep 17 00:00:00 2001 From: Alexander Beregalov Date: Fri, 15 May 2009 18:00:38 +0400 Subject: ALSA: parisc/harmony: fix printk format warning Fix this warning: sound/parisc/harmony.c:938: warning: format '%lx' expects type 'long unsigned int', but argument 2 has type 'resource_size_t' Signed-off-by: Alexander Beregalov Signed-off-by: Takashi Iwai --- sound/parisc/harmony.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 6055fd6..63ae0f9 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -935,7 +935,7 @@ snd_harmony_create(struct snd_card *card, h->iobase = ioremap_nocache(padev->hpa.start, HARMONY_SIZE); if (h->iobase == NULL) { printk(KERN_ERR PFX "unable to remap hpa 0x%lx\n", - padev->hpa.start); + (unsigned long)padev->hpa.start); err = -EBUSY; goto free_and_ret; } -- cgit v1.1 From 812a2cca295ee7f56cd1b988a0f93646285c214a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 16 May 2009 10:00:49 +0200 Subject: ALSA: hda - Split codec->name to vendor and chip name strings Split the name string in hda_codec struct to vendor_name and chip_name strings to be stored directly from the preset name. Since mostly only the chip name is referred in many patch_*.c, this results in the reduction of many codes in the end. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 36 ++++++++++++++++--------- sound/pci/hda/hda_codec.h | 3 ++- sound/pci/hda/hda_hwdep.c | 9 ++++--- sound/pci/hda/hda_proc.c | 8 ++++-- sound/pci/hda/patch_realtek.c | 63 ++++++------------------------------------- 5 files changed, 46 insertions(+), 73 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b91f6ed..77385de 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -623,7 +623,10 @@ static int get_codec_name(struct hda_codec *codec) const struct hda_vendor_id *c; const char *vendor = NULL; u16 vendor_id = codec->vendor_id >> 16; - char tmp[16], name[32]; + char tmp[16]; + + if (codec->vendor_name) + goto get_chip_name; for (c = hda_vendor_ids; c->id; c++) { if (c->id == vendor_id) { @@ -635,14 +638,21 @@ static int get_codec_name(struct hda_codec *codec) sprintf(tmp, "Generic %04x", vendor_id); vendor = tmp; } + codec->vendor_name = kstrdup(vendor, GFP_KERNEL); + if (!codec->vendor_name) + return -ENOMEM; + + get_chip_name: + if (codec->chip_name) + return 0; + if (codec->preset && codec->preset->name) - snprintf(name, sizeof(name), "%s %s", vendor, - codec->preset->name); - else - snprintf(name, sizeof(name), "%s ID %x", vendor, - codec->vendor_id & 0xffff); - codec->name = kstrdup(name, GFP_KERNEL); - if (!codec->name) + codec->chip_name = kstrdup(codec->preset->name, GFP_KERNEL); + else { + sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); + codec->chip_name = kstrdup(tmp, GFP_KERNEL); + } + if (!codec->chip_name) return -ENOMEM; return 0; } @@ -848,7 +858,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); - kfree(codec->name); + kfree(codec->vendor_name); + kfree(codec->chip_name); kfree(codec->modelname); kfree(codec->wcaps); kfree(codec); @@ -989,15 +1000,16 @@ int snd_hda_codec_configure(struct hda_codec *codec) int err; codec->preset = find_codec_preset(codec); - if (!codec->name) { + if (!codec->vendor_name || !codec->chip_name) { err = get_codec_name(codec); if (err < 0) return err; } /* audio codec should override the mixer name */ if (codec->afg || !*codec->bus->card->mixername) - strlcpy(codec->bus->card->mixername, codec->name, - sizeof(codec->bus->card->mixername)); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); if (is_generic_config(codec)) { err = snd_hda_parse_generic_codec(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index cd8979c..c5bd40f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -748,7 +748,8 @@ struct hda_codec { /* detected preset */ const struct hda_codec_preset *preset; struct module *owner; - const char *name; /* codec name */ + const char *vendor_name; /* codec vendor name */ + const char *chip_name; /* codec chip name */ const char *modelname; /* model name for preset */ /* set by patch */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 1c57505..6812fbe 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -242,7 +242,8 @@ CODEC_INFO_SHOW(subsystem_id); CODEC_INFO_SHOW(revision_id); CODEC_INFO_SHOW(afg); CODEC_INFO_SHOW(mfg); -CODEC_INFO_STR_SHOW(name); +CODEC_INFO_STR_SHOW(vendor_name); +CODEC_INFO_STR_SHOW(chip_name); CODEC_INFO_STR_SHOW(modelname); #define CODEC_INFO_STORE(type) \ @@ -275,7 +276,8 @@ static ssize_t type##_store(struct device *dev, \ CODEC_INFO_STORE(vendor_id); CODEC_INFO_STORE(subsystem_id); CODEC_INFO_STORE(revision_id); -CODEC_INFO_STR_STORE(name); +CODEC_INFO_STR_STORE(vendor_name); +CODEC_INFO_STR_STORE(chip_name); CODEC_INFO_STR_STORE(modelname); #define CODEC_ACTION_STORE(type) \ @@ -499,7 +501,8 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RW(revision_id), CODEC_ATTR_RO(afg), CODEC_ATTR_RO(mfg), - CODEC_ATTR_RW(name), + CODEC_ATTR_RW(vendor_name), + CODEC_ATTR_RW(chip_name), CODEC_ATTR_RW(modelname), CODEC_ATTR_RW(init_verbs), CODEC_ATTR_RW(hints), diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 93d7499..418c5d1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -466,8 +466,12 @@ static void print_codec_info(struct snd_info_entry *entry, hda_nid_t nid; int i, nodes; - snd_iprintf(buffer, "Codec: %s\n", - codec->name ? codec->name : "Not Set"); + snd_iprintf(buffer, "Codec: "); + if (codec->vendor_name && codec->chip_name) + snd_iprintf(buffer, "%s %s\n", + codec->vendor_name, codec->chip_name); + else + snd_iprintf(buffer, "Not Set\n"); snd_iprintf(buffer, "Address: %d\n", codec->addr); snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1bf9d05..4bc3e7e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -277,13 +277,13 @@ struct alc_spec { */ unsigned int num_init_verbs; - char *stream_name_analog; /* analog PCM stream */ + char stream_name_analog[16]; /* analog PCM stream */ struct hda_pcm_stream *stream_analog_playback; struct hda_pcm_stream *stream_analog_capture; struct hda_pcm_stream *stream_analog_alt_playback; struct hda_pcm_stream *stream_analog_alt_capture; - char *stream_name_digital; /* digital PCM stream */ + char stream_name_digital[16]; /* digital PCM stream */ struct hda_pcm_stream *stream_digital_playback; struct hda_pcm_stream *stream_digital_capture; @@ -3169,7 +3169,10 @@ static int alc_build_pcms(struct hda_codec *codec) if (spec->no_analog) goto skip_analog; + snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog), + "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; + if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) return -EINVAL; @@ -3195,6 +3198,9 @@ static int alc_build_pcms(struct hda_codec *codec) skip_analog: /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { + snprintf(spec->stream_name_digital, + sizeof(spec->stream_name_digital), + "%s Digital", codec->chip_name); codec->num_pcms = 2; codec->slave_dig_outs = spec->multiout.slave_dig_outs; info = spec->pcm_rec + 1; @@ -4432,12 +4438,10 @@ static int patch_alc880(struct hda_codec *codec) if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); - spec->stream_name_analog = "ALC880 Analog"; spec->stream_analog_playback = &alc880_pcm_analog_playback; spec->stream_analog_capture = &alc880_pcm_analog_capture; spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC880 Digital"; spec->stream_digital_playback = &alc880_pcm_digital_playback; spec->stream_digital_capture = &alc880_pcm_digital_capture; @@ -6078,11 +6082,9 @@ static int patch_alc260(struct hda_codec *codec) if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); - spec->stream_name_analog = "ALC260 Analog"; spec->stream_analog_playback = &alc260_pcm_analog_playback; spec->stream_analog_capture = &alc260_pcm_analog_capture; - spec->stream_name_digital = "ALC260 Digital"; spec->stream_digital_playback = &alc260_pcm_digital_playback; spec->stream_digital_capture = &alc260_pcm_digital_capture; @@ -7337,14 +7339,6 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (codec->vendor_id == 0x10ec0885) { - spec->stream_name_analog = "ALC885 Analog"; - spec->stream_name_digital = "ALC885 Digital"; - } else { - spec->stream_name_analog = "ALC882 Analog"; - spec->stream_name_digital = "ALC882 Digital"; - } - spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; /* FIXME: setup DAC5 */ @@ -9232,13 +9226,6 @@ static int patch_alc883(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0888: - if (codec->revision_id == 0x100101) { - spec->stream_name_analog = "ALC1200 Analog"; - spec->stream_name_digital = "ALC1200 Digital"; - } else { - spec->stream_name_analog = "ALC888 Analog"; - spec->stream_name_digital = "ALC888 Digital"; - } if (!spec->num_adc_nids) { spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); spec->adc_nids = alc883_adc_nids; @@ -9249,8 +9236,6 @@ static int patch_alc883(struct hda_codec *codec) spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ break; case 0x10ec0889: - spec->stream_name_analog = "ALC889 Analog"; - spec->stream_name_digital = "ALC889 Digital"; if (!spec->num_adc_nids) { spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); spec->adc_nids = alc889_adc_nids; @@ -9261,8 +9246,6 @@ static int patch_alc883(struct hda_codec *codec) capture */ break; default: - spec->stream_name_analog = "ALC883 Analog"; - spec->stream_name_digital = "ALC883 Digital"; if (!spec->num_adc_nids) { spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); spec->adc_nids = alc883_adc_nids; @@ -11087,11 +11070,9 @@ static int patch_alc262(struct hda_codec *codec) if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); - spec->stream_name_analog = "ALC262 Analog"; spec->stream_analog_playback = &alc262_pcm_analog_playback; spec->stream_analog_capture = &alc262_pcm_analog_capture; - spec->stream_name_digital = "ALC262 Digital"; spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; @@ -12117,14 +12098,6 @@ static int patch_alc268(struct hda_codec *codec) if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); - if (codec->vendor_id == 0x10ec0267) { - spec->stream_name_analog = "ALC267 Analog"; - spec->stream_name_digital = "ALC267 Digital"; - } else { - spec->stream_name_analog = "ALC268 Analog"; - spec->stream_name_digital = "ALC268 Digital"; - } - spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture; @@ -12979,7 +12952,6 @@ static int patch_alc269(struct hda_codec *codec) if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); - spec->stream_name_analog = "ALC269 Analog"; if (codec->subsystem_id == 0x17aa3bf8) { /* Due to a hardware problem on Lenovo Ideadpad, we need to * fix the sample rate of analog I/O to 44.1kHz @@ -12990,7 +12962,6 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_analog_playback = &alc269_pcm_analog_playback; spec->stream_analog_capture = &alc269_pcm_analog_capture; } - spec->stream_name_digital = "ALC269 Digital"; spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; @@ -14080,11 +14051,9 @@ static int patch_alc861(struct hda_codec *codec) if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); - spec->stream_name_analog = "ALC861 Analog"; spec->stream_analog_playback = &alc861_pcm_analog_playback; spec->stream_analog_capture = &alc861_pcm_analog_capture; - spec->stream_name_digital = "ALC861 Digital"; spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; @@ -15007,13 +14976,8 @@ static int patch_alc861vd(struct hda_codec *codec) setup_preset(spec, &alc861vd_presets[board_config]); if (codec->vendor_id == 0x10ec0660) { - spec->stream_name_analog = "ALC660-VD Analog"; - spec->stream_name_digital = "ALC660-VD Digital"; /* always turn on EAPD */ add_verb(spec, alc660vd_eapd_verbs); - } else { - spec->stream_name_analog = "ALC861VD Analog"; - spec->stream_name_digital = "ALC861VD Digital"; } spec->stream_analog_playback = &alc861vd_pcm_analog_playback; @@ -16936,17 +16900,6 @@ static int patch_alc662(struct hda_codec *codec) if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); - if (codec->vendor_id == 0x10ec0663) { - spec->stream_name_analog = "ALC663 Analog"; - spec->stream_name_digital = "ALC663 Digital"; - } else if (codec->vendor_id == 0x10ec0272) { - spec->stream_name_analog = "ALC272 Analog"; - spec->stream_name_digital = "ALC272 Digital"; - } else { - spec->stream_name_analog = "ALC662 Analog"; - spec->stream_name_digital = "ALC662 Digital"; - } - spec->stream_analog_playback = &alc662_pcm_analog_playback; spec->stream_analog_capture = &alc662_pcm_analog_capture; -- cgit v1.1 From 5b740ea975c4ce3da12ac21b56f9e43354ca4327 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 17 May 2009 11:29:21 +0200 Subject: sound: use dev_set_drvdata Eliminate direct accesses to the driver_data field. cf 82ab13b26f15f49be45f15ccc96bfa0b81dfd015 The semantic patch that makes this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // @@ struct device *dev; expression E; type T; @@ - dev->driver_data = (T)E + dev_set_drvdata(dev, E) @@ struct device *dev; type T; @@ - (T)dev->driver_data + dev_get_drvdata(dev) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3711d84..47afaa9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -674,7 +674,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) ssi_private->dev = ssi_info->dev; ssi_private->asynchronous = ssi_info->asynchronous; - ssi_private->dev->driver_data = fsl_ssi_dai; + dev_set_drvdata(ssi_private->dev, fsl_ssi_dai); /* Initialize the the device_attribute structure */ dev_attr->attr.name = "ssi-stats"; -- cgit v1.1 From b7a755a8a145a7e34e735bda9c452317de7a538a Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Sun, 17 May 2009 20:02:31 -0500 Subject: ASoC: TWL4030: Enable/disable voice digital filters Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 2 +- 2 files changed, 38 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index eaf91ab..e4d683d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1629,6 +1629,28 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1665,6 +1687,17 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, return 0; } +static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 0); +} + static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -1673,6 +1706,9 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u8 old_mode, mode; + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 1); + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~(TWL4030_CODECPDZ); @@ -1780,6 +1816,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, + .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 3441115..9668bdf 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -110,7 +110,7 @@ #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 #define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x04 +#define TWL4030_SEL_16K 0x08 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 #define TWL4030_OPTION_1 (1 << 0) -- cgit v1.1 From 6c627f3978bf3059d45713e2e1240b7c43cc85f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 May 2009 12:33:36 +0200 Subject: ALSA: hda - Show the actual chip name in 'unkown model' messages Show the actual chip name in 'unknown model..' info messages for Realtek codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4bc3e7e..62f4d0c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4410,8 +4410,8 @@ static int patch_alc880(struct hda_codec *codec) alc880_models, alc880_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC880, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC880_AUTO; } @@ -6054,8 +6054,9 @@ static int patch_alc260(struct hda_codec *codec) alc260_models, alc260_cfg_tbl); if (board_config < 0) { - snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, " - "trying auto-probe from BIOS...\n"); + snd_printd(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", + codec->chip_name); board_config = ALC260_AUTO; } @@ -7308,8 +7309,9 @@ static int patch_alc882(struct hda_codec *codec) alc_free(codec); return patch_alc883(codec); } - printk(KERN_INFO "hda_codec: Unknown model for ALC882, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", + codec->chip_name); board_config = ALC882_AUTO; } } @@ -9196,8 +9198,8 @@ static int patch_alc883(struct hda_codec *codec) alc883_models, alc883_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC883, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC883_AUTO; } @@ -11040,8 +11042,8 @@ static int patch_alc262(struct hda_codec *codec) alc262_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC262, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC262_AUTO; } @@ -12076,8 +12078,8 @@ static int patch_alc268(struct hda_codec *codec) alc268_cfg_tbl); if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for ALC268, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC268_AUTO; } @@ -12924,8 +12926,8 @@ static int patch_alc269(struct hda_codec *codec) alc269_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC269, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC269_AUTO; } @@ -14023,8 +14025,8 @@ static int patch_alc861(struct hda_codec *codec) alc861_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC861, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC861_AUTO; } @@ -14947,8 +14949,8 @@ static int patch_alc861vd(struct hda_codec *codec) alc861vd_cfg_tbl); if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for ALC660VD/" - "ALC861VD, trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC861VD_AUTO; } @@ -16872,8 +16874,8 @@ static int patch_alc662(struct hda_codec *codec) alc662_models, alc662_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC662, " - "trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: Unknown model for %s, " + "trying auto-probe from BIOS...\n", codec->chip_name); board_config = ALC662_AUTO; } -- cgit v1.1 From 313f6e2d40bd67e394a65e7d64acd0cf9c9d248d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 May 2009 12:40:52 +0200 Subject: ALSA: hda - Avoid conflicts with snd-ctxfi driver The PCI entries of Creative with HD-audio class can be the devices with emu20k1/emu20k2 chips. These are supported better by snd-ctxfi driver. With that driver, the device will mutate from HD-audio to its native class. This patch adds a simple ifdef to avoid the conflict of device probe between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems still OK to be added as it has no emu20kx chip, and is a pure HD-audio device. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 04f19f8..7bb6dd2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2517,10 +2517,19 @@ static struct pci_device_id azx_ids[] = { /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* Creative X-Fi (CA0110-IBG) */ +#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE) + /* the following entry conflicts with snd-ctxfi driver, + * as ctxfi driver mutates from HD-audio to native mode with + * a special command sequence. + */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_GENERIC }, +#else + /* this entry seems still valid -- i.e. without emu20kx chip */ + { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, +#endif /* AMD Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- cgit v1.1 From eb4c41d30ba1f973c8b10be3644571ab997ae0d6 Mon Sep 17 00:00:00 2001 From: Torben Schulz Date: Mon, 18 May 2009 15:02:35 +0200 Subject: ALSA: hda - Improved MacBook 3,1 support This patch adds support for MacBook 3,1 sound by adding a model new "mb31" with the appropriate init verbs, mixers and channel modes to the ALC883 configuration. patch_alc882() and patch_alc883() are modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A) correctly. Signed-off-by: Torben Schulz Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 151 ++++++++++++++++++++++++++- 2 files changed, 147 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 14ed1d1..ebb444d 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -157,6 +157,7 @@ ALC883/888 fujitsu-xa3530 Fujitsu AMILO XA3530 3stack-6ch-intel Intel DG33* boards asus-p5q ASUS P5Q-EM boards + mb31 MacBook 3,1 auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62f4d0c..b942019 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -240,6 +240,7 @@ enum { ALC883_3ST_6ch_INTEL, ALC888_ASUS_M90V, ALC888_ASUS_EEE1601, + ALC889A_MB31, ALC1200_ASUS_P5Q, ALC883_AUTO, ALC883_MODEL_LAST, @@ -7291,7 +7292,7 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ - case 0x106b3600: /* Macbook 3.1 */ + /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */ case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; @@ -7493,6 +7494,17 @@ static struct hda_input_mux alc883_asus_eee1601_capture_source = { }, }; +static struct hda_input_mux alc889A_mb31_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + /* Front Mic (0x01) unused */ + { "Line", 0x2 }, + /* Line 2 (0x03) unused */ + /* CD (0x04) unsused? */ + }, +}; + /* * 2ch mode */ @@ -7611,6 +7623,49 @@ static struct hda_channel_mode alc883_sixstack_modes[2] = { { 8, alc883_sixstack_ch8_init }, }; +/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ +static struct hda_verb alc889A_mb31_ch2_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ + { } /* end */ +}; + +/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ +static struct hda_verb alc889A_mb31_ch4_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ + { } /* end */ +}; + +/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ +static struct hda_verb alc889A_mb31_ch5_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ + { } /* end */ +}; + +/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ +static struct hda_verb alc889A_mb31_ch6_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ + { } /* end */ +}; + +static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { + { 2, alc889A_mb31_ch2_init }, + { 4, alc889A_mb31_ch4_init }, + { 5, alc889A_mb31_ch5_init }, + { 6, alc889A_mb31_ch6_init }, +}; + static struct hda_verb alc883_medion_eapd_verbs[] = { /* eanable EAPD on medion laptop */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -7889,6 +7944,32 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc889A_mb31_mixer[] = { + /* Output mixers */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00, + HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT), + /* Output switches */ + HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), + /* Boost mixers */ + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT), + /* Input mixers */ + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + static struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -8561,6 +8642,42 @@ static void alc883_eee1601_inithook(struct hda_codec *codec) alc_automute_pin(codec); } +static struct hda_verb alc889A_mb31_verbs[] = { + /* Init rear pin (used as headphone output) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Init line pin (used as output in 4ch and 6ch mode) */ + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */ + /* Init line 2 pin (used as headphone out by default) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */ + { } /* end */ +}; + +/* Mute speakers according to the headphone jack state */ +static void alc889A_mb31_automute(struct hda_codec *codec) +{ + unsigned int present; + + /* Mute only in 2ch or 4ch mode */ + if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) + == 0x00) { + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + } +} + +static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc889A_mb31_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc883_loopbacks alc880_loopbacks #endif @@ -8601,6 +8718,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC1200_ASUS_P5Q] = "asus-p5q", + [ALC889A_MB31] = "mb31", [ALC883_AUTO] = "auto", }; @@ -9052,6 +9170,21 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC889A_MB31] = { + .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, + .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, + alc880_gpio1_init_verbs }, + .adc_nids = alc883_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .dac_nids = alc883_dac_nids, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .channel_mode = alc889A_mb31_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes), + .input_mux = &alc889A_mb31_capture_source, + .dig_out_nid = ALC883_DIGOUT_NID, + .unsol_event = alc889A_mb31_unsol_event, + .init_hook = alc889A_mb31_automute, + }, }; @@ -9197,10 +9330,18 @@ static int patch_alc883(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, alc883_models, alc883_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); - board_config = ALC883_AUTO; + if (board_config < 0 || board_config >= ALC883_MODEL_LAST) { + /* Pick up systems that don't supply PCI SSID */ + switch (codec->subsystem_id) { + case 0x106b3600: /* Macbook 3.1 */ + board_config = ALC889A_MB31; + break; + default: + printk(KERN_INFO + "hda_codec: Unknown model for %s, trying " + "auto-probe from BIOS...\n", codec->chip_name); + board_config = ALC883_AUTO; + } } if (board_config == ALC883_AUTO) { -- cgit v1.1 From 6d3ddc81f5762d54ce7d1db70eb757c6c12fabbc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 16 May 2009 17:47:29 +0100 Subject: ASoC: Split DAPM power checks from sequencing of power changes DAPM has always applied any changes to the power state of widgets as soon as it has determined that they are required. Instead of doing this store all the changes that are required on lists of widgets to power up and down, then iterate over those lists and apply the changes. This changes the sequence in which changes are implemented, doing all power downs before power ups and always using the up/down sequences (previously they were only used when changes were due to DAC/ADC power events). The error handling is also changed so that we continue attempting to power widgets if some changes fail. The main benefit of this is to allow future changes to do optimisations over the whole power sequence and to reduce the number of walks of the widget graph required to check the power status of widgets. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 ++ include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 81 +++++++++++++++++++++++++++++++++--------------- 3 files changed, 61 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 533f9f2..b3f789d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -385,6 +385,9 @@ struct snd_soc_dapm_widget { /* widget input and outputs */ struct list_head sources; struct list_head sinks; + + /* used during DAPM updates */ + struct list_head power_list; }; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 6ab80bf..8309ce8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -372,6 +372,8 @@ struct snd_soc_codec { enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; + struct list_head up_list; + struct list_head down_list; /* codec DAI's */ struct snd_soc_dai *dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7847f80..04ef841 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -658,7 +658,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) static int dapm_power_widget(struct snd_soc_codec *codec, int event, struct snd_soc_dapm_widget *w) { - int power, ret; + int ret; switch (w->id) { case snd_soc_dapm_pre: @@ -696,18 +696,8 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, return 0; default: - break; + return dapm_generic_apply_power(w); } - - if (!w->power_check) - return 0; - - power = w->power_check(w); - if (w->power == power) - return 0; - w->power = power; - - return dapm_generic_apply_power(w); } /* @@ -722,27 +712,68 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { struct snd_soc_dapm_widget *w; - int i, c = 1, *seq = NULL, ret = 0; - - /* do we have a sequenced stream event */ - if (event == SND_SOC_DAPM_STREAM_START) { - c = ARRAY_SIZE(dapm_up_seq); - seq = dapm_up_seq; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - c = ARRAY_SIZE(dapm_down_seq); - seq = dapm_down_seq; + int ret = 0; + int i, power; + + INIT_LIST_HEAD(&codec->up_list); + INIT_LIST_HEAD(&codec->down_list); + + /* Check which widgets we need to power and store them in + * lists indicating if they should be powered up or down. + */ + list_for_each_entry(w, &codec->dapm_widgets, list) { + switch (w->id) { + case snd_soc_dapm_pre: + list_add_tail(&codec->down_list, &w->power_list); + break; + case snd_soc_dapm_post: + list_add_tail(&codec->up_list, &w->power_list); + break; + + default: + if (!w->power_check) + continue; + + power = w->power_check(w); + if (w->power == power) + continue; + + if (power) + list_add_tail(&w->power_list, &codec->up_list); + else + list_add_tail(&w->power_list, + &codec->down_list); + + w->power = power; + break; + } } - for (i = 0; i < c; i++) { - list_for_each_entry(w, &codec->dapm_widgets, list) { + /* Power down widgets first; try to avoid amplifying pops. */ + for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) { + list_for_each_entry(w, &codec->down_list, power_list) { + /* is widget in stream order */ + if (w->id != dapm_down_seq[i]) + continue; + + ret = dapm_power_widget(codec, event, w); + if (ret != 0) + pr_err("Failed to power down %s: %d\n", + w->name, ret); + } + } + /* Now power up. */ + for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) { + list_for_each_entry(w, &codec->up_list, power_list) { /* is widget in stream order */ - if (seq && seq[i] && w->id != seq[i]) + if (w->id != dapm_up_seq[i]) continue; ret = dapm_power_widget(codec, event, w); if (ret != 0) - return ret; + pr_err("Failed to power up %s: %d\n", + w->name, ret); } } -- cgit v1.1 From aef908434cd24dd5529065bf5d781773fad21125 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 16 May 2009 17:53:16 +0100 Subject: ASoC: Make DAPM sysfs entries non-optional sysfs is so standard these days there's no point. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 04ef841..d130602 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -67,10 +67,6 @@ static int dapm_down_seq[] = { snd_soc_dapm_post }; -static int dapm_status = 1; -module_param(dapm_status, int, 0); -MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); - static void pop_wait(u32 pop_time) { if (pop_time) @@ -974,16 +970,12 @@ static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) { - if (!dapm_status) - return 0; return device_create_file(dev, &dev_attr_dapm_widget); } static void snd_soc_dapm_sys_remove(struct device *dev) { - if (dapm_status) { - device_remove_file(dev, &dev_attr_dapm_widget); - } + device_remove_file(dev, &dev_attr_dapm_widget); } /* free all dapm widgets and resources */ -- cgit v1.1 From 452c5eaa0d5162e02ffee742ea17540887bc2904 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 17 May 2009 21:41:23 +0100 Subject: ASoC: Integrate bias management with DAPM power management Rather than managing the bias level of the system based on if there is an active audio stream manage it based on there being an active DAPM widget. This simplifies the code a little, moving the power handling into one place, and improves audio performance for bypass paths when no playbacks or captures are active. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 -- include/sound/soc.h | 1 + sound/soc/soc-core.c | 61 +++++++------------------------------ sound/soc/soc-dapm.c | 78 +++++++++++++++++++++++++++++++++--------------- 4 files changed, 65 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index b3f789d..ec8a45f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,8 +279,6 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); -int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, - enum snd_soc_bias_level level); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index 8309ce8..2af3213 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -339,6 +339,7 @@ struct snd_soc_codec { struct module *owner; struct mutex mutex; struct device *dev; + struct snd_soc_device *socdev; struct list_head list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c0e7066..4aa8e2d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -299,7 +299,6 @@ static void close_delayed_work(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, delayed_work.work); - struct snd_soc_device *socdev = card->socdev; struct snd_soc_codec *codec = card->codec; struct snd_soc_dai *codec_dai; int i; @@ -315,27 +314,10 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { - - /* Reduce power if no longer active */ - if (codec->active == 0) { - pr_debug("pop wq D1 %s %s\n", codec->name, - codec_dai->playback.stream_name); - snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_PREPARE); - } - codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP); - - /* Fall into standby if no longer active */ - if (codec->active == 0) { - pr_debug("pop wq D3 %s %s\n", codec->name, - codec_dai->playback.stream_name); - snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_STANDBY); - } } } mutex_unlock(&pcm_mutex); @@ -399,10 +381,6 @@ static int soc_codec_close(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP); - - if (codec->active == 0 && codec_dai->pop_wait == 0) - snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_STANDBY); } mutex_unlock(&pcm_mutex); @@ -467,36 +445,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&card->delayed_work); } - /* do we need to power up codec */ - if (codec->bias_level != SND_SOC_BIAS_ON) { - snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_PREPARE); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, - codec_dai->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - snd_soc_dai_digital_mute(codec_dai, 0); - - } else { - /* codec already powered - power on widgets */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, - codec_dai->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, - SND_SOC_DAPM_STREAM_START); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(codec, + codec_dai->playback.stream_name, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(codec, + codec_dai->capture.stream_name, + SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); - } + snd_soc_dai_digital_mute(codec_dai, 0); out: mutex_unlock(&pcm_mutex); @@ -1372,6 +1330,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) return ret; } + codec->socdev = socdev; codec->card->dev = socdev->dev; codec->card->private_data = codec; strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d130602..4ca5e56 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -94,6 +94,30 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +/** + * snd_soc_dapm_set_bias_level - set the bias level for the system + * @socdev: audio device + * @level: level to configure + * + * Configure the bias (power) levels for the SoC audio device. + * + * Returns 0 for success else error. + */ +static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level) +{ + struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = socdev->card->codec; + int ret = 0; + + if (card->set_bias_level) + ret = card->set_bias_level(card, level); + if (ret == 0 && codec->set_bias_level) + ret = codec->set_bias_level(codec, level); + + return ret; +} + /* set up initial codec paths */ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *p, int i) @@ -707,9 +731,11 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, */ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { + struct snd_soc_device *socdev = codec->socdev; struct snd_soc_dapm_widget *w; int ret = 0; int i, power; + int sys_power = 0; INIT_LIST_HEAD(&codec->up_list); INIT_LIST_HEAD(&codec->down_list); @@ -731,6 +757,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; power = w->power_check(w); + if (power) + sys_power = 1; + if (w->power == power) continue; @@ -745,6 +774,15 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } } + /* If we're changing to all on or all off then prepare */ + if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || + (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); + if (ret != 0) + pr_err("Failed to prepare bias: %d\n", ret); + } + /* Power down widgets first; try to avoid amplifying pops. */ for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) { list_for_each_entry(w, &codec->down_list, power_list) { @@ -773,6 +811,22 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } } + /* If we just powered the last thing off drop to standby bias */ + if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to apply standby bias: %d\n", ret); + } + + /* If we just powered up then move to active bias */ + if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_ON); + if (ret != 0) + pr_err("Failed to apply active bias: %d\n", ret); + } + return 0; } @@ -1721,30 +1775,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** - * snd_soc_dapm_set_bias_level - set the bias level for the system - * @socdev: audio device - * @level: level to configure - * - * Configure the bias (power) levels for the SoC audio device. - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, - enum snd_soc_bias_level level) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = socdev->card->codec; - int ret = 0; - - if (card->set_bias_level) - ret = card->set_bias_level(card, level); - if (ret == 0 && codec->set_bias_level) - ret = codec->set_bias_level(codec, level); - - return ret; -} - -/** * snd_soc_dapm_enable_pin - enable pin. * @codec: SoC codec * @pin: pin name -- cgit v1.1 From f83fba8baab9e95fff0fe2be0e1e32a1650bdd7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 18 May 2009 15:44:43 +0100 Subject: ASoC: Add debug trace for bias level transitions A standard way of making sure we know when the bias level changes. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4ca5e56..39a63f9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -110,6 +110,24 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; + switch (level) { + case SND_SOC_BIAS_ON: + dev_dbg(socdev->dev, "Setting full bias\n"); + break; + case SND_SOC_BIAS_PREPARE: + dev_dbg(socdev->dev, "Setting bias prepare\n"); + break; + case SND_SOC_BIAS_STANDBY: + dev_dbg(socdev->dev, "Setting standby bias\n"); + break; + case SND_SOC_BIAS_OFF: + dev_dbg(socdev->dev, "Setting bias off\n"); + break; + default: + dev_err(socdev->dev, "Setting invalid bias %d\n", level); + return -EINVAL; + } + if (card->set_bias_level) ret = card->set_bias_level(card, level); if (ret == 0 && codec->set_bias_level) -- cgit v1.1 From 181da78cd048ce866b05a2e0208ea09d2f80e721 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 19 May 2009 10:51:03 +0300 Subject: ASoC: TWL4030: Fix Analog capture path for AUXR AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d683d..abf6914 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -432,7 +432,7 @@ static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { /* Right analog microphone selection */ static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), - SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0), + SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0), }; /* TX1 L/R Analog/Digital microphone selection */ -- cgit v1.1 From b74bd40fa4ae018898c8a6429c2a7daf61516524 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Mon, 18 May 2009 11:52:55 -0500 Subject: ASoC: TWL4030: Add control for selecting codec operation mode Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujflausi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index abf6914..731534c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -814,6 +814,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +/* Codec operation modes */ +static const char *twl4030_op_modes_texts[] = { + "Option 2 (voice/audio)", "Option 1 (audio)" +}; + +static const struct soc_enum twl4030_op_modes_enum = + SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, + ARRAY_SIZE(twl4030_op_modes_texts), + twl4030_op_modes_texts); + +int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl4030_priv *twl4030 = codec->private_data; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val; + unsigned short mask, bitmask; + + if (twl4030->configured) { + printk(KERN_ERR "twl4030 operation mode cannot be " + "changed on-the-fly\n"); + return -EBUSY; + } + + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) + ; + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + + val = ucontrol->value.enumerated.item[0] << e->shift_l; + mask = (bitmask - 1) << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= ucontrol->value.enumerated.item[1] << e->shift_r; + mask |= (bitmask - 1) << e->shift_r; + } + + return snd_soc_update_bits(codec, e->reg, mask, val); +} + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -895,6 +937,11 @@ static const struct soc_enum twl4030_vibradir_enum = twl4030_vibradir_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + /* Codec operation mode control */ + SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, + snd_soc_get_enum_double, + snd_soc_put_twl4030_opmode_enum_double), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, -- cgit v1.1 From 11a728110633320d95935a1ba79c038db303596f Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Mon, 18 May 2009 11:53:04 -0500 Subject: ASoC: SDP3430: Connect twl4030 voice DAI to McBSP3 Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver. Voice DAI init function enables corresponding interface by writting directly to VOICE_IF codec register. Signed-off-by: Misael Lopez Cruz Acked-by: Peter Ujflausi Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 87 ++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 77 insertions(+), 10 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 1c79741..19966a7 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -84,6 +84,49 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFM); + if (ret) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_voice_ops = { + .hw_params = sdp3430_hw_voice_params, +}; + /* Headset jack */ static struct snd_soc_jack hs_jack; @@ -192,22 +235,45 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +static int sdp3430_twl4030_voice_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Enable voice interface */ + reg = codec->read(codec, TWL4030_REG_VOICE_IF); + reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; + codec->write(codec, TWL4030_REG_VOICE_IF, reg); + + return 0; +} + + /* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sdp3430_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], - .init = sdp3430_twl4030_init, - .ops = &sdp3430_ops, +static struct snd_soc_dai_link sdp3430_dai[] = { + { + .name = "TWL4030 I2S", + .stream_name = "TWL4030 Audio", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .init = sdp3430_twl4030_init, + .ops = &sdp3430_ops, + }, + { + .name = "TWL4030 PCM", + .stream_name = "TWL4030 Voice", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE], + .init = sdp3430_twl4030_voice_init, + .ops = &sdp3430_voice_ops, + }, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", .platform = &omap_soc_platform, - .dai_link = &sdp3430_dai, - .num_links = 1, + .dai_link = sdp3430_dai, + .num_links = ARRAY_SIZE(sdp3430_dai), }; /* Audio subsystem */ @@ -236,7 +302,8 @@ static int __init sdp3430_soc_init(void) platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; - *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ ret = platform_device_add(sdp3430_snd_device); if (ret) -- cgit v1.1 From 4abc1cc2f9fe4b6bb3acc1d78e2c15af47b8133d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 May 2009 12:16:46 +0200 Subject: ALSA: hda - Add prefix to kernel messages Add proper prefix to each kernel message in hda_intel.c. Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used together with snd_print*(). Reference: bko#13207 http://bugzilla.kernel.org/show_bug.cgi?id=13207 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7bb6dd2..49fd973 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -128,8 +128,11 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{ULI, M5461}}"); MODULE_DESCRIPTION("Intel HDA driver"); +#ifdef CONFIG_SND_VERBOSE_PRINTK +#define SFX /* nop */ +#else #define SFX "hda-intel: " - +#endif /* * registers @@ -620,7 +623,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) } if (chip->msi) { - snd_printk(KERN_WARNING "hda_intel: No response from codec, " + snd_printk(KERN_WARNING SFX "No response from codec, " "disabling MSI: last cmd=0x%08x\n", chip->last_cmd); free_irq(chip->irq, chip); chip->irq = -1; @@ -634,7 +637,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) } if (!chip->polling_mode) { - snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, " + snd_printk(KERN_WARNING SFX "azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", chip->last_cmd); chip->polling_mode = 1; @@ -649,7 +652,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) return -1; } - snd_printk(KERN_ERR "hda_intel: azx_get_response timeout (ERROR): " + snd_printk(KERN_ERR SFX "azx_get_response timeout (ERROR): " "last cmd=0x%08x\n", chip->last_cmd); spin_lock_irq(&chip->reg_lock); chip->rirb.cmds = 0; /* reset the index */ @@ -776,7 +779,7 @@ static int azx_reset(struct azx *chip) /* check to see if controller is ready */ if (!azx_readb(chip, GCTL)) { - snd_printd("azx_reset: controller not ready!\n"); + snd_printd(SFX "azx_reset: controller not ready!\n"); return -EBUSY; } @@ -786,7 +789,7 @@ static int azx_reset(struct azx *chip) /* detect codecs */ if (!chip->codec_mask) { chip->codec_mask = azx_readw(chip, STATESTS); - snd_printdd("codec_mask = 0x%x\n", chip->codec_mask); + snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask); } return 0; @@ -954,12 +957,12 @@ static void azx_init_pci(struct azx *chip) case AZX_DRIVER_SCH: pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) { - pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, \ + pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP)); pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); - snd_printdd("HDA snoop disabled, enabling ... %s\n",\ - (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) \ + snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n", + (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) ? "Failed" : "OK"); } break; @@ -1099,7 +1102,7 @@ static int azx_setup_periods(struct azx *chip, pos_align; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { - snd_printk(KERN_WARNING "Too big adjustment %d\n", + snd_printk(KERN_WARNING SFX "Too big adjustment %d\n", bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { @@ -1123,7 +1126,7 @@ static int azx_setup_periods(struct azx *chip, return 0; error: - snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", + snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); return -EINVAL; } @@ -1216,7 +1219,7 @@ static int probe_codec(struct azx *chip, int addr) chip->probing = 0; if (res == -1) return -EIO; - snd_printdd("hda_intel: codec #%d probed OK\n", addr); + snd_printdd(SFX "codec #%d probed OK\n", addr); return 0; } @@ -1271,8 +1274,8 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* Some BIOSen give you wrong codec addresses * that don't exist */ - snd_printk(KERN_WARNING - "hda_intel: Codec #%d probe error; " + snd_printk(KERN_WARNING SFX + "Codec #%d probe error; " "disabling it...\n", c); chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing @@ -1488,7 +1491,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) bufsize = snd_pcm_lib_buffer_bytes(substream); period_bytes = snd_pcm_lib_period_bytes(substream); - snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", + snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", bufsize, format_val); if (bufsize != azx_dev->bufsize || @@ -2265,7 +2268,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, synchronize_irq(chip->irq); gcap = azx_readw(chip, GCAP); - snd_printdd("chipset global capabilities = 0x%x\n", gcap); + snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); /* ATI chips seems buggy about 64bit DMA addresses */ if (chip->driver_type == AZX_DRIVER_ATI) @@ -2309,7 +2312,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); if (!chip->azx_dev) { - snd_printk(KERN_ERR "cannot malloc azx_dev\n"); + snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n"); goto errout; } -- cgit v1.1 From fa7979663190240b838ab8c8bad7f59e618bf77c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 May 2009 12:50:04 +0200 Subject: ALSA: hda - Fix digital beep tone calculation The digital beep tone is calculated in two different ways depending on the codec chip. The standard one is using a divider, and another one is a linear tone for IDT/STAC codecs. Currently, only the latter type is used for all codecs, which resulted in a wrong tone pitch. This patch adds the calculation of the standard HD-audio type. Also clean-up the fields in hda_beep struct. Reference: bko#13162 http://bugzilla.kernel.org/show_bug.cgi?id=13162 Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 55 +++++++++++++++++++++++++++++++++--------- sound/pci/hda/hda_beep.h | 5 ++-- sound/pci/hda/patch_sigmatel.c | 2 ++ 3 files changed, 49 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 4de5bac..29272f2 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -45,6 +45,46 @@ static void snd_hda_generate_beep(struct work_struct *work) AC_VERB_SET_BEEP_CONTROL, beep->tone); } +/* (non-standard) Linear beep tone calculation for IDT/STAC codecs + * + * The tone frequency of beep generator on IDT/STAC codecs is + * defined from the 8bit tone parameter, in Hz, + * freq = 48000 * (257 - tone) / 1024 + * that is from 12kHz to 93.75kHz in step of 46.875 hz + */ +static int beep_linear_tone(struct hda_beep *beep, int hz) +{ + hz *= 1000; /* fixed point */ + hz = hz - DIGBEEP_HZ_MIN; + if (hz < 0) + hz = 0; /* turn off PC beep*/ + else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) + hz = 0xff; + else { + hz /= DIGBEEP_HZ_STEP; + hz++; + } + return hz; +} + +/* HD-audio standard beep tone parameter calculation + * + * The tone frequency in Hz is calculated as + * freq = 48000 / (tone * 4) + * from 47Hz to 12kHz + */ +static int beep_standard_tone(struct hda_beep *beep, int hz) +{ + if (hz <= 0) + return 0; /* disabled */ + hz = 12000 / hz; + if (hz > 0xff) + return 0xff; + if (hz <= 0) + return 1; + return hz; +} + static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, unsigned int code, int hz) { @@ -55,21 +95,14 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, if (hz) hz = 1000; case SND_TONE: - hz *= 1000; /* fixed point */ - hz = hz - DIGBEEP_HZ_MIN; - if (hz < 0) - hz = 0; /* turn off PC beep*/ - else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) - hz = 0xff; - else { - hz /= DIGBEEP_HZ_STEP; - hz++; - } + if (beep->linear_tone) + beep->tone = beep_linear_tone(beep, hz); + else + beep->tone = beep_standard_tone(beep, hz); break; default: return -1; } - beep->tone = hz; /* schedule beep event */ schedule_work(&beep->beep_work); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 51bf6a5..0c3de78 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -30,8 +30,9 @@ struct hda_beep { struct hda_codec *codec; char phys[32]; int tone; - int nid; - int enabled; + hda_nid_t nid; + unsigned int enabled:1; + unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ struct work_struct beep_work; /* scheduled task for beep event */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0295098..1731081 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3737,6 +3737,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = snd_hda_attach_beep_device(codec, nid); if (err < 0) return err; + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = 1; /* if no beep switch is available, make its own one */ caps = query_amp_caps(codec, nid, HDA_OUTPUT); if (codec->beep && -- cgit v1.1 From e24805dd85283ac0912b9c400768a4d171b400ff Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Tue, 19 May 2009 22:12:15 +0900 Subject: ASoC: Add TXx9 AC link controller driver (v3) This patch adds support for the integrated ACLC of the TXx9 family. Signed-off-by: Atsushi Nemoto Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/txx9/Kconfig | 29 +++ sound/soc/txx9/Makefile | 11 + sound/soc/txx9/txx9aclc-ac97.c | 255 ++++++++++++++++++++++ sound/soc/txx9/txx9aclc-generic.c | 98 +++++++++ sound/soc/txx9/txx9aclc.c | 430 ++++++++++++++++++++++++++++++++++++++ sound/soc/txx9/txx9aclc.h | 83 ++++++++ 8 files changed, 908 insertions(+) create mode 100644 sound/soc/txx9/Kconfig create mode 100644 sound/soc/txx9/Makefile create mode 100644 sound/soc/txx9/txx9aclc-ac97.c create mode 100644 sound/soc/txx9/txx9aclc-generic.c create mode 100644 sound/soc/txx9/txx9aclc.c create mode 100644 sound/soc/txx9/txx9aclc.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3304f9d..d3e786a 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -34,6 +34,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/txx9/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8943a14..6f1e28d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -12,3 +12,4 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig new file mode 100644 index 0000000..ebc9327 --- /dev/null +++ b/sound/soc/txx9/Kconfig @@ -0,0 +1,29 @@ +## +## TXx9 ACLC +## +config SND_SOC_TXX9ACLC + tristate "SoC Audio for TXx9" + depends on HAS_TXX9_ACLC && TXX9_DMAC + help + This option enables support for the AC Link Controllers in TXx9 SoC. + +config HAS_TXX9_ACLC + bool + +config SND_SOC_TXX9ACLC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_TXX9ACLC_GENERIC + tristate "Generic TXx9 ACLC sound machine" + depends on SND_SOC_TXX9ACLC + select SND_SOC_TXX9ACLC_AC97 + select SND_SOC_AC97_CODEC + help + This is a generic AC97 sound machine for use in TXx9 based systems. diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile new file mode 100644 index 0000000..551f16c --- /dev/null +++ b/sound/soc/txx9/Makefile @@ -0,0 +1,11 @@ +# Platform +snd-soc-txx9aclc-objs := txx9aclc.o +snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o +obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o + +# Machine +snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o + +obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c new file mode 100644 index 0000000..0f83bdb9 --- /dev/null +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -0,0 +1,255 @@ +/* + * TXx9 ACLC AC97 driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "txx9aclc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#ifdef __BIG_ENDIAN +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_BE +#else +#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_LE +#endif + +static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq); + +/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */ +static struct txx9aclc_soc_device *txx9aclc_soc_dev; + +static int txx9aclc_regready(struct txx9aclc_soc_device *dev) +{ + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + + return __raw_readl(drvdata->base + ACINTSTS) & ACINT_REGACCRDY; +} + +/* AC97 controller reads codec register */ +static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + u32 dat; + + if (!(__raw_readl(base + ACINTSTS) & ACINT_CODECRDY(ac97->num))) + return 0xffff; + reg |= ac97->num << 7; + dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ; + __raw_writel(dat, base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); + dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg); + dat = 0xffff; + goto done; + } + dat = __raw_readl(base + ACREGACC); + if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) { + dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n", + dat, reg); + dat = 0xffff; + goto done; + } + dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff; +done: + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); + return dat; +} + +/* AC97 controller writes to codec register */ +static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + + __raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) | + (val << ACREGACC_DAT_SHIFT), + base + ACREGACC); + __raw_writel(ACINT_REGACCRDY, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) { + dev_err(dev->soc_dev.dev, + "ac97 write timeout (reg %#x)\n", reg); + } + __raw_writel(ACINT_REGACCRDY, base + ACINTDIS); +} + +static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct txx9aclc_soc_device *dev = txx9aclc_soc_dev; + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY; + + __raw_writel(ACCTL_ENLINK, base + ACCTLDIS); + mmiowb(); + udelay(1); + __raw_writel(ACCTL_ENLINK, base + ACCTLEN); + /* wait for primary codec ready status */ + __raw_writel(ready, base + ACINTEN); + if (!wait_event_timeout(ac97_waitq, + (__raw_readl(base + ACINTSTS) & ready) == ready, + HZ)) { + dev_err(&ac97->dev, "primary codec is not ready " + "(status %#x)\n", + __raw_readl(base + ACINTSTS)); + } + __raw_writel(ACINT_REGACCRDY, base + ACINTSTS); + __raw_writel(ready, base + ACINTDIS); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = txx9aclc_ac97_read, + .write = txx9aclc_ac97_write, + .reset = txx9aclc_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id) +{ + struct txx9aclc_plat_drvdata *drvdata = dev_id; + void __iomem *base = drvdata->base; + + __raw_writel(__raw_readl(base + ACINTMSTS), base + ACINTDIS); + wake_up(&ac97_waitq); + return IRQ_HANDLED; +} + +static int txx9aclc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + + dev->aclc_pdev = to_platform_device(dai->dev); + txx9aclc_soc_dev = dev; + return 0; +} + +static void txx9aclc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct platform_device *aclc_pdev = to_platform_device(dai->dev); + struct txx9aclc_plat_drvdata *drvdata = platform_get_drvdata(aclc_pdev); + + /* disable AC-link */ + __raw_writel(ACCTL_ENLINK, drvdata->base + ACCTLDIS); + txx9aclc_soc_dev = NULL; +} + +struct snd_soc_dai txx9aclc_ac97_dai = { + .name = "txx9aclc_ac97", + .ac97_control = 1, + .probe = txx9aclc_ac97_probe, + .remove = txx9aclc_ac97_remove, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, +}; +EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai); + +static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev) +{ + struct txx9aclc_plat_drvdata *drvdata; + struct resource *r; + int err; + int irq; + + irq = platform_get_irq(pdev, 0); + if (irq < 0) + return irq; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) + return -EBUSY; + + if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r), + dev_name(&pdev->dev))) + return -EBUSY; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + platform_set_drvdata(pdev, drvdata); + drvdata->physbase = r->start; + if (sizeof(drvdata->physbase) > sizeof(r->start) && + r->start >= TXX9_DIRECTMAP_BASE && + r->start < TXX9_DIRECTMAP_BASE + 0x400000) + drvdata->physbase |= 0xf00000000ull; + drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r)); + if (!drvdata->base) + return -EBUSY; + err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, + IRQF_DISABLED, dev_name(&pdev->dev), drvdata); + if (err < 0) + return err; + + txx9aclc_ac97_dai.dev = &pdev->dev; + return snd_soc_register_dai(&txx9aclc_ac97_dai); +} + +static int __devexit txx9aclc_ac97_dev_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&txx9aclc_ac97_dai); + return 0; +} + +static struct platform_driver txx9aclc_ac97_driver = { + .probe = txx9aclc_ac97_dev_probe, + .remove = __devexit_p(txx9aclc_ac97_dev_remove), + .driver = { + .name = "txx9aclc-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_ac97_init(void) +{ + return platform_driver_register(&txx9aclc_ac97_driver); +} + +static void __exit txx9aclc_ac97_exit(void) +{ + platform_driver_unregister(&txx9aclc_ac97_driver); +} + +module_init(txx9aclc_ac97_init); +module_exit(txx9aclc_ac97_exit); + +MODULE_AUTHOR("Atsushi Nemoto "); +MODULE_DESCRIPTION("TXx9 ACLC AC97 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c new file mode 100644 index 0000000..3175de9 --- /dev/null +++ b/sound/soc/txx9/txx9aclc-generic.c @@ -0,0 +1,98 @@ +/* + * Generic TXx9 ACLC machine driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at ACLC (e.g. RBTX49XX boards). + */ + +#include +#include +#include +#include +#include +#include "../codecs/ac97.h" +#include "txx9aclc.h" + +static struct snd_soc_dai_link txx9aclc_generic_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &txx9aclc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card txx9aclc_generic_card = { + .name = "Generic TXx9 ACLC Audio", + .platform = &txx9aclc_soc_platform, + .dai_link = &txx9aclc_generic_dai, + .num_links = 1, +}; + +static struct txx9aclc_soc_device txx9aclc_generic_soc_device = { + .soc_dev = { + .card = &txx9aclc_generic_card, + .codec_dev = &soc_codec_dev_ac97, + }, +}; + +static int __init txx9aclc_generic_probe(struct platform_device *pdev) +{ + struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device; + struct platform_device *soc_pdev; + int ret; + + soc_pdev = platform_device_alloc("soc-audio", -1); + if (!soc_pdev) + return -ENOMEM; + platform_set_drvdata(soc_pdev, &dev->soc_dev); + dev->soc_dev.dev = &soc_pdev->dev; + ret = platform_device_add(soc_pdev); + if (ret) { + platform_device_put(soc_pdev); + return ret; + } + platform_set_drvdata(pdev, soc_pdev); + return 0; +} + +static int __exit txx9aclc_generic_remove(struct platform_device *pdev) +{ + struct platform_device *soc_pdev = platform_get_drvdata(pdev); + + platform_device_unregister(soc_pdev); + return 0; +} + +static struct platform_driver txx9aclc_generic_driver = { + .remove = txx9aclc_generic_remove, + .driver = { + .name = "txx9aclc-generic", + .owner = THIS_MODULE, + }, +}; + +static int __init txx9aclc_generic_init(void) +{ + return platform_driver_probe(&txx9aclc_generic_driver, + txx9aclc_generic_probe); +} + +static void __exit txx9aclc_generic_exit(void) +{ + platform_driver_unregister(&txx9aclc_generic_driver); +} + +module_init(txx9aclc_generic_init); +module_exit(txx9aclc_generic_exit); + +MODULE_AUTHOR("Atsushi Nemoto "); +MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c new file mode 100644 index 0000000..fa33661 --- /dev/null +++ b/sound/soc/txx9/txx9aclc.c @@ -0,0 +1,430 @@ +/* + * Generic TXx9 ACLC platform driver + * + * Copyright (C) 2009 Atsushi Nemoto + * + * Based on RBTX49xx patch from CELF patch archive. + * (C) Copyright TOSHIBA CORPORATION 2004-2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "txx9aclc.h" + +static const struct snd_pcm_hardware txx9aclc_pcm_hardware = { + /* + * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID + * needs more works for noncoherent MIPS. + */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE, +#ifdef __BIG_ENDIAN + .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif + .period_bytes_min = 1024, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 4096, + .buffer_bytes_max = 32 * 1024, +}; + +static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_device *socdev = rtd->socdev; + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + dev_dbg(socdev->dev, + "runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd " + "runtime->min_align %ld\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + dev_dbg(socdev->dev, + "periods %d period_bytes %d stream %d\n", + params_periods(params), params_period_bytes(params), + substream->stream); + + dmadata->substream = substream; + dmadata->pos = 0; + return 0; +} + +static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct txx9aclc_dmadata *dmadata = runtime->private_data; + + dmadata->dma_addr = runtime->dma_addr; + dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + dmadata->period_bytes = snd_pcm_lib_period_bytes(substream); + + if (dmadata->buffer_bytes == dmadata->period_bytes) { + dmadata->frag_bytes = dmadata->period_bytes >> 1; + dmadata->frags = 2; + } else { + dmadata->frag_bytes = dmadata->period_bytes; + dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes; + } + dmadata->frag_count = 0; + dmadata->pos = 0; + return 0; +} + +static void txx9aclc_dma_complete(void *arg) +{ + struct txx9aclc_dmadata *dmadata = arg; + unsigned long flags; + + /* dma completion handler cannot submit new operations */ + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count >= 0) { + dmadata->dmacount--; + BUG_ON(dmadata->dmacount < 0); + tasklet_schedule(&dmadata->tasklet); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static struct dma_async_tx_descriptor * +txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) +{ + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct scatterlist sg; + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)), + dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1)); + sg_dma_address(&sg) = buf_dma_addr; + desc = chan->device->device_prep_slave_sg(chan, &sg, 1, + dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return NULL; + } + desc->callback = txx9aclc_dma_complete; + desc->callback_param = dmadata; + desc->tx_submit(desc); + return desc; +} + +#define NR_DMA_CHAIN 2 + +static void txx9aclc_dma_tasklet(unsigned long data) +{ + struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data; + struct dma_chan *chan = dmadata->dma_chan; + struct dma_async_tx_descriptor *desc; + struct snd_pcm_substream *substream = dmadata->substream; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + int i; + unsigned long flags; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + if (dmadata->frag_count < 0) { + struct txx9aclc_soc_device *dev = + container_of(dmadata, struct txx9aclc_soc_device, + dmadata[substream->stream]); + struct txx9aclc_plat_drvdata *drvdata = + txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + chan->device->device_terminate_all(chan); + /* first time */ + for (i = 0; i < NR_DMA_CHAIN; i++) { + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + i * dmadata->frag_bytes); + if (!desc) + return; + } + dmadata->dmacount = NR_DMA_CHAIN; + chan->device->device_issue_pending(chan); + spin_lock_irqsave(&dmadata->dma_lock, flags); + __raw_writel(ctlbit, base + ACCTLEN); + dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return; + } + BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN); + while (dmadata->dmacount < NR_DMA_CHAIN) { + dmadata->dmacount++; + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + desc = txx9aclc_dma_submit(dmadata, + dmadata->dma_addr + + dmadata->frag_count * dmadata->frag_bytes); + if (!desc) + return; + chan->device->device_issue_pending(chan); + + spin_lock_irqsave(&dmadata->dma_lock, flags); + dmadata->frag_count++; + dmadata->frag_count %= dmadata->frags; + dmadata->pos += dmadata->frag_bytes; + dmadata->pos %= dmadata->buffer_bytes; + if ((dmadata->frag_count * dmadata->frag_bytes) % + dmadata->period_bytes == 0) + snd_pcm_period_elapsed(substream); + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); +} + +static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + unsigned long flags; + int ret = 0; + u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + ACCTL_AUDODMA : ACCTL_AUDIDMA; + + spin_lock_irqsave(&dmadata->dma_lock, flags); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dmadata->frag_count = -1; + tasklet_schedule(&dmadata->tasklet); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + __raw_writel(ctlbit, base + ACCTLDIS); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + __raw_writel(ctlbit, base + ACCTLEN); + break; + default: + ret = -EINVAL; + } + spin_unlock_irqrestore(&dmadata->dma_lock, flags); + return ret; +} + +static snd_pcm_uframes_t +txx9aclc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + + return bytes_to_frames(substream->runtime, dmadata->pos); +} + +static int txx9aclc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct txx9aclc_soc_device *dev = + container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream]; + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware); + if (ret) + return ret; + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + substream->runtime->private_data = dmadata; + return 0; +} + +static int txx9aclc_pcm_close(struct snd_pcm_substream *substream) +{ + struct txx9aclc_dmadata *dmadata = substream->runtime->private_data; + struct dma_chan *chan = dmadata->dma_chan; + + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + return 0; +} + +static struct snd_pcm_ops txx9aclc_pcm_ops = { + .open = txx9aclc_pcm_open, + .close = txx9aclc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = txx9aclc_pcm_hw_params, + .hw_free = txx9aclc_pcm_hw_free, + .prepare = txx9aclc_pcm_prepare, + .trigger = txx9aclc_pcm_trigger, + .pointer = txx9aclc_pcm_pointer, +}; + +static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, 64 * 1024, 4 * 1024 * 1024); +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct txx9aclc_dmadata *dmadata = param; + char devname[BUS_ID_SIZE + 2]; + + sprintf(devname, "%s.%d", dmadata->dma_res->name, + (int)dmadata->dma_res->start); + if (strcmp(dev_name(chan->device->dev), devname) == 0) { + chan->private = &dmadata->dma_slave; + return true; + } + return false; +} + +static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, + struct txx9aclc_dmadata *dmadata) +{ + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + struct txx9dmac_slave *ds = &dmadata->dma_slave; + dma_cap_mask_t mask; + + spin_lock_init(&dmadata->dma_lock); + + ds->reg_width = sizeof(u32); + if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ds->tx_reg = drvdata->physbase + ACAUDODAT; + ds->rx_reg = 0; + } else { + ds->tx_reg = 0; + ds->rx_reg = drvdata->physbase + ACAUDIDAT; + } + + /* Try to grab a DMA channel */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dmadata->dma_chan = dma_request_channel(mask, filter, dmadata); + if (!dmadata->dma_chan) { + dev_err(dev->soc_dev.dev, + "DMA channel for %s is not available\n", + dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return -EBUSY; + } + tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet, + (unsigned long)dmadata); + return 0; +} + +static int txx9aclc_pcm_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + struct resource *r; + int i; + int ret; + + dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK; + dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE; + for (i = 0; i < 2; i++) { + r = platform_get_resource(dev->aclc_pdev, IORESOURCE_DMA, i); + if (!r) { + ret = -EBUSY; + goto exit; + } + dev->dmadata[i].dma_res = r; + ret = txx9aclc_dma_init(dev, &dev->dmadata[i]); + if (ret) + goto exit; + } + return 0; + +exit: + for (i = 0; i < 2; i++) { + if (dev->dmadata[i].dma_chan) + dma_release_channel(dev->dmadata[i].dma_chan); + dev->dmadata[i].dma_chan = NULL; + } + return ret; +} + +static int txx9aclc_pcm_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct txx9aclc_soc_device *dev = + container_of(socdev, struct txx9aclc_soc_device, soc_dev); + struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev); + void __iomem *base = drvdata->base; + int i; + + /* disable all FIFO DMAs */ + __raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, base + ACCTLDIS); + /* dummy R/W to clear pending DMAREQ if any */ + __raw_writel(__raw_readl(base + ACAUDIDAT), base + ACAUDODAT); + + for (i = 0; i < 2; i++) { + struct txx9aclc_dmadata *dmadata = &dev->dmadata[i]; + struct dma_chan *chan = dmadata->dma_chan; + if (chan) { + dmadata->frag_count = -1; + chan->device->device_terminate_all(chan); + dma_release_channel(chan); + } + dev->dmadata[i].dma_chan = NULL; + } + return 0; +} + +struct snd_soc_platform txx9aclc_soc_platform = { + .name = "txx9aclc-audio", + .probe = txx9aclc_pcm_probe, + .remove = txx9aclc_pcm_remove, + .pcm_ops = &txx9aclc_pcm_ops, + .pcm_new = txx9aclc_pcm_new, + .pcm_free = txx9aclc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(txx9aclc_soc_platform); + +static int __init txx9aclc_soc_platform_init(void) +{ + return snd_soc_register_platform(&txx9aclc_soc_platform); +} + +static void __exit txx9aclc_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&txx9aclc_soc_platform); +} + +module_init(txx9aclc_soc_platform_init); +module_exit(txx9aclc_soc_platform_exit); + +MODULE_AUTHOR("Atsushi Nemoto "); +MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h new file mode 100644 index 0000000..6769aab --- /dev/null +++ b/sound/soc/txx9/txx9aclc.h @@ -0,0 +1,83 @@ +/* + * TXx9 SoC AC Link Controller + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __TXX9ACLC_H +#define __TXX9ACLC_H + +#include +#include + +#define ACCTLEN 0x00 /* control enable */ +#define ACCTLDIS 0x04 /* control disable */ +#define ACCTL_ENLINK 0x00000001 /* enable/disable AC-link */ +#define ACCTL_AUDODMA 0x00000100 /* AUDODMA enable/disable */ +#define ACCTL_AUDIDMA 0x00001000 /* AUDIDMA enable/disable */ +#define ACCTL_AUDOEHLT 0x00010000 /* AUDO error halt + enable/disable */ +#define ACCTL_AUDIEHLT 0x00100000 /* AUDI error halt + enable/disable */ +#define ACREGACC 0x08 /* codec register access */ +#define ACREGACC_DAT_SHIFT 0 /* data field */ +#define ACREGACC_REG_SHIFT 16 /* address field */ +#define ACREGACC_CODECID_SHIFT 24 /* CODEC ID field */ +#define ACREGACC_READ 0x80000000 /* CODEC read */ +#define ACREGACC_WRITE 0x00000000 /* CODEC write */ +#define ACINTSTS 0x10 /* interrupt status */ +#define ACINTMSTS 0x14 /* interrupt masked status */ +#define ACINTEN 0x18 /* interrupt enable */ +#define ACINTDIS 0x1c /* interrupt disable */ +#define ACINT_CODECRDY(n) (0x00000001 << (n)) /* CODECn ready */ +#define ACINT_REGACCRDY 0x00000010 /* ACREGACC ready */ +#define ACINT_AUDOERR 0x00000100 /* AUDO underrun error */ +#define ACINT_AUDIERR 0x00001000 /* AUDI overrun error */ +#define ACDMASTS 0x80 /* DMA request status */ +#define ACDMA_AUDO 0x00000001 /* AUDODMA pending */ +#define ACDMA_AUDI 0x00000010 /* AUDIDMA pending */ +#define ACAUDODAT 0xa0 /* audio out data */ +#define ACAUDIDAT 0xb0 /* audio in data */ +#define ACREVID 0xfc /* revision ID */ + +struct txx9aclc_dmadata { + struct resource *dma_res; + struct txx9dmac_slave dma_slave; + struct dma_chan *dma_chan; + struct tasklet_struct tasklet; + spinlock_t dma_lock; + int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */ + struct snd_pcm_substream *substream; + unsigned long pos; + dma_addr_t dma_addr; + unsigned long buffer_bytes; + unsigned long period_bytes; + unsigned long frag_bytes; + int frags; + int frag_count; + int dmacount; +}; + +struct txx9aclc_plat_drvdata { + void __iomem *base; + u64 physbase; +}; + +struct txx9aclc_soc_device { + struct snd_soc_device soc_dev; + struct platform_device *aclc_pdev; /* for ioresources, drvdata */ + struct txx9aclc_dmadata dmadata[2]; +}; + +static inline struct txx9aclc_plat_drvdata *txx9aclc_get_plat_drvdata( + struct txx9aclc_soc_device *sdev) +{ + return platform_get_drvdata(sdev->aclc_pdev); +} + +extern struct snd_soc_platform txx9aclc_soc_platform; +extern struct snd_soc_dai txx9aclc_ac97_dai; + +#endif /* __TXX9ACLC_H */ -- cgit v1.1 From 4005d39a5f5549f1f6afe88abceed78b2ab225b6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 18 May 2009 16:02:04 +0300 Subject: ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs Restructuring the twl4030 codec's DAPM routing to be able to handle the power sequences correctly. The twl4030 codec internal implementation have this order: DAC -> Analog PGA -> Mixer/Mux While the ASoC framework expects the following order: DAC -> Mixer -> Analog PGA This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and adds two levels of mixer to handle the digital and analog loopback functionality. Now the analog loopback does not powers on any of the DACs. Signed-off-by: Peter Ujfalusi Tested-by: Anuj Aggarwal Tested-by: Jarkko Nikula Tested-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 141 +++++++++++++++++++++++---------------------- 1 file changed, 71 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 731534c..99fe44f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1051,18 +1051,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", SND_SOC_NOPM, 0, 0), - /* Analog PGAs */ - SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, - 0, 0, NULL, 0), - /* Analog bypasses */ SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, &twl4030_dapm_abypassr1_control, bypass_event, @@ -1091,16 +1079,29 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassv_control, bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 1, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 3, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL, - 4, 0, NULL, 0), + /* Digital mixers, power control for the physical DACs */ + SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer", + TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer", + TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0), + + /* Analog mixers, power control for the physical PGAs */ + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", + TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", + TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", + TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", + TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", + TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ @@ -1194,60 +1195,60 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { - {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, - {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, - {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, - {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, - {"Analog Voice Playback Mixer", NULL, "DAC Voice"}, - - {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, - {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, - {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, - {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, - {"VDL_APGA", NULL, "Analog Voice Playback Mixer"}, + {"Digital L1 Playback Mixer", NULL, "DAC Left1"}, + {"Digital R1 Playback Mixer", NULL, "DAC Right1"}, + {"Digital L2 Playback Mixer", NULL, "DAC Left2"}, + {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, + {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, + + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, + {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, + {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, + {"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"}, + {"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mixer", "Voice", "VDL_APGA"}, - {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, - {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, - {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"}, /* PreDrivL */ - {"PredriveL Mixer", "Voice", "VDL_APGA"}, - {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, - {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, - {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"}, /* PreDrivR */ - {"PredriveR Mixer", "Voice", "VDL_APGA"}, - {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, - {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, - {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, + {"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"}, /* HeadsetL */ - {"HeadsetL Mixer", "Voice", "VDL_APGA"}, - {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, - {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, /* HeadsetR */ - {"HeadsetR Mixer", "Voice", "VDL_APGA"}, - {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, - {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, /* CarkitL */ - {"CarkitL Mixer", "Voice", "VDL_APGA"}, - {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, - {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, + {"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, /* CarkitR */ - {"CarkitR Mixer", "Voice", "VDL_APGA"}, - {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, - {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"}, + {"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, /* HandsfreeL */ - {"HandsfreeL Mux", "Voice", "VDL_APGA"}, - {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"}, + {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, + {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, + {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, /* HandsfreeR */ - {"HandsfreeR Mux", "Voice", "VDL_APGA"}, - {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, + {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, + {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, + {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, @@ -1255,8 +1256,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Vibra Mux", "AudioR2", "DAC Right2"}, /* outputs */ - {"OUTL", NULL, "ARXL2_APGA"}, - {"OUTR", NULL, "ARXR2_APGA"}, + {"OUTL", NULL, "Analog L2 Playback Mixer"}, + {"OUTR", NULL, "Analog R2 Playback Mixer"}, {"EARPIECE", NULL, "Earpiece Mixer"}, {"PREDRIVEL", NULL, "PredriveL Mixer"}, {"PREDRIVER", NULL, "PredriveR Mixer"}, @@ -1320,9 +1321,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, {"Voice Digital Loopback", "Volume", "TX2 Capture Route"}, - {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, - {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, - {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"}, + {"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"}, + {"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"}, + {"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"}, }; -- cgit v1.1 From 6943c92e87c4aa2a6d7a1f4dbd79cf4a0b5fd67b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 18 May 2009 16:02:05 +0300 Subject: ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles the headset ramp up and down sequences needed for the pop noise removal. With this patch the order of the internal components in the twl4030 codec is turned on and off in a correct order. Signed-off-by: Peter Ujfalusi Tested-by: Anuj Aggarwal Tested-by: Jarkko Nikula Tested-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 116 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 91 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 99fe44f..f554672 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -130,6 +130,12 @@ struct twl4030_priv { unsigned int rate; unsigned int sample_bits; unsigned int channels; + + unsigned int sysclk; + + /* Headset output state handling */ + unsigned int hsl_enabled; + unsigned int hsr_enabled; }; /* @@ -564,39 +570,85 @@ static int handsfree_event(struct snd_soc_dapm_widget *w, return 0; } -static int headsetl_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static void headset_ramp(struct snd_soc_codec *codec, int ramp) { unsigned char hs_gain, hs_pop; + struct twl4030_priv *twl4030 = codec->private_data; + /* Base values for ramp delay calculation: 2^19 - 2^26 */ + unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304, + 8388608, 16777216, 33554432, 67108864}; - /* Save the current volume */ - hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET); - hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET); + hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); + hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* Do the anti-pop/bias ramp enable according to the TRM */ + if (ramp) { + /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); - /* Is this needed? Can we just use whatever gain here? */ - twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, - (hs_gain & (~0x0f)) | 0x0a); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); - - /* Restore the original volume */ - twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain); - break; - case SND_SOC_DAPM_POST_PMD: - /* Do the anti-pop/bias ramp disable according to the TRM */ + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } else { + /* Headset ramp-down _not_ according to + * the TRM, but in a way that it is working */ hs_pop &= ~TWL4030_RAMP_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); /* Bypass the reg_cache to mute the headset */ twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain & (~0x0f), TWL4030_REG_HS_GAIN_SET); + hs_pop &= ~TWL4030_VMID_EN; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } +} + +static int headsetlpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = w->codec->private_data; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Do the ramp-up only once */ + if (!twl4030->hsr_enabled) + headset_ramp(w->codec, 1); + + twl4030->hsl_enabled = 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* Do the ramp-down only if both headsetL/R is disabled */ + if (!twl4030->hsr_enabled) + headset_ramp(w->codec, 0); + + twl4030->hsl_enabled = 0; + break; + } + return 0; +} + +static int headsetrpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = w->codec->private_data; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Do the ramp-up only once */ + if (!twl4030->hsl_enabled) + headset_ramp(w->codec, 1); + + twl4030->hsr_enabled = 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* Do the ramp-down only if both headsetL/R is disabled */ + if (!twl4030->hsl_enabled) + headset_ramp(w->codec, 0); + + twl4030->hsr_enabled = 0; break; } return 0; @@ -1116,13 +1168,18 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_predriver_controls[0], ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsol_controls[0], - ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + ARRAY_SIZE(twl4030_dapm_hsol_controls)), + SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, headsetlpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsor_controls[0], ARRAY_SIZE(twl4030_dapm_hsor_controls)), + SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, headsetrpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* CarkitL/R */ SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitl_controls[0], @@ -1227,10 +1284,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"HeadsetL PGA", NULL, "HeadsetL Mixer"}, /* HeadsetR */ {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"}, {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, + {"HeadsetR PGA", NULL, "HeadsetR Mixer"}, /* CarkitL */ {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, @@ -1261,8 +1320,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"EARPIECE", NULL, "Earpiece Mixer"}, {"PREDRIVEL", NULL, "PredriveL Mixer"}, {"PREDRIVER", NULL, "PredriveR Mixer"}, - {"HSOL", NULL, "HeadsetL Mixer"}, - {"HSOR", NULL, "HeadsetR Mixer"}, + {"HSOL", NULL, "HeadsetL PGA"}, + {"HSOR", NULL, "HeadsetR PGA"}, {"CARKITL", NULL, "CarkitL Mixer"}, {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, @@ -1601,17 +1660,21 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; + struct twl4030_priv *twl4030 = codec->private_data; u8 infreq; switch (freq) { case 19200000: infreq = TWL4030_APLL_INFREQ_19200KHZ; + twl4030->sysclk = 19200; break; case 26000000: infreq = TWL4030_APLL_INFREQ_26000KHZ; + twl4030->sysclk = 26000; break; case 38400000: infreq = TWL4030_APLL_INFREQ_38400KHZ; + twl4030->sysclk = 38400; break; default: printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", @@ -2000,6 +2063,9 @@ static int twl4030_probe(struct platform_device *pdev) kfree(codec); return -ENOMEM; } + /* Set default sysclk (used by the headsetl/rpga_event callback for + * pop-attenuation) */ + twl4030->sysclk = 26000; codec->private_data = twl4030; socdev->card->codec = codec; -- cgit v1.1 From 3e3ee6dc949d25423f72bb5ee5f58dfa79ac85c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 May 2009 16:47:59 +0200 Subject: ALSA: ctxfi - Add depends on X86 The ctxfi driver requires explicitly the 4k page size, and gives a build error on architectures with non-4k pages. As a workaround, just add the kconfig dependency on X86, which is the only architecture ever tested. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 3a7640f..2d7fef5 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -277,6 +277,7 @@ config SND_CS5535AUDIO config SND_CTXFI tristate "Creative Sound Blaster X-Fi" + depends on X86 select SND_PCM help If you want to use soundcards based on Creative Sound Blastr X-Fi -- cgit v1.1 From 5c82f56736e4c3a9eaf53c94366b056c8622d79e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 May 2009 09:41:30 +0100 Subject: AsoC: Make snd_soc_read() and snd_soc_write() functions Should be no impact on the generated code but it helps the compiler print clearer messages. Signed-off-by: Mark Brown --- include/sound/soc.h | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2af3213..cf6111d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,10 +214,6 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); #endif -/* codec IO */ -#define snd_soc_read(codec, reg) codec->read(codec, reg) -#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) - /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned short mask, unsigned short value); @@ -507,6 +503,19 @@ struct soc_enum { void *dapm; }; +/* codec IO */ +static inline unsigned int snd_soc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + return codec->read(codec, reg); +} + +static inline unsigned int snd_soc_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val) +{ + return codec->write(codec, reg, val); +} + #include #endif -- cgit v1.1 From 9da28c7b38170882b1c43d7d133ddce34e25f161 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 May 2009 10:13:15 +0300 Subject: ASoC: TWL4030: Add support for platform dependent configuration twl4030_setup_data structure can be passed from platform drivers to the codec via the snd_soc_device->codec_data pointer. Currently the setup data has support for the Headset pop-removal related configuration, which differs from board to board. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 22 +++++++++++++++++++--- sound/soc/codecs/twl4030.h | 5 +++++ 2 files changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f554672..584507f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1997,6 +1997,8 @@ static int twl4030_resume(struct platform_device *pdev) static int twl4030_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_setup_data *setup = socdev->codec_data; + struct twl4030_priv *twl4030 = codec->private_data; int ret = 0; printk(KERN_INFO "TWL4030 Audio Codec init \n"); @@ -2014,6 +2016,23 @@ static int twl4030_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + /* Configuration for headset ramp delay from setup data */ + if (setup) { + unsigned char hs_pop; + + if (setup->sysclk) + twl4030->sysclk = setup->sysclk; + else + twl4030->sysclk = 26000; + + hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + hs_pop &= ~TWL4030_RAMP_DELAY; + hs_pop |= (setup->ramp_delay_value << 2); + twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } else { + twl4030->sysclk = 26000; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -2063,9 +2082,6 @@ static int twl4030_probe(struct platform_device *pdev) kfree(codec); return -ENOMEM; } - /* Set default sysclk (used by the headsetl/rpga_event callback for - * pop-attenuation) */ - twl4030->sysclk = 26000; codec->private_data = twl4030; socdev->card->codec = codec; diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 9668bdf..48326e2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -266,4 +266,9 @@ extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; +struct twl4030_setup_data { + unsigned int ramp_delay_value; + unsigned int sysclk; +}; + #endif /* End of __TWL4030_AUDIO_H__ */ -- cgit v1.1 From 7385ba44f8bcea15bf0d75ae2814f0cec63140b9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 May 2009 10:13:16 +0300 Subject: ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal With this patch the initial headset pop-removal related values are configured for the twl4030 codec (ramp delay and sysclk). Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 19966a7..b719e5d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -276,10 +276,17 @@ static struct snd_soc_card snd_soc_sdp3430 = { .num_links = ARRAY_SIZE(sdp3430_dai), }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 3, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device sdp3430_snd_devdata = { .card = &snd_soc_sdp3430, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *sdp3430_snd_device; -- cgit v1.1 From b4852b793a1dd74ccde5572d8a8f73e948a5b1a1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 May 2009 15:12:15 +0300 Subject: ASoC: TWL4030: Differentiate the playback streams Give unique stream names for the two playback streams so DAPM can figure out which codec_dai is in use. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 584507f..9197fdd 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1092,13 +1092,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIBRA"), /* DACs */ - SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", + SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", + SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", + SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", + SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", SND_SOC_NOPM, 0, 0), @@ -1937,7 +1937,7 @@ struct snd_soc_dai twl4030_dai[] = { { .name = "twl4030", .playback = { - .stream_name = "Playback", + .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, @@ -1953,7 +1953,7 @@ struct snd_soc_dai twl4030_dai[] = { { .name = "twl4030 Voice", .playback = { - .stream_name = "Playback", + .stream_name = "Voice Playback", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, -- cgit v1.1 From 86ed3669f068b514ab85ffd548456a342b3fb8d3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 May 2009 15:01:19 +0100 Subject: ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver The WM9081 is designed to provide high power output at low distortion levels in space-constrained portable applications. Signed-off-by: Mark Brown --- include/sound/wm9081.h | 25 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm9081.c | 1532 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9081.h | 787 +++++++++++++++++++++++ 5 files changed, 2350 insertions(+) create mode 100644 include/sound/wm9081.h create mode 100644 sound/soc/codecs/wm9081.c create mode 100644 sound/soc/codecs/wm9081.h diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h new file mode 100644 index 0000000..e173ddb --- /dev/null +++ b/include/sound/wm9081.h @@ -0,0 +1,25 @@ +/* + * linux/sound/wm9081.h -- Platform data for WM9081 + * + * Copyright 2009 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_WM_9081_H +#define __LINUX_SND_WM_9081_H + +struct wm9081_retune_mobile_setting { + const char *name; + unsigned int rate; + u16 config[20]; +}; + +struct wm9081_retune_mobile_config { + struct wm9081_retune_mobile_setting *configs; + int num_configs; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c19ad5..7f78b65 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8971 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C + select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS @@ -156,6 +157,9 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate +config SND_SOC_WM9081 + tristate + config SND_SOC_WM9705 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d31b6b..70c55fa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm9081-objs := wm9081.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -62,6 +63,7 @@ obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c new file mode 100644 index 0000000..83e3148 --- /dev/null +++ b/sound/soc/codecs/wm9081.c @@ -0,0 +1,1532 @@ +/* + * wm9081.c -- WM9081 ALSA SoC Audio driver + * + * Author: Mark Brown + * + * Copyright 2009 Wolfson Microelectronics plc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "wm9081.h" + +static u16 wm9081_reg_defaults[] = { + 0x0000, /* R0 - Software Reset */ + 0x0000, /* R1 */ + 0x00B9, /* R2 - Analogue Lineout */ + 0x00B9, /* R3 - Analogue Speaker PGA */ + 0x0001, /* R4 - VMID Control */ + 0x0068, /* R5 - Bias Control 1 */ + 0x0000, /* R6 */ + 0x0000, /* R7 - Analogue Mixer */ + 0x0000, /* R8 - Anti Pop Control */ + 0x01DB, /* R9 - Analogue Speaker 1 */ + 0x0018, /* R10 - Analogue Speaker 2 */ + 0x0180, /* R11 - Power Management */ + 0x0000, /* R12 - Clock Control 1 */ + 0x0038, /* R13 - Clock Control 2 */ + 0x4000, /* R14 - Clock Control 3 */ + 0x0000, /* R15 */ + 0x0000, /* R16 - FLL Control 1 */ + 0x0200, /* R17 - FLL Control 2 */ + 0x0000, /* R18 - FLL Control 3 */ + 0x0204, /* R19 - FLL Control 4 */ + 0x0000, /* R20 - FLL Control 5 */ + 0x0000, /* R21 */ + 0x0000, /* R22 - Audio Interface 1 */ + 0x0002, /* R23 - Audio Interface 2 */ + 0x0008, /* R24 - Audio Interface 3 */ + 0x0022, /* R25 - Audio Interface 4 */ + 0x0000, /* R26 - Interrupt Status */ + 0x0006, /* R27 - Interrupt Status Mask */ + 0x0000, /* R28 - Interrupt Polarity */ + 0x0000, /* R29 - Interrupt Control */ + 0x00C0, /* R30 - DAC Digital 1 */ + 0x0008, /* R31 - DAC Digital 2 */ + 0x09AF, /* R32 - DRC 1 */ + 0x4201, /* R33 - DRC 2 */ + 0x0000, /* R34 - DRC 3 */ + 0x0000, /* R35 - DRC 4 */ + 0x0000, /* R36 */ + 0x0000, /* R37 */ + 0x0000, /* R38 - Write Sequencer 1 */ + 0x0000, /* R39 - Write Sequencer 2 */ + 0x0002, /* R40 - MW Slave 1 */ + 0x0000, /* R41 */ + 0x0000, /* R42 - EQ 1 */ + 0x0000, /* R43 - EQ 2 */ + 0x0FCA, /* R44 - EQ 3 */ + 0x0400, /* R45 - EQ 4 */ + 0x00B8, /* R46 - EQ 5 */ + 0x1EB5, /* R47 - EQ 6 */ + 0xF145, /* R48 - EQ 7 */ + 0x0B75, /* R49 - EQ 8 */ + 0x01C5, /* R50 - EQ 9 */ + 0x169E, /* R51 - EQ 10 */ + 0xF829, /* R52 - EQ 11 */ + 0x07AD, /* R53 - EQ 12 */ + 0x1103, /* R54 - EQ 13 */ + 0x1C58, /* R55 - EQ 14 */ + 0xF373, /* R56 - EQ 15 */ + 0x0A54, /* R57 - EQ 16 */ + 0x0558, /* R58 - EQ 17 */ + 0x0564, /* R59 - EQ 18 */ + 0x0559, /* R60 - EQ 19 */ + 0x4000, /* R61 - EQ 20 */ +}; + +static struct { + int ratio; + int clk_sys_rate; +} clk_sys_rates[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 768, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +static struct { + int rate; + int sample_rate; +} sample_rates[] = { + { 8000, 0 }, + { 11025, 1 }, + { 12000, 2 }, + { 16000, 3 }, + { 22050, 4 }, + { 24000, 5 }, + { 32000, 6 }, + { 44100, 7 }, + { 48000, 8 }, + { 88200, 9 }, + { 96000, 10 }, +}; + +static struct { + int div; /* *10 due to .5s */ + int bclk_div; +} bclk_divs[] = { + { 10, 0 }, + { 15, 1 }, + { 20, 2 }, + { 30, 3 }, + { 40, 4 }, + { 50, 5 }, + { 55, 6 }, + { 60, 7 }, + { 80, 8 }, + { 100, 9 }, + { 110, 10 }, + { 120, 11 }, + { 160, 12 }, + { 200, 13 }, + { 220, 14 }, + { 240, 15 }, + { 250, 16 }, + { 300, 17 }, + { 320, 18 }, + { 440, 19 }, + { 480, 20 }, +}; + +struct wm9081_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM9081_MAX_REGISTER + 1]; + int sysclk_source; + int mclk_rate; + int sysclk_rate; + int fs; + int bclk; + int master; + int fll_fref; + int fll_fout; + struct wm9081_retune_mobile_config *retune; +}; + +static int wm9081_reg_is_volatile(int reg) +{ + switch (reg) { + default: + return 0; + } +} + +static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > WM9081_MAX_REGISTER); + return cache[reg]; +} + +static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg) +{ + struct i2c_msg xfer[2]; + u16 data; + int ret; + struct i2c_client *client = codec->control_data; + + BUG_ON(reg > WM9081_MAX_REGISTER); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} + +static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg) +{ + if (wm9081_reg_is_volatile(reg)) + return wm9081_read_hw(codec, reg); + else + return wm9081_read_reg_cache(codec, reg); +} + +static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[3]; + + BUG_ON(reg > WM9081_MAX_REGISTER); + + if (!wm9081_reg_is_volatile(reg)) + cache[reg] = value; + + data[0] = reg; + data[1] = value >> 8; + data[2] = value & 0x00ff; + + if (codec->hw_write(codec->control_data, data, 3) == 3) + return 0; + else + return -EIO; +} + +static int wm9081_reset(struct snd_soc_codec *codec) +{ + return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0); +static const DECLARE_TLV_DB_SCALE(drc_out_tlv, -2250, 75, 0); +static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); +static unsigned int drc_max_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(1200, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(1800, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -300, 50, 0); + +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(in_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); + +static const char *drc_high_text[] = { + "1", + "1/2", + "1/4", + "1/8", + "1/16", + "0", +}; + +static const struct soc_enum drc_high = + SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text); + +static const char *drc_low_text[] = { + "1", + "1/2", + "1/4", + "1/8", + "0", +}; + +static const struct soc_enum drc_low = + SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text); + +static const char *drc_atk_text[] = { + "181us", + "181us", + "363us", + "726us", + "1.45ms", + "2.9ms", + "5.8ms", + "11.6ms", + "23.2ms", + "46.4ms", + "92.8ms", + "185.6ms", +}; + +static const struct soc_enum drc_atk = + SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text); + +static const char *drc_dcy_text[] = { + "186ms", + "372ms", + "743ms", + "1.49s", + "2.97s", + "5.94s", + "11.89s", + "23.78s", + "47.56s", +}; + +static const struct soc_enum drc_dcy = + SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text); + +static const char *drc_qr_dcy_text[] = { + "0.725ms", + "1.45ms", + "5.8ms", +}; + +static const struct soc_enum drc_qr_dcy = + SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text); + +static const char *dac_deemph_text[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", +}; + +static const struct soc_enum dac_deemph = + SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text); + +static const char *speaker_mode_text[] = { + "Class D", + "Class AB", +}; + +static const struct soc_enum speaker_mode = + SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text); + +static int speaker_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg; + + reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2); + if (reg & WM9081_SPK_MODE) + ucontrol->value.integer.value[0] = 1; + else + ucontrol->value.integer.value[0] = 0; + + return 0; +} + +/* + * Stop any attempts to change speaker mode while the speaker is enabled. + * + * We also have some special anti-pop controls dependant on speaker + * mode which must be changed along with the mode. + */ +static int speaker_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT); + unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2); + + /* Are we changing anything? */ + if (ucontrol->value.integer.value[0] == + ((reg2 & WM9081_SPK_MODE) != 0)) + return 0; + + /* Don't try to change modes while enabled */ + if (reg_pwr & WM9081_SPK_ENA) + return -EINVAL; + + if (ucontrol->value.integer.value[0]) { + /* Class AB */ + reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL); + reg2 |= WM9081_SPK_MODE; + } else { + /* Class D */ + reg2 |= WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL; + reg2 &= ~WM9081_SPK_MODE; + } + + wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2); + + return 0; +} + +static const struct snd_kcontrol_new wm9081_snd_controls[] = { +SOC_SINGLE_TLV("IN1 Volume", WM9081_ANALOGUE_MIXER, 1, 1, 1, in_tlv), +SOC_SINGLE_TLV("IN2 Volume", WM9081_ANALOGUE_MIXER, 3, 1, 1, in_tlv), + +SOC_SINGLE_TLV("Playback Volume", WM9081_DAC_DIGITAL_1, 1, 96, 0, dac_tlv), + +SOC_SINGLE("LINEOUT Switch", WM9081_ANALOGUE_LINEOUT, 7, 1, 1), +SOC_SINGLE("LINEOUT ZC Switch", WM9081_ANALOGUE_LINEOUT, 6, 1, 0), +SOC_SINGLE_TLV("LINEOUT Volume", WM9081_ANALOGUE_LINEOUT, 0, 63, 0, out_tlv), + +SOC_SINGLE("DRC Switch", WM9081_DRC_1, 15, 1, 0), +SOC_ENUM("DRC High Slope", drc_high), +SOC_ENUM("DRC Low Slope", drc_low), +SOC_SINGLE_TLV("DRC Input Volume", WM9081_DRC_4, 5, 60, 1, drc_in_tlv), +SOC_SINGLE_TLV("DRC Output Volume", WM9081_DRC_4, 0, 30, 1, drc_out_tlv), +SOC_SINGLE_TLV("DRC Minimum Volume", WM9081_DRC_2, 2, 3, 1, drc_min_tlv), +SOC_SINGLE_TLV("DRC Maximum Volume", WM9081_DRC_2, 0, 3, 0, drc_max_tlv), +SOC_ENUM("DRC Attack", drc_atk), +SOC_ENUM("DRC Decay", drc_dcy), +SOC_SINGLE("DRC Quick Release Switch", WM9081_DRC_1, 2, 1, 0), +SOC_SINGLE_TLV("DRC Quick Release Volume", WM9081_DRC_2, 6, 3, 0, drc_qr_tlv), +SOC_ENUM("DRC Quick Release Decay", drc_qr_dcy), +SOC_SINGLE_TLV("DRC Startup Volume", WM9081_DRC_1, 6, 18, 0, drc_startup_tlv), + +SOC_SINGLE("EQ Switch", WM9081_EQ_1, 0, 1, 0), + +SOC_SINGLE("Speaker DC Volume", WM9081_ANALOGUE_SPEAKER_1, 3, 5, 0), +SOC_SINGLE("Speaker AC Volume", WM9081_ANALOGUE_SPEAKER_1, 0, 5, 0), +SOC_SINGLE("Speaker Switch", WM9081_ANALOGUE_SPEAKER_PGA, 7, 1, 1), +SOC_SINGLE("Speaker ZC Switch", WM9081_ANALOGUE_SPEAKER_PGA, 6, 1, 0), +SOC_SINGLE_TLV("Speaker Volume", WM9081_ANALOGUE_SPEAKER_PGA, 0, 63, 0, + out_tlv), +SOC_ENUM("DAC Deemphasis", dac_deemph), +SOC_ENUM_EXT("Speaker Mode", speaker_mode, speaker_mode_get, speaker_mode_put), +}; + +static const struct snd_kcontrol_new wm9081_eq_controls[] = { +SOC_SINGLE_TLV("EQ1 Volume", WM9081_EQ_1, 11, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Volume", WM9081_EQ_1, 6, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Volume", WM9081_EQ_1, 1, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Volume", WM9081_EQ_2, 11, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ5 Volume", WM9081_EQ_2, 6, 24, 0, eq_tlv), +}; + +static const struct snd_kcontrol_new mixer[] = { +SOC_DAPM_SINGLE("IN1 Switch", WM9081_ANALOGUE_MIXER, 0, 1, 0), +SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0), +}; + +static int speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg |= WM9081_SPK_ENA; + break; + + case SND_SOC_DAPM_PRE_PMD: + reg &= ~WM9081_SPK_ENA; + break; + } + + wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg); + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_clk_ref_div; + u16 n; + u16 k; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + unsigned int div; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + while ((Fref / div) > 13500000) { + div *= 2; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + fll_div->fll_clk_ref_div = div / 2; + + pr_debug("Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 0; + target = Fout * 2; + while (target < 90000000) { + div++; + target *= 2; + if (div > 7) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + fll_div->fll_outdiv = div; + + pr_debug("Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + target /= fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + fll_div->n = Ndiv; + Nmod = target % Fref; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll_div->k = K / 10; + + pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n", + fll_div->n, fll_div->k, + fll_div->fll_fratio, fll_div->fll_outdiv, + fll_div->fll_clk_ref_div); + + return 0; +} + +static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id, + unsigned int Fref, unsigned int Fout) +{ + struct wm9081_priv *wm9081 = codec->private_data; + u16 reg1, reg4, reg5; + struct _fll_div fll_div; + int ret; + int clk_sys_reg; + + /* Any change? */ + if (Fref == wm9081->fll_fref && Fout == wm9081->fll_fout) + return 0; + + /* Disable the FLL */ + if (Fout == 0) { + dev_dbg(codec->dev, "FLL disabled\n"); + wm9081->fll_fref = 0; + wm9081->fll_fout = 0; + + return 0; + } + + ret = fll_factors(&fll_div, Fref, Fout); + if (ret != 0) + return ret; + + reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5); + reg5 &= ~WM9081_FLL_CLK_SRC_MASK; + + switch (fll_id) { + case WM9081_SYSCLK_FLL_MCLK: + reg5 |= 0x1; + break; + + default: + dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id); + return -EINVAL; + } + + /* Disable CLK_SYS while we reconfigure */ + clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3); + if (clk_sys_reg & WM9081_CLK_SYS_ENA) + wm9081_write(codec, WM9081_CLOCK_CONTROL_3, + clk_sys_reg & ~WM9081_CLK_SYS_ENA); + + /* Any FLL configuration change requires that the FLL be + * disabled first. */ + reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1); + reg1 &= ~WM9081_FLL_ENA; + wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1); + + /* Apply the configuration */ + if (fll_div.k) + reg1 |= WM9081_FLL_FRAC_MASK; + else + reg1 &= ~WM9081_FLL_FRAC_MASK; + wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1); + + wm9081_write(codec, WM9081_FLL_CONTROL_2, + (fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT)); + wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k); + + reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4); + reg4 &= ~WM9081_FLL_N_MASK; + reg4 |= fll_div.n << WM9081_FLL_N_SHIFT; + wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4); + + reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK; + reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT; + wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5); + + /* Enable the FLL */ + wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA); + + /* Then bring CLK_SYS up again if it was disabled */ + if (clk_sys_reg & WM9081_CLK_SYS_ENA) + wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg); + + dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); + + wm9081->fll_fref = Fref; + wm9081->fll_fout = Fout; + + return 0; +} + +static int configure_clock(struct snd_soc_codec *codec) +{ + struct wm9081_priv *wm9081 = codec->private_data; + int new_sysclk, i, target; + unsigned int reg; + int ret = 0; + int mclkdiv = 0; + int fll = 0; + + switch (wm9081->sysclk_source) { + case WM9081_SYSCLK_MCLK: + if (wm9081->mclk_rate > 12225000) { + mclkdiv = 1; + wm9081->sysclk_rate = wm9081->mclk_rate / 2; + } else { + wm9081->sysclk_rate = wm9081->mclk_rate; + } + wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, 0, 0); + break; + + case WM9081_SYSCLK_FLL_MCLK: + /* If we have a sample rate calculate a CLK_SYS that + * gives us a suitable DAC configuration, plus BCLK. + * Ideally we would check to see if we can clock + * directly from MCLK and only use the FLL if this is + * not the case, though care must be taken with free + * running mode. + */ + if (wm9081->master && wm9081->bclk) { + /* Make sure we can generate CLK_SYS and BCLK + * and that we've got 3MHz for optimal + * performance. */ + for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { + target = wm9081->fs * clk_sys_rates[i].ratio; + if (target >= wm9081->bclk && + target > 3000000) + new_sysclk = target; + } + } else if (wm9081->fs) { + for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { + new_sysclk = clk_sys_rates[i].ratio + * wm9081->fs; + if (new_sysclk > 3000000) + break; + } + } else { + new_sysclk = 12288000; + } + + ret = wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, + wm9081->mclk_rate, new_sysclk); + if (ret == 0) { + wm9081->sysclk_rate = new_sysclk; + + /* Switch SYSCLK over to FLL */ + fll = 1; + } else { + wm9081->sysclk_rate = wm9081->mclk_rate; + } + break; + + default: + return -EINVAL; + } + + reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1); + if (mclkdiv) + reg |= WM9081_MCLKDIV2; + else + reg &= ~WM9081_MCLKDIV2; + wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg); + + reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3); + if (fll) + reg |= WM9081_CLK_SRC_SEL; + else + reg &= ~WM9081_CLK_SRC_SEL; + wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg); + + dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate); + + return ret; +} + +static int clk_sys_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm9081_priv *wm9081 = codec->private_data; + + /* This should be done on init() for bypass paths */ + switch (wm9081->sysclk_source) { + case WM9081_SYSCLK_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK\n", wm9081->mclk_rate); + break; + case WM9081_SYSCLK_FLL_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK with FLL\n", + wm9081->mclk_rate); + break; + default: + dev_err(codec->dev, "System clock not configured\n"); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + configure_clock(codec); + break; + + case SND_SOC_DAPM_POST_PMD: + /* Disable the FLL if it's running */ + wm9081_set_fll(codec, 0, 0, 0); + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN1"), +SND_SOC_DAPM_INPUT("IN2"), + +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0), + +SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, + mixer, ARRAY_SIZE(mixer)), + +SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0, + speaker_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("LINEOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), + +SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), +}; + + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "CLK_SYS" }, + { "DAC", NULL, "CLK_DSP" }, + + { "Mixer", "IN1 Switch", "IN1" }, + { "Mixer", "IN2 Switch", "IN2" }, + { "Mixer", "Playback Switch", "DAC" }, + + { "LINEOUT PGA", NULL, "Mixer" }, + { "LINEOUT PGA", NULL, "TOCLK" }, + { "LINEOUT PGA", NULL, "CLK_SYS" }, + + { "LINEOUT", NULL, "LINEOUT PGA" }, + + { "Speaker PGA", NULL, "Mixer" }, + { "Speaker PGA", NULL, "TOCLK" }, + { "Speaker PGA", NULL, "CLK_SYS" }, + + { "SPKN", NULL, "Speaker PGA" }, + { "SPKP", NULL, "Speaker PGA" }, +}; + +static int wm9081_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*40k */ + reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg &= ~WM9081_VMID_SEL_MASK; + reg |= 0x2; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Normal bias current */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg &= ~WM9081_STBY_BIAS_ENA; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + break; + + case SND_SOC_BIAS_STANDBY: + /* Initial cold start */ + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Disable LINEOUT discharge */ + reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL); + reg &= ~WM9081_LINEOUT_DISCH; + wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg); + + /* Select startup bias source */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + + /* VMID 2*4k; Soft VMID ramp enable */ + reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg |= WM9081_VMID_RAMP | 0x6; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + mdelay(100); + + /* Normal bias enable & soft start off */ + reg |= WM9081_BIAS_ENA; + reg &= ~WM9081_VMID_RAMP; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Standard bias source */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg &= ~WM9081_BIAS_SRC; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + } + + /* VMID 2*240k */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg &= ~WM9081_VMID_SEL_MASK; + reg |= 0x40; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Standby bias current on */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg |= WM9081_STBY_BIAS_ENA; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + break; + + case SND_SOC_BIAS_OFF: + /* Startup bias source */ + reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg |= WM9081_BIAS_SRC; + wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + + /* Disable VMID and biases with soft ramping */ + reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); + reg |= WM9081_VMID_RAMP; + wm9081_write(codec, WM9081_VMID_CONTROL, reg); + + /* Actively discharge LINEOUT */ + reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL); + reg |= WM9081_LINEOUT_DISCH; + wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg); + break; + } + + codec->bias_level = level; + + return 0; +} + +static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm9081_priv *wm9081 = codec->private_data; + unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2); + + aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV | + WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + wm9081->master = 0; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif2 |= WM9081_LRCLK_DIR; + wm9081->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif2 |= WM9081_BCLK_DIR; + wm9081->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif2 |= WM9081_LRCLK_DIR | WM9081_BCLK_DIR; + wm9081->master = 1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif2 |= WM9081_AIF_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif2 |= 0x3; + break; + case SND_SOC_DAIFMT_I2S: + aif2 |= 0x2; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif2 |= 0x1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif2 |= WM9081_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif2 |= WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif2 |= WM9081_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif2 |= WM9081_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); + + return 0; +} + +static int wm9081_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm9081_priv *wm9081 = codec->private_data; + int ret, i, best, best_val, cur_val; + unsigned int clk_ctrl2, aif1, aif2, aif3, aif4; + + clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2); + clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK); + + aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); + + aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2); + aif2 &= ~WM9081_AIF_WL_MASK; + + aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3); + aif3 &= ~WM9081_BCLK_DIV_MASK; + + aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4); + aif4 &= ~WM9081_LRCLK_RATE_MASK; + + /* What BCLK do we need? */ + wm9081->fs = params_rate(params); + wm9081->bclk = 2 * wm9081->fs; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wm9081->bclk *= 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wm9081->bclk *= 20; + aif2 |= 0x4; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wm9081->bclk *= 24; + aif2 |= 0x8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wm9081->bclk *= 32; + aif2 |= 0xc; + break; + default: + return -EINVAL; + } + + if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) { + int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >> + WM9081_AIFDAC_TDM_MODE_SHIFT) + 1; + wm9081->bclk *= slots; + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk); + + ret = configure_clock(codec); + if (ret != 0) + return ret; + + /* Select nearest CLK_SYS_RATE */ + best = 0; + best_val = abs((wm9081->sysclk_rate / clk_sys_rates[0].ratio) + - wm9081->fs); + for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { + cur_val = abs((wm9081->sysclk_rate / + clk_sys_rates[i].ratio) - wm9081->fs);; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n", + clk_sys_rates[best].ratio); + clk_ctrl2 |= (clk_sys_rates[best].clk_sys_rate + << WM9081_CLK_SYS_RATE_SHIFT); + + /* SAMPLE_RATE */ + best = 0; + best_val = abs(wm9081->fs - sample_rates[0].rate); + for (i = 1; i < ARRAY_SIZE(sample_rates); i++) { + /* Closest match */ + cur_val = abs(wm9081->fs - sample_rates[i].rate); + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", + sample_rates[best].rate); + clk_ctrl2 |= (sample_rates[i].sample_rate << WM9081_SAMPLE_RATE_SHIFT); + + /* BCLK_DIV */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = ((wm9081->sysclk_rate * 10) / bclk_divs[i].div) + - wm9081->bclk; + if (cur_val < 0) /* Table is sorted */ + break; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + wm9081->bclk = (wm9081->sysclk_rate * 10) / bclk_divs[best].div; + dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n", + bclk_divs[best].div, wm9081->bclk); + aif3 |= bclk_divs[best].bclk_div; + + /* LRCLK is a simple fraction of BCLK */ + dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm9081->bclk / wm9081->fs); + aif4 |= wm9081->bclk / wm9081->fs; + + /* Apply a ReTune Mobile configuration if it's in use */ + if (wm9081->retune) { + struct wm9081_retune_mobile_config *retune = wm9081->retune; + struct wm9081_retune_mobile_setting *s; + int eq1; + + best = 0; + best_val = abs(retune->configs[0].rate - wm9081->fs); + for (i = 0; i < retune->num_configs; i++) { + cur_val = abs(retune->configs[i].rate - wm9081->fs); + if (cur_val < best_val) { + best_val = cur_val; + best = i; + } + } + s = &retune->configs[best]; + + dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", + s->name, s->rate); + + /* If the EQ is enabled then disable it while we write out */ + eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA; + if (eq1 & WM9081_EQ_ENA) + wm9081_write(codec, WM9081_EQ_1, 0); + + /* Write out the other values */ + for (i = 1; i < ARRAY_SIZE(s->config); i++) + wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]); + + eq1 |= (s->config[0] & ~WM9081_EQ_ENA); + wm9081_write(codec, WM9081_EQ_1, eq1); + } + + wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2); + wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); + wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3); + wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4); + + return 0; +} + +static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg; + + reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2); + + if (mute) + reg |= WM9081_DAC_MUTE; + else + reg &= ~WM9081_DAC_MUTE; + + wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg); + + return 0; +} + +static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm9081_priv *wm9081 = codec->private_data; + + switch (clk_id) { + case WM9081_SYSCLK_MCLK: + case WM9081_SYSCLK_FLL_MCLK: + wm9081->sysclk_source = clk_id; + wm9081->mclk_rate = freq; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); + + aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK); + + if (slots < 1 || slots > 4) + return -EINVAL; + + aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; + + switch (mask) { + case 1: + break; + case 2: + aif1 |= 0x10; + break; + case 4: + aif1 |= 0x20; + break; + case 8: + aif1 |= 0x30; + break; + default: + return -EINVAL; + } + + wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1); + + return 0; +} + +#define WM9081_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM9081_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm9081_dai_ops = { + .hw_params = wm9081_hw_params, + .set_sysclk = wm9081_set_sysclk, + .set_fmt = wm9081_set_dai_fmt, + .digital_mute = wm9081_digital_mute, + .set_tdm_slot = wm9081_set_tdm_slot, +}; + +/* We report two channels because the CODEC processes a stereo signal, even + * though it is only capable of handling a mono output. + */ +struct snd_soc_dai wm9081_dai = { + .name = "WM9081", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9081_RATES, + .formats = WM9081_FORMATS, + }, + .ops = &wm9081_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm9081_dai); + + +static struct snd_soc_codec *wm9081_codec; + +static int wm9081_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct wm9081_priv *wm9081; + int ret = 0; + + if (wm9081_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm9081_codec; + codec = wm9081_codec; + wm9081 = codec->private_data; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm9081_snd_controls, + ARRAY_SIZE(wm9081_snd_controls)); + if (!wm9081->retune) { + dev_dbg(codec->dev, + "No ReTune Mobile data, using normal EQ\n"); + snd_soc_add_controls(codec, wm9081_eq_controls, + ARRAY_SIZE(wm9081_eq_controls)); + } + + snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets, + ARRAY_SIZE(wm9081_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm9081_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM +static int wm9081_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm9081_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 *reg_cache = codec->reg_cache; + int i; + + for (i = 0; i < codec->reg_cache_size; i++) { + if (i == WM9081_SOFTWARE_RESET) + continue; + + wm9081_write(codec, i, reg_cache[i]); + } + + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm9081_suspend NULL +#define wm9081_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_wm9081 = { + .probe = wm9081_probe, + .remove = wm9081_remove, + .suspend = wm9081_suspend, + .resume = wm9081_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081); + +static int wm9081_register(struct wm9081_priv *wm9081) +{ + struct snd_soc_codec *codec = &wm9081->codec; + int ret; + u16 reg; + + if (wm9081_codec) { + dev_err(codec->dev, "Another WM9081 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm9081; + codec->name = "WM9081"; + codec->owner = THIS_MODULE; + codec->read = wm9081_read; + codec->write = wm9081_write; + codec->dai = &wm9081_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache); + codec->reg_cache = &wm9081->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm9081_set_bias_level; + + memcpy(codec->reg_cache, wm9081_reg_defaults, + sizeof(wm9081_reg_defaults)); + + reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET); + if (reg != 0x9081) { + dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg); + ret = -EINVAL; + goto err; + } + + ret = wm9081_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Enable zero cross by default */ + reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT); + wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); + reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); + wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, + reg | WM9081_SPKPGAZC); + + wm9081_dai.dev = codec->dev; + + wm9081_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm9081_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm9081); + return ret; +} + +static void wm9081_unregister(struct wm9081_priv *wm9081) +{ + wm9081_set_bias_level(&wm9081->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm9081_dai); + snd_soc_unregister_codec(&wm9081->codec); + kfree(wm9081); + wm9081_codec = NULL; +} + +static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm9081_priv *wm9081; + struct snd_soc_codec *codec; + + wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL); + if (wm9081 == NULL) + return -ENOMEM; + + codec = &wm9081->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + wm9081->retune = i2c->dev.platform_data; + + i2c_set_clientdata(i2c, wm9081); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm9081_register(wm9081); +} + +static __devexit int wm9081_i2c_remove(struct i2c_client *client) +{ + struct wm9081_priv *wm9081 = i2c_get_clientdata(client); + wm9081_unregister(wm9081); + return 0; +} + +static const struct i2c_device_id wm9081_i2c_id[] = { + { "wm9081", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); + +static struct i2c_driver wm9081_i2c_driver = { + .driver = { + .name = "wm9081", + .owner = THIS_MODULE, + }, + .probe = wm9081_i2c_probe, + .remove = __devexit_p(wm9081_i2c_remove), + .id_table = wm9081_i2c_id, +}; + +static int __init wm9081_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm9081_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm9081_modinit); + +static void __exit wm9081_exit(void) +{ + i2c_del_driver(&wm9081_i2c_driver); +} +module_exit(wm9081_exit); + + +MODULE_DESCRIPTION("ASoC WM9081 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm9081.h b/sound/soc/codecs/wm9081.h new file mode 100644 index 0000000..42d3bc7 --- /dev/null +++ b/sound/soc/codecs/wm9081.h @@ -0,0 +1,787 @@ +#ifndef WM9081_H +#define WM9081_H + +/* + * wm9081.c -- WM9081 ALSA SoC Audio driver + * + * Author: Mark Brown + * + * Copyright 2009 Wolfson Microelectronics plc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include + +extern struct snd_soc_dai wm9081_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm9081; + +/* + * SYSCLK sources + */ +#define WM9081_SYSCLK_MCLK 1 /* Use MCLK without FLL */ +#define WM9081_SYSCLK_FLL_MCLK 2 /* Use MCLK, enabling FLL if required */ + +/* + * Register values. + */ +#define WM9081_SOFTWARE_RESET 0x00 +#define WM9081_ANALOGUE_LINEOUT 0x02 +#define WM9081_ANALOGUE_SPEAKER_PGA 0x03 +#define WM9081_VMID_CONTROL 0x04 +#define WM9081_BIAS_CONTROL_1 0x05 +#define WM9081_ANALOGUE_MIXER 0x07 +#define WM9081_ANTI_POP_CONTROL 0x08 +#define WM9081_ANALOGUE_SPEAKER_1 0x09 +#define WM9081_ANALOGUE_SPEAKER_2 0x0A +#define WM9081_POWER_MANAGEMENT 0x0B +#define WM9081_CLOCK_CONTROL_1 0x0C +#define WM9081_CLOCK_CONTROL_2 0x0D +#define WM9081_CLOCK_CONTROL_3 0x0E +#define WM9081_FLL_CONTROL_1 0x10 +#define WM9081_FLL_CONTROL_2 0x11 +#define WM9081_FLL_CONTROL_3 0x12 +#define WM9081_FLL_CONTROL_4 0x13 +#define WM9081_FLL_CONTROL_5 0x14 +#define WM9081_AUDIO_INTERFACE_1 0x16 +#define WM9081_AUDIO_INTERFACE_2 0x17 +#define WM9081_AUDIO_INTERFACE_3 0x18 +#define WM9081_AUDIO_INTERFACE_4 0x19 +#define WM9081_INTERRUPT_STATUS 0x1A +#define WM9081_INTERRUPT_STATUS_MASK 0x1B +#define WM9081_INTERRUPT_POLARITY 0x1C +#define WM9081_INTERRUPT_CONTROL 0x1D +#define WM9081_DAC_DIGITAL_1 0x1E +#define WM9081_DAC_DIGITAL_2 0x1F +#define WM9081_DRC_1 0x20 +#define WM9081_DRC_2 0x21 +#define WM9081_DRC_3 0x22 +#define WM9081_DRC_4 0x23 +#define WM9081_WRITE_SEQUENCER_1 0x26 +#define WM9081_WRITE_SEQUENCER_2 0x27 +#define WM9081_MW_SLAVE_1 0x28 +#define WM9081_EQ_1 0x2A +#define WM9081_EQ_2 0x2B +#define WM9081_EQ_3 0x2C +#define WM9081_EQ_4 0x2D +#define WM9081_EQ_5 0x2E +#define WM9081_EQ_6 0x2F +#define WM9081_EQ_7 0x30 +#define WM9081_EQ_8 0x31 +#define WM9081_EQ_9 0x32 +#define WM9081_EQ_10 0x33 +#define WM9081_EQ_11 0x34 +#define WM9081_EQ_12 0x35 +#define WM9081_EQ_13 0x36 +#define WM9081_EQ_14 0x37 +#define WM9081_EQ_15 0x38 +#define WM9081_EQ_16 0x39 +#define WM9081_EQ_17 0x3A +#define WM9081_EQ_18 0x3B +#define WM9081_EQ_19 0x3C +#define WM9081_EQ_20 0x3D + +#define WM9081_REGISTER_COUNT 55 +#define WM9081_MAX_REGISTER 0x3D + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Software Reset + */ +#define WM9081_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM9081_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM9081_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R2 (0x02) - Analogue Lineout + */ +#define WM9081_LINEOUT_MUTE 0x0080 /* LINEOUT_MUTE */ +#define WM9081_LINEOUT_MUTE_MASK 0x0080 /* LINEOUT_MUTE */ +#define WM9081_LINEOUT_MUTE_SHIFT 7 /* LINEOUT_MUTE */ +#define WM9081_LINEOUT_MUTE_WIDTH 1 /* LINEOUT_MUTE */ +#define WM9081_LINEOUTZC 0x0040 /* LINEOUTZC */ +#define WM9081_LINEOUTZC_MASK 0x0040 /* LINEOUTZC */ +#define WM9081_LINEOUTZC_SHIFT 6 /* LINEOUTZC */ +#define WM9081_LINEOUTZC_WIDTH 1 /* LINEOUTZC */ +#define WM9081_LINEOUT_VOL_MASK 0x003F /* LINEOUT_VOL - [5:0] */ +#define WM9081_LINEOUT_VOL_SHIFT 0 /* LINEOUT_VOL - [5:0] */ +#define WM9081_LINEOUT_VOL_WIDTH 6 /* LINEOUT_VOL - [5:0] */ + +/* + * R3 (0x03) - Analogue Speaker PGA + */ +#define WM9081_SPKPGA_MUTE 0x0080 /* SPKPGA_MUTE */ +#define WM9081_SPKPGA_MUTE_MASK 0x0080 /* SPKPGA_MUTE */ +#define WM9081_SPKPGA_MUTE_SHIFT 7 /* SPKPGA_MUTE */ +#define WM9081_SPKPGA_MUTE_WIDTH 1 /* SPKPGA_MUTE */ +#define WM9081_SPKPGAZC 0x0040 /* SPKPGAZC */ +#define WM9081_SPKPGAZC_MASK 0x0040 /* SPKPGAZC */ +#define WM9081_SPKPGAZC_SHIFT 6 /* SPKPGAZC */ +#define WM9081_SPKPGAZC_WIDTH 1 /* SPKPGAZC */ +#define WM9081_SPKPGA_VOL_MASK 0x003F /* SPKPGA_VOL - [5:0] */ +#define WM9081_SPKPGA_VOL_SHIFT 0 /* SPKPGA_VOL - [5:0] */ +#define WM9081_SPKPGA_VOL_WIDTH 6 /* SPKPGA_VOL - [5:0] */ + +/* + * R4 (0x04) - VMID Control + */ +#define WM9081_VMID_BUF_ENA 0x0020 /* VMID_BUF_ENA */ +#define WM9081_VMID_BUF_ENA_MASK 0x0020 /* VMID_BUF_ENA */ +#define WM9081_VMID_BUF_ENA_SHIFT 5 /* VMID_BUF_ENA */ +#define WM9081_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */ +#define WM9081_VMID_RAMP 0x0008 /* VMID_RAMP */ +#define WM9081_VMID_RAMP_MASK 0x0008 /* VMID_RAMP */ +#define WM9081_VMID_RAMP_SHIFT 3 /* VMID_RAMP */ +#define WM9081_VMID_RAMP_WIDTH 1 /* VMID_RAMP */ +#define WM9081_VMID_SEL_MASK 0x0006 /* VMID_SEL - [2:1] */ +#define WM9081_VMID_SEL_SHIFT 1 /* VMID_SEL - [2:1] */ +#define WM9081_VMID_SEL_WIDTH 2 /* VMID_SEL - [2:1] */ +#define WM9081_VMID_FAST_ST 0x0001 /* VMID_FAST_ST */ +#define WM9081_VMID_FAST_ST_MASK 0x0001 /* VMID_FAST_ST */ +#define WM9081_VMID_FAST_ST_SHIFT 0 /* VMID_FAST_ST */ +#define WM9081_VMID_FAST_ST_WIDTH 1 /* VMID_FAST_ST */ + +/* + * R5 (0x05) - Bias Control 1 + */ +#define WM9081_BIAS_SRC 0x0040 /* BIAS_SRC */ +#define WM9081_BIAS_SRC_MASK 0x0040 /* BIAS_SRC */ +#define WM9081_BIAS_SRC_SHIFT 6 /* BIAS_SRC */ +#define WM9081_BIAS_SRC_WIDTH 1 /* BIAS_SRC */ +#define WM9081_STBY_BIAS_LVL 0x0020 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_LVL_MASK 0x0020 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_LVL_SHIFT 5 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_LVL_WIDTH 1 /* STBY_BIAS_LVL */ +#define WM9081_STBY_BIAS_ENA 0x0010 /* STBY_BIAS_ENA */ +#define WM9081_STBY_BIAS_ENA_MASK 0x0010 /* STBY_BIAS_ENA */ +#define WM9081_STBY_BIAS_ENA_SHIFT 4 /* STBY_BIAS_ENA */ +#define WM9081_STBY_BIAS_ENA_WIDTH 1 /* STBY_BIAS_ENA */ +#define WM9081_BIAS_LVL_MASK 0x000C /* BIAS_LVL - [3:2] */ +#define WM9081_BIAS_LVL_SHIFT 2 /* BIAS_LVL - [3:2] */ +#define WM9081_BIAS_LVL_WIDTH 2 /* BIAS_LVL - [3:2] */ +#define WM9081_BIAS_ENA 0x0002 /* BIAS_ENA */ +#define WM9081_BIAS_ENA_MASK 0x0002 /* BIAS_ENA */ +#define WM9081_BIAS_ENA_SHIFT 1 /* BIAS_ENA */ +#define WM9081_BIAS_ENA_WIDTH 1 /* BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA 0x0001 /* STARTUP_BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA_MASK 0x0001 /* STARTUP_BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA_SHIFT 0 /* STARTUP_BIAS_ENA */ +#define WM9081_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */ + +/* + * R7 (0x07) - Analogue Mixer + */ +#define WM9081_DAC_SEL 0x0010 /* DAC_SEL */ +#define WM9081_DAC_SEL_MASK 0x0010 /* DAC_SEL */ +#define WM9081_DAC_SEL_SHIFT 4 /* DAC_SEL */ +#define WM9081_DAC_SEL_WIDTH 1 /* DAC_SEL */ +#define WM9081_IN2_VOL 0x0008 /* IN2_VOL */ +#define WM9081_IN2_VOL_MASK 0x0008 /* IN2_VOL */ +#define WM9081_IN2_VOL_SHIFT 3 /* IN2_VOL */ +#define WM9081_IN2_VOL_WIDTH 1 /* IN2_VOL */ +#define WM9081_IN2_ENA 0x0004 /* IN2_ENA */ +#define WM9081_IN2_ENA_MASK 0x0004 /* IN2_ENA */ +#define WM9081_IN2_ENA_SHIFT 2 /* IN2_ENA */ +#define WM9081_IN2_ENA_WIDTH 1 /* IN2_ENA */ +#define WM9081_IN1_VOL 0x0002 /* IN1_VOL */ +#define WM9081_IN1_VOL_MASK 0x0002 /* IN1_VOL */ +#define WM9081_IN1_VOL_SHIFT 1 /* IN1_VOL */ +#define WM9081_IN1_VOL_WIDTH 1 /* IN1_VOL */ +#define WM9081_IN1_ENA 0x0001 /* IN1_ENA */ +#define WM9081_IN1_ENA_MASK 0x0001 /* IN1_ENA */ +#define WM9081_IN1_ENA_SHIFT 0 /* IN1_ENA */ +#define WM9081_IN1_ENA_WIDTH 1 /* IN1_ENA */ + +/* + * R8 (0x08) - Anti Pop Control + */ +#define WM9081_LINEOUT_DISCH 0x0004 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_DISCH_MASK 0x0004 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_DISCH_SHIFT 2 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_DISCH_WIDTH 1 /* LINEOUT_DISCH */ +#define WM9081_LINEOUT_VROI 0x0002 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_VROI_MASK 0x0002 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_VROI_SHIFT 1 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_VROI_WIDTH 1 /* LINEOUT_VROI */ +#define WM9081_LINEOUT_CLAMP 0x0001 /* LINEOUT_CLAMP */ +#define WM9081_LINEOUT_CLAMP_MASK 0x0001 /* LINEOUT_CLAMP */ +#define WM9081_LINEOUT_CLAMP_SHIFT 0 /* LINEOUT_CLAMP */ +#define WM9081_LINEOUT_CLAMP_WIDTH 1 /* LINEOUT_CLAMP */ + +/* + * R9 (0x09) - Analogue Speaker 1 + */ +#define WM9081_SPK_DCGAIN_MASK 0x0038 /* SPK_DCGAIN - [5:3] */ +#define WM9081_SPK_DCGAIN_SHIFT 3 /* SPK_DCGAIN - [5:3] */ +#define WM9081_SPK_DCGAIN_WIDTH 3 /* SPK_DCGAIN - [5:3] */ +#define WM9081_SPK_ACGAIN_MASK 0x0007 /* SPK_ACGAIN - [2:0] */ +#define WM9081_SPK_ACGAIN_SHIFT 0 /* SPK_ACGAIN - [2:0] */ +#define WM9081_SPK_ACGAIN_WIDTH 3 /* SPK_ACGAIN - [2:0] */ + +/* + * R10 (0x0A) - Analogue Speaker 2 + */ +#define WM9081_SPK_MODE 0x0040 /* SPK_MODE */ +#define WM9081_SPK_MODE_MASK 0x0040 /* SPK_MODE */ +#define WM9081_SPK_MODE_SHIFT 6 /* SPK_MODE */ +#define WM9081_SPK_MODE_WIDTH 1 /* SPK_MODE */ +#define WM9081_SPK_INV_MUTE 0x0010 /* SPK_INV_MUTE */ +#define WM9081_SPK_INV_MUTE_MASK 0x0010 /* SPK_INV_MUTE */ +#define WM9081_SPK_INV_MUTE_SHIFT 4 /* SPK_INV_MUTE */ +#define WM9081_SPK_INV_MUTE_WIDTH 1 /* SPK_INV_MUTE */ +#define WM9081_OUT_SPK_CTRL 0x0008 /* OUT_SPK_CTRL */ +#define WM9081_OUT_SPK_CTRL_MASK 0x0008 /* OUT_SPK_CTRL */ +#define WM9081_OUT_SPK_CTRL_SHIFT 3 /* OUT_SPK_CTRL */ +#define WM9081_OUT_SPK_CTRL_WIDTH 1 /* OUT_SPK_CTRL */ + +/* + * R11 (0x0B) - Power Management + */ +#define WM9081_TSHUT_ENA 0x0100 /* TSHUT_ENA */ +#define WM9081_TSHUT_ENA_MASK 0x0100 /* TSHUT_ENA */ +#define WM9081_TSHUT_ENA_SHIFT 8 /* TSHUT_ENA */ +#define WM9081_TSHUT_ENA_WIDTH 1 /* TSHUT_ENA */ +#define WM9081_TSENSE_ENA 0x0080 /* TSENSE_ENA */ +#define WM9081_TSENSE_ENA_MASK 0x0080 /* TSENSE_ENA */ +#define WM9081_TSENSE_ENA_SHIFT 7 /* TSENSE_ENA */ +#define WM9081_TSENSE_ENA_WIDTH 1 /* TSENSE_ENA */ +#define WM9081_TEMP_SHUT 0x0040 /* TEMP_SHUT */ +#define WM9081_TEMP_SHUT_MASK 0x0040 /* TEMP_SHUT */ +#define WM9081_TEMP_SHUT_SHIFT 6 /* TEMP_SHUT */ +#define WM9081_TEMP_SHUT_WIDTH 1 /* TEMP_SHUT */ +#define WM9081_LINEOUT_ENA 0x0010 /* LINEOUT_ENA */ +#define WM9081_LINEOUT_ENA_MASK 0x0010 /* LINEOUT_ENA */ +#define WM9081_LINEOUT_ENA_SHIFT 4 /* LINEOUT_ENA */ +#define WM9081_LINEOUT_ENA_WIDTH 1 /* LINEOUT_ENA */ +#define WM9081_SPKPGA_ENA 0x0004 /* SPKPGA_ENA */ +#define WM9081_SPKPGA_ENA_MASK 0x0004 /* SPKPGA_ENA */ +#define WM9081_SPKPGA_ENA_SHIFT 2 /* SPKPGA_ENA */ +#define WM9081_SPKPGA_ENA_WIDTH 1 /* SPKPGA_ENA */ +#define WM9081_SPK_ENA 0x0002 /* SPK_ENA */ +#define WM9081_SPK_ENA_MASK 0x0002 /* SPK_ENA */ +#define WM9081_SPK_ENA_SHIFT 1 /* SPK_ENA */ +#define WM9081_SPK_ENA_WIDTH 1 /* SPK_ENA */ +#define WM9081_DAC_ENA 0x0001 /* DAC_ENA */ +#define WM9081_DAC_ENA_MASK 0x0001 /* DAC_ENA */ +#define WM9081_DAC_ENA_SHIFT 0 /* DAC_ENA */ +#define WM9081_DAC_ENA_WIDTH 1 /* DAC_ENA */ + +/* + * R12 (0x0C) - Clock Control 1 + */ +#define WM9081_CLK_OP_DIV_MASK 0x1C00 /* CLK_OP_DIV - [12:10] */ +#define WM9081_CLK_OP_DIV_SHIFT 10 /* CLK_OP_DIV - [12:10] */ +#define WM9081_CLK_OP_DIV_WIDTH 3 /* CLK_OP_DIV - [12:10] */ +#define WM9081_CLK_TO_DIV_MASK 0x0300 /* CLK_TO_DIV - [9:8] */ +#define WM9081_CLK_TO_DIV_SHIFT 8 /* CLK_TO_DIV - [9:8] */ +#define WM9081_CLK_TO_DIV_WIDTH 2 /* CLK_TO_DIV - [9:8] */ +#define WM9081_MCLKDIV2 0x0080 /* MCLKDIV2 */ +#define WM9081_MCLKDIV2_MASK 0x0080 /* MCLKDIV2 */ +#define WM9081_MCLKDIV2_SHIFT 7 /* MCLKDIV2 */ +#define WM9081_MCLKDIV2_WIDTH 1 /* MCLKDIV2 */ + +/* + * R13 (0x0D) - Clock Control 2 + */ +#define WM9081_CLK_SYS_RATE_MASK 0x00F0 /* CLK_SYS_RATE - [7:4] */ +#define WM9081_CLK_SYS_RATE_SHIFT 4 /* CLK_SYS_RATE - [7:4] */ +#define WM9081_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [7:4] */ +#define WM9081_SAMPLE_RATE_MASK 0x000F /* SAMPLE_RATE - [3:0] */ +#define WM9081_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [3:0] */ +#define WM9081_SAMPLE_RATE_WIDTH 4 /* SAMPLE_RATE - [3:0] */ + +/* + * R14 (0x0E) - Clock Control 3 + */ +#define WM9081_CLK_SRC_SEL 0x2000 /* CLK_SRC_SEL */ +#define WM9081_CLK_SRC_SEL_MASK 0x2000 /* CLK_SRC_SEL */ +#define WM9081_CLK_SRC_SEL_SHIFT 13 /* CLK_SRC_SEL */ +#define WM9081_CLK_SRC_SEL_WIDTH 1 /* CLK_SRC_SEL */ +#define WM9081_CLK_OP_ENA 0x0020 /* CLK_OP_ENA */ +#define WM9081_CLK_OP_ENA_MASK 0x0020 /* CLK_OP_ENA */ +#define WM9081_CLK_OP_ENA_SHIFT 5 /* CLK_OP_ENA */ +#define WM9081_CLK_OP_ENA_WIDTH 1 /* CLK_OP_ENA */ +#define WM9081_CLK_TO_ENA 0x0004 /* CLK_TO_ENA */ +#define WM9081_CLK_TO_ENA_MASK 0x0004 /* CLK_TO_ENA */ +#define WM9081_CLK_TO_ENA_SHIFT 2 /* CLK_TO_ENA */ +#define WM9081_CLK_TO_ENA_WIDTH 1 /* CLK_TO_ENA */ +#define WM9081_CLK_DSP_ENA 0x0002 /* CLK_DSP_ENA */ +#define WM9081_CLK_DSP_ENA_MASK 0x0002 /* CLK_DSP_ENA */ +#define WM9081_CLK_DSP_ENA_SHIFT 1 /* CLK_DSP_ENA */ +#define WM9081_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ +#define WM9081_CLK_SYS_ENA 0x0001 /* CLK_SYS_ENA */ +#define WM9081_CLK_SYS_ENA_MASK 0x0001 /* CLK_SYS_ENA */ +#define WM9081_CLK_SYS_ENA_SHIFT 0 /* CLK_SYS_ENA */ +#define WM9081_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ + +/* + * R16 (0x10) - FLL Control 1 + */ +#define WM9081_FLL_HOLD 0x0008 /* FLL_HOLD */ +#define WM9081_FLL_HOLD_MASK 0x0008 /* FLL_HOLD */ +#define WM9081_FLL_HOLD_SHIFT 3 /* FLL_HOLD */ +#define WM9081_FLL_HOLD_WIDTH 1 /* FLL_HOLD */ +#define WM9081_FLL_FRAC 0x0004 /* FLL_FRAC */ +#define WM9081_FLL_FRAC_MASK 0x0004 /* FLL_FRAC */ +#define WM9081_FLL_FRAC_SHIFT 2 /* FLL_FRAC */ +#define WM9081_FLL_FRAC_WIDTH 1 /* FLL_FRAC */ +#define WM9081_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM9081_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM9081_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM9081_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R17 (0x11) - FLL Control 2 + */ +#define WM9081_FLL_OUTDIV_MASK 0x0700 /* FLL_OUTDIV - [10:8] */ +#define WM9081_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [10:8] */ +#define WM9081_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [10:8] */ +#define WM9081_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */ +#define WM9081_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */ +#define WM9081_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */ +#define WM9081_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM9081_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM9081_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R18 (0x12) - FLL Control 3 + */ +#define WM9081_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */ +#define WM9081_FLL_K_SHIFT 0 /* FLL_K - [15:0] */ +#define WM9081_FLL_K_WIDTH 16 /* FLL_K - [15:0] */ + +/* + * R19 (0x13) - FLL Control 4 + */ +#define WM9081_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */ +#define WM9081_FLL_N_SHIFT 5 /* FLL_N - [14:5] */ +#define WM9081_FLL_N_WIDTH 10 /* FLL_N - [14:5] */ +#define WM9081_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */ +#define WM9081_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */ +#define WM9081_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */ + +/* + * R20 (0x14) - FLL Control 5 + */ +#define WM9081_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM9081_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM9081_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM9081_FLL_CLK_SRC_MASK 0x0003 /* FLL_CLK_SRC - [1:0] */ +#define WM9081_FLL_CLK_SRC_SHIFT 0 /* FLL_CLK_SRC - [1:0] */ +#define WM9081_FLL_CLK_SRC_WIDTH 2 /* FLL_CLK_SRC - [1:0] */ + +/* + * R22 (0x16) - Audio Interface 1 + */ +#define WM9081_AIFDAC_CHAN 0x0040 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_CHAN_MASK 0x0040 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_CHAN_SHIFT 6 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_CHAN_WIDTH 1 /* AIFDAC_CHAN */ +#define WM9081_AIFDAC_TDM_SLOT_MASK 0x0030 /* AIFDAC_TDM_SLOT - [5:4] */ +#define WM9081_AIFDAC_TDM_SLOT_SHIFT 4 /* AIFDAC_TDM_SLOT - [5:4] */ +#define WM9081_AIFDAC_TDM_SLOT_WIDTH 2 /* AIFDAC_TDM_SLOT - [5:4] */ +#define WM9081_AIFDAC_TDM_MODE_MASK 0x000C /* AIFDAC_TDM_MODE - [3:2] */ +#define WM9081_AIFDAC_TDM_MODE_SHIFT 2 /* AIFDAC_TDM_MODE - [3:2] */ +#define WM9081_AIFDAC_TDM_MODE_WIDTH 2 /* AIFDAC_TDM_MODE - [3:2] */ +#define WM9081_DAC_COMP 0x0002 /* DAC_COMP */ +#define WM9081_DAC_COMP_MASK 0x0002 /* DAC_COMP */ +#define WM9081_DAC_COMP_SHIFT 1 /* DAC_COMP */ +#define WM9081_DAC_COMP_WIDTH 1 /* DAC_COMP */ +#define WM9081_DAC_COMPMODE 0x0001 /* DAC_COMPMODE */ +#define WM9081_DAC_COMPMODE_MASK 0x0001 /* DAC_COMPMODE */ +#define WM9081_DAC_COMPMODE_SHIFT 0 /* DAC_COMPMODE */ +#define WM9081_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */ + +/* + * R23 (0x17) - Audio Interface 2 + */ +#define WM9081_AIF_TRIS 0x0200 /* AIF_TRIS */ +#define WM9081_AIF_TRIS_MASK 0x0200 /* AIF_TRIS */ +#define WM9081_AIF_TRIS_SHIFT 9 /* AIF_TRIS */ +#define WM9081_AIF_TRIS_WIDTH 1 /* AIF_TRIS */ +#define WM9081_DAC_DAT_INV 0x0100 /* DAC_DAT_INV */ +#define WM9081_DAC_DAT_INV_MASK 0x0100 /* DAC_DAT_INV */ +#define WM9081_DAC_DAT_INV_SHIFT 8 /* DAC_DAT_INV */ +#define WM9081_DAC_DAT_INV_WIDTH 1 /* DAC_DAT_INV */ +#define WM9081_AIF_BCLK_INV 0x0080 /* AIF_BCLK_INV */ +#define WM9081_AIF_BCLK_INV_MASK 0x0080 /* AIF_BCLK_INV */ +#define WM9081_AIF_BCLK_INV_SHIFT 7 /* AIF_BCLK_INV */ +#define WM9081_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */ +#define WM9081_BCLK_DIR 0x0040 /* BCLK_DIR */ +#define WM9081_BCLK_DIR_MASK 0x0040 /* BCLK_DIR */ +#define WM9081_BCLK_DIR_SHIFT 6 /* BCLK_DIR */ +#define WM9081_BCLK_DIR_WIDTH 1 /* BCLK_DIR */ +#define WM9081_LRCLK_DIR 0x0020 /* LRCLK_DIR */ +#define WM9081_LRCLK_DIR_MASK 0x0020 /* LRCLK_DIR */ +#define WM9081_LRCLK_DIR_SHIFT 5 /* LRCLK_DIR */ +#define WM9081_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */ +#define WM9081_AIF_LRCLK_INV 0x0010 /* AIF_LRCLK_INV */ +#define WM9081_AIF_LRCLK_INV_MASK 0x0010 /* AIF_LRCLK_INV */ +#define WM9081_AIF_LRCLK_INV_SHIFT 4 /* AIF_LRCLK_INV */ +#define WM9081_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */ +#define WM9081_AIF_WL_MASK 0x000C /* AIF_WL - [3:2] */ +#define WM9081_AIF_WL_SHIFT 2 /* AIF_WL - [3:2] */ +#define WM9081_AIF_WL_WIDTH 2 /* AIF_WL - [3:2] */ +#define WM9081_AIF_FMT_MASK 0x0003 /* AIF_FMT - [1:0] */ +#define WM9081_AIF_FMT_SHIFT 0 /* AIF_FMT - [1:0] */ +#define WM9081_AIF_FMT_WIDTH 2 /* AIF_FMT - [1:0] */ + +/* + * R24 (0x18) - Audio Interface 3 + */ +#define WM9081_BCLK_DIV_MASK 0x001F /* BCLK_DIV - [4:0] */ +#define WM9081_BCLK_DIV_SHIFT 0 /* BCLK_DIV - [4:0] */ +#define WM9081_BCLK_DIV_WIDTH 5 /* BCLK_DIV - [4:0] */ + +/* + * R25 (0x19) - Audio Interface 4 + */ +#define WM9081_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */ +#define WM9081_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */ +#define WM9081_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */ + +/* + * R26 (0x1A) - Interrupt Status + */ +#define WM9081_WSEQ_BUSY_EINT 0x0004 /* WSEQ_BUSY_EINT */ +#define WM9081_WSEQ_BUSY_EINT_MASK 0x0004 /* WSEQ_BUSY_EINT */ +#define WM9081_WSEQ_BUSY_EINT_SHIFT 2 /* WSEQ_BUSY_EINT */ +#define WM9081_WSEQ_BUSY_EINT_WIDTH 1 /* WSEQ_BUSY_EINT */ +#define WM9081_TSHUT_EINT 0x0001 /* TSHUT_EINT */ +#define WM9081_TSHUT_EINT_MASK 0x0001 /* TSHUT_EINT */ +#define WM9081_TSHUT_EINT_SHIFT 0 /* TSHUT_EINT */ +#define WM9081_TSHUT_EINT_WIDTH 1 /* TSHUT_EINT */ + +/* + * R27 (0x1B) - Interrupt Status Mask + */ +#define WM9081_IM_WSEQ_BUSY_EINT 0x0004 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_WSEQ_BUSY_EINT_MASK 0x0004 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_WSEQ_BUSY_EINT_SHIFT 2 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_WSEQ_BUSY_EINT_WIDTH 1 /* IM_WSEQ_BUSY_EINT */ +#define WM9081_IM_TSHUT_EINT 0x0001 /* IM_TSHUT_EINT */ +#define WM9081_IM_TSHUT_EINT_MASK 0x0001 /* IM_TSHUT_EINT */ +#define WM9081_IM_TSHUT_EINT_SHIFT 0 /* IM_TSHUT_EINT */ +#define WM9081_IM_TSHUT_EINT_WIDTH 1 /* IM_TSHUT_EINT */ + +/* + * R28 (0x1C) - Interrupt Polarity + */ +#define WM9081_TSHUT_INV 0x0001 /* TSHUT_INV */ +#define WM9081_TSHUT_INV_MASK 0x0001 /* TSHUT_INV */ +#define WM9081_TSHUT_INV_SHIFT 0 /* TSHUT_INV */ +#define WM9081_TSHUT_INV_WIDTH 1 /* TSHUT_INV */ + +/* + * R29 (0x1D) - Interrupt Control + */ +#define WM9081_IRQ_POL 0x8000 /* IRQ_POL */ +#define WM9081_IRQ_POL_MASK 0x8000 /* IRQ_POL */ +#define WM9081_IRQ_POL_SHIFT 15 /* IRQ_POL */ +#define WM9081_IRQ_POL_WIDTH 1 /* IRQ_POL */ +#define WM9081_IRQ_OP_CTRL 0x0001 /* IRQ_OP_CTRL */ +#define WM9081_IRQ_OP_CTRL_MASK 0x0001 /* IRQ_OP_CTRL */ +#define WM9081_IRQ_OP_CTRL_SHIFT 0 /* IRQ_OP_CTRL */ +#define WM9081_IRQ_OP_CTRL_WIDTH 1 /* IRQ_OP_CTRL */ + +/* + * R30 (0x1E) - DAC Digital 1 + */ +#define WM9081_DAC_VOL_MASK 0x00FF /* DAC_VOL - [7:0] */ +#define WM9081_DAC_VOL_SHIFT 0 /* DAC_VOL - [7:0] */ +#define WM9081_DAC_VOL_WIDTH 8 /* DAC_VOL - [7:0] */ + +/* + * R31 (0x1F) - DAC Digital 2 + */ +#define WM9081_DAC_MUTERATE 0x0400 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTERATE_MASK 0x0400 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTERATE_SHIFT 10 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */ +#define WM9081_DAC_MUTEMODE 0x0200 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTEMODE_MASK 0x0200 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTEMODE_SHIFT 9 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTEMODE_WIDTH 1 /* DAC_MUTEMODE */ +#define WM9081_DAC_MUTE 0x0008 /* DAC_MUTE */ +#define WM9081_DAC_MUTE_MASK 0x0008 /* DAC_MUTE */ +#define WM9081_DAC_MUTE_SHIFT 3 /* DAC_MUTE */ +#define WM9081_DAC_MUTE_WIDTH 1 /* DAC_MUTE */ +#define WM9081_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM9081_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM9081_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ + +/* + * R32 (0x20) - DRC 1 + */ +#define WM9081_DRC_ENA 0x8000 /* DRC_ENA */ +#define WM9081_DRC_ENA_MASK 0x8000 /* DRC_ENA */ +#define WM9081_DRC_ENA_SHIFT 15 /* DRC_ENA */ +#define WM9081_DRC_ENA_WIDTH 1 /* DRC_ENA */ +#define WM9081_DRC_STARTUP_GAIN_MASK 0x07C0 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM9081_DRC_STARTUP_GAIN_SHIFT 6 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM9081_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [10:6] */ +#define WM9081_DRC_FF_DLY 0x0020 /* DRC_FF_DLY */ +#define WM9081_DRC_FF_DLY_MASK 0x0020 /* DRC_FF_DLY */ +#define WM9081_DRC_FF_DLY_SHIFT 5 /* DRC_FF_DLY */ +#define WM9081_DRC_FF_DLY_WIDTH 1 /* DRC_FF_DLY */ +#define WM9081_DRC_QR 0x0004 /* DRC_QR */ +#define WM9081_DRC_QR_MASK 0x0004 /* DRC_QR */ +#define WM9081_DRC_QR_SHIFT 2 /* DRC_QR */ +#define WM9081_DRC_QR_WIDTH 1 /* DRC_QR */ +#define WM9081_DRC_ANTICLIP 0x0002 /* DRC_ANTICLIP */ +#define WM9081_DRC_ANTICLIP_MASK 0x0002 /* DRC_ANTICLIP */ +#define WM9081_DRC_ANTICLIP_SHIFT 1 /* DRC_ANTICLIP */ +#define WM9081_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */ + +/* + * R33 (0x21) - DRC 2 + */ +#define WM9081_DRC_ATK_MASK 0xF000 /* DRC_ATK - [15:12] */ +#define WM9081_DRC_ATK_SHIFT 12 /* DRC_ATK - [15:12] */ +#define WM9081_DRC_ATK_WIDTH 4 /* DRC_ATK - [15:12] */ +#define WM9081_DRC_DCY_MASK 0x0F00 /* DRC_DCY - [11:8] */ +#define WM9081_DRC_DCY_SHIFT 8 /* DRC_DCY - [11:8] */ +#define WM9081_DRC_DCY_WIDTH 4 /* DRC_DCY - [11:8] */ +#define WM9081_DRC_QR_THR_MASK 0x00C0 /* DRC_QR_THR - [7:6] */ +#define WM9081_DRC_QR_THR_SHIFT 6 /* DRC_QR_THR - [7:6] */ +#define WM9081_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [7:6] */ +#define WM9081_DRC_QR_DCY_MASK 0x0030 /* DRC_QR_DCY - [5:4] */ +#define WM9081_DRC_QR_DCY_SHIFT 4 /* DRC_QR_DCY - [5:4] */ +#define WM9081_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [5:4] */ +#define WM9081_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */ +#define WM9081_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */ +#define WM9081_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */ +#define WM9081_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM9081_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM9081_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R34 (0x22) - DRC 3 + */ +#define WM9081_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */ +#define WM9081_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */ +#define WM9081_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */ +#define WM9081_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */ +#define WM9081_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */ +#define WM9081_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */ + +/* + * R35 (0x23) - DRC 4 + */ +#define WM9081_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */ +#define WM9081_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */ +#define WM9081_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */ +#define WM9081_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */ +#define WM9081_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */ +#define WM9081_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */ + +/* + * R38 (0x26) - Write Sequencer 1 + */ +#define WM9081_WSEQ_ENA 0x8000 /* WSEQ_ENA */ +#define WM9081_WSEQ_ENA_MASK 0x8000 /* WSEQ_ENA */ +#define WM9081_WSEQ_ENA_SHIFT 15 /* WSEQ_ENA */ +#define WM9081_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM9081_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */ +#define WM9081_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */ +#define WM9081_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */ +#define WM9081_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM9081_WSEQ_START 0x0100 /* WSEQ_START */ +#define WM9081_WSEQ_START_MASK 0x0100 /* WSEQ_START */ +#define WM9081_WSEQ_START_SHIFT 8 /* WSEQ_START */ +#define WM9081_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM9081_WSEQ_START_INDEX_MASK 0x007F /* WSEQ_START_INDEX - [6:0] */ +#define WM9081_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [6:0] */ +#define WM9081_WSEQ_START_INDEX_WIDTH 7 /* WSEQ_START_INDEX - [6:0] */ + +/* + * R39 (0x27) - Write Sequencer 2 + */ +#define WM9081_WSEQ_CURRENT_INDEX_MASK 0x07F0 /* WSEQ_CURRENT_INDEX - [10:4] */ +#define WM9081_WSEQ_CURRENT_INDEX_SHIFT 4 /* WSEQ_CURRENT_INDEX - [10:4] */ +#define WM9081_WSEQ_CURRENT_INDEX_WIDTH 7 /* WSEQ_CURRENT_INDEX - [10:4] */ +#define WM9081_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM9081_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM9081_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM9081_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R40 (0x28) - MW Slave 1 + */ +#define WM9081_SPI_CFG 0x0020 /* SPI_CFG */ +#define WM9081_SPI_CFG_MASK 0x0020 /* SPI_CFG */ +#define WM9081_SPI_CFG_SHIFT 5 /* SPI_CFG */ +#define WM9081_SPI_CFG_WIDTH 1 /* SPI_CFG */ +#define WM9081_SPI_4WIRE 0x0010 /* SPI_4WIRE */ +#define WM9081_SPI_4WIRE_MASK 0x0010 /* SPI_4WIRE */ +#define WM9081_SPI_4WIRE_SHIFT 4 /* SPI_4WIRE */ +#define WM9081_SPI_4WIRE_WIDTH 1 /* SPI_4WIRE */ +#define WM9081_ARA_ENA 0x0008 /* ARA_ENA */ +#define WM9081_ARA_ENA_MASK 0x0008 /* ARA_ENA */ +#define WM9081_ARA_ENA_SHIFT 3 /* ARA_ENA */ +#define WM9081_ARA_ENA_WIDTH 1 /* ARA_ENA */ +#define WM9081_AUTO_INC 0x0002 /* AUTO_INC */ +#define WM9081_AUTO_INC_MASK 0x0002 /* AUTO_INC */ +#define WM9081_AUTO_INC_SHIFT 1 /* AUTO_INC */ +#define WM9081_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +/* + * R42 (0x2A) - EQ 1 + */ +#define WM9081_EQ_B1_GAIN_MASK 0xF800 /* EQ_B1_GAIN - [15:11] */ +#define WM9081_EQ_B1_GAIN_SHIFT 11 /* EQ_B1_GAIN - [15:11] */ +#define WM9081_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [15:11] */ +#define WM9081_EQ_B2_GAIN_MASK 0x07C0 /* EQ_B2_GAIN - [10:6] */ +#define WM9081_EQ_B2_GAIN_SHIFT 6 /* EQ_B2_GAIN - [10:6] */ +#define WM9081_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [10:6] */ +#define WM9081_EQ_B4_GAIN_MASK 0x003E /* EQ_B4_GAIN - [5:1] */ +#define WM9081_EQ_B4_GAIN_SHIFT 1 /* EQ_B4_GAIN - [5:1] */ +#define WM9081_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [5:1] */ +#define WM9081_EQ_ENA 0x0001 /* EQ_ENA */ +#define WM9081_EQ_ENA_MASK 0x0001 /* EQ_ENA */ +#define WM9081_EQ_ENA_SHIFT 0 /* EQ_ENA */ +#define WM9081_EQ_ENA_WIDTH 1 /* EQ_ENA */ + +/* + * R43 (0x2B) - EQ 2 + */ +#define WM9081_EQ_B3_GAIN_MASK 0xF800 /* EQ_B3_GAIN - [15:11] */ +#define WM9081_EQ_B3_GAIN_SHIFT 11 /* EQ_B3_GAIN - [15:11] */ +#define WM9081_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [15:11] */ +#define WM9081_EQ_B5_GAIN_MASK 0x07C0 /* EQ_B5_GAIN - [10:6] */ +#define WM9081_EQ_B5_GAIN_SHIFT 6 /* EQ_B5_GAIN - [10:6] */ +#define WM9081_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [10:6] */ + +/* + * R44 (0x2C) - EQ 3 + */ +#define WM9081_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */ +#define WM9081_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */ +#define WM9081_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */ + +/* + * R45 (0x2D) - EQ 4 + */ +#define WM9081_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */ +#define WM9081_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */ +#define WM9081_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */ + +/* + * R46 (0x2E) - EQ 5 + */ +#define WM9081_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */ +#define WM9081_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */ +#define WM9081_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */ + +/* + * R47 (0x2F) - EQ 6 + */ +#define WM9081_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */ +#define WM9081_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */ +#define WM9081_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */ + +/* + * R48 (0x30) - EQ 7 + */ +#define WM9081_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */ +#define WM9081_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */ +#define WM9081_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */ + +/* + * R49 (0x31) - EQ 8 + */ +#define WM9081_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */ +#define WM9081_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */ +#define WM9081_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */ + +/* + * R50 (0x32) - EQ 9 + */ +#define WM9081_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */ +#define WM9081_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */ +#define WM9081_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */ + +/* + * R51 (0x33) - EQ 10 + */ +#define WM9081_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */ +#define WM9081_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */ +#define WM9081_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */ + +/* + * R52 (0x34) - EQ 11 + */ +#define WM9081_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */ +#define WM9081_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */ +#define WM9081_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */ + +/* + * R53 (0x35) - EQ 12 + */ +#define WM9081_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */ +#define WM9081_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */ +#define WM9081_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */ + +/* + * R54 (0x36) - EQ 13 + */ +#define WM9081_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */ +#define WM9081_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */ +#define WM9081_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */ + +/* + * R55 (0x37) - EQ 14 + */ +#define WM9081_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */ +#define WM9081_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */ +#define WM9081_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */ + +/* + * R56 (0x38) - EQ 15 + */ +#define WM9081_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */ +#define WM9081_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */ +#define WM9081_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */ + +/* + * R57 (0x39) - EQ 16 + */ +#define WM9081_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */ +#define WM9081_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */ +#define WM9081_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */ + +/* + * R58 (0x3A) - EQ 17 + */ +#define WM9081_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */ +#define WM9081_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */ +#define WM9081_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */ + +/* + * R59 (0x3B) - EQ 18 + */ +#define WM9081_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */ +#define WM9081_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */ +#define WM9081_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */ + +/* + * R60 (0x3C) - EQ 19 + */ +#define WM9081_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */ +#define WM9081_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */ +#define WM9081_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */ + +/* + * R61 (0x3D) - EQ 20 + */ +#define WM9081_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */ +#define WM9081_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */ +#define WM9081_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */ + + +#endif -- cgit v1.1 From 0154724d487586241c1ad57cfd348ed2ff2274e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 23 May 2009 00:01:05 +0100 Subject: ASoC: Fix WM9081 PowerPC compiler issues Ensure that we always set a new sysclk when using the FLL in master mode and pick out the correct value for the sample rate in hw_params(). Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 83e3148..86fc57e 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -702,9 +702,10 @@ static int configure_clock(struct snd_soc_codec *codec) * performance. */ for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { target = wm9081->fs * clk_sys_rates[i].ratio; + new_sysclk = target; if (target >= wm9081->bclk && target > 3000000) - new_sysclk = target; + break; } } else if (wm9081->fs) { for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { @@ -1102,7 +1103,8 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, } dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", sample_rates[best].rate); - clk_ctrl2 |= (sample_rates[i].sample_rate << WM9081_SAMPLE_RATE_SHIFT); + clk_ctrl2 |= (sample_rates[best].sample_rate + << WM9081_SAMPLE_RATE_SHIFT); /* BCLK_DIV */ best = 0; -- cgit v1.1 From 3c166c7f1828f226c7f478758bf6c8ce8be1623f Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Sat, 23 May 2009 19:13:07 -0400 Subject: ASoC: Codec for STAC9766 used on the Efika Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/stac9766.c | 470 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/stac9766.h | 21 ++ 4 files changed, 497 insertions(+) create mode 100644 sound/soc/codecs/stac9766.c create mode 100644 sound/soc/codecs/stac9766.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif -- cgit v1.1 From 05e1efa2deb42b1bd548208e5c43f471e2cf0da1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 May 2009 13:32:24 +0100 Subject: ASoC: Fix minor issues in STAC9766 driver Fairly minor issues: - Don't register the DAIs, it's not required for AC97 devices. - Make unexported functions static. - Wrap some excessively long lines. - Undo tab/space breakage. Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 65 ++++++++++++++++++++------------------------- 1 file changed, 29 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 7740cd5..8ad4b7b 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -52,12 +52,14 @@ static const u16 stac9766_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ }; -static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; -static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_record_all_mux[] = {"All analog", + "Analog plus DAC"}; static const char *stac9766_boost1[] = {"0dB", "10dB"}; static const char *stac9766_boost2[] = {"0dB", "20dB"}; static const char *stac9766_stereo_mic[] = {"Off", "On"}; @@ -73,7 +75,8 @@ static const struct soc_enum stac9766_SPDIF_enum = static const struct soc_enum stac9766_popbypass_enum = SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); static const struct soc_enum stac9766_record_all_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, + stac9766_record_all_mux); static const struct soc_enum stac9766_boost1_enum = SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ static const struct soc_enum stac9766_boost2_enum = @@ -89,9 +92,11 @@ static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), - SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, + master_tlv), SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), - SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, + master_tlv), SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), @@ -133,8 +138,8 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), }; -int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) { u16 *cache = codec->reg_cache; @@ -152,7 +157,8 @@ int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, + unsigned int reg) { u16 val = 0, *cache = codec->reg_cache; @@ -176,7 +182,7 @@ unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) } static int ac97_analog_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; @@ -197,7 +203,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, } static int ac97_digital_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; @@ -216,7 +222,7 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, } static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) + int cmd, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; unsigned short vra; @@ -232,7 +238,7 @@ static int ac97_digital_trigger(struct snd_pcm_substream *substream, } static int stac9766_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: /* full On */ @@ -249,7 +255,7 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, return 0; } -int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); @@ -266,7 +272,7 @@ int stac9766_reset(struct snd_soc_codec *codec, int try_warm) } static int stac9766_codec_suspend(struct platform_device *pdev, - pm_message_t state) + pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -303,13 +309,11 @@ reset: return 0; } -static struct snd_soc_dai_ops stac9766_dai_ops_analog = -{ +static struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; -static struct snd_soc_dai_ops stac9766_dai_ops_digital = -{ +static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, .trigger = ac97_digital_trigger, }; @@ -354,7 +358,8 @@ struct snd_soc_dai stac9766_dai[] = { }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, -}}; +} +}; EXPORT_SYMBOL_GPL(stac9766_dai); static int stac9766_codec_probe(struct platform_device *pdev) @@ -371,7 +376,8 @@ static int stac9766_codec_probe(struct platform_device *pdev) codec = socdev->card->codec; mutex_init(&codec->mutex); - codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), + GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; @@ -409,8 +415,8 @@ static int stac9766_codec_probe(struct platform_device *pdev) stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( - stac9766_snd_ac97_controls)); + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, + ARRAY_SIZE(stac9766_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) @@ -444,8 +450,7 @@ static int stac9766_codec_remove(struct platform_device *pdev) return 0; } -struct snd_soc_codec_device soc_codec_dev_stac9766 = -{ +struct snd_soc_codec_device soc_codec_dev_stac9766 = { .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, @@ -453,18 +458,6 @@ struct snd_soc_codec_device soc_codec_dev_stac9766 = }; EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); -static int __init stac9766_modinit(void) -{ - return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); -} -module_init(stac9766_modinit); - -static void __exit stac9766_exit(void) -{ - snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); -} -module_exit(stac9766_exit); - MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl "); MODULE_LICENSE("GPL"); -- cgit v1.1 From 89dd08425273773fd33fc85d48d152c5679b2fb4 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Sat, 23 May 2009 19:12:59 -0400 Subject: ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2s Basic split of mpc5200 DMA code out from i2s into a standalone file. Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/mpc5200_dma.c | 458 +++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 81 +++++++ sound/soc/fsl/mpc5200_psc_i2s.c | 485 +--------------------------------------- 5 files changed, 547 insertions(+), 483 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 9fc9082..dc79bdf 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,8 @@ config SND_SOC_OF_SIMPLE tristate + +config SND_MPC52xx_DMA + tristate # ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers # for the SSI and the Elo DMA controller. You will still need to select @@ -23,6 +26,7 @@ config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM select SND_SOC_OF_SIMPLE + select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index f85134c..7731ef2 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -10,5 +10,7 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-dma-objs := fsl_dma.o obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o +# MPC5200 Platform Support +obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c new file mode 100644 index 0000000..4bae8d6 --- /dev/null +++ b/sound/soc/fsl/mpc5200_dma.c @@ -0,0 +1,458 @@ +/* + * Freescale MPC5200 PSC DMA + * ALSA SoC Platform driver + * + * Copyright (C) 2008 Secret Lab Technologies Ltd. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "mpc5200_dma.h" + +MODULE_AUTHOR("Grant Likely "); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); +MODULE_LICENSE("GPL"); + +/* + * Interrupt handlers + */ +static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) +{ + struct psc_i2s *psc_i2s = _psc_i2s; + struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; + u16 isr; + + isr = in_be16(®s->mpc52xx_psc_isr); + + /* Playback underrun error */ + if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) + psc_i2s->stats.underrun_count++; + + /* Capture overrun error */ + if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) + psc_i2s->stats.overrun_count++; + + out_8(®s->command, 4 << 4); /* reset the error status */ + + return IRQ_HANDLED; +} + +/** + * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer + * @s: pointer to stream private data structure + * + * Enqueues another audio period buffer into the bestcomm queue. + * + * Note: The routine must only be called when there is space available in + * the queue. Otherwise the enqueue will fail and the audio ring buffer + * will get out of sync + */ +static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +{ + struct bcom_bd *bd; + + /* Prepare and enqueue the next buffer descriptor */ + bd = bcom_prepare_next_buffer(s->bcom_task); + bd->status = s->period_bytes; + bd->data[0] = s->period_next_pt; + bcom_submit_next_buffer(s->bcom_task, NULL); + + /* Update for next period */ + s->period_next_pt += s->period_bytes; + if (s->period_next_pt >= s->period_end) + s->period_next_pt = s->period_start; +} + +/* Bestcomm DMA irq handler */ +static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +{ + struct psc_i2s_stream *s = _psc_i2s_stream; + + /* For each finished period, dequeue the completed period buffer + * and enqueue a new one in it's place. */ + while (bcom_buffer_done(s->bcom_task)) { + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + s->period_current_pt += s->period_bytes; + if (s->period_current_pt >= s->period_end) + s->period_current_pt = s->period_start; + psc_i2s_bcom_enqueue_next_buffer(s); + bcom_enable(s->bcom_task); + } + + /* If the stream is active, then also inform the PCM middle layer + * of the period finished event. */ + if (s->active) + snd_pcm_period_elapsed(s->stream); + + return IRQ_HANDLED; +} + +/** + * psc_i2s_startup: create a new substream + * + * This is the first function called when a stream is opened. + * + * If this is the first stream open, then grab the IRQ and program most of + * the PSC registers. + */ +int psc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + int rc; + + dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream); + + if (!psc_i2s->playback.active && + !psc_i2s->capture.active) { + /* Setup the IRQs */ + rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED, + "psc-i2s-status", psc_i2s); + rc |= request_irq(psc_i2s->capture.irq, + &psc_i2s_bcom_irq, IRQF_SHARED, + "psc-i2s-capture", &psc_i2s->capture); + rc |= request_irq(psc_i2s->playback.irq, + &psc_i2s_bcom_irq, IRQF_SHARED, + "psc-i2s-playback", &psc_i2s->playback); + if (rc) { + free_irq(psc_i2s->irq, psc_i2s); + free_irq(psc_i2s->capture.irq, + &psc_i2s->capture); + free_irq(psc_i2s->playback.irq, + &psc_i2s->playback); + return -ENODEV; + } + } + + return 0; +} + +int psc_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +/** + * psc_i2s_trigger: start and stop the DMA transfer. + * + * This function is called by ALSA to start, stop, pause, and resume the DMA + * transfer of data. + */ +int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct psc_i2s_stream *s; + struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; + u16 imr; + u8 psc_cmd; + unsigned long flags; + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)" + " stream_id=%i\n", + substream, cmd, substream->pstr->stream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + s->period_bytes = frames_to_bytes(runtime, + runtime->period_size); + s->period_start = virt_to_phys(runtime->dma_area); + s->period_end = s->period_start + + (s->period_bytes * runtime->periods); + s->period_next_pt = s->period_start; + s->period_current_pt = s->period_start; + s->active = 1; + + /* First; reset everything */ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + out_8(®s->command, MPC52xx_PSC_RST_RX); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + } else { + out_8(®s->command, MPC52xx_PSC_RST_TX); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + } + + /* Next, fill up the bestcomm bd queue and enable DMA. + * This will begin filling the PSC's fifo. */ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + bcom_gen_bd_rx_reset(s->bcom_task); + else + bcom_gen_bd_tx_reset(s->bcom_task); + while (!bcom_queue_full(s->bcom_task)) + psc_i2s_bcom_enqueue_next_buffer(s); + bcom_enable(s->bcom_task); + + /* Due to errata in the i2s mode; need to line up enabling + * the transmitter with a transition on the frame sync + * line */ + + spin_lock_irqsave(&psc_i2s->lock, flags); + /* first make sure it is low */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) + ; + /* then wait for the transition to high */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) + ; + /* Finally, enable the PSC. + * Receiver must always be enabled; even when we only want + * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ + psc_cmd = MPC52xx_PSC_RX_ENABLE; + if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) + psc_cmd |= MPC52xx_PSC_TX_ENABLE; + out_8(®s->command, psc_cmd); + spin_unlock_irqrestore(&psc_i2s->lock, flags); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + /* Turn off the PSC */ + s->active = 0; + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!psc_i2s->playback.active) { + out_8(®s->command, 2 << 4); /* reset rx */ + out_8(®s->command, 3 << 4); /* reset tx */ + out_8(®s->command, 4 << 4); /* reset err */ + } + } else { + out_8(®s->command, 3 << 4); /* reset tx */ + out_8(®s->command, 4 << 4); /* reset err */ + if (!psc_i2s->capture.active) + out_8(®s->command, 2 << 4); /* reset rx */ + } + + bcom_disable(s->bcom_task); + while (!bcom_queue_empty(s->bcom_task)) + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + + break; + + default: + dev_dbg(psc_i2s->dev, "invalid command\n"); + return -EINVAL; + } + + /* Update interrupt enable settings */ + imr = 0; + if (psc_i2s->playback.active) + imr |= MPC52xx_PSC_IMR_TXEMP; + if (psc_i2s->capture.active) + imr |= MPC52xx_PSC_IMR_ORERR; + out_be16(®s->isr_imr.imr, imr); + + return 0; +} + +/** + * psc_i2s_shutdown: shutdown the data transfer on a stream + * + * Shutdown the PSC if there are no other substreams open. + */ +void psc_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream); + + /* + * If this is the last active substream, disable the PSC and release + * the IRQ. + */ + if (!psc_i2s->playback.active && + !psc_i2s->capture.active) { + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); + out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */ + out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */ + out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ + out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ + + /* Release irqs */ + free_irq(psc_i2s->irq, psc_i2s); + free_irq(psc_i2s->capture.irq, &psc_i2s->capture); + free_irq(psc_i2s->playback.irq, &psc_i2s->playback); + } +} + +/* --------------------------------------------------------------------- + * The PSC DMA 'ASoC platform' driver + * + * Can be referenced by an 'ASoC machine' driver + * This driver only deals with the audio bus; it doesn't have any + * interaction with the attached codec + */ + +static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_max = 1024 * 1024, + .period_bytes_min = 32, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 2 * 1024 * 1024, + .fifo_size = 0, +}; + +static int psc_i2s_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_i2s_stream *s; + + dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream); + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware); + + s->stream = substream; + return 0; +} + +static int psc_i2s_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_i2s_stream *s; + + dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream); + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + s->stream = NULL; + return 0; +} + +static snd_pcm_uframes_t +psc_i2s_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_i2s_stream *s; + dma_addr_t count; + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + s = &psc_i2s->capture; + else + s = &psc_i2s->playback; + + count = s->period_current_pt - s->period_start; + + return bytes_to_frames(substream->runtime, count); +} + +static struct snd_pcm_ops psc_i2s_pcm_ops = { + .open = psc_i2s_pcm_open, + .close = psc_i2s_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .pointer = psc_i2s_pcm_pointer, +}; + +static u64 psc_i2s_pcm_dmamask = 0xffffffff; +static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + size_t size = psc_i2s_pcm_hardware.buffer_bytes_max; + int rc = 0; + + dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n", + card, dai, pcm); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &psc_i2s_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (pcm->streams[0].substream) { + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, + &pcm->streams[0].substream->dma_buffer); + if (rc) + goto playback_alloc_err; + } + + if (pcm->streams[1].substream) { + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, + &pcm->streams[1].substream->dma_buffer); + if (rc) + goto capture_alloc_err; + } + + return 0; + + capture_alloc_err: + if (pcm->streams[0].substream) + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + playback_alloc_err: + dev_err(card->dev, "Cannot allocate buffer(s)\n"); + return -ENOMEM; +} + +static void psc_i2s_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_pcm_substream *substream; + int stream; + + dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +struct snd_soc_platform psc_i2s_pcm_soc_platform = { + .name = "mpc5200-psc-audio", + .pcm_ops = &psc_i2s_pcm_ops, + .pcm_new = &psc_i2s_pcm_new, + .pcm_free = &psc_i2s_pcm_free, +}; + diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h new file mode 100644 index 0000000..9a19e8a --- /dev/null +++ b/sound/soc/fsl/mpc5200_dma.h @@ -0,0 +1,81 @@ +/* + * Freescale MPC5200 Audio DMA driver + */ + +#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__ +#define __SOUND_SOC_FSL_MPC5200_DMA_H__ + +/** + * psc_i2s_stream - Data specific to a single stream (playback or capture) + * @active: flag indicating if the stream is active + * @psc_i2s: pointer back to parent psc_i2s data structure + * @bcom_task: bestcomm task structure + * @irq: irq number for bestcomm task + * @period_start: physical address of start of DMA region + * @period_end: physical address of end of DMA region + * @period_next_pt: physical address of next DMA buffer to enqueue + * @period_bytes: size of DMA period in bytes + */ +struct psc_i2s_stream { + int active; + struct psc_i2s *psc_i2s; + struct bcom_task *bcom_task; + int irq; + struct snd_pcm_substream *stream; + dma_addr_t period_start; + dma_addr_t period_end; + dma_addr_t period_next_pt; + dma_addr_t period_current_pt; + int period_bytes; +}; + +/** + * psc_i2s - Private driver data + * @name: short name for this device ("PSC0", "PSC1", etc) + * @psc_regs: pointer to the PSC's registers + * @fifo_regs: pointer to the PSC's FIFO registers + * @irq: IRQ of this PSC + * @dev: struct device pointer + * @dai: the CPU DAI for this device + * @sicr: Base value used in serial interface control register; mode is ORed + * with this value. + * @playback: Playback stream context data + * @capture: Capture stream context data + */ +struct psc_i2s { + char name[32]; + struct mpc52xx_psc __iomem *psc_regs; + struct mpc52xx_psc_fifo __iomem *fifo_regs; + unsigned int irq; + struct device *dev; + struct snd_soc_dai dai; + spinlock_t lock; + u32 sicr; + + /* per-stream data */ + struct psc_i2s_stream playback; + struct psc_i2s_stream capture; + + /* Statistics */ + struct { + int overrun_count; + int underrun_count; + } stats; +}; + + +int psc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +int psc_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +void psc_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai); + +extern struct snd_soc_platform psc_i2s_pcm_soc_platform; + +#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 1111c71..8974b53 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -25,6 +25,8 @@ #include #include +#include "mpc5200_dma.h" + MODULE_AUTHOR("Grant Likely "); MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); MODULE_LICENSE("GPL"); @@ -47,179 +49,6 @@ MODULE_LICENSE("GPL"); SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \ SNDRV_PCM_FMTBIT_S32_BE) -/** - * psc_i2s_stream - Data specific to a single stream (playback or capture) - * @active: flag indicating if the stream is active - * @psc_i2s: pointer back to parent psc_i2s data structure - * @bcom_task: bestcomm task structure - * @irq: irq number for bestcomm task - * @period_start: physical address of start of DMA region - * @period_end: physical address of end of DMA region - * @period_next_pt: physical address of next DMA buffer to enqueue - * @period_bytes: size of DMA period in bytes - */ -struct psc_i2s_stream { - int active; - struct psc_i2s *psc_i2s; - struct bcom_task *bcom_task; - int irq; - struct snd_pcm_substream *stream; - dma_addr_t period_start; - dma_addr_t period_end; - dma_addr_t period_next_pt; - dma_addr_t period_current_pt; - int period_bytes; -}; - -/** - * psc_i2s - Private driver data - * @name: short name for this device ("PSC0", "PSC1", etc) - * @psc_regs: pointer to the PSC's registers - * @fifo_regs: pointer to the PSC's FIFO registers - * @irq: IRQ of this PSC - * @dev: struct device pointer - * @dai: the CPU DAI for this device - * @sicr: Base value used in serial interface control register; mode is ORed - * with this value. - * @playback: Playback stream context data - * @capture: Capture stream context data - */ -struct psc_i2s { - char name[32]; - struct mpc52xx_psc __iomem *psc_regs; - struct mpc52xx_psc_fifo __iomem *fifo_regs; - unsigned int irq; - struct device *dev; - struct snd_soc_dai dai; - spinlock_t lock; - u32 sicr; - - /* per-stream data */ - struct psc_i2s_stream playback; - struct psc_i2s_stream capture; - - /* Statistics */ - struct { - int overrun_count; - int underrun_count; - } stats; -}; - -/* - * Interrupt handlers - */ -static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) -{ - struct psc_i2s *psc_i2s = _psc_i2s; - struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; - u16 isr; - - isr = in_be16(®s->mpc52xx_psc_isr); - - /* Playback underrun error */ - if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) - psc_i2s->stats.underrun_count++; - - /* Capture overrun error */ - if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) - psc_i2s->stats.overrun_count++; - - out_8(®s->command, 4 << 4); /* reset the error status */ - - return IRQ_HANDLED; -} - -/** - * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer - * @s: pointer to stream private data structure - * - * Enqueues another audio period buffer into the bestcomm queue. - * - * Note: The routine must only be called when there is space available in - * the queue. Otherwise the enqueue will fail and the audio ring buffer - * will get out of sync - */ -static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) -{ - struct bcom_bd *bd; - - /* Prepare and enqueue the next buffer descriptor */ - bd = bcom_prepare_next_buffer(s->bcom_task); - bd->status = s->period_bytes; - bd->data[0] = s->period_next_pt; - bcom_submit_next_buffer(s->bcom_task, NULL); - - /* Update for next period */ - s->period_next_pt += s->period_bytes; - if (s->period_next_pt >= s->period_end) - s->period_next_pt = s->period_start; -} - -/* Bestcomm DMA irq handler */ -static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) -{ - struct psc_i2s_stream *s = _psc_i2s_stream; - - /* For each finished period, dequeue the completed period buffer - * and enqueue a new one in it's place. */ - while (bcom_buffer_done(s->bcom_task)) { - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; - psc_i2s_bcom_enqueue_next_buffer(s); - bcom_enable(s->bcom_task); - } - - /* If the stream is active, then also inform the PCM middle layer - * of the period finished event. */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - return IRQ_HANDLED; -} - -/** - * psc_i2s_startup: create a new substream - * - * This is the first function called when a stream is opened. - * - * If this is the first stream open, then grab the IRQ and program most of - * the PSC registers. - */ -static int psc_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - int rc; - - dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream); - - if (!psc_i2s->playback.active && - !psc_i2s->capture.active) { - /* Setup the IRQs */ - rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED, - "psc-i2s-status", psc_i2s); - rc |= request_irq(psc_i2s->capture.irq, - &psc_i2s_bcom_irq, IRQF_SHARED, - "psc-i2s-capture", &psc_i2s->capture); - rc |= request_irq(psc_i2s->playback.irq, - &psc_i2s_bcom_irq, IRQF_SHARED, - "psc-i2s-playback", &psc_i2s->playback); - if (rc) { - free_irq(psc_i2s->irq, psc_i2s); - free_irq(psc_i2s->capture.irq, - &psc_i2s->capture); - free_irq(psc_i2s->playback.irq, - &psc_i2s->playback); - return -ENODEV; - } - } - - return 0; -} - static int psc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -258,164 +87,6 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int psc_i2s_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - return 0; -} - -/** - * psc_i2s_trigger: start and stop the DMA transfer. - * - * This function is called by ALSA to start, stop, pause, and resume the DMA - * transfer of data. - */ -static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_i2s_stream *s; - struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; - u16 imr; - u8 psc_cmd; - unsigned long flags; - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)" - " stream_id=%i\n", - substream, cmd, substream->pstr->stream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - s->period_bytes = frames_to_bytes(runtime, - runtime->period_size); - s->period_start = virt_to_phys(runtime->dma_area); - s->period_end = s->period_start + - (s->period_bytes * runtime->periods); - s->period_next_pt = s->period_start; - s->period_current_pt = s->period_start; - s->active = 1; - - /* First; reset everything */ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - out_8(®s->command, MPC52xx_PSC_RST_RX); - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); - } else { - out_8(®s->command, MPC52xx_PSC_RST_TX); - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); - } - - /* Next, fill up the bestcomm bd queue and enable DMA. - * This will begin filling the PSC's fifo. */ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - bcom_gen_bd_rx_reset(s->bcom_task); - else - bcom_gen_bd_tx_reset(s->bcom_task); - while (!bcom_queue_full(s->bcom_task)) - psc_i2s_bcom_enqueue_next_buffer(s); - bcom_enable(s->bcom_task); - - /* Due to errata in the i2s mode; need to line up enabling - * the transmitter with a transition on the frame sync - * line */ - - spin_lock_irqsave(&psc_i2s->lock, flags); - /* first make sure it is low */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) - ; - /* then wait for the transition to high */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) - ; - /* Finally, enable the PSC. - * Receiver must always be enabled; even when we only want - * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ - psc_cmd = MPC52xx_PSC_RX_ENABLE; - if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) - psc_cmd |= MPC52xx_PSC_TX_ENABLE; - out_8(®s->command, psc_cmd); - spin_unlock_irqrestore(&psc_i2s->lock, flags); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - /* Turn off the PSC */ - s->active = 0; - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (!psc_i2s->playback.active) { - out_8(®s->command, 2 << 4); /* reset rx */ - out_8(®s->command, 3 << 4); /* reset tx */ - out_8(®s->command, 4 << 4); /* reset err */ - } - } else { - out_8(®s->command, 3 << 4); /* reset tx */ - out_8(®s->command, 4 << 4); /* reset err */ - if (!psc_i2s->capture.active) - out_8(®s->command, 2 << 4); /* reset rx */ - } - - bcom_disable(s->bcom_task); - while (!bcom_queue_empty(s->bcom_task)) - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - break; - - default: - dev_dbg(psc_i2s->dev, "invalid command\n"); - return -EINVAL; - } - - /* Update interrupt enable settings */ - imr = 0; - if (psc_i2s->playback.active) - imr |= MPC52xx_PSC_IMR_TXEMP; - if (psc_i2s->capture.active) - imr |= MPC52xx_PSC_IMR_ORERR; - out_be16(®s->isr_imr.imr, imr); - - return 0; -} - -/** - * psc_i2s_shutdown: shutdown the data transfer on a stream - * - * Shutdown the PSC if there are no other substreams open. - */ -static void psc_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - - dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream); - - /* - * If this is the last active substream, disable the PSC and release - * the IRQ. - */ - if (!psc_i2s->playback.active && - !psc_i2s->capture.active) { - - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); - out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */ - out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */ - out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ - - /* Release irqs */ - free_irq(psc_i2s->irq, psc_i2s); - free_irq(psc_i2s->capture.irq, &psc_i2s->capture); - free_irq(psc_i2s->playback.irq, &psc_i2s->playback); - } -} - /** * psc_i2s_set_sysclk: set the clock frequency and direction * @@ -495,158 +166,6 @@ static struct snd_soc_dai psc_i2s_dai_template = { }; /* --------------------------------------------------------------------- - * The PSC I2S 'ASoC platform' driver - * - * Can be referenced by an 'ASoC machine' driver - * This driver only deals with the audio bus; it doesn't have any - * interaction with the attached codec - */ - -static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_BATCH, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .period_bytes_max = 1024 * 1024, - .period_bytes_min = 32, - .periods_min = 2, - .periods_max = 256, - .buffer_bytes_max = 2 * 1024 * 1024, - .fifo_size = 0, -}; - -static int psc_i2s_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; - - dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware); - - s->stream = substream; - return 0; -} - -static int psc_i2s_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; - - dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - s->stream = NULL; - return 0; -} - -static snd_pcm_uframes_t -psc_i2s_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; - dma_addr_t count; - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; - else - s = &psc_i2s->playback; - - count = s->period_current_pt - s->period_start; - - return bytes_to_frames(substream->runtime, count); -} - -static struct snd_pcm_ops psc_i2s_pcm_ops = { - .open = psc_i2s_pcm_open, - .close = psc_i2s_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .pointer = psc_i2s_pcm_pointer, -}; - -static u64 psc_i2s_pcm_dmamask = 0xffffffff; -static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - size_t size = psc_i2s_pcm_hardware.buffer_bytes_max; - int rc = 0; - - dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n", - card, dai, pcm); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &psc_i2s_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (pcm->streams[0].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, - &pcm->streams[0].substream->dma_buffer); - if (rc) - goto playback_alloc_err; - } - - if (pcm->streams[1].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, - &pcm->streams[1].substream->dma_buffer); - if (rc) - goto capture_alloc_err; - } - - return 0; - - capture_alloc_err: - if (pcm->streams[0].substream) - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - playback_alloc_err: - dev_err(card->dev, "Cannot allocate buffer(s)\n"); - return -ENOMEM; -} - -static void psc_i2s_pcm_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_pcm_substream *substream; - int stream; - - dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (substream) { - snd_dma_free_pages(&substream->dma_buffer); - substream->dma_buffer.area = NULL; - substream->dma_buffer.addr = 0; - } - } -} - -struct snd_soc_platform psc_i2s_pcm_soc_platform = { - .name = "mpc5200-psc-audio", - .pcm_ops = &psc_i2s_pcm_ops, - .pcm_new = &psc_i2s_pcm_new, - .pcm_free = &psc_i2s_pcm_free, -}; - -/* --------------------------------------------------------------------- * Sysfs attributes for debugging */ -- cgit v1.1 From cebe77674cab51a9ff1deaa077ab74aff3996764 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Sat, 23 May 2009 19:13:01 -0400 Subject: ASoC: Rename the PSC functions to DMA Rename the functions in the mpc5200 DMA file from i2s based names to dma ones to reflect the file they are in. Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 194 ++++++++++++++++++++-------------------- sound/soc/fsl/mpc5200_dma.h | 26 +++--- sound/soc/fsl/mpc5200_psc_i2s.c | 160 ++++++++++++++++----------------- 3 files changed, 190 insertions(+), 190 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 4bae8d6..6850392 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -34,21 +34,21 @@ MODULE_LICENSE("GPL"); /* * Interrupt handlers */ -static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) +static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) { - struct psc_i2s *psc_i2s = _psc_i2s; - struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; + struct psc_dma *psc_dma = _psc_dma; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 isr; isr = in_be16(®s->mpc52xx_psc_isr); /* Playback underrun error */ - if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) - psc_i2s->stats.underrun_count++; + if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP)) + psc_dma->stats.underrun_count++; /* Capture overrun error */ - if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) - psc_i2s->stats.overrun_count++; + if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) + psc_dma->stats.overrun_count++; out_8(®s->command, 4 << 4); /* reset the error status */ @@ -56,7 +56,7 @@ static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) } /** - * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer + * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer * @s: pointer to stream private data structure * * Enqueues another audio period buffer into the bestcomm queue. @@ -65,7 +65,7 @@ static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s) * the queue. Otherwise the enqueue will fail and the audio ring buffer * will get out of sync */ -static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) +static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) { struct bcom_bd *bd; @@ -82,9 +82,9 @@ static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s) } /* Bestcomm DMA irq handler */ -static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) +static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { - struct psc_i2s_stream *s = _psc_i2s_stream; + struct psc_dma_stream *s = _psc_dma_stream; /* For each finished period, dequeue the completed period buffer * and enqueue a new one in it's place. */ @@ -93,7 +93,7 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) s->period_current_pt += s->period_bytes; if (s->period_current_pt >= s->period_end) s->period_current_pt = s->period_start; - psc_i2s_bcom_enqueue_next_buffer(s); + psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); } @@ -106,39 +106,39 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) } /** - * psc_i2s_startup: create a new substream + * psc_dma_startup: create a new substream * * This is the first function called when a stream is opened. * * If this is the first stream open, then grab the IRQ and program most of * the PSC registers. */ -int psc_i2s_startup(struct snd_pcm_substream *substream, +int psc_dma_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; int rc; - dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream); + dev_dbg(psc_dma->dev, "psc_dma_startup(substream=%p)\n", substream); - if (!psc_i2s->playback.active && - !psc_i2s->capture.active) { + if (!psc_dma->playback.active && + !psc_dma->capture.active) { /* Setup the IRQs */ - rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED, - "psc-i2s-status", psc_i2s); - rc |= request_irq(psc_i2s->capture.irq, - &psc_i2s_bcom_irq, IRQF_SHARED, - "psc-i2s-capture", &psc_i2s->capture); - rc |= request_irq(psc_i2s->playback.irq, - &psc_i2s_bcom_irq, IRQF_SHARED, - "psc-i2s-playback", &psc_i2s->playback); + rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, + "psc-dma-status", psc_dma); + rc |= request_irq(psc_dma->capture.irq, + &psc_dma_bcom_irq, IRQF_SHARED, + "psc-dma-capture", &psc_dma->capture); + rc |= request_irq(psc_dma->playback.irq, + &psc_dma_bcom_irq, IRQF_SHARED, + "psc-dma-playback", &psc_dma->playback); if (rc) { - free_irq(psc_i2s->irq, psc_i2s); - free_irq(psc_i2s->capture.irq, - &psc_i2s->capture); - free_irq(psc_i2s->playback.irq, - &psc_i2s->playback); + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, + &psc_dma->capture); + free_irq(psc_dma->playback.irq, + &psc_dma->playback); return -ENODEV; } } @@ -146,7 +146,7 @@ int psc_i2s_startup(struct snd_pcm_substream *substream, return 0; } -int psc_i2s_hw_free(struct snd_pcm_substream *substream, +int psc_dma_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { snd_pcm_set_runtime_buffer(substream, NULL); @@ -154,29 +154,29 @@ int psc_i2s_hw_free(struct snd_pcm_substream *substream, } /** - * psc_i2s_trigger: start and stop the DMA transfer. + * psc_dma_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ -int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, +int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_i2s_stream *s; - struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; + struct psc_dma_stream *s; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; u8 psc_cmd; unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; + s = &psc_dma->capture; else - s = &psc_i2s->playback; + s = &psc_dma->playback; - dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)" + dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)" " stream_id=%i\n", substream, cmd, substream->pstr->stream); @@ -207,14 +207,14 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else bcom_gen_bd_tx_reset(s->bcom_task); while (!bcom_queue_full(s->bcom_task)) - psc_i2s_bcom_enqueue_next_buffer(s); + psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); - /* Due to errata in the i2s mode; need to line up enabling + /* Due to errata in the dma mode; need to line up enabling * the transmitter with a transition on the frame sync * line */ - spin_lock_irqsave(&psc_i2s->lock, flags); + spin_lock_irqsave(&psc_dma->lock, flags); /* first make sure it is low */ while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) ; @@ -228,7 +228,7 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) psc_cmd |= MPC52xx_PSC_TX_ENABLE; out_8(®s->command, psc_cmd); - spin_unlock_irqrestore(&psc_i2s->lock, flags); + spin_unlock_irqrestore(&psc_dma->lock, flags); break; @@ -236,7 +236,7 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, /* Turn off the PSC */ s->active = 0; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (!psc_i2s->playback.active) { + if (!psc_dma->playback.active) { out_8(®s->command, 2 << 4); /* reset rx */ out_8(®s->command, 3 << 4); /* reset tx */ out_8(®s->command, 4 << 4); /* reset err */ @@ -244,7 +244,7 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, } else { out_8(®s->command, 3 << 4); /* reset tx */ out_8(®s->command, 4 << 4); /* reset err */ - if (!psc_i2s->capture.active) + if (!psc_dma->capture.active) out_8(®s->command, 2 << 4); /* reset rx */ } @@ -255,15 +255,15 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, break; default: - dev_dbg(psc_i2s->dev, "invalid command\n"); + dev_dbg(psc_dma->dev, "invalid command\n"); return -EINVAL; } /* Update interrupt enable settings */ imr = 0; - if (psc_i2s->playback.active) + if (psc_dma->playback.active) imr |= MPC52xx_PSC_IMR_TXEMP; - if (psc_i2s->capture.active) + if (psc_dma->capture.active) imr |= MPC52xx_PSC_IMR_ORERR; out_be16(®s->isr_imr.imr, imr); @@ -271,36 +271,36 @@ int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, } /** - * psc_i2s_shutdown: shutdown the data transfer on a stream + * psc_dma_shutdown: shutdown the data transfer on a stream * * Shutdown the PSC if there are no other substreams open. */ -void psc_i2s_shutdown(struct snd_pcm_substream *substream, +void psc_dma_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; - dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream); + dev_dbg(psc_dma->dev, "psc_dma_shutdown(substream=%p)\n", substream); /* * If this is the last active substream, disable the PSC and release * the IRQ. */ - if (!psc_i2s->playback.active && - !psc_i2s->capture.active) { + if (!psc_dma->playback.active && + !psc_dma->capture.active) { /* Disable all interrupts and reset the PSC */ - out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); - out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */ - out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */ - out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ + out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); + out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset tx */ + out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */ + out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ + out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ /* Release irqs */ - free_irq(psc_i2s->irq, psc_i2s); - free_irq(psc_i2s->capture.irq, &psc_i2s->capture); - free_irq(psc_i2s->playback.irq, &psc_i2s->playback); + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); } } @@ -312,7 +312,7 @@ void psc_i2s_shutdown(struct snd_pcm_substream *substream, * interaction with the attached codec */ -static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { +static const struct snd_pcm_hardware psc_dma_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH, @@ -330,80 +330,80 @@ static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { .fifo_size = 0, }; -static int psc_i2s_pcm_open(struct snd_pcm_substream *substream) +static int psc_dma_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + struct psc_dma_stream *s; - dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream); + dev_dbg(psc_dma->dev, "psc_dma_pcm_open(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; + s = &psc_dma->capture; else - s = &psc_i2s->playback; + s = &psc_dma->playback; - snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &psc_dma_pcm_hardware); s->stream = substream; return 0; } -static int psc_i2s_pcm_close(struct snd_pcm_substream *substream) +static int psc_dma_pcm_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + struct psc_dma_stream *s; - dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream); + dev_dbg(psc_dma->dev, "psc_dma_pcm_close(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; + s = &psc_dma->capture; else - s = &psc_i2s->playback; + s = &psc_dma->playback; s->stream = NULL; return 0; } static snd_pcm_uframes_t -psc_i2s_pcm_pointer(struct snd_pcm_substream *substream) +psc_dma_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; - struct psc_i2s_stream *s; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + struct psc_dma_stream *s; dma_addr_t count; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_i2s->capture; + s = &psc_dma->capture; else - s = &psc_i2s->playback; + s = &psc_dma->playback; count = s->period_current_pt - s->period_start; return bytes_to_frames(substream->runtime, count); } -static struct snd_pcm_ops psc_i2s_pcm_ops = { - .open = psc_i2s_pcm_open, - .close = psc_i2s_pcm_close, +static struct snd_pcm_ops psc_dma_pcm_ops = { + .open = psc_dma_pcm_open, + .close = psc_dma_pcm_close, .ioctl = snd_pcm_lib_ioctl, - .pointer = psc_i2s_pcm_pointer, + .pointer = psc_dma_pcm_pointer, }; -static u64 psc_i2s_pcm_dmamask = 0xffffffff; -static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static u64 psc_dma_pcm_dmamask = 0xffffffff; +static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; - size_t size = psc_i2s_pcm_hardware.buffer_bytes_max; + size_t size = psc_dma_pcm_hardware.buffer_bytes_max; int rc = 0; - dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n", + dev_dbg(rtd->socdev->dev, "psc_dma_pcm_new(card=%p, dai=%p, pcm=%p)\n", card, dai, pcm); if (!card->dev->dma_mask) - card->dev->dma_mask = &psc_i2s_pcm_dmamask; + card->dev->dma_mask = &psc_dma_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; @@ -431,13 +431,13 @@ static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, return -ENOMEM; } -static void psc_i2s_pcm_free(struct snd_pcm *pcm) +static void psc_dma_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_pcm_substream *substream; int stream; - dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm); + dev_dbg(rtd->socdev->dev, "psc_dma_pcm_free(pcm=%p)\n", pcm); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -449,10 +449,10 @@ static void psc_i2s_pcm_free(struct snd_pcm *pcm) } } -struct snd_soc_platform psc_i2s_pcm_soc_platform = { +struct snd_soc_platform psc_dma_pcm_soc_platform = { .name = "mpc5200-psc-audio", - .pcm_ops = &psc_i2s_pcm_ops, - .pcm_new = &psc_i2s_pcm_new, - .pcm_free = &psc_i2s_pcm_free, + .pcm_ops = &psc_dma_pcm_ops, + .pcm_new = &psc_dma_pcm_new, + .pcm_free = &psc_dma_pcm_free, }; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 9a19e8a..a33232c 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -6,9 +6,9 @@ #define __SOUND_SOC_FSL_MPC5200_DMA_H__ /** - * psc_i2s_stream - Data specific to a single stream (playback or capture) + * psc_dma_stream - Data specific to a single stream (playback or capture) * @active: flag indicating if the stream is active - * @psc_i2s: pointer back to parent psc_i2s data structure + * @psc_dma: pointer back to parent psc_dma data structure * @bcom_task: bestcomm task structure * @irq: irq number for bestcomm task * @period_start: physical address of start of DMA region @@ -16,9 +16,9 @@ * @period_next_pt: physical address of next DMA buffer to enqueue * @period_bytes: size of DMA period in bytes */ -struct psc_i2s_stream { +struct psc_dma_stream { int active; - struct psc_i2s *psc_i2s; + struct psc_dma *psc_dma; struct bcom_task *bcom_task; int irq; struct snd_pcm_substream *stream; @@ -30,7 +30,7 @@ struct psc_i2s_stream { }; /** - * psc_i2s - Private driver data + * psc_dma - Private driver data * @name: short name for this device ("PSC0", "PSC1", etc) * @psc_regs: pointer to the PSC's registers * @fifo_regs: pointer to the PSC's FIFO registers @@ -42,7 +42,7 @@ struct psc_i2s_stream { * @playback: Playback stream context data * @capture: Capture stream context data */ -struct psc_i2s { +struct psc_dma { char name[32]; struct mpc52xx_psc __iomem *psc_regs; struct mpc52xx_psc_fifo __iomem *fifo_regs; @@ -53,8 +53,8 @@ struct psc_i2s { u32 sicr; /* per-stream data */ - struct psc_i2s_stream playback; - struct psc_i2s_stream capture; + struct psc_dma_stream playback; + struct psc_dma_stream capture; /* Statistics */ struct { @@ -64,18 +64,18 @@ struct psc_i2s { }; -int psc_i2s_startup(struct snd_pcm_substream *substream, +int psc_dma_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); -int psc_i2s_hw_free(struct snd_pcm_substream *substream, +int psc_dma_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); -void psc_i2s_shutdown(struct snd_pcm_substream *substream, +void psc_dma_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); -int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, +int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai); -extern struct snd_soc_platform psc_i2s_pcm_soc_platform; +extern struct snd_soc_platform psc_dma_pcm_soc_platform; #endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 8974b53..12a7917 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -54,10 +54,10 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; u32 mode; - dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i\n", __func__, substream, params_period_size(params), params_period_bytes(params), params_periods(params), @@ -77,10 +77,10 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, mode = MPC52xx_PSC_SICR_SIM_CODEC_32; break; default: - dev_dbg(psc_i2s->dev, "invalid format\n"); + dev_dbg(psc_dma->dev, "invalid format\n"); return -EINVAL; } - out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode); + out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); @@ -104,8 +104,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct psc_i2s *psc_i2s = cpu_dai->private_data; - dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", + struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", cpu_dai, dir); return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; } @@ -123,8 +123,8 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, */ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { - struct psc_i2s *psc_i2s = cpu_dai->private_data; - dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n", + struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n", cpu_dai, format); return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -140,11 +140,11 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) * psc_i2s_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai_ops psc_i2s_dai_ops = { - .startup = psc_i2s_startup, + .startup = psc_dma_startup, .hw_params = psc_i2s_hw_params, - .hw_free = psc_i2s_hw_free, - .shutdown = psc_i2s_shutdown, - .trigger = psc_i2s_trigger, + .hw_free = psc_dma_hw_free, + .shutdown = psc_dma_shutdown, + .trigger = psc_dma_trigger, .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }; @@ -172,24 +172,24 @@ static struct snd_soc_dai psc_i2s_dai_template = { static ssize_t psc_i2s_status_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + struct psc_dma *psc_dma = dev_get_drvdata(dev); return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x " "tfnum=%i tfstat=0x%.4x\n", - in_be16(&psc_i2s->psc_regs->sr_csr.status), - in_be32(&psc_i2s->psc_regs->sicr), - in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff, - in_be16(&psc_i2s->fifo_regs->rfstat), - in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff, - in_be16(&psc_i2s->fifo_regs->tfstat)); + in_be16(&psc_dma->psc_regs->sr_csr.status), + in_be32(&psc_dma->psc_regs->sicr), + in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff, + in_be16(&psc_dma->fifo_regs->rfstat), + in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff, + in_be16(&psc_dma->fifo_regs->tfstat)); } -static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name) +static int *psc_i2s_get_stat_attr(struct psc_dma *psc_dma, const char *name) { if (strcmp(name, "playback_underrun") == 0) - return &psc_i2s->stats.underrun_count; + return &psc_dma->stats.underrun_count; if (strcmp(name, "capture_overrun") == 0) - return &psc_i2s->stats.overrun_count; + return &psc_dma->stats.overrun_count; return NULL; } @@ -197,10 +197,10 @@ static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name) static ssize_t psc_i2s_stat_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + struct psc_dma *psc_dma = dev_get_drvdata(dev); int *attrib; - attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); + attrib = psc_i2s_get_stat_attr(psc_dma, attr->attr.name); if (!attrib) return 0; @@ -212,10 +212,10 @@ static ssize_t psc_i2s_stat_store(struct device *dev, const char *buf, size_t count) { - struct psc_i2s *psc_i2s = dev_get_drvdata(dev); + struct psc_dma *psc_dma = dev_get_drvdata(dev); int *attrib; - attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name); + attrib = psc_i2s_get_stat_attr(psc_dma, attr->attr.name); if (!attrib) return 0; @@ -238,7 +238,7 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, const struct of_device_id *match) { phys_addr_t fifo; - struct psc_i2s *psc_i2s; + struct psc_dma *psc_dma; struct resource res; int size, psc_id, irq, rc; const __be32 *prop; @@ -265,56 +265,56 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, } /* Allocate and initialize the driver private data */ - psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL); - if (!psc_i2s) { + psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); + if (!psc_dma) { iounmap(regs); return -ENOMEM; } - spin_lock_init(&psc_i2s->lock); - psc_i2s->irq = irq; - psc_i2s->psc_regs = regs; - psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs; - psc_i2s->dev = &op->dev; - psc_i2s->playback.psc_i2s = psc_i2s; - psc_i2s->capture.psc_i2s = psc_i2s; - snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1); + spin_lock_init(&psc_dma->lock); + psc_dma->irq = irq; + psc_dma->psc_regs = regs; + psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; + psc_dma->dev = &op->dev; + psc_dma->playback.psc_dma = psc_dma; + psc_dma->capture.psc_dma = psc_dma; + snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_id+1); /* Fill out the CPU DAI structure */ - memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai); - psc_i2s->dai.private_data = psc_i2s; - psc_i2s->dai.name = psc_i2s->name; - psc_i2s->dai.id = psc_id; + memcpy(&psc_dma->dai, &psc_i2s_dai_template, sizeof psc_dma->dai); + psc_dma->dai.private_data = psc_dma; + psc_dma->dai.name = psc_dma->name; + psc_dma->dai.id = psc_id; /* Find the address of the fifo data registers and setup the * DMA tasks */ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); - psc_i2s->capture.bcom_task = + psc_dma->capture.bcom_task = bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); - psc_i2s->playback.bcom_task = + psc_dma->playback.bcom_task = bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); - if (!psc_i2s->capture.bcom_task || - !psc_i2s->playback.bcom_task) { + if (!psc_dma->capture.bcom_task || + !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); iounmap(regs); - kfree(psc_i2s); + kfree(psc_dma); return -ENODEV; } /* Disable all interrupts and reset the PSC */ - out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0); - out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */ - out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */ - out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */ + out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); + out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset transmitter */ + out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset receiver */ + out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ + out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ /* Configure the serial interface mode; defaulting to CODEC8 mode */ - psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | + psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL; if (of_get_property(op->node, "fsl,cellslave", NULL)) - psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | + psc_dma->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | MPC52xx_PSC_SICR_GENCLK; - out_be32(&psc_i2s->psc_regs->sicr, - psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); + out_be32(&psc_dma->psc_regs->sicr, + psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); /* Check for the codec handle. If it is not present then we * are done */ @@ -325,54 +325,54 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, * First write: RxRdy (FIFO Alarm) generates rx FIFO irq * Second write: register Normal mode for non loopback */ - out_8(&psc_i2s->psc_regs->mode, 0); - out_8(&psc_i2s->psc_regs->mode, 0); + out_8(&psc_dma->psc_regs->mode, 0); + out_8(&psc_dma->psc_regs->mode, 0); /* Set the TX and RX fifo alarm thresholds */ - out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100); - out_8(&psc_i2s->fifo_regs->rfcntl, 0x4); - out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100); - out_8(&psc_i2s->fifo_regs->tfcntl, 0x7); + out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); + out_8(&psc_dma->fifo_regs->rfcntl, 0x4); + out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); + out_8(&psc_dma->fifo_regs->tfcntl, 0x7); /* Lookup the IRQ numbers */ - psc_i2s->playback.irq = - bcom_get_task_irq(psc_i2s->playback.bcom_task); - psc_i2s->capture.irq = - bcom_get_task_irq(psc_i2s->capture.bcom_task); + psc_dma->playback.irq = + bcom_get_task_irq(psc_dma->playback.bcom_task); + psc_dma->capture.irq = + bcom_get_task_irq(psc_dma->capture.bcom_task); /* Save what we've done so it can be found again later */ - dev_set_drvdata(&op->dev, psc_i2s); + dev_set_drvdata(&op->dev, psc_dma); /* Register the SYSFS files */ - rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + rc = device_create_file(psc_dma->dev, &dev_attr_status); + rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun); if (rc) - dev_info(psc_i2s->dev, "error creating sysfs files\n"); + dev_info(psc_dma->dev, "error creating sysfs files\n"); - snd_soc_register_platform(&psc_i2s_pcm_soc_platform); + snd_soc_register_platform(&psc_dma_pcm_soc_platform); /* Tell the ASoC OF helpers about it */ - of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node, - &psc_i2s->dai); + of_snd_soc_register_platform(&psc_dma_pcm_soc_platform, op->node, + &psc_dma->dai); return 0; } static int __devexit psc_i2s_of_remove(struct of_device *op) { - struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev); + struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); dev_dbg(&op->dev, "psc_i2s_remove()\n"); - snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform); + snd_soc_unregister_platform(&psc_dma_pcm_soc_platform); - bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task); - bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task); + bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); + bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); - iounmap(psc_i2s->psc_regs); - iounmap(psc_i2s->fifo_regs); - kfree(psc_i2s); + iounmap(psc_dma->psc_regs); + iounmap(psc_dma->fifo_regs); + kfree(psc_dma); dev_set_drvdata(&op->dev, NULL); return 0; -- cgit v1.1 From 0bc53a67ac831ec84f730a657dbcadd80a589ef5 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Sat, 23 May 2009 19:13:03 -0400 Subject: ASoC: Add a few more mpc5200 PSC defines Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown --- arch/powerpc/include/asm/mpc52xx_psc.h | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) -- cgit v1.1 From 345c03ef0f2bd0c933a1eca27732af7edf5e8d4d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 May 2009 10:05:00 +0200 Subject: sound: oxygen: reset DMA when stream is closed When a PCM stream is closed, flush the corresponding DMA channel. Otherwise, the DMA controller would continue to output the last sample which would result in a DC offset on the output. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_pcm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index c262049..3b5ca70 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -487,10 +487,14 @@ static int oxygen_hw_free(struct snd_pcm_substream *substream) { struct oxygen *chip = snd_pcm_substream_chip(substream); unsigned int channel = oxygen_substream_channel(substream); + unsigned int channel_mask = 1 << channel; spin_lock_irq(&chip->reg_lock); - chip->interrupt_mask &= ~(1 << channel); + chip->interrupt_mask &= ~channel_mask; oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + + oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); + oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask); spin_unlock_irq(&chip->reg_lock); return snd_pcm_lib_free_pages(substream); -- cgit v1.1 From 53bb705d12e0f642c131cdcab2c8e3be7364e505 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 May 2009 10:05:43 +0200 Subject: sound: virtuoso: add another DX PCI ID Add another PCI ID for a second revision of the Xonar DX. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bc5ce11..ebc4e85 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -188,6 +188,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV }, + { OXYGEN_PCI_SUBID(0x1043, 0x8327), .driver_data = MODEL_DX }, { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, { OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, -- cgit v1.1 From b990ae963a3f80a659b30562c1e3214b386ecce3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 May 2009 10:06:22 +0200 Subject: sound: virtuoso: enable HDAV S/PDIF input The Xonar HDAV1.3 has a digital input jack, so enable the corresponding device. This is not related to the HDMI stuff, which stays unsupported. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index ebc4e85..71e38f4 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -1022,7 +1022,8 @@ static const struct oxygen_model model_xonar_hdav = { .model_data_size = sizeof(struct xonar_data), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, .dac_channels = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, -- cgit v1.1 From 04f9890df1bad2115665b7027e664aaffa44088d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 25 May 2009 10:11:29 +0200 Subject: sound: virtuoso: add Xonar Essence ST support Add support for the Asus Xonar Essence ST and its daughterboard. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 +- sound/pci/Kconfig | 4 +- sound/pci/oxygen/virtuoso.c | 60 +++++++++++++++++-------- 3 files changed, 45 insertions(+), 22 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 012858d..68ef84f 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1859,7 +1859,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), and Essence STX. + i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST + (Deluxe) and Essence STX. This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93422e3..699c280 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -764,8 +764,8 @@ config SND_VIRTUOSO select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, and - Essence STX. + Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, + Essence ST (Deluxe), and Essence STX. Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 71e38f4..bf971f7 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -113,8 +113,8 @@ */ /* - * Xonar Essence STX - * ----------------- + * Xonar Essence ST (Deluxe)/STX + * ----------------------------- * * CMI8788: * @@ -180,6 +180,8 @@ enum { MODEL_DX, MODEL_HDAV, /* without daughterboard */ MODEL_HDAV_H6, /* with H6 daughterboard */ + MODEL_ST, + MODEL_ST_H6, MODEL_STX, }; @@ -191,6 +193,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x8327), .driver_data = MODEL_DX }, { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, { OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX }, + { OXYGEN_PCI_SUBID(0x1043, 0x835d), .driver_data = MODEL_ST }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; @@ -211,9 +214,9 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_DX_FRONT_PANEL 0x0002 #define GPIO_DX_INPUT_ROUTE 0x0100 -#define GPIO_HDAV_DB_MASK 0x0030 -#define GPIO_HDAV_DB_H6 0x0000 -#define GPIO_HDAV_DB_XX 0x0020 +#define GPIO_DB_MASK 0x0030 +#define GPIO_DB_H6 0x0000 +#define GPIO_DB_XX 0x0020 #define GPIO_ST_HP_REAR 0x0002 #define GPIO_ST_HP 0x0080 @@ -531,7 +534,7 @@ static void xonar_hdav_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } -static void xonar_stx_init(struct oxygen *chip) +static void xonar_st_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -540,12 +543,11 @@ static void xonar_stx_init(struct oxygen *chip) OXYGEN_2WIRE_INTERRUPT_MASK | OXYGEN_2WIRE_SPEED_FAST); + if (chip->model.private_data == MODEL_ST_H6) + chip->model.dac_channels = 8; data->anti_pop_delay = 100; - data->dacs = 1; + data->dacs = chip->model.private_data == MODEL_ST_H6 ? 4 : 1; data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; data->pcm1796_oversampling = PCM1796_OS_64; pcm1796_init(chip); @@ -561,6 +563,17 @@ static void xonar_stx_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void xonar_stx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + + xonar_st_init(chip); +} + static void xonar_disable_output(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -1036,7 +1049,7 @@ static const struct oxygen_model model_xonar_hdav = { static const struct oxygen_model model_xonar_st = { .longname = "Asus Virtuoso 100", .chip = "AV200", - .init = xonar_stx_init, + .init = xonar_st_init, .control_filter = xonar_st_control_filter, .mixer_init = xonar_st_mixer_init, .cleanup = xonar_st_cleanup, @@ -1069,6 +1082,7 @@ static int __devinit get_xonar_model(struct oxygen *chip, [MODEL_D2] = &model_xonar_d2, [MODEL_D2X] = &model_xonar_d2, [MODEL_HDAV] = &model_xonar_hdav, + [MODEL_ST] = &model_xonar_st, [MODEL_STX] = &model_xonar_st, }; static const char *const names[] = { @@ -1078,6 +1092,8 @@ static int __devinit get_xonar_model(struct oxygen *chip, [MODEL_D2X] = "Xonar D2X", [MODEL_HDAV] = "Xonar HDAV1.3", [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", + [MODEL_ST] = "Xonar Essence ST", + [MODEL_ST_H6] = "Xonar Essence ST+H6", [MODEL_STX] = "Xonar Essence STX", }; unsigned int model = id->driver_data; @@ -1094,21 +1110,27 @@ static int __devinit get_xonar_model(struct oxygen *chip, chip->model.init = xonar_dx_init; break; case MODEL_HDAV: - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_HDAV_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & - GPIO_HDAV_DB_MASK) { - case GPIO_HDAV_DB_H6: + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { + case GPIO_DB_H6: model = MODEL_HDAV_H6; break; - case GPIO_HDAV_DB_XX: + case GPIO_DB_XX: snd_printk(KERN_ERR "unknown daughterboard\n"); return -ENODEV; } break; + case MODEL_ST: + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { + case GPIO_DB_H6: + model = MODEL_ST_H6; + break; + } + break; case MODEL_STX: - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_HDAV_DB_MASK); + chip->model.init = xonar_stx_init; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); break; } -- cgit v1.1 From 5a2e9a48b1d6de35ae5efea35d117133c3eb30f2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 25 May 2009 11:12:11 +0300 Subject: ASoC: TWL4030: Handsfree pop removal redesign Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E to a more appropriate DAPM_PGA_E widget. Also fix the power-up sequence to match with the TRM. The power-down sequence is not described in the TRM, so do it in a way, which seams like the correct sequence. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 78 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 59 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 9197fdd..17ddcb2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -546,27 +546,61 @@ static int micpath_event(struct snd_soc_dapm_widget *w, return 0; } -static int handsfree_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp) { - struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; unsigned char hs_ctl; - hs_ctl = twl4030_read_reg_cache(w->codec, e->reg); + hs_ctl = twl4030_read_reg_cache(codec, reg); - if (hs_ctl & TWL4030_HF_CTL_REF_EN) { + if (ramp) { + /* HF ramp-up */ + hs_ctl |= TWL4030_HF_CTL_REF_EN; + twl4030_write(codec, reg, hs_ctl); + udelay(10); hs_ctl |= TWL4030_HF_CTL_RAMP_EN; - twl4030_write(w->codec, e->reg, hs_ctl); + twl4030_write(codec, reg, hs_ctl); + udelay(40); hs_ctl |= TWL4030_HF_CTL_LOOP_EN; - twl4030_write(w->codec, e->reg, hs_ctl); hs_ctl |= TWL4030_HF_CTL_HB_EN; - twl4030_write(w->codec, e->reg, hs_ctl); + twl4030_write(codec, reg, hs_ctl); } else { - hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN - | TWL4030_HF_CTL_HB_EN); - twl4030_write(w->codec, e->reg, hs_ctl); + /* HF ramp-down */ + hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN; + hs_ctl &= ~TWL4030_HF_CTL_HB_EN; + twl4030_write(codec, reg, hs_ctl); + hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN; + twl4030_write(codec, reg, hs_ctl); + udelay(40); + hs_ctl &= ~TWL4030_HF_CTL_REF_EN; + twl4030_write(codec, reg, hs_ctl); } +} +static int handsfreelpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1); + break; + case SND_SOC_DAPM_POST_PMD: + handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0); + break; + } + return 0; +} + +static int handsfreerpga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1); + break; + case SND_SOC_DAPM_POST_PMD: + handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0); + break; + } return 0; } @@ -1190,12 +1224,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Output MUX controls */ /* HandsfreeL/R */ - SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, - &twl4030_dapm_handsfreel_control, handsfree_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, - &twl4030_dapm_handsfreer_control, handsfree_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_handsfreel_control), + SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, handsfreelpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, + &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, handsfreerpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Vibra */ SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, &twl4030_dapm_vibra_control), @@ -1303,11 +1341,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, + {"HandsfreeL PGA", NULL, "HandsfreeL Mux"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, + {"HandsfreeR PGA", NULL, "HandsfreeR Mux"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, @@ -1324,8 +1364,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOR", NULL, "HeadsetR PGA"}, {"CARKITL", NULL, "CarkitL Mixer"}, {"CARKITR", NULL, "CarkitR Mixer"}, - {"HFL", NULL, "HandsfreeL Mux"}, - {"HFR", NULL, "HandsfreeR Mux"}, + {"HFL", NULL, "HandsfreeL PGA"}, + {"HFR", NULL, "HandsfreeR PGA"}, {"Vibra Route", "Audio", "Vibra Mux"}, {"VIBRA", NULL, "Vibra Route"}, -- cgit v1.1 From f3b5d3002d5b43d277dedc1e044d02f2a40a43c5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 25 May 2009 11:12:12 +0300 Subject: ASoC: TWL4030: Add shadow register Shadow, non HW register for dealing with the HandsfreeL/R muting. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 7 ++++++- sound/soc/codecs/twl4030.h | 7 ++++++- 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 17ddcb2..989446d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -115,6 +115,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_PWM_SET (0x47) */ 0x00, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ + 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; /* codec private data */ @@ -172,7 +173,11 @@ static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { twl4030_write_reg_cache(codec, reg, value); - return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); + if (likely(reg < TWL4030_REG_SW_SHADOW)) + return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, + reg); + else + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 48326e2..fe5f395 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -92,8 +92,9 @@ #define TWL4030_REG_VIBRA_PWM_SET 0x47 #define TWL4030_REG_ANAMIC_GAIN 0x48 #define TWL4030_REG_MISC_SET_2 0x49 +#define TWL4030_REG_SW_SHADOW 0x4A -#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) +#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) /* Bitfield Definitions */ @@ -260,6 +261,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 +/* TWL4030_REG_SW_SHADOW (0x4A) Fields */ +#define TWL4030_HFL_EN 0x01 +#define TWL4030_HFR_EN 0x02 + #define TWL4030_DAI_HIFI 0 #define TWL4030_DAI_VOICE 1 -- cgit v1.1 From 0f89bdcac61536c5cb2a095a514657019573afb4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 25 May 2009 11:12:13 +0300 Subject: ASoC: TWL4030: HandsfreeL/R mute DAPM switch Add DAPM switch for HeadsetL/R mute. Since all bits are are needed for the HFL/R pop removal to work the switch is using the SW_SHADOW no HW register for the HandsfreeL/R mute. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 989446d..63ebd17 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -395,6 +395,10 @@ static const struct soc_enum twl4030_handsfreel_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control = SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); +/* Handsfree Left virtual mute */ +static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0); + /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = {"Voice", "AudioR1", "AudioR2", "AudioL2"}; @@ -407,6 +411,10 @@ static const struct soc_enum twl4030_handsfreer_enum = static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); +/* Handsfree Right virtual mute */ +static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0); + /* Vibra */ /* Vibra audio path selection */ static const char *twl4030_vibra_texts[] = @@ -1231,11 +1239,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* HandsfreeL/R */ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreel_control), + SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_handsfreelmute_control), SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreelpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_handsfreermute_control), SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreerpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), @@ -1346,13 +1358,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, - {"HandsfreeL PGA", NULL, "HandsfreeL Mux"}, + {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"}, + {"HandsfreeL PGA", NULL, "HandsfreeL Switch"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, - {"HandsfreeR PGA", NULL, "HandsfreeR Mux"}, + {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"}, + {"HandsfreeR PGA", NULL, "HandsfreeR Switch"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, -- cgit v1.1 From 4fcd39207f4c91185cc89e3e6a28cbb643034ff1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 May 2009 18:34:52 +0200 Subject: ALSA: hda - Reset CORB/RIRB at retrying the verb communication When a codec communication error occurs, the CORB/RIRB counters should be reset first before re-issuing the verb. Simply call azx_free_cmd_io() and azx_init_cmd_io() to achieve that. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 49fd973..3fc75e2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -522,6 +522,7 @@ static void azx_init_cmd_io(struct azx *chip) /* RIRB set up */ chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + chip->rirb.wp = chip->rirb.rp = chip->rirb.cmds = 0; azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr)); @@ -533,7 +534,6 @@ static void azx_init_cmd_io(struct azx *chip) azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN); - chip->rirb.rp = chip->rirb.cmds = 0; } static void azx_free_cmd_io(struct azx *chip) @@ -654,9 +654,11 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) snd_printk(KERN_ERR SFX "azx_get_response timeout (ERROR): " "last cmd=0x%08x\n", chip->last_cmd); + /* re-initialize CORB/RIRB */ spin_lock_irq(&chip->reg_lock); - chip->rirb.cmds = 0; /* reset the index */ bus->rirb_error = 1; + azx_free_cmd_io(chip); + azx_init_cmd_io(chip); spin_unlock_irq(&chip->reg_lock); return -1; } -- cgit v1.1 From a693a26fe0807c3c406d6b5295d222b646677288 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 12:58:58 +0200 Subject: ALSA: riptide - Code clean up A code clean up, coding style fixes. The firmware loading routine is split to an own function to improve the readability. Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 286 +++++++++++++++++++++----------------------- 1 file changed, 134 insertions(+), 152 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e51a5ef..460be03 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -507,41 +507,19 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); */ static struct pci_device_id snd_riptide_ids[] = { - { - .vendor = 0x127a,.device = 0x4310, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, - { - .vendor = 0x127a,.device = 0x4320, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, - { - .vendor = 0x127a,.device = 0x4330, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, - { - .vendor = 0x127a,.device = 0x4340, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, + { PCI_DEVICE(0x127a, 0x4310) }, + { PCI_DEVICE(0x127a, 0x4320) }, + { PCI_DEVICE(0x127a, 0x4330) }, + { PCI_DEVICE(0x127a, 0x4340) }, {0,}, }; #ifdef SUPPORT_JOYSTICK static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = { - { - .vendor = 0x127a,.device = 0x4312, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, - { - .vendor = 0x127a,.device = 0x4322, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, - {.vendor = 0x127a,.device = 0x4332, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, - {.vendor = 0x127a,.device = 0x4342, - .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID, - }, + { PCI_DEVICE(0x127a, 0x4312) }, + { PCI_DEVICE(0x127a, 0x4322) }, + { PCI_DEVICE(0x127a, 0x4332) }, + { PCI_DEVICE(0x127a, 0x4342) }, {0,}, }; #endif @@ -1209,12 +1187,79 @@ static int riptide_resume(struct pci_dev *pci) } #endif +static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) +{ + union firmware_version firmware = { .ret = CMDRET_ZERO }; + int i, timeout, err; + + for (i = 0; i < 2; i++) { + WRITE_PORT_ULONG(cif->hwport->port[i].data1, 0); + WRITE_PORT_ULONG(cif->hwport->port[i].data2, 0); + } + SET_GRESET(cif->hwport); + udelay(100); + UNSET_GRESET(cif->hwport); + udelay(100); + + for (timeout = 100000; --timeout; udelay(10)) { + if (IS_READY(cif->hwport) && !IS_GERR(cif->hwport)) + break; + } + if (!timeout) { + snd_printk(KERN_ERR + "Riptide: device not ready, audio status: 0x%x " + "ready: %d gerr: %d\n", + READ_AUDIO_STATUS(cif->hwport), + IS_READY(cif->hwport), IS_GERR(cif->hwport)); + return -EIO; + } else { + snd_printdd + ("Riptide: audio status: 0x%x ready: %d gerr: %d\n", + READ_AUDIO_STATUS(cif->hwport), + IS_READY(cif->hwport), IS_GERR(cif->hwport)); + } + + SEND_GETV(cif, &firmware.ret); + snd_printdd("Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n", + firmware.firmware.ASIC, firmware.firmware.CODEC, + firmware.firmware.AUXDSP, firmware.firmware.PROG); + + for (i = 0; i < FIRMWARE_VERSIONS; i++) { + if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware))) + break; + } + if (i >= FIRMWARE_VERSIONS) + return 0; /* no match */ + + if (!chip) + return 1; /* OK */ + + snd_printdd("Writing Firmware\n"); + if (!chip->fw_entry) { + err = request_firmware(&chip->fw_entry, "riptide.hex", + &chip->pci->dev); + if (err) { + snd_printk(KERN_ERR + "Riptide: Firmware not available %d\n", err); + return -EIO; + } + } + err = loadfirmware(cif, chip->fw_entry->data, chip->fw_entry->size); + if (err) { + snd_printk(KERN_ERR + "Riptide: Could not load firmware %d\n", err); + return err; + } + + chip->firmware = firmware; + + return 1; /* OK */ +} + static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip) { - int timeout, tries; union cmdret rptr = CMDRET_ZERO; - union firmware_version firmware; - int i, j, err, has_firmware; + int err, tries; if (!cif) return -EINVAL; @@ -1227,75 +1272,11 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip) cif->is_reset = 0; tries = RESET_TRIES; - has_firmware = 0; - while (has_firmware == 0 && tries-- > 0) { - for (i = 0; i < 2; i++) { - WRITE_PORT_ULONG(cif->hwport->port[i].data1, 0); - WRITE_PORT_ULONG(cif->hwport->port[i].data2, 0); - } - SET_GRESET(cif->hwport); - udelay(100); - UNSET_GRESET(cif->hwport); - udelay(100); - - for (timeout = 100000; --timeout; udelay(10)) { - if (IS_READY(cif->hwport) && !IS_GERR(cif->hwport)) - break; - } - if (timeout == 0) { - snd_printk(KERN_ERR - "Riptide: device not ready, audio status: 0x%x ready: %d gerr: %d\n", - READ_AUDIO_STATUS(cif->hwport), - IS_READY(cif->hwport), IS_GERR(cif->hwport)); - return -EIO; - } else { - snd_printdd - ("Riptide: audio status: 0x%x ready: %d gerr: %d\n", - READ_AUDIO_STATUS(cif->hwport), - IS_READY(cif->hwport), IS_GERR(cif->hwport)); - } - - SEND_GETV(cif, &rptr); - for (i = 0; i < 4; i++) - firmware.ret.retwords[i] = rptr.retwords[i]; - - snd_printdd - ("Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n", - firmware.firmware.ASIC, firmware.firmware.CODEC, - firmware.firmware.AUXDSP, firmware.firmware.PROG); - - for (j = 0; j < FIRMWARE_VERSIONS; j++) { - has_firmware = 1; - for (i = 0; i < 4; i++) { - if (firmware_versions[j].ret.retwords[i] != - firmware.ret.retwords[i]) - has_firmware = 0; - } - if (has_firmware) - break; - } - - if (chip != NULL && has_firmware == 0) { - snd_printdd("Writing Firmware\n"); - if (!chip->fw_entry) { - if ((err = - request_firmware(&chip->fw_entry, - "riptide.hex", - &chip->pci->dev)) != 0) { - snd_printk(KERN_ERR - "Riptide: Firmware not available %d\n", - err); - return -EIO; - } - } - err = loadfirmware(cif, chip->fw_entry->data, - chip->fw_entry->size); - if (err) - snd_printk(KERN_ERR - "Riptide: Could not load firmware %d\n", - err); - } - } + do { + err = try_to_load_firmware(cif, chip); + if (err < 0) + return err; + } while (!err && --tries); SEND_SACR(cif, 0, AC97_RESET); SEND_RACR(cif, AC97_RESET, &rptr); @@ -1337,11 +1318,6 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip) SET_AIE(cif->hwport); SET_AIACK(cif->hwport); cif->is_reset = 1; - if (chip) { - for (i = 0; i < 4; i++) - chip->firmware.ret.retwords[i] = - firmware.ret.retwords[i]; - } return 0; } @@ -2094,8 +2070,8 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) static int dev; struct snd_card *card; struct snd_riptide *chip; - unsigned short addr; - int err = 0; + unsigned short val; + int err; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -2107,60 +2083,63 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) return err; - if ((err = snd_riptide_create(card, pci, &chip)) < 0) { - snd_card_free(card); - return err; - } + err = snd_riptide_create(card, pci, &chip); + if (err < 0) + goto error; card->private_data = chip; - if ((err = snd_riptide_pcm(chip, 0, NULL)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_riptide_mixer(chip)) < 0) { - snd_card_free(card); - return err; - } - pci_write_config_word(chip->pci, PCI_EXT_Legacy_Mask, LEGACY_ENABLE_ALL - | (opl3_port[dev] ? LEGACY_ENABLE_FM : 0) + err = snd_riptide_pcm(chip, 0, NULL); + if (err < 0) + goto error; + err = snd_riptide_mixer(chip); + if (err < 0) + goto error; + + val = LEGACY_ENABLE_ALL; + if (opl3_port[dev]) + val |= LEGACY_ENABLE_FM; #ifdef SUPPORT_JOYSTICK - | (joystick_port[dev] ? LEGACY_ENABLE_GAMEPORT : - 0) + if (joystick_port[dev]) + val |= LEGACY_ENABLE_GAMEPORT; #endif - | (mpu_port[dev] - ? (LEGACY_ENABLE_MPU_INT | LEGACY_ENABLE_MPU) : - 0) - | ((chip->irq << 4) & 0xF0)); - if ((addr = mpu_port[dev]) != 0) { - pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, addr); - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE, - addr, 0, chip->irq, 0, - &chip->rmidi)) < 0) + if (mpu_port[dev]) + val |= LEGACY_ENABLE_MPU_INT | LEGACY_ENABLE_MPU; + val |= (chip->irq << 4) & 0xf0; + pci_write_config_word(chip->pci, PCI_EXT_Legacy_Mask, val); + if (mpu_port[dev]) { + val = mpu_port[dev]; + pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE, + val, 0, chip->irq, 0, + &chip->rmidi); + if (err < 0) snd_printk(KERN_WARNING "Riptide: Can't Allocate MPU at 0x%x\n", - addr); + val); else - chip->mpuaddr = addr; + chip->mpuaddr = val; } - if ((addr = opl3_port[dev]) != 0) { - pci_write_config_word(chip->pci, PCI_EXT_FM_Base, addr); - if ((err = snd_opl3_create(card, addr, addr + 2, - OPL3_HW_RIPTIDE, 0, - &chip->opl3)) < 0) + if (opl3_port[dev]) { + val = opl3_port[dev]; + pci_write_config_word(chip->pci, PCI_EXT_FM_Base, val); + err = snd_opl3_create(card, val, val + 2, + OPL3_HW_RIPTIDE, 0, &chip->opl3); + if (err < 0) snd_printk(KERN_WARNING "Riptide: Can't Allocate OPL3 at 0x%x\n", - addr); + val); else { - chip->opladdr = addr; - if ((err = - snd_opl3_hwdep_new(chip->opl3, 0, 1, NULL)) < 0) + chip->opladdr = val; + err = snd_opl3_hwdep_new(chip->opl3, 0, 1, NULL); + if (err < 0) snd_printk(KERN_WARNING "Riptide: Can't Allocate OPL3-HWDEP\n"); } } #ifdef SUPPORT_JOYSTICK - if ((addr = joystick_port[dev]) != 0) { - pci_write_config_word(chip->pci, PCI_EXT_Game_Base, addr); - chip->gameaddr = addr; + if (joystick_port[dev]) { + val = joystick_port[dev]; + pci_write_config_word(chip->pci, PCI_EXT_Game_Base, val); + chip->gameaddr = val; } #endif @@ -2178,13 +2157,16 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) chip->opladdr); #endif snd_riptide_proc_init(chip); - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); - return err; - } + err = snd_card_register(card); + if (err < 0) + goto error; pci_set_drvdata(pci, card); dev++; return 0; + + error: + snd_card_free(card); + return err; } static void __devexit snd_card_riptide_remove(struct pci_dev *pci) -- cgit v1.1 From db1005ec6ff8f250bccbc87387a955078891bfe4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 13:55:42 +0200 Subject: ALSA: riptide - Fix joystick resource handling The current code doesn't handle the multiple gameports properly, and uses unnecessary global static variables to store the data. This patch changes the probe / remove routines to use the driver data assigned to the dedicated pci device, and adds the support of multiple devices. Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 61 +++++++++++++++++++-------------------------- 1 file changed, 26 insertions(+), 35 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 460be03..235a71e 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2014,14 +2014,12 @@ static int __devinit snd_riptide_mixer(struct snd_riptide *chip) } #ifdef SUPPORT_JOYSTICK -static int have_joystick; -static struct pci_dev *riptide_gameport_pci; -static struct gameport *riptide_gameport; static int __devinit snd_riptide_joystick_probe(struct pci_dev *pci, const struct pci_device_id *id) { static int dev; + struct gameport *gameport; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -2030,36 +2028,33 @@ snd_riptide_joystick_probe(struct pci_dev *pci, const struct pci_device_id *id) return -ENOENT; } - if (joystick_port[dev]) { - riptide_gameport = gameport_allocate_port(); - if (riptide_gameport) { - if (!request_region - (joystick_port[dev], 8, "Riptide gameport")) { - snd_printk(KERN_WARNING - "Riptide: cannot grab gameport 0x%x\n", - joystick_port[dev]); - gameport_free_port(riptide_gameport); - riptide_gameport = NULL; - } else { - riptide_gameport_pci = pci; - riptide_gameport->io = joystick_port[dev]; - gameport_register_port(riptide_gameport); - } - } + if (!joystick_port[dev++]) + return 0; + + gameport = gameport_allocate_port(); + if (!gameport) + return -ENOMEM; + if (!request_region(joystick_port[dev], 8, "Riptide gameport")) { + snd_printk(KERN_WARNING + "Riptide: cannot grab gameport 0x%x\n", + joystick_port[dev]); + gameport_free_port(gameport); + return -EBUSY; } - dev++; + + gameport->io = joystick_port[dev]; + gameport_register_port(gameport); + pci_set_drvdata(pci, gameport); return 0; } static void __devexit snd_riptide_joystick_remove(struct pci_dev *pci) { - if (riptide_gameport) { - if (riptide_gameport_pci == pci) { - release_region(riptide_gameport->io, 8); - riptide_gameport_pci = NULL; - gameport_unregister_port(riptide_gameport); - riptide_gameport = NULL; - } + struct gameport *gameport = pci_get_drvdata(pci); + if (gameport) { + release_region(gameport->io, 8); + gameport_unregister_port(gameport); + pci_set_drvdata(pci, NULL); } } #endif @@ -2198,14 +2193,11 @@ static struct pci_driver joystick_driver = { static int __init alsa_card_riptide_init(void) { int err; - if ((err = pci_register_driver(&driver)) < 0) + err = pci_register_driver(&driver); + if (err < 0) return err; #if defined(SUPPORT_JOYSTICK) - if (pci_register_driver(&joystick_driver) < 0) { - have_joystick = 0; - snd_printk(KERN_INFO "no joystick found\n"); - } else - have_joystick = 1; + pci_register_driver(&joystick_driver); #endif return 0; } @@ -2214,8 +2206,7 @@ static void __exit alsa_card_riptide_exit(void) { pci_unregister_driver(&driver); #if defined(SUPPORT_JOYSTICK) - if (have_joystick) - pci_unregister_driver(&joystick_driver); + pci_unregister_driver(&joystick_driver); #endif } -- cgit v1.1 From 86d190e77c44cb057742dcc871b12ebd4633c387 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 15:18:58 +0200 Subject: ALSA: hda - Minor clean up of patch_sigmatel.c - Remove unneeded semicolons - Introduce spec->gpio_led to specify the GPIO bit for LED control Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 48 ++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1731081..26d8707 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -193,6 +193,7 @@ struct sigmatel_spec { unsigned int gpio_dir; unsigned int gpio_data; unsigned int gpio_mute; + unsigned int gpio_led; /* stream */ unsigned int stream_delay; @@ -4651,22 +4652,13 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio_bit = 0; /* gets rid of compiler warning */ - - switch (spec->board_config) { - case STAC_HP_DV4_1222NR: - gpio_bit = 0x01; - break; - case STAC_HP_HDX: - gpio_bit = 0x08; - } if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data &= ~gpio_bit; /* orange */ + spec->gpio_data |= spec->gpio_led; /* white */ else - spec->gpio_data |= gpio_bit; /* white */ + spec->gpio_data &= ~spec->gpio_led; /* orange */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, @@ -5352,14 +5344,7 @@ again: */ spec->num_smuxes = 1; spec->num_dmuxes = 1; -#ifdef CONFIG_SND_HDA_POWER_SAVE - /* This controls MUTE LED */ - spec->gpio_mask |= 0x01; - spec->gpio_dir |= 0x01; - spec->gpio_data |= 0x01; - codec->patch_ops.check_power_status = - stac92xx_hp_check_power_status; -#endif + spec->gpio_led = 0x01; /* fallthrough */ case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); @@ -5369,22 +5354,21 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; - /* - * For controlling MUTE LED on HP HDX16/HDX18 notebooks, - * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. - */ -#ifdef CONFIG_SND_HDA_POWER_SAVE /* orange/white mute led on GPIO3, orange=0, white=1 */ - spec->gpio_mask |= 0x08; - spec->gpio_dir |= 0x08; - spec->gpio_data |= 0x08; /* set to white */ + spec->gpio_led = 0x08; + break; + } +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (spec->gpio_led) { + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + spec->gpio_data |= spec->gpio_led; /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_check_power_status; + stac92xx_hp_check_power_status; + } #endif - break; - }; spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) @@ -5409,7 +5393,7 @@ again: codec->proc_widget_hook = stac92hd7x_proc_hook; return 0; -}; +} static int patch_stac922x(struct hda_codec *codec) { @@ -5564,7 +5548,7 @@ static int patch_stac927x(struct hda_codec *codec) /* correct the device field to SPDIF out */ snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070); break; - }; + } /* configure the analog microphone on some laptops */ snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ -- cgit v1.1 From 8174086167d43d0fd7b21928074145ae1d15bbab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 15:22:00 +0200 Subject: ALSA: hda - Allow concurrent RIRB access in single_cmd mode In the single_cmd mode, the current driver code doesn't do any update for RIRB just for any safety reason. But, actually the RIRB and single_cmd mode don't conflict. Unsolicited events can be delivered even while using the single_cmd mode. This patch allows the handling of unsolicited events with single_cmd mode, just always checking RIRB independent from single_cmd flag. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 82d5218..01d8d97 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -901,8 +901,7 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - if (!chip->single_cmd) - azx_init_cmd_io(chip); + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); @@ -1018,7 +1017,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { - if (!chip->single_cmd && (status & RIRB_INT_RESPONSE)) + if (status & RIRB_INT_RESPONSE) azx_update_rirb(chip); azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } @@ -2338,11 +2337,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } /* allocate CORB/RIRB */ - if (!chip->single_cmd) { - err = azx_alloc_cmd_io(chip); - if (err < 0) - goto errout; - } + err = azx_alloc_cmd_io(chip); + if (err < 0) + goto errout; /* initialize streams */ azx_init_stream(chip); -- cgit v1.1 From aa2936f5fe060e95ae06685149645b234085a468 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 16:07:57 +0200 Subject: ALSA: hda - Support sync after writing a verb This patch adds a debug mode to make the codec communication synchronous. Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c, and the call of snd_hda_codec_write*() will become synchronous, i.e. wait for the reply from the codec at each time issuing a verb. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 113 +++++++++++++++++++++++++--------------------- 1 file changed, 61 insertions(+), 52 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 77385de..d1d5fb9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -158,6 +158,38 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, return val; } +/* + * Send and receive a verb + */ +static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, + unsigned int *res) +{ + struct hda_bus *bus = codec->bus; + int err, repeated = 0; + + if (res) + *res = -1; + snd_hda_power_up(codec); + mutex_lock(&bus->cmd_mutex); + again: + err = bus->ops.command(bus, cmd); + if (!err) { + if (res) { + *res = bus->ops.get_response(bus); + if (*res == -1 && bus->rirb_error) { + if (repeated++ < 1) { + snd_printd(KERN_WARNING "hda_codec: " + "Trying verb 0x%08x again\n", cmd); + goto again; + } + } + } + } + mutex_unlock(&bus->cmd_mutex); + snd_hda_power_down(codec); + return err; +} + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec @@ -174,31 +206,18 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { - struct hda_bus *bus = codec->bus; - unsigned int cmd, res; - int repeated = 0; - - cmd = make_codec_cmd(codec, nid, direct, verb, parm); - snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); - again: - if (!bus->ops.command(bus, cmd)) { - res = bus->ops.get_response(bus); - if (res == -1 && bus->rirb_error) { - if (repeated++ < 1) { - snd_printd(KERN_WARNING "hda_codec: " - "Trying verb 0x%08x again\n", cmd); - goto again; - } - } - } else - res = (unsigned int)-1; - mutex_unlock(&bus->cmd_mutex); - snd_hda_power_down(codec); + unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int res; + codec_exec_verb(codec, cmd, &res); return res; } EXPORT_SYMBOL_HDA(snd_hda_codec_read); +/* Define the below to send and receive verbs synchronously. + * If you often get any codec communication errors, this is worth to try. + */ +/* #define SND_HDA_SUPPORT_SYNC_WRITE */ + /** * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec @@ -214,17 +233,13 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { - struct hda_bus *bus = codec->bus; + unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); +#ifdef SND_HDA_SUPPORT_SYNC_WRITE unsigned int res; - int err; - - res = make_codec_cmd(codec, nid, direct, verb, parm); - snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); - err = bus->ops.command(bus, res); - mutex_unlock(&bus->cmd_mutex); - snd_hda_power_down(codec); - return err; + return codec_exec_verb(codec, cmd, &res); +#else + return codec_exec_verb(codec, cmd, NULL); +#endif } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -2281,28 +2296,22 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { - struct hda_bus *bus = codec->bus; - unsigned int res; - int err; + int err = snd_hda_codec_write(codec, nid, direct, verb, parm); + struct hda_cache_head *c; + u32 key; - res = make_codec_cmd(codec, nid, direct, verb, parm); - snd_hda_power_up(codec); - mutex_lock(&bus->cmd_mutex); - err = bus->ops.command(bus, res); - if (!err) { - struct hda_cache_head *c; - u32 key; - /* parm may contain the verb stuff for get/set amp */ - verb = verb | (parm >> 8); - parm &= 0xff; - key = build_cmd_cache_key(nid, verb); - c = get_alloc_hash(&codec->cmd_cache, key); - if (c) - c->val = parm; - } - mutex_unlock(&bus->cmd_mutex); - snd_hda_power_down(codec); - return err; + if (err < 0) + return err; + /* parm may contain the verb stuff for get/set amp */ + verb = verb | (parm >> 8); + parm &= 0xff; + key = build_cmd_cache_key(nid, verb); + mutex_lock(&codec->bus->cmd_mutex); + c = get_alloc_hash(&codec->cmd_cache, key); + if (c) + c->val = parm; + mutex_unlock(&codec->bus->cmd_mutex); + return 0; } EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); -- cgit v1.1 From aae80dc24aeddec4e2b6e182a43491942f8667d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 May 2009 18:35:27 +0200 Subject: ALSA: ctxfi - Add missing module parameter definitions Added missing module_param*() and MODULE_PARM*(). Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/xfi.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 19310ed..e31e29e 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -23,13 +23,22 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); static unsigned int reference_rate = 48000; static unsigned int multiple = 2; +MODULE_PARM_DESC(reference_rate, "Reference rate (default=48000)"); module_param(reference_rate, uint, S_IRUGO); +MODULE_PARM_DESC(multiple, "Rate multiplier (default=2)"); module_param(multiple, uint, S_IRUGO); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Creative X-Fi driver"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Creative X-Fi driver"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); + static struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1) }, -- cgit v1.1 From dbcc34756234596993a3b1304636f032e66d401f Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Tue, 26 May 2009 08:34:08 -0400 Subject: ASoC: Main rewite of the mpc5200 audio DMA code Rewrite the mpc5200 audio DMA code to support both I2S and AC97. Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/mpc5200_dma.c | 442 +++++++++++++++++++++++++--------------- sound/soc/fsl/mpc5200_dma.h | 33 ++- sound/soc/fsl/mpc5200_psc_i2s.c | 247 ++++------------------ sound/soc/fsl/mpc5200_psc_i2s.h | 12 ++ 5 files changed, 344 insertions(+), 391 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index dc79bdf..1918c78 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -25,7 +25,6 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select SND_SOC_OF_SIMPLE select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 6850392..efec33a 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -3,23 +3,13 @@ * ALSA SoC Platform driver * * Copyright (C) 2008 Secret Lab Technologies Ltd. + * Copyright (C) 2009 Jon Smirl, Digispeaker */ -#include #include -#include -#include -#include #include -#include -#include -#include -#include -#include -#include #include -#include #include #include @@ -27,10 +17,6 @@ #include "mpc5200_dma.h" -MODULE_AUTHOR("Grant Likely "); -MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); -MODULE_LICENSE("GPL"); - /* * Interrupt handlers */ @@ -50,7 +36,7 @@ static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma) if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR)) psc_dma->stats.overrun_count++; - out_8(®s->command, 4 << 4); /* reset the error status */ + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); return IRQ_HANDLED; } @@ -81,21 +67,36 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) s->period_next_pt = s->period_start; } +static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) +{ + while (s->appl_ptr < s->runtime->control->appl_ptr) { + + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } +} + /* Bestcomm DMA irq handler */ -static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) +static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; + spin_lock(&s->psc_dma->lock); /* For each finished period, dequeue the completed period buffer * and enqueue a new one in it's place. */ while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + s->period_current_pt += s->period_bytes; if (s->period_current_pt >= s->period_end) s->period_current_pt = s->period_start; - psc_dma_bcom_enqueue_next_buffer(s); - bcom_enable(s->bcom_task); } + psc_dma_bcom_enqueue_tx(s); + spin_unlock(&s->psc_dma->lock); /* If the stream is active, then also inform the PCM middle layer * of the period finished event. */ @@ -105,49 +106,33 @@ static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) return IRQ_HANDLED; } -/** - * psc_dma_startup: create a new substream - * - * This is the first function called when a stream is opened. - * - * If this is the first stream open, then grab the IRQ and program most of - * the PSC registers. - */ -int psc_dma_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; - int rc; + struct psc_dma_stream *s = _psc_dma_stream; - dev_dbg(psc_dma->dev, "psc_dma_startup(substream=%p)\n", substream); + spin_lock(&s->psc_dma->lock); + /* For each finished period, dequeue the completed period buffer + * and enqueue a new one in it's place. */ + while (bcom_buffer_done(s->bcom_task)) { + bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - if (!psc_dma->playback.active && - !psc_dma->capture.active) { - /* Setup the IRQs */ - rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, - "psc-dma-status", psc_dma); - rc |= request_irq(psc_dma->capture.irq, - &psc_dma_bcom_irq, IRQF_SHARED, - "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, - &psc_dma_bcom_irq, IRQF_SHARED, - "psc-dma-playback", &psc_dma->playback); - if (rc) { - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, - &psc_dma->capture); - free_irq(psc_dma->playback.irq, - &psc_dma->playback); - return -ENODEV; - } + s->period_current_pt += s->period_bytes; + if (s->period_current_pt >= s->period_end) + s->period_current_pt = s->period_start; + + psc_dma_bcom_enqueue_next_buffer(s); } + spin_unlock(&s->psc_dma->lock); - return 0; + /* If the stream is active, then also inform the PCM middle layer + * of the period finished event. */ + if (s->active) + snd_pcm_period_elapsed(s->stream); + + return IRQ_HANDLED; } -int psc_dma_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int psc_dma_hw_free(struct snd_pcm_substream *substream) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -159,8 +144,7 @@ int psc_dma_hw_free(struct snd_pcm_substream *substream, * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ -int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) +static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; @@ -168,8 +152,8 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, struct psc_dma_stream *s; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; - u8 psc_cmd; unsigned long flags; + int i; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; @@ -189,68 +173,48 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, (s->period_bytes * runtime->periods); s->period_next_pt = s->period_start; s->period_current_pt = s->period_start; + s->period_size = runtime->period_size; s->active = 1; - /* First; reset everything */ + /* track appl_ptr so that we have a better chance of detecting + * end of stream and not over running it. + */ + s->runtime = runtime; + s->appl_ptr = s->runtime->control->appl_ptr - + (runtime->period_size * runtime->periods); + + /* Fill up the bestcomm bd queue and enable DMA. + * This will begin filling the PSC's fifo. + */ + spin_lock_irqsave(&psc_dma->lock, flags); + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - out_8(®s->command, MPC52xx_PSC_RST_RX); - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + bcom_gen_bd_rx_reset(s->bcom_task); + for (i = 0; i < runtime->periods; i++) + if (!bcom_queue_full(s->bcom_task)) + psc_dma_bcom_enqueue_next_buffer(s); } else { - out_8(®s->command, MPC52xx_PSC_RST_TX); - out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + bcom_gen_bd_tx_reset(s->bcom_task); + psc_dma_bcom_enqueue_tx(s); } - /* Next, fill up the bestcomm bd queue and enable DMA. - * This will begin filling the PSC's fifo. */ - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - bcom_gen_bd_rx_reset(s->bcom_task); - else - bcom_gen_bd_tx_reset(s->bcom_task); - while (!bcom_queue_full(s->bcom_task)) - psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); - - /* Due to errata in the dma mode; need to line up enabling - * the transmitter with a transition on the frame sync - * line */ - - spin_lock_irqsave(&psc_dma->lock, flags); - /* first make sure it is low */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) - ; - /* then wait for the transition to high */ - while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) - ; - /* Finally, enable the PSC. - * Receiver must always be enabled; even when we only want - * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ - psc_cmd = MPC52xx_PSC_RX_ENABLE; - if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) - psc_cmd |= MPC52xx_PSC_TX_ENABLE; - out_8(®s->command, psc_cmd); spin_unlock_irqrestore(&psc_dma->lock, flags); + out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT); + break; case SNDRV_PCM_TRIGGER_STOP: - /* Turn off the PSC */ s->active = 0; - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (!psc_dma->playback.active) { - out_8(®s->command, 2 << 4); /* reset rx */ - out_8(®s->command, 3 << 4); /* reset tx */ - out_8(®s->command, 4 << 4); /* reset err */ - } - } else { - out_8(®s->command, 3 << 4); /* reset tx */ - out_8(®s->command, 4 << 4); /* reset err */ - if (!psc_dma->capture.active) - out_8(®s->command, 2 << 4); /* reset rx */ - } + spin_lock_irqsave(&psc_dma->lock, flags); bcom_disable(s->bcom_task); - while (!bcom_queue_empty(s->bcom_task)) - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + bcom_gen_bd_rx_reset(s->bcom_task); + else + bcom_gen_bd_tx_reset(s->bcom_task); + spin_unlock_irqrestore(&psc_dma->lock, flags); break; @@ -265,44 +229,11 @@ int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, imr |= MPC52xx_PSC_IMR_TXEMP; if (psc_dma->capture.active) imr |= MPC52xx_PSC_IMR_ORERR; - out_be16(®s->isr_imr.imr, imr); + out_be16(®s->isr_imr.imr, psc_dma->imr | imr); return 0; } -/** - * psc_dma_shutdown: shutdown the data transfer on a stream - * - * Shutdown the PSC if there are no other substreams open. - */ -void psc_dma_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; - - dev_dbg(psc_dma->dev, "psc_dma_shutdown(substream=%p)\n", substream); - - /* - * If this is the last active substream, disable the PSC and release - * the IRQ. - */ - if (!psc_dma->playback.active && - !psc_dma->capture.active) { - - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); - out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset tx */ - out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset rx */ - out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ - - /* Release irqs */ - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, &psc_dma->capture); - free_irq(psc_dma->playback.irq, &psc_dma->playback); - } -} /* --------------------------------------------------------------------- * The PSC DMA 'ASoC platform' driver @@ -312,62 +243,78 @@ void psc_dma_shutdown(struct snd_pcm_substream *substream, * interaction with the attached codec */ -static const struct snd_pcm_hardware psc_dma_pcm_hardware = { +static const struct snd_pcm_hardware psc_dma_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, .rate_max = 48000, - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .period_bytes_max = 1024 * 1024, .period_bytes_min = 32, .periods_min = 2, .periods_max = 256, .buffer_bytes_max = 2 * 1024 * 1024, - .fifo_size = 0, + .fifo_size = 512, }; -static int psc_dma_pcm_open(struct snd_pcm_substream *substream) +static int psc_dma_open(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; + int rc; - dev_dbg(psc_dma->dev, "psc_dma_pcm_open(substream=%p)\n", substream); + dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; - snd_soc_set_runtime_hwparams(substream, &psc_dma_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware); + + rc = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (rc < 0) { + dev_err(substream->pcm->card->dev, "invalid buffer size\n"); + return rc; + } s->stream = substream; return 0; } -static int psc_dma_pcm_close(struct snd_pcm_substream *substream) +static int psc_dma_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct psc_dma_stream *s; - dev_dbg(psc_dma->dev, "psc_dma_pcm_close(substream=%p)\n", substream); + dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream); if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_dma->capture; else s = &psc_dma->playback; + if (!psc_dma->playback.active && + !psc_dma->capture.active) { + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ + } s->stream = NULL; return 0; } static snd_pcm_uframes_t -psc_dma_pcm_pointer(struct snd_pcm_substream *substream) +psc_dma_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; @@ -384,60 +331,78 @@ psc_dma_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(substream->runtime, count); } -static struct snd_pcm_ops psc_dma_pcm_ops = { - .open = psc_dma_pcm_open, - .close = psc_dma_pcm_close, +static int +psc_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static struct snd_pcm_ops psc_dma_ops = { + .open = psc_dma_open, + .close = psc_dma_close, + .hw_free = psc_dma_hw_free, .ioctl = snd_pcm_lib_ioctl, - .pointer = psc_dma_pcm_pointer, + .pointer = psc_dma_pointer, + .trigger = psc_dma_trigger, + .hw_params = psc_dma_hw_params, }; -static u64 psc_dma_pcm_dmamask = 0xffffffff; -static int psc_dma_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static u64 psc_dma_dmamask = 0xffffffff; +static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; - size_t size = psc_dma_pcm_hardware.buffer_bytes_max; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + size_t size = psc_dma_hardware.buffer_bytes_max; int rc = 0; - dev_dbg(rtd->socdev->dev, "psc_dma_pcm_new(card=%p, dai=%p, pcm=%p)\n", + dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n", card, dai, pcm); if (!card->dev->dma_mask) - card->dev->dma_mask = &psc_dma_pcm_dmamask; + card->dev->dma_mask = &psc_dma_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (pcm->streams[0].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, - &pcm->streams[0].substream->dma_buffer); + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + size, &pcm->streams[0].substream->dma_buffer); if (rc) goto playback_alloc_err; } if (pcm->streams[1].substream) { - rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size, - &pcm->streams[1].substream->dma_buffer); + rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, + size, &pcm->streams[1].substream->dma_buffer); if (rc) goto capture_alloc_err; } + if (rtd->socdev->card->codec->ac97) + rtd->socdev->card->codec->ac97->private_data = psc_dma; + return 0; capture_alloc_err: if (pcm->streams[0].substream) snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + playback_alloc_err: dev_err(card->dev, "Cannot allocate buffer(s)\n"); + return -ENOMEM; } -static void psc_dma_pcm_free(struct snd_pcm *pcm) +static void psc_dma_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_pcm_substream *substream; int stream; - dev_dbg(rtd->socdev->dev, "psc_dma_pcm_free(pcm=%p)\n", pcm); + dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -449,10 +414,151 @@ static void psc_dma_pcm_free(struct snd_pcm *pcm) } } -struct snd_soc_platform psc_dma_pcm_soc_platform = { +struct snd_soc_platform mpc5200_audio_dma_platform = { .name = "mpc5200-psc-audio", - .pcm_ops = &psc_dma_pcm_ops, - .pcm_new = &psc_dma_pcm_new, - .pcm_free = &psc_dma_pcm_free, + .pcm_ops = &psc_dma_ops, + .pcm_new = &psc_dma_new, + .pcm_free = &psc_dma_free, }; +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform); + +int mpc5200_audio_dma_create(struct of_device *op) +{ + phys_addr_t fifo; + struct psc_dma *psc_dma; + struct resource res; + int size, irq, rc; + const __be32 *prop; + void __iomem *regs; + + /* Fetch the registers and IRQ of the PSC */ + irq = irq_of_parse_and_map(op->node, 0); + if (of_address_to_resource(op->node, 0, &res)) { + dev_err(&op->dev, "Missing reg property\n"); + return -ENODEV; + } + regs = ioremap(res.start, 1 + res.end - res.start); + if (!regs) { + dev_err(&op->dev, "Could not map registers\n"); + return -ENODEV; + } + + /* Allocate and initialize the driver private data */ + psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); + if (!psc_dma) { + iounmap(regs); + return -ENOMEM; + } + + /* Get the PSC ID */ + prop = of_get_property(op->node, "cell-index", &size); + if (!prop || size < sizeof *prop) + return -ENODEV; + + spin_lock_init(&psc_dma->lock); + psc_dma->id = be32_to_cpu(*prop); + psc_dma->irq = irq; + psc_dma->psc_regs = regs; + psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; + psc_dma->dev = &op->dev; + psc_dma->playback.psc_dma = psc_dma; + psc_dma->capture.psc_dma = psc_dma; + snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id); + + /* Find the address of the fifo data registers and setup the + * DMA tasks */ + fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); + psc_dma->capture.bcom_task = + bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512); + psc_dma->playback.bcom_task = + bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo); + if (!psc_dma->capture.bcom_task || + !psc_dma->playback.bcom_task) { + dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); + iounmap(regs); + kfree(psc_dma); + return -ENODEV; + } + + /* Disable all interrupts and reset the PSC */ + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + /* reset receiver */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX); + /* reset transmitter */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX); + /* reset error */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT); + /* reset mode */ + out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1); + + /* Set up mode register; + * First write: RxRdy (FIFO Alarm) generates rx FIFO irq + * Second write: register Normal mode for non loopback + */ + out_8(&psc_dma->psc_regs->mode, 0); + out_8(&psc_dma->psc_regs->mode, 0); + + /* Set the TX and RX fifo alarm thresholds */ + out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); + out_8(&psc_dma->fifo_regs->rfcntl, 0x4); + out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); + out_8(&psc_dma->fifo_regs->tfcntl, 0x7); + + /* Lookup the IRQ numbers */ + psc_dma->playback.irq = + bcom_get_task_irq(psc_dma->playback.bcom_task); + psc_dma->capture.irq = + bcom_get_task_irq(psc_dma->capture.bcom_task); + + rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, + "psc-dma-status", psc_dma); + rc |= request_irq(psc_dma->capture.irq, + &psc_dma_bcom_irq_rx, IRQF_SHARED, + "psc-dma-capture", &psc_dma->capture); + rc |= request_irq(psc_dma->playback.irq, + &psc_dma_bcom_irq_tx, IRQF_SHARED, + "psc-dma-playback", &psc_dma->playback); + if (rc) { + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, + &psc_dma->capture); + free_irq(psc_dma->playback.irq, + &psc_dma->playback); + return -ENODEV; + } + /* Save what we've done so it can be found again later */ + dev_set_drvdata(&op->dev, psc_dma); + + /* Tell the ASoC OF helpers about it */ + return snd_soc_register_platform(&mpc5200_audio_dma_platform); +} +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create); + +int mpc5200_audio_dma_destroy(struct of_device *op) +{ + struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); + + dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n"); + + snd_soc_unregister_platform(&mpc5200_audio_dma_platform); + + bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); + bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); + + /* Release irqs */ + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); + + iounmap(psc_dma->psc_regs); + kfree(psc_dma); + dev_set_drvdata(&op->dev, NULL); + + return 0; +} +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy); + +MODULE_AUTHOR("Grant Likely "); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index a33232c..2000803 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -5,8 +5,10 @@ #ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__ #define __SOUND_SOC_FSL_MPC5200_DMA_H__ +#define PSC_STREAM_NAME_LEN 32 + /** - * psc_dma_stream - Data specific to a single stream (playback or capture) + * psc_ac97_stream - Data specific to a single stream (playback or capture) * @active: flag indicating if the stream is active * @psc_dma: pointer back to parent psc_dma data structure * @bcom_task: bestcomm task structure @@ -17,6 +19,9 @@ * @period_bytes: size of DMA period in bytes */ struct psc_dma_stream { + struct snd_pcm_runtime *runtime; + snd_pcm_uframes_t appl_ptr; + int active; struct psc_dma *psc_dma; struct bcom_task *bcom_task; @@ -27,6 +32,7 @@ struct psc_dma_stream { dma_addr_t period_next_pt; dma_addr_t period_current_pt; int period_bytes; + int period_size; }; /** @@ -48,9 +54,12 @@ struct psc_dma { struct mpc52xx_psc_fifo __iomem *fifo_regs; unsigned int irq; struct device *dev; - struct snd_soc_dai dai; spinlock_t lock; u32 sicr; + uint sysclk; + int imr; + int id; + unsigned int slots; /* per-stream data */ struct psc_dma_stream playback; @@ -58,24 +67,14 @@ struct psc_dma { /* Statistics */ struct { - int overrun_count; - int underrun_count; + unsigned long overrun_count; + unsigned long underrun_count; } stats; }; +int mpc5200_audio_dma_create(struct of_device *op); +int mpc5200_audio_dma_destroy(struct of_device *op); -int psc_dma_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); - -int psc_dma_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); - -void psc_dma_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); - -int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai); - -extern struct snd_soc_platform psc_dma_pcm_soc_platform; +extern struct snd_soc_platform mpc5200_audio_dma_platform; #endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 12a7917..ce8de90 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -3,34 +3,22 @@ * ALSA SoC Digital Audio Interface (DAI) driver * * Copyright (C) 2008 Secret Lab Technologies Ltd. + * Copyright (C) 2009 Jon Smirl, Digispeaker */ -#include #include -#include -#include -#include #include #include -#include -#include #include #include -#include #include -#include -#include -#include #include +#include "mpc5200_psc_i2s.h" #include "mpc5200_dma.h" -MODULE_AUTHOR("Grant Likely "); -MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); -MODULE_LICENSE("GPL"); - /** * PSC_I2S_RATES: sample rates supported by the I2S * @@ -46,8 +34,7 @@ MODULE_LICENSE("GPL"); * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode */ #define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \ - SNDRV_PCM_FMTBIT_S32_BE) + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) static int psc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, @@ -82,8 +69,6 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, } out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode); - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - return 0; } @@ -140,16 +125,13 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) * psc_i2s_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai_ops psc_i2s_dai_ops = { - .startup = psc_dma_startup, .hw_params = psc_i2s_hw_params, - .hw_free = psc_dma_hw_free, - .shutdown = psc_dma_shutdown, - .trigger = psc_dma_trigger, .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }; -static struct snd_soc_dai psc_i2s_dai_template = { +struct snd_soc_dai psc_i2s_dai[] = {{ + .name = "I2S", .playback = { .channels_min = 2, .channels_max = 2, @@ -163,71 +145,8 @@ static struct snd_soc_dai psc_i2s_dai_template = { .formats = PSC_I2S_FORMATS, }, .ops = &psc_i2s_dai_ops, -}; - -/* --------------------------------------------------------------------- - * Sysfs attributes for debugging - */ - -static ssize_t psc_i2s_status_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct psc_dma *psc_dma = dev_get_drvdata(dev); - - return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x " - "tfnum=%i tfstat=0x%.4x\n", - in_be16(&psc_dma->psc_regs->sr_csr.status), - in_be32(&psc_dma->psc_regs->sicr), - in_be16(&psc_dma->fifo_regs->rfnum) & 0x1ff, - in_be16(&psc_dma->fifo_regs->rfstat), - in_be16(&psc_dma->fifo_regs->tfnum) & 0x1ff, - in_be16(&psc_dma->fifo_regs->tfstat)); -} - -static int *psc_i2s_get_stat_attr(struct psc_dma *psc_dma, const char *name) -{ - if (strcmp(name, "playback_underrun") == 0) - return &psc_dma->stats.underrun_count; - if (strcmp(name, "capture_overrun") == 0) - return &psc_dma->stats.overrun_count; - - return NULL; -} - -static ssize_t psc_i2s_stat_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct psc_dma *psc_dma = dev_get_drvdata(dev); - int *attrib; - - attrib = psc_i2s_get_stat_attr(psc_dma, attr->attr.name); - if (!attrib) - return 0; - - return sprintf(buf, "%i\n", *attrib); -} - -static ssize_t psc_i2s_stat_store(struct device *dev, - struct device_attribute *attr, - const char *buf, - size_t count) -{ - struct psc_dma *psc_dma = dev_get_drvdata(dev); - int *attrib; - - attrib = psc_i2s_get_stat_attr(psc_dma, attr->attr.name); - if (!attrib) - return 0; - - *attrib = simple_strtoul(buf, NULL, 0); - return count; -} - -static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, - psc_i2s_stat_store); -static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, - psc_i2s_stat_store); +} }; +EXPORT_SYMBOL_GPL(psc_i2s_dai); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -237,82 +156,26 @@ static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, static int __devinit psc_i2s_of_probe(struct of_device *op, const struct of_device_id *match) { - phys_addr_t fifo; + int rc; struct psc_dma *psc_dma; - struct resource res; - int size, psc_id, irq, rc; - const __be32 *prop; - void __iomem *regs; - - dev_dbg(&op->dev, "probing psc i2s device\n"); - - /* Get the PSC ID */ - prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; - psc_id = be32_to_cpu(*prop); - - /* Fetch the registers and IRQ of the PSC */ - irq = irq_of_parse_and_map(op->node, 0); - if (of_address_to_resource(op->node, 0, &res)) { - dev_err(&op->dev, "Missing reg property\n"); - return -ENODEV; - } - regs = ioremap(res.start, 1 + res.end - res.start); - if (!regs) { - dev_err(&op->dev, "Could not map registers\n"); - return -ENODEV; - } + struct mpc52xx_psc __iomem *regs; - /* Allocate and initialize the driver private data */ - psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); - if (!psc_dma) { - iounmap(regs); - return -ENOMEM; - } - spin_lock_init(&psc_dma->lock); - psc_dma->irq = irq; - psc_dma->psc_regs = regs; - psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs; - psc_dma->dev = &op->dev; - psc_dma->playback.psc_dma = psc_dma; - psc_dma->capture.psc_dma = psc_dma; - snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_id+1); - - /* Fill out the CPU DAI structure */ - memcpy(&psc_dma->dai, &psc_i2s_dai_template, sizeof psc_dma->dai); - psc_dma->dai.private_data = psc_dma; - psc_dma->dai.name = psc_dma->name; - psc_dma->dai.id = psc_id; - - /* Find the address of the fifo data registers and setup the - * DMA tasks */ - fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32); - psc_dma->capture.bcom_task = - bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512); - psc_dma->playback.bcom_task = - bcom_psc_gen_bd_tx_init(psc_id, 10, fifo); - if (!psc_dma->capture.bcom_task || - !psc_dma->playback.bcom_task) { - dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_dma); - return -ENODEV; + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; } - /* Disable all interrupts and reset the PSC */ - out_be16(&psc_dma->psc_regs->isr_imr.imr, 0); - out_8(&psc_dma->psc_regs->command, 3 << 4); /* reset transmitter */ - out_8(&psc_dma->psc_regs->command, 2 << 4); /* reset receiver */ - out_8(&psc_dma->psc_regs->command, 1 << 4); /* reset mode */ - out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */ + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; /* Configure the serial interface mode; defaulting to CODEC8 mode */ psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S | MPC52xx_PSC_SICR_CLKPOL; - if (of_get_property(op->node, "fsl,cellslave", NULL)) - psc_dma->sicr |= MPC52xx_PSC_SICR_CELLSLAVE | - MPC52xx_PSC_SICR_GENCLK; out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8); @@ -321,66 +184,37 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, if (!of_get_property(op->node, "codec-handle", NULL)) return 0; - /* Set up mode register; - * First write: RxRdy (FIFO Alarm) generates rx FIFO irq - * Second write: register Normal mode for non loopback - */ - out_8(&psc_dma->psc_regs->mode, 0); - out_8(&psc_dma->psc_regs->mode, 0); - - /* Set the TX and RX fifo alarm thresholds */ - out_be16(&psc_dma->fifo_regs->rfalarm, 0x100); - out_8(&psc_dma->fifo_regs->rfcntl, 0x4); - out_be16(&psc_dma->fifo_regs->tfalarm, 0x100); - out_8(&psc_dma->fifo_regs->tfcntl, 0x7); - - /* Lookup the IRQ numbers */ - psc_dma->playback.irq = - bcom_get_task_irq(psc_dma->playback.bcom_task); - psc_dma->capture.irq = - bcom_get_task_irq(psc_dma->capture.bcom_task); - - /* Save what we've done so it can be found again later */ - dev_set_drvdata(&op->dev, psc_dma); - - /* Register the SYSFS files */ - rc = device_create_file(psc_dma->dev, &dev_attr_status); - rc |= device_create_file(psc_dma->dev, &dev_attr_capture_overrun); - rc |= device_create_file(psc_dma->dev, &dev_attr_playback_underrun); - if (rc) - dev_info(psc_dma->dev, "error creating sysfs files\n"); - - snd_soc_register_platform(&psc_dma_pcm_soc_platform); - - /* Tell the ASoC OF helpers about it */ - of_snd_soc_register_platform(&psc_dma_pcm_soc_platform, op->node, - &psc_dma->dai); + /* Due to errata in the dma mode; need to line up enabling + * the transmitter with a transition on the frame sync + * line */ + + /* first make sure it is low */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) + ; + /* then wait for the transition to high */ + while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) + ; + /* Finally, enable the PSC. + * Receiver must always be enabled; even when we only want + * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */ + + /* Go */ + out_8(&psc_dma->psc_regs->command, + MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); return 0; + } static int __devexit psc_i2s_of_remove(struct of_device *op) { - struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); - - dev_dbg(&op->dev, "psc_i2s_remove()\n"); - - snd_soc_unregister_platform(&psc_dma_pcm_soc_platform); - - bcom_gen_bd_rx_release(psc_dma->capture.bcom_task); - bcom_gen_bd_tx_release(psc_dma->playback.bcom_task); - - iounmap(psc_dma->psc_regs); - iounmap(psc_dma->fifo_regs); - kfree(psc_dma); - dev_set_drvdata(&op->dev, NULL); - - return 0; + return mpc5200_audio_dma_destroy(op); } /* Match table for of_platform binding */ static struct of_device_id psc_i2s_match[] __devinitdata = { { .compatible = "fsl,mpc5200-psc-i2s", }, + { .compatible = "fsl,mpc5200b-psc-i2s", }, {} }; MODULE_DEVICE_TABLE(of, psc_i2s_match); @@ -411,4 +245,7 @@ static void __exit psc_i2s_exit(void) } module_exit(psc_i2s_exit); +MODULE_AUTHOR("Grant Likely "); +MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h new file mode 100644 index 0000000..ce55e07 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_i2s.h @@ -0,0 +1,12 @@ +/* + * Freescale MPC5200 PSC in I2S mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ + +extern struct snd_soc_dai psc_i2s_dai[]; + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */ -- cgit v1.1 From 20d0e1520ed1ba8aad05f416245446de0f7ec4bb Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Tue, 26 May 2009 08:34:10 -0400 Subject: ASoC: AC97 driver for mpc5200 I've implemented retries for when the AC97 hardware doesn't reset on first try. About 10% of the time both the Efika and pcm030 AC97 codecs don't reset on first try and need to be poked multiple times. Failure is indicated by not having the link clock start ticking. Every once in a while even five pokes won't get the link started and I have to power cycle. Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 11 ++ sound/soc/fsl/Makefile | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 331 +++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 ++ 4 files changed, 358 insertions(+) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..9f2df15 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,331 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include +#include +#include + +#include +#include +#include + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int rc; + unsigned int val; + + /* Wait for command send status zero = ready */ + spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); + if (rc == 0) { + pr_err("timeout on ac97 bus (rdy)\n"); + return -ENODEV; + } + /* Send the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL), 100, 0, rc); + if (rc == 0) { + pr_err("timeout on ac97 read (val) %x\n", + in_be16(&psc_dma->psc_regs->sr_csr.status)); + return -ENODEV; + } + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if (((val >> 24) & 0x7f) != reg) { + pr_err("reg echo error on ac97 read\n"); + return -ENODEV; + } + val = (val >> 8) & 0xffff; + + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + int rc; + + /* Wait for command status zero = ready */ + spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); + if (rc == 0) { + pr_err("timeout on ac97 bus (write)\n"); + return; + } + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, + ((reg & 0x7f) << 24) | (val << 8)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + int rc; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + spin_event_timeout(0, 3, 0, rc); + out_be32(®s->sicr, psc_dma->sicr); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + int rc; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(®s->op1, MPC52xx_PSC_OP_RES); + spin_event_timeout(0, 10, 0, rc); + out_8(®s->op0, MPC52xx_PSC_OP_RES); + spin_event_timeout(0, 50, 0, rc); + psc_ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), + params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + break; + } + return 0; +} + +static int psc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct psc_dma *psc_dma = cpu_dai->private_data; + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Go */ + out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .ac97_control = 1, + .probe = psc_ac97_probe, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .ac97_control = 1, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +} }; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + dev_err(&op->dev, "Failed to register DAI\n"); + return rc; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(®s->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(®s->ac97_slots, 0x00000000); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl "); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ -- cgit v1.1 From a9262c4fd404654acd3684699047fa63206518c8 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Tue, 26 May 2009 08:34:12 -0400 Subject: ASoC: Support for AC97 on Phytec pmc030 base board. A wm9712 AC97 codec is used. Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 7 +++ sound/soc/fsl/Makefile | 3 ++ sound/soc/fsl/pcm030-audio-fabric.c | 90 +++++++++++++++++++++++++++++++++++++ 3 files changed, 100 insertions(+) create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3bce952..79579ae 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -39,4 +39,11 @@ config SND_SOC_MPC5200_AC97 help Say Y here to support the MPC5200 PSCs in AC97 mode. +config SND_MPC52xx_SOC_PCM030 + tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" + depends on PPC_MPC5200_SIMPLE + select SND_SOC_MPC5200_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for sound on the Phytec pcm030 baseboard. diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 14631a1..66d88c8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -15,3 +15,6 @@ obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o +# MPC5200 Machine Support +obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o + diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c new file mode 100644 index 0000000..8766f7a --- /dev/null +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -0,0 +1,90 @@ +/* + * Phytec pcm030 driver for the PSC of the Freescale MPC52xx + * configured as AC97 interface + * + * Copyright 2008 Jon Smirl, Digispeaker + * Author: Jon Smirl + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" +#include "../codecs/wm9712.h" + +static struct snd_soc_device device; +static struct snd_soc_card card; + +static struct snd_soc_dai_link pcm030_fabric_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 Analog", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL], +}, +{ + .name = "AC97", + .stream_name = "AC97 IEC958", + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF], +}, +}; + +static __init int pcm030_fabric_init(void) +{ + struct platform_device *pdev; + int rc; + + if (!machine_is_compatible("phytec,pcm030")) + return -ENODEV; + + card.platform = &mpc5200_audio_dma_platform; + card.name = "pcm030"; + card.dai_link = pcm030_fabric_dai; + card.num_links = ARRAY_SIZE(pcm030_fabric_dai); + + device.card = &card; + device.codec_dev = &soc_codec_dev_wm9712; + + pdev = platform_device_alloc("soc-audio", 1); + if (!pdev) { + pr_err("pcm030_fabric_init: platform_device_alloc() failed\n"); + return -ENODEV; + } + + platform_set_drvdata(pdev, &device); + device.dev = &pdev->dev; + + rc = platform_device_add(pdev); + if (rc) { + pr_err("pcm030_fabric_init: platform_device_add() failed\n"); + return -ENODEV; + } + return 0; +} + +module_init(pcm030_fabric_init); + + +MODULE_AUTHOR("Jon Smirl "); +MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver"); +MODULE_LICENSE("GPL"); + -- cgit v1.1 From 6ffee43ecf8bfbe0bd74c9084c9772a59097d53b Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Tue, 26 May 2009 08:34:14 -0400 Subject: ASoC: Fabric bindings for STAC9766 on the Efika Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 8 ++++ sound/soc/fsl/Makefile | 1 + sound/soc/fsl/efika-audio-fabric.c | 90 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 99 insertions(+) create mode 100644 sound/soc/fsl/efika-audio-fabric.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 79579ae..f571c6e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -47,3 +47,11 @@ config SND_MPC52xx_SOC_PCM030 help Say Y if you want to add support for sound on the Phytec pcm030 baseboard. +config SND_MPC52xx_SOC_EFIKA + tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" + depends on PPC_EFIKA + select SND_SOC_MPC5200_AC97 + select SND_SOC_STAC9766 + help + Say Y if you want to add support for sound on the Efika. + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 66d88c8..a83a739 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -17,4 +17,5 @@ obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o # MPC5200 Machine Support obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o +obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c new file mode 100644 index 0000000..85b0e75 --- /dev/null +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -0,0 +1,90 @@ +/* + * Efika driver for the PSC of the Freescale MPC52xx + * configured as AC97 interface + * + * Copyright 2008 Jon Smirl, Digispeaker + * Author: Jon Smirl + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" +#include "../codecs/stac9766.h" + +static struct snd_soc_device device; +static struct snd_soc_card card; + +static struct snd_soc_dai_link efika_fabric_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 Analog", + .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL], +}, +{ + .name = "AC97", + .stream_name = "AC97 IEC958", + .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL], + .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF], +}, +}; + +static __init int efika_fabric_init(void) +{ + struct platform_device *pdev; + int rc; + + if (!machine_is_compatible("bplan,efika")) + return -ENODEV; + + card.platform = &mpc5200_audio_dma_platform; + card.name = "Efika"; + card.dai_link = efika_fabric_dai; + card.num_links = ARRAY_SIZE(efika_fabric_dai); + + device.card = &card; + device.codec_dev = &soc_codec_dev_stac9766; + + pdev = platform_device_alloc("soc-audio", 1); + if (!pdev) { + pr_err("efika_fabric_init: platform_device_alloc() failed\n"); + return -ENODEV; + } + + platform_set_drvdata(pdev, &device); + device.dev = &pdev->dev; + + rc = platform_device_add(pdev); + if (rc) { + pr_err("efika_fabric_init: platform_device_add() failed\n"); + return -ENODEV; + } + return 0; +} + +module_init(efika_fabric_init); + + +MODULE_AUTHOR("Jon Smirl "); +MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver"); +MODULE_LICENSE("GPL"); + -- cgit v1.1 From 0c0e09e21a9e7bc6ca54e06ef3d497255ca26383 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 May 2009 21:14:59 +0100 Subject: ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved These drivers use spin_event_timeout() which is only present in the PowerPC tree at present and which is undergoing some API revisions so temporarily mark them as BROKEN until these issues are sorted out. Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f571c6e..5dbebf8 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -41,15 +41,16 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE + depends on PPC_MPC5200_SIMPLE && BROKEN select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help - Say Y if you want to add support for sound on the Phytec pcm030 baseboard. + Say Y if you want to add support for sound on the Phytec pcm030 + baseboard. config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA + depends on PPC_EFIKA && BROKEN select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help -- cgit v1.1 From 08d15f034e94251606479d7ca9070994c2e2fcf0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 23 May 2009 10:41:05 +0100 Subject: ASoC: Switch FSL SSI DAI over to symmetric_rates The effect of symmetric_constraints should provide a standard way to enforce the use of the same sample rate for both directions. Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/fsl/fsl_ssi.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 47afaa9..93f0f38 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -375,18 +375,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_pcm_runtime *first_runtime = ssi_private->first_stream->runtime; - if (!first_runtime->rate || !first_runtime->sample_bits) { + if (!first_runtime->sample_bits) { dev_err(substream->pcm->card->dev, - "set sample rate and size in %s stream first\n", + "set sample size in %s stream first\n", substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "capture" : "playback"); return -EAGAIN; } - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - first_runtime->rate, first_runtime->rate); - /* If we're in synchronous mode, then we need to constrain * the sample size as well. We don't support independent sample * rates in asynchronous mode. @@ -693,6 +689,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) fsl_ssi_dai->name = ssi_private->name; fsl_ssi_dai->id = ssi_info->id; fsl_ssi_dai->dev = ssi_info->dev; + fsl_ssi_dai->symmetric_rates = 1; ret = snd_soc_register_dai(fsl_ssi_dai); if (ret != 0) { -- cgit v1.1 From ea8b27ad0cc2573776c6cd87617a37aaf603b8bd Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Wed, 27 May 2009 01:06:19 -0400 Subject: ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout() The function signature for spin_event_timeout() has changed in version V9. Adjust the mpc5200 AC97 driver to use the new function. Signed-off-by: Jon Smirl Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_ac97.c | 30 ++++++++++++++---------------- 1 file changed, 14 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 9f2df15..794a247 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -31,13 +31,13 @@ static struct psc_dma *psc_dma; static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - int rc; + int status; unsigned int val; /* Wait for command send status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); return -ENODEV; } @@ -45,9 +45,9 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); /* Wait for the answer */ - spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_DATA_VAL), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL), 100, 0); + if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); return -ENODEV; @@ -66,12 +66,12 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - int rc; + int status; /* Wait for command status zero = ready */ - spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & - MPC52xx_PSC_SR_CMDSEND), 100, 0, rc); - if (rc == 0) { + status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND), 100, 0); + if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); return; } @@ -82,24 +82,22 @@ static void psc_ac97_write(struct snd_ac97 *ac97, static void psc_ac97_warm_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); - spin_event_timeout(0, 3, 0, rc); + udelay(3); out_be32(®s->sicr, psc_dma->sicr); } static void psc_ac97_cold_reset(struct snd_ac97 *ac97) { - int rc; struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; /* Do a cold reset */ out_8(®s->op1, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 10, 0, rc); + udelay(10); out_8(®s->op0, MPC52xx_PSC_OP_RES); - spin_event_timeout(0, 50, 0, rc); + udelay(50); psc_ac97_warm_reset(ac97); } -- cgit v1.1 From 449bd54dcbd0b60070ce4129fedaf0f4ae044099 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 27 May 2009 17:08:39 -0700 Subject: ASoC: correct print specifiers for unsigneds Unsigned variables should use `%u' rather than `%d'. Signed-off-by: Roel Kluin Signed-off-by: Andrew Morton Signed-off-by: Mark Brown --- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/tlv320aic23.c | 4 ++-- sound/soc/codecs/uda134x.c | 4 ++-- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 4 ++-- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 6 +++--- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 ++-- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 ++-- sound/soc/sh/ssi.c | 2 +- 14 files changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 7065753..9eb610c 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( * Find actual rate, compare to requested rate */ actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", rate, actual_rate); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 21f69df..9fcbb9c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, */ if ((reg < 0 || reg > 9) && (reg != 15)) { - printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } @@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; - printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__, value, reg); return -EIO; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index ddefb8f..269b108 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %d", + printk(KERN_ERR "%s unkown register: reg: %u", __func__, reg); return -EINVAL; } @@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct uda134x_priv *uda134x = codec->private_data; - pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__, clk_id, freq, dir); /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0275321..e7348d3 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, if (ret < 0) return ret; dev_dbg(wm8350->dev, - "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d", + "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d", freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div, fll_div.ratio); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index e4547de..502eefa 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->outdiv *= 2; if (factors->outdiv > 32) { dev_err(wm8400->wm8400->dev, - "Unsupported FLL output frequency %dHz\n", + "Unsupported FLL output frequency %uHz\n", Fout); return -EINVAL; } @@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, factors->k = K / 10; dev_dbg(wm8400->wm8400->dev, - "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n", + "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n", Fref, Fout, factors->n, factors->k, factors->fratio, factors->outdiv); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6a4cea0..c8b8dba 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source) if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8510 N value %d outwith recommended range!d\n", + "WM8510 N value %u outwith recommended range!d\n", Ndiv); pll_div.n = Ndiv; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 9f6be3d..86c4b24d 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -415,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, unsigned int K, Ndiv, Nmod; int i; - pr_debug("wm8580: PLL %dHz->%dHz\n", source, target); + pr_debug("wm8580: PLL %uHz->%uHz\n", source, target); /* Scale the output frequency up; the PLL should run in the * region of 90-100MHz. @@ -447,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 5) || (Ndiv > 13)) { printk(KERN_ERR - "WM8580 N=%d outside supported range\n", Ndiv); + "WM8580 N=%u outside supported range\n", Ndiv); return -EINVAL; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d121e58..d28eeac 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "wm8753: unsupported N = %d\n", Ndiv); + "wm8753: unsupported N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 46c5ea1..3c78945 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, } if (target > 100000000) - printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d" - " Fout=%d\n", target, Fref, Fout); + printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u" + " Fout=%u\n", target, Fref, Fout); if (div > 32) { printk(KERN_ERR "wm8900: Invalid FLL division rate %u, " - "Fref=%d, Fout=%d, target=%d\n", + "Fref=%u, Fout=%u, target=%u\n", div, Fref, Fout, target); return -EINVAL; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 40cd274..d029818 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + "WM8990 N value outwith recommended range! N = %u\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index d1744e9..abed37a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -710,7 +710,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source) Ndiv = target / source; if ((Ndiv < 5) || (Ndiv > 12)) printk(KERN_WARNING - "WM9713 PLL N value %d out of recommended range!\n", + "WM9713 PLL N value %u out of recommended range!\n", Ndiv); pll_div->n = Ndiv; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc7876..19c4540 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -208,7 +208,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", cpu_dai->id, clk_id, freq); switch (clk_id) { @@ -357,7 +357,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, ssacd |= (0x6 << 4); dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %dHz\n", + "Using SSACDD %x to supply %uHz\n", val, freq_out); break; } diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 972c276..1a28317 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -547,7 +547,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, actual = clkrate / (fsdiv * div); deviation = actual - rate; - printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", fsdiv, div, actual, deviation); deviation = abs(deviation); @@ -563,7 +563,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, break; } - printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", best_fs, best_div, best_rate); info->fs_div = best_fs; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 56fa087..b378096 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -145,7 +145,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); - pr_debug("bits: %d channels: %d\n", bits, channels); + pr_debug("bits: %u channels: %u\n", bits, channels); ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | CR_SWL_MASK); -- cgit v1.1 From b05a7d4fed7e51dca37d0a31baf1466de30b1f01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 May 2009 11:59:12 +0200 Subject: ALSA: hda - Always sync writes in single_cmd mode In the single_cmd mode, the hardware cannot store the multiple replies like on RIRB, thus each verb has to sync and wait for the response no matter whether the return value is needed or not. Otherwise it may result in a wrong return value from the previous verb. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 01d8d97..31a695e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -673,6 +673,27 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) * I left the codes, however, for debugging/testing purposes. */ +/* receive a response */ +static int azx_single_wait_for_response(struct azx *chip) +{ + int timeout = 50; + + while (timeout--) { + /* check IRV busy bit */ + if (azx_readw(chip, IRS) & ICH6_IRS_VALID) { + /* reuse rirb.res as the response return value */ + chip->rirb.res = azx_readl(chip, IR); + return 0; + } + udelay(1); + } + if (printk_ratelimit()) + snd_printd(SFX "get_response timeout: IRS=0x%x\n", + azx_readw(chip, IRS)); + chip->rirb.res = -1; + return -EIO; +} + /* send a command */ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) { @@ -688,7 +709,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) azx_writel(chip, IC, val); azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_BUSY); - return 0; + return azx_single_wait_for_response(chip); } udelay(1); } @@ -702,18 +723,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) static unsigned int azx_single_get_response(struct hda_bus *bus) { struct azx *chip = bus->private_data; - int timeout = 50; - - while (timeout--) { - /* check IRV busy bit */ - if (azx_readw(chip, IRS) & ICH6_IRS_VALID) - return azx_readl(chip, IR); - udelay(1); - } - if (printk_ratelimit()) - snd_printd(SFX "get_response timeout: IRS=0x%x\n", - azx_readw(chip, IRS)); - return (unsigned int)-1; + return chip->rirb.res; } /* -- cgit v1.1 From b21fadb9c1852c91622ca1dccfeb144bc535e36e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 May 2009 12:26:15 +0200 Subject: ALSA: hda - Add more register bits definitions Added some missing register bits definitions to reduce magic numbers. Also renamed some to follow the names on the datasheet. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 43 ++++++++++++++++++++++++++----------------- 1 file changed, 26 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 31a695e..f63bc65 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -138,14 +138,23 @@ MODULE_DESCRIPTION("Intel HDA driver"); * registers */ #define ICH6_REG_GCAP 0x00 +#define ICH6_GCAP_64OK (1 << 0) /* 64bit address support */ +#define ICH6_GCAP_NSDO (3 << 1) /* # of serial data out signals */ +#define ICH6_GCAP_BSS (31 << 3) /* # of bidirectional streams */ +#define ICH6_GCAP_ISS (15 << 8) /* # of input streams */ +#define ICH6_GCAP_OSS (15 << 12) /* # of output streams */ #define ICH6_REG_VMIN 0x02 #define ICH6_REG_VMAJ 0x03 #define ICH6_REG_OUTPAY 0x04 #define ICH6_REG_INPAY 0x06 #define ICH6_REG_GCTL 0x08 +#define ICH6_GCTL_RESET (1 << 1) /* controller reset */ +#define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */ +#define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ #define ICH6_REG_WAKEEN 0x0c #define ICH6_REG_STATESTS 0x0e #define ICH6_REG_GSTS 0x10 +#define ICH6_GSTS_FSTS (1 << 1) /* flush status */ #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 #define ICH6_REG_WALCLK 0x30 @@ -153,17 +162,27 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 #define ICH6_REG_CORBWP 0x48 -#define ICH6_REG_CORBRP 0x4A +#define ICH6_REG_CORBRP 0x4a +#define ICH6_CORBRP_RST (1 << 15) /* read pointer reset */ #define ICH6_REG_CORBCTL 0x4c +#define ICH6_CORBCTL_RUN (1 << 1) /* enable DMA */ +#define ICH6_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ #define ICH6_REG_CORBSTS 0x4d +#define ICH6_CORBSTS_CMEI (1 << 0) /* memory error indication */ #define ICH6_REG_CORBSIZE 0x4e #define ICH6_REG_RIRBLBASE 0x50 #define ICH6_REG_RIRBUBASE 0x54 #define ICH6_REG_RIRBWP 0x58 +#define ICH6_RIRBWP_RST (1 << 15) /* write pointer reset */ #define ICH6_REG_RINTCNT 0x5a #define ICH6_REG_RIRBCTL 0x5c +#define ICH6_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ +#define ICH6_RBCTL_DMA_EN (1 << 1) /* enable DMA */ +#define ICH6_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ #define ICH6_REG_RIRBSTS 0x5d +#define ICH6_RBSTS_IRQ (1 << 0) /* response irq */ +#define ICH6_RBSTS_OVERRUN (1 << 2) /* overrun irq */ #define ICH6_REG_RIRBSIZE 0x5e #define ICH6_REG_IC 0x60 @@ -260,16 +279,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ #define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ -/* GCTL unsolicited response enable bit */ -#define ICH6_GCTL_UREN (1<<8) - -/* GCTL reset bit */ -#define ICH6_GCTL_RESET (1<<0) - -/* CORB/RIRB control, read/write pointer */ -#define ICH6_RBCTL_DMA_EN 0x02 /* enable DMA */ -#define ICH6_RBCTL_IRQ_EN 0x01 /* enable IRQ */ -#define ICH6_RBRWP_CLR 0x8000 /* read/write pointer clear */ /* below are so far hardcoded - should read registers in future */ #define ICH6_MAX_CORB_ENTRIES 256 #define ICH6_MAX_RIRB_ENTRIES 256 @@ -515,9 +524,9 @@ static void azx_init_cmd_io(struct azx *chip) /* set the corb write pointer to 0 */ azx_writew(chip, CORBWP, 0); /* reset the corb hw read pointer */ - azx_writew(chip, CORBRP, ICH6_RBRWP_CLR); + azx_writew(chip, CORBRP, ICH6_CORBRP_RST); /* enable corb dma */ - azx_writeb(chip, CORBCTL, ICH6_RBCTL_DMA_EN); + azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN); /* RIRB set up */ chip->rirb.addr = chip->rb.addr + 2048; @@ -529,7 +538,7 @@ static void azx_init_cmd_io(struct azx *chip) /* set the rirb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, RIRBSIZE, 0x02); /* reset the rirb hw write pointer */ - azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR); + azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST); /* set N=1, get RIRB response interrupt for new entry */ azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ @@ -796,7 +805,7 @@ static int azx_reset(struct azx *chip) } /* Accept unsolicited responses */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN); + azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL); /* detect codecs */ if (!chip->codec_mask) { @@ -2284,10 +2293,10 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, /* ATI chips seems buggy about 64bit DMA addresses */ if (chip->driver_type == AZX_DRIVER_ATI) - gcap &= ~0x01; + gcap &= ~ICH6_GCAP_64OK; /* allow 64bit DMA address if supported by H/W */ - if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) + if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64)); else { pci_set_dma_mask(pci, DMA_BIT_MASK(32)); -- cgit v1.1 From be461ba836770263826457624bc4a5173a1f5040 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Thu, 28 May 2009 05:10:50 -0400 Subject: ASoC: Add dummy S/PDIF codec support McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. Signed-off-by: Steve Chen Signed-off-by: Pavel Kiryukhin Signed-off-by: Naresh Medisetty Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 +++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/spdif_transciever.c | 69 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 +++++++++ 4 files changed, 92 insertions(+) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb07d9b..bbc97fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 + select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C @@ -91,6 +92,9 @@ config SND_SOC_L3 config SND_SOC_PCM3008 tristate +config SND_SOC_SPDIF + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 46c007c..8b75305 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c new file mode 100644 index 0000000..118e976 --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. TI DaVinci Audio controller uses this driver. + * + * Author: Steve Chen, + * Copyright: (C) 2009 MontaVista Software, Inc., + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +struct snd_soc_dai dit_stub_dai = { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(&dit_stub_dai); +} + +static int spdif_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&dit_stub_dai); + return 0; +} + +static struct platform_driver spdif_dit_driver = { + .probe = spdif_dit_probe, + .remove = spdif_dit_remove, + .driver = { + .name = "spdif-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&spdif_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&spdif_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h new file mode 100644 index 0000000..296f2eb --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, + * Copyright: (C) 2008 MontaVista Software, Inc., + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +extern struct snd_soc_dai dit_stub_dai; + +#endif /* CODEC_STUBS_H */ -- cgit v1.1 From 203350c1a8e23adf17fd9a96d8bfc7adf63c1ff6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 May 2009 14:51:00 +0100 Subject: ASoC: Initialise dev for the dummy S/PDIF DAI Also include the header to make sure the DAI is prototyped. Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transciever.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 118e976..218b33a 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -19,10 +19,11 @@ #include #include +#include "spdif_transciever.h" + #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE - struct snd_soc_dai dit_stub_dai = { .name = "DIT", .playback = { @@ -36,6 +37,7 @@ struct snd_soc_dai dit_stub_dai = { static int spdif_dit_probe(struct platform_device *pdev) { + dit_stub_dai.dev = &pdev->dev; return snd_soc_register_dai(&dit_stub_dai); } -- cgit v1.1 From 3fd43858c7937801134bd70ef1d411e44f9c0c1c Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 28 May 2009 14:12:27 +0200 Subject: ALSA: au88x0: fix .pointer callback Appearently, the used mask in the .pointer callback is invalid. It should be in period_bytes range. The period_bytes is pow(2), so simple bitwise operation is used. Idea was taken from ALSA bug#4455. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 3906f5a..f84bb19 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1255,8 +1255,8 @@ static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma) int temp; temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2)); - temp = (dma->period_virt * dma->period_bytes) + (temp & POS_MASK); - return (temp); + temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1)); + return temp; } static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma) @@ -1504,8 +1504,7 @@ static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) int temp; temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)); - //temp = (temp & POS_MASK) + (((temp>>WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK)*(dma->cfg0&POS_MASK)); - temp = (temp & POS_MASK) + ((dma->period_virt) * (dma->period_bytes)); + temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1)); return temp; } -- cgit v1.1 From e9ab33d03eb721a632214c0bbaa18652de88aa2d Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 28 May 2009 14:20:00 +0200 Subject: ALSA: au88x0: fix wrong period_elapsed() call The period_elapsed() call should be called when position moves. The idea was taken from ALSA bug#4455. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index f84bb19..23f49f3 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2440,7 +2440,8 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) spin_lock(&vortex->lock); for (i = 0; i < NR_ADB; i++) { if (vortex->dma_adb[i].fifo_status == FIFO_START) { - if (vortex_adbdma_bufshift(vortex, i)) ; + if (!vortex_adbdma_bufshift(vortex, i)) + continue; spin_unlock(&vortex->lock); snd_pcm_period_elapsed(vortex->dma_adb[i]. substream); -- cgit v1.1 From 8bea869c5e56234990e6bad92a543437115bfc18 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 27 Apr 2009 09:44:40 +0200 Subject: ALSA: PCM midlevel: improve fifo_size handling Move the fifo_size assignment to hw->ioctl callback to allow lowlevel drivers overwrite the default behaviour. fifo_size is in frames not bytes as specified in asound.h and alsa-lib's documentation, but most hardware have fixed byte based FIFOs. Introduce internal SNDRV_PCM_INFO_FIFO_IN_FRAMES. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/asound.h | 1 + include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 19 +++++++++++++++++++ sound/core/pcm_native.c | 15 ++++++++++++--- 4 files changed, 33 insertions(+), 3 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 6add80f..82aed3f 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -255,6 +255,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000 /* only half duplex */ #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ +#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 267effd..8a69b5c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -98,6 +98,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_IOCTL1_INFO 1 #define SNDRV_PCM_IOCTL1_CHANNEL_INFO 2 #define SNDRV_PCM_IOCTL1_GSTATE 3 +#define SNDRV_PCM_IOCTL1_FIFO_SIZE 4 #define SNDRV_PCM_TRIGGER_STOP 0 #define SNDRV_PCM_TRIGGER_START 1 diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d659995..adc2b0b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1524,6 +1524,23 @@ static int snd_pcm_lib_ioctl_channel_info(struct snd_pcm_substream *substream, return 0; } +static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, + void *arg) +{ + struct snd_pcm_hw_params *params = arg; + snd_pcm_format_t format; + int channels, width; + + params->fifo_size = substream->runtime->hw.fifo_size; + if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_FIFO_IN_FRAMES)) { + format = params_format(params); + channels = params_channels(params); + width = snd_pcm_format_physical_width(format); + params->fifo_size /= width * channels; + } + return 0; +} + /** * snd_pcm_lib_ioctl - a generic PCM ioctl callback * @substream: the pcm substream instance @@ -1545,6 +1562,8 @@ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, return snd_pcm_lib_ioctl_reset(substream, arg); case SNDRV_PCM_IOCTL1_CHANNEL_INFO: return snd_pcm_lib_ioctl_channel_info(substream, arg); + case SNDRV_PCM_IOCTL1_FIFO_SIZE: + return snd_pcm_lib_ioctl_fifo_size(substream, arg); } return -ENXIO; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 45dc53f..84da3ba 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -312,9 +312,18 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, hw = &substream->runtime->hw; if (!params->info) - params->info = hw->info; - if (!params->fifo_size) - params->fifo_size = hw->fifo_size; + params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; + if (!params->fifo_size) { + if (snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) == + snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) && + snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) == + snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) { + changed = substream->ops->ioctl(substream, + SNDRV_PCM_IOCTL1_FIFO_SIZE, params); + if (params < 0) + return changed; + } + } params->rmask = 0; return 0; } -- cgit v1.1 From c62a01ad6e746fae9c93f51ea67e0abfd8d94b58 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 28 May 2009 11:21:52 +0200 Subject: ALSA: PCM midlevel: introduce mask for xrun_debug() macro For debugging purposes, it is better to separate actions. Bit-values: 1: show bad PCM ring buffer pointer 2: show also stack (to debug kernel latency issues) 4: check pointer against system jiffies Example: 5: show bad PCM ring buffer pointer and do jiffies check Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index adc2b0b..25cb367 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,20 +126,20 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -#define xrun_debug(substream) ((substream)->pstr->xrun_debug) +#define xrun_debug(substream, mask) ((substream)->pstr->xrun_debug & (mask)) #else -#define xrun_debug(substream) 0 +#define xrun_debug(substream, mask) 0 #endif -#define dump_stack_on_xrun(substream) do { \ - if (xrun_debug(substream) > 1) \ - dump_stack(); \ +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream, 2)) \ + dump_stack(); \ } while (0) static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - if (xrun_debug(substream)) { + if (xrun_debug(substream, 1)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, substream->pcm->device, @@ -197,7 +197,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, #define hw_ptr_error(substream, fmt, args...) \ do { \ - if (xrun_debug(substream)) { \ + if (xrun_debug(substream, 1)) { \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -251,7 +251,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream)) + if (!xrun_debug(substream, 4)) goto no_jiffies_check; /* Skip the jiffies check for hardwares with BATCH flag. @@ -342,7 +342,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } /* Do jiffies check only in xrun_debug mode */ - if (xrun_debug(substream) && + if (xrun_debug(substream, 4) && ((delta * HZ) / runtime->rate) > jdelta + HZ/100) { hw_ptr_error(substream, "hw_ptr skipping! " -- cgit v1.1 From 13f040f9e55d41e92e485389123654971e03b819 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 28 May 2009 11:31:20 +0200 Subject: ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed Some hardware might have bigger FIFOs and DMA pointer value will be updated in large chunks. Do not update hw_ptr_jiffies and position timestamp when hw_ptr value was not changed. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 25cb367..0f299a5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -138,6 +138,10 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram static void xrun(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); if (xrun_debug(substream, 1)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", @@ -154,8 +158,6 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, { snd_pcm_uframes_t pos; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) return pos; /* XRUN */ @@ -298,10 +300,15 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + if (runtime->status->hw_ptr == new_hw_ptr) + return 0; + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; runtime->hw_ptr_interrupt = hw_ptr_interrupt; + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); return snd_pcm_update_hw_ptr_post(substream, runtime); } @@ -356,9 +363,14 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + if (runtime->status->hw_ptr != new_hw_ptr) + return 0; + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); return snd_pcm_update_hw_ptr_post(substream, runtime); } -- cgit v1.1 From a4444da31ec92f89cd6923579c20a9c240439cfc Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 28 May 2009 12:31:56 +0200 Subject: ALSA: PCM midlevel: lower jiffies check margin using runtime->delay value When hardware has large FIFO, it is necessary to lower jiffies margin by count of queued samples. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0f299a5..dd9126b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -263,6 +263,9 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; hdelta = new_hw_ptr - old_hw_ptr; + if (hdelta < runtime->delay) + goto no_jiffies_check; + hdelta -= runtime->delay; jdelta = jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { delta = jdelta / @@ -349,8 +352,12 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } /* Do jiffies check only in xrun_debug mode */ - if (xrun_debug(substream, 4) && - ((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + if (!xrun_debug(substream, 4)) + goto no_jiffies_check; + if (delta < runtime->delay) + goto no_jiffies_check; + delta -= runtime->delay; + if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", @@ -359,6 +366,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) ((delta * HZ) / runtime->rate)); return 0; } + no_jiffies_check: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); -- cgit v1.1 From 0528c7494e67c30329d086df141d2dd691f01556 Mon Sep 17 00:00:00 2001 From: Michal Marek Date: Tue, 26 May 2009 17:07:52 +0200 Subject: ALSA: clean up the logic for building sequencer modules Instead of mangling the CONFIG_* variables in the makefiles over and over, set a few helper variables in Kconfig. Signed-off-by: Michal Marek Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 2 ++ sound/core/seq/Kconfig | 16 ++++++++++++++++ sound/core/seq/Makefile | 18 +++++------------- sound/drivers/opl3/Makefile | 10 +--------- sound/drivers/opl4/Makefile | 10 +--------- sound/isa/sb/Makefile | 10 +--------- sound/pci/emu10k1/Makefile | 10 +--------- sound/synth/Makefile | 12 ++---------- sound/synth/emux/Makefile | 12 ++---------- 9 files changed, 31 insertions(+), 69 deletions(-) create mode 100644 sound/core/seq/Kconfig diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 7bbdda0..6061fb5 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -205,3 +205,5 @@ config SND_PCM_XRUN_DEBUG config SND_VMASTER bool + +source "sound/core/seq/Kconfig" diff --git a/sound/core/seq/Kconfig b/sound/core/seq/Kconfig new file mode 100644 index 0000000..b851fd8 --- /dev/null +++ b/sound/core/seq/Kconfig @@ -0,0 +1,16 @@ +# define SND_XXX_SEQ to min(SND_SEQUENCER,SND_XXX) + +config SND_RAWMIDI_SEQ + def_tristate SND_SEQUENCER && SND_RAWMIDI + +config SND_OPL3_LIB_SEQ + def_tristate SND_SEQUENCER && SND_OPL3_LIB + +config SND_OPL4_LIB_SEQ + def_tristate SND_SEQUENCER && SND_OPL4_LIB + +config SND_SBAWE_SEQ + def_tristate SND_SEQUENCER && SND_SBAWE + +config SND_EMU10K1_SEQ + def_tristate SND_SEQUENCER && SND_EMU10K1 diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 0695937..1bcb360 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -17,14 +17,6 @@ snd-seq-midi-event-objs := seq_midi_event.o snd-seq-dummy-objs := seq_dummy.o snd-seq-virmidi-objs := seq_virmidi.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o @@ -33,8 +25,8 @@ obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o # Toplevel Module Dependency obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o +obj-$(CONFIG_SND_RAWMIDI_SEQ) += snd-seq-midi.o snd-seq-midi-event.o +obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 19767a6..7f2c2a1 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -7,14 +7,6 @@ snd-opl3-lib-objs := opl3_lib.o opl3_synth.o snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-opl3-synth.o +obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-opl3-synth.o diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile index d178b39..b94009b 100644 --- a/sound/drivers/opl4/Makefile +++ b/sound/drivers/opl4/Makefile @@ -6,13 +6,5 @@ snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-opl4-synth.o +obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-opl4-synth.o diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index 1098a56..faeffceb 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -13,14 +13,6 @@ snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o @@ -33,4 +25,4 @@ ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o endif -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emu8000-synth.o diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile index cf2d563..fc5591e 100644 --- a/sound/pci/emu10k1/Makefile +++ b/sound/pci/emu10k1/Makefile @@ -9,15 +9,7 @@ snd-emu10k1-objs := emu10k1.o emu10k1_main.o \ snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o snd-emu10k1x-objs := emu10k1x.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emu10k1-synth.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emu10k1-synth.o obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o diff --git a/sound/synth/Makefile b/sound/synth/Makefile index e99fd76..11eb06a 100644 --- a/sound/synth/Makefile +++ b/sound/synth/Makefile @@ -5,16 +5,8 @@ snd-util-mem-objs := util_mem.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o obj-$(CONFIG_SND_TRIDENT) += snd-util-mem.o -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-util-mem.o -obj-$(call sequencer,$(CONFIG_SND)) += emux/ +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-util-mem.o +obj-$(CONFIG_SND_SEQUENCER) += emux/ diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile index b690352..328594e 100644 --- a/sound/synth/emux/Makefile +++ b/sound/synth/emux/Makefile @@ -7,14 +7,6 @@ snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ emux_effect.o emux_proc.o emux_hwdep.o soundfont.o \ $(if $(CONFIG_SND_SEQUENCER_OSS),emux_oss.o) -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependencies -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emux-synth.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emux-synth.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emux-synth.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emux-synth.o -- cgit v1.1 From 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 May 2009 09:22:37 +0300 Subject: ASoC: TWL4030: Use reg_cache in twl4030_init_chip Use the codec->reg_cache instead of the array directly in twl4030_init_chip for setting the default values. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 63ebd17..df474a5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -145,7 +145,6 @@ struct twl4030_priv { static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { - u8 *cache = codec->reg_cache; if (reg >= TWL4030_CACHEREGNUM) return -EIO; @@ -204,6 +203,7 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) static void twl4030_init_chip(struct snd_soc_codec *codec) { + u8 *cache = codec->reg_cache; int i; /* clear CODECPDZ prior to setting register defaults */ @@ -211,7 +211,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) - twl4030_write(codec, i, twl4030_reg[i]); + twl4030_write(codec, i, cache[i]); } -- cgit v1.1 From ba84bfcd2b6fdc5a9ac53a4ab103088c99f84a12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 30 May 2009 08:59:03 +0200 Subject: ALSA: hda - Fix reverted LED setup for HP The commit 86d190e77c44cb057742dcc871b12ebd4633c387 reverted the bit flip of LED GPIO for HP DX and DV4-1222nr. Fixed now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a1b4c94..a915f40 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4666,9 +4666,9 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) - spec->gpio_data |= spec->gpio_led; /* white */ - else spec->gpio_data &= ~spec->gpio_led; /* orange */ + else + spec->gpio_data |= spec->gpio_led; /* white */ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, -- cgit v1.1 From 8a933ece41a59ce077eeffe5b9bf08b14d173c58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 31 May 2009 09:28:12 +0200 Subject: ALSA: hda - Fix a typo in the previous patch ICH6_GCTL_RESET was wrongly set to another bit by the commit b21fadb9c1852c91622ca1dccfeb144bc535e36e. This caused a problem when the codec needs really a reset (e.g. recovering from the communication error at probe). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f63bc65..b063d0e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -148,7 +148,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_OUTPAY 0x04 #define ICH6_REG_INPAY 0x06 #define ICH6_REG_GCTL 0x08 -#define ICH6_GCTL_RESET (1 << 1) /* controller reset */ +#define ICH6_GCTL_RESET (1 << 0) /* controller reset */ #define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */ #define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ #define ICH6_REG_WAKEEN 0x0c -- cgit v1.1 From 93bfd01227408a62006a4e4f640a6056abc6af7a Mon Sep 17 00:00:00 2001 From: Andrea Borgia Date: Mon, 1 Jun 2009 10:48:54 +0200 Subject: ALSA: usb-audio - quirk for USB Aureon cards Add quirk to provide proper naming of the Terratec Aureon 5.1 MkII USB card. Signed-off-by: Andrea Borgia Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 5d955aa..d84d6f3 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1951,6 +1951,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "TerraTec", + .product_name = "Aureon 5.1 MkII", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ USB_DEVICE(0x0ccd, 0x0035), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", -- cgit v1.1 From 6efd2cd5e8c566b9c2b4c19830e5e120b442d040 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 1 Jun 2009 10:59:51 +0200 Subject: ALSA: usb-audio - Add quirk for Roland/Edirol M-16DX Added a half-working quirk for Roland/Edirol M-16DX. This enables the capture on the device but the playback on it seems still problematic becuase of lack of sync with the capture clock. Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index d84d6f3..58f24f5 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1470,6 +1470,41 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Edirol M-16DX */ + /* FIXME: This quirk gives a good-working capture stream but the + * playback seems problematic because of lacking of sync + * with capture stream. It needs to sync with the capture + * clock. As now, you'll get frequent sound distortions + * via the playback. + */ + USB_DEVICE(0x0582, 0x00c4), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* BOSS GT-10 */ USB_DEVICE(0x0582, 0x00da), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.1 From a3d6ab9723060e8d67e28a8d9142e6d08953d07b Mon Sep 17 00:00:00 2001 From: Wei Ni Date: Mon, 1 Jun 2009 16:37:28 +0800 Subject: ALSA: hda - Support NVIDIA 8 channel HDMI audio Support 8 channel HDMI audio for MCP78/7A Signed-off-by: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 279 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 248 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index d57d813..f5792e2 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -35,9 +35,28 @@ struct nvhdmi_spec { struct hda_pcm pcm_rec; }; +#define Nv_VERB_SET_Channel_Allocation 0xF79 +#define Nv_VERB_SET_Info_Frame_Checksum 0xF7A +#define Nv_VERB_SET_Audio_Protection_On 0xF98 +#define Nv_VERB_SET_Audio_Protection_Off 0xF99 + +#define Nv_Master_Convert_nid 0x04 +#define Nv_Master_Pin_nid 0x05 + +static hda_nid_t nvhdmi_convert_nids[4] = { + /*front, rear, clfe, rear_surr */ + 0x6, 0x8, 0xa, 0xc, +}; + static struct hda_verb nvhdmi_basic_init[] = { + /* set audio protect on */ + { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, /* enable digital output on pin widget */ - { 0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0x7, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0x9, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0xb, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0xd, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, {} /* terminator */ }; @@ -66,48 +85,205 @@ static int nvhdmi_init(struct hda_codec *codec) * Digital out */ static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); } -static int nvhdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int nvhdmi_dig_playback_pcm_close_8ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; + int i; + + snd_hda_codec_write(codec, Nv_Master_Convert_nid, + 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + for (i = 0; i < 4; i++) { + /* set the stream id */ + snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0, + AC_VERB_SET_CHANNEL_STREAMID, 0); + /* set the stream format */ + snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0, + AC_VERB_SET_STREAM_FORMAT, 0); + } + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } -static int nvhdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) +static int nvhdmi_dig_playback_pcm_close_2ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct nvhdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + int chs; + unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + int i; + + mutex_lock(&codec->spdif_mutex); + + chs = substream->runtime->channels; + chan = chs ? (chs - 1) : 1; + + switch (chs) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; + dataDCC2 = 0x2; + + /* set the Audio InforFrame Channel Allocation */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write(codec, + Nv_Master_Convert_nid, + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + + /* set the stream id */ + snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | 0x0); + + /* set the stream format */ + snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); + + /* turn on again (if needed) */ + /* enable and set the channel status audio/data flag */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { + snd_hda_codec_write(codec, + Nv_Master_Convert_nid, + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + snd_hda_codec_write(codec, + Nv_Master_Convert_nid, + 0, + AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); + } + + for (i = 0; i < 4; i++) { + if (chs == 2) + channel_id = 0; + else + channel_id = i * 2; + + /* turn off SPDIF once; + *otherwise the IEC958 bits won't be updated + */ + if (codec->spdif_status_reset && + (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + /* set the stream id */ + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_CHANNEL_STREAMID, + (stream_tag << 4) | channel_id); + /* set the stream format */ + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_STREAM_FORMAT, + format); + /* turn on again (if needed) */ + /* enable and set the channel status audio/data flag */ + if (codec->spdif_status_reset && + (codec->spdif_ctls & AC_DIG1_ENABLE)) { + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + snd_hda_codec_write(codec, + nvhdmi_convert_nids[i], + 0, + AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); + } + } + + /* set the Audio Info Frame Checksum */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); + + mutex_unlock(&codec->spdif_mutex); + return 0; +} + +static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) { struct nvhdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); + format, substream); } -static struct hda_pcm_stream nvhdmi_pcm_digital_playback = { +static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = Nv_Master_Convert_nid, + .rates = SNDRV_PCM_RATE_48000, + .maxbps = 16, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .ops = { + .open = nvhdmi_dig_playback_pcm_open, + .close = nvhdmi_dig_playback_pcm_close_8ch, + .prepare = nvhdmi_dig_playback_pcm_prepare_8ch + }, +}; + +static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .nid = 0x4, /* NID to query formats and rates and setup streams */ + .nid = Nv_Master_Convert_nid, .rates = SNDRV_PCM_RATE_48000, .maxbps = 16, .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = nvhdmi_dig_playback_pcm_open, - .close = nvhdmi_dig_playback_pcm_close, - .prepare = nvhdmi_dig_playback_pcm_prepare + .close = nvhdmi_dig_playback_pcm_close_2ch, + .prepare = nvhdmi_dig_playback_pcm_prepare_2ch }, }; -static int nvhdmi_build_pcms(struct hda_codec *codec) +static int nvhdmi_build_pcms_8ch(struct hda_codec *codec) { struct nvhdmi_spec *spec = codec->spec; struct hda_pcm *info = &spec->pcm_rec; @@ -117,7 +293,24 @@ static int nvhdmi_build_pcms(struct hda_codec *codec) info->name = "NVIDIA HDMI"; info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] + = nvhdmi_pcm_digital_playback_8ch; + + return 0; +} + +static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "NVIDIA HDMI"; + info->pcm_type = HDA_PCM_TYPE_HDMI; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] + = nvhdmi_pcm_digital_playback_2ch; return 0; } @@ -127,14 +320,40 @@ static void nvhdmi_free(struct hda_codec *codec) kfree(codec->spec); } -static struct hda_codec_ops nvhdmi_patch_ops = { +static struct hda_codec_ops nvhdmi_patch_ops_8ch = { + .build_controls = nvhdmi_build_controls, + .build_pcms = nvhdmi_build_pcms_8ch, + .init = nvhdmi_init, + .free = nvhdmi_free, +}; + +static struct hda_codec_ops nvhdmi_patch_ops_2ch = { .build_controls = nvhdmi_build_controls, - .build_pcms = nvhdmi_build_pcms, + .build_pcms = nvhdmi_build_pcms_2ch, .init = nvhdmi_init, .free = nvhdmi_free, }; -static int patch_nvhdmi(struct hda_codec *codec) +static int patch_nvhdmi_8ch(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 8; + spec->multiout.dig_out_nid = Nv_Master_Convert_nid; + + codec->patch_ops = nvhdmi_patch_ops_8ch; + + return 0; +} + +static int patch_nvhdmi_2ch(struct hda_codec *codec) { struct nvhdmi_spec *spec; @@ -144,13 +363,11 @@ static int patch_nvhdmi(struct hda_codec *codec) codec->spec = spec; - spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = 0x4; /* NID for copying analog to digital, - * seems to be unused in pure-digital - * case. */ + spec->multiout.dig_out_nid = Nv_Master_Convert_nid; - codec->patch_ops = nvhdmi_patch_ops; + codec->patch_ops = nvhdmi_patch_ops_2ch; return 0; } @@ -159,11 +376,11 @@ static int patch_nvhdmi(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; -- cgit v1.1 From 7c922de709b90badc19705e4f998e6d5b44c419b Mon Sep 17 00:00:00 2001 From: Nickolas Lloyd Date: Mon, 1 Jun 2009 11:12:29 +0200 Subject: ALSA: hda - Jack Mode changes for Sigmatel boards This patch changes Line In as Out Switch and Mic In as Out Switch to enums for consistency, and causes all mic and line in ports to be probed and controls to be added appropriately. Signed-off-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 219 ++++++++++++++++++++++++++--------------- 1 file changed, 140 insertions(+), 79 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a915f40..48f4a36 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -670,62 +670,6 @@ static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) return vref; } -static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int new_vref; - unsigned int error; - - if (ucontrol->value.enumerated.item[0] == 0) - new_vref = AC_PINCTL_VREF_80; - else if (ucontrol->value.enumerated.item[0] == 1) - new_vref = AC_PINCTL_VREF_GRD; - else - new_vref = AC_PINCTL_VREF_HIZ; - - if (new_vref != stac92xx_vref_get(codec, kcontrol->private_value)) { - error = stac92xx_vref_set(codec, - kcontrol->private_value, new_vref); - return error; - } - - return 0; -} - -static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int vref = stac92xx_vref_get(codec, kcontrol->private_value); - if (vref == AC_PINCTL_VREF_80) - ucontrol->value.enumerated.item[0] = 0; - else if (vref == AC_PINCTL_VREF_GRD) - ucontrol->value.enumerated.item[0] = 1; - else if (vref == AC_PINCTL_VREF_HIZ) - ucontrol->value.enumerated.item[0] = 2; - - return 0; -} - -static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static char *texts[] = { - "Mic In", "Line In", "Line Out" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 3; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= 3) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -2652,7 +2596,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec) return 0; } -static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid) +static unsigned int stac92xx_get_default_vref(struct hda_codec *codec, + hda_nid_t nid) { unsigned int pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; @@ -2706,15 +2651,108 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, return 1; } -#define stac92xx_io_switch_info snd_ctl_boolean_mono_info +static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int i; + static char *texts[] = { + "Mic In", "Line In", "Line Out" + }; + + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = kcontrol->private_value; + + if (nid == spec->mic_switch || nid == spec->line_switch) + i = 3; + else + i = 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = i; + uinfo->count = 1; + if (uinfo->value.enumerated.item >= i) + uinfo->value.enumerated.item = i-1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value; + unsigned int vref = stac92xx_vref_get(codec, nid); + + if (vref == stac92xx_get_default_vref(codec, nid)) + ucontrol->value.enumerated.item[0] = 0; + else if (vref == AC_PINCTL_VREF_GRD) + ucontrol->value.enumerated.item[0] = 1; + else if (vref == AC_PINCTL_VREF_HIZ) + ucontrol->value.enumerated.item[0] = 2; + + return 0; +} + +static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int new_vref = 0; + unsigned int error; + hda_nid_t nid = kcontrol->private_value; + + if (ucontrol->value.enumerated.item[0] == 0) + new_vref = stac92xx_get_default_vref(codec, nid); + else if (ucontrol->value.enumerated.item[0] == 1) + new_vref = AC_PINCTL_VREF_GRD; + else if (ucontrol->value.enumerated.item[0] == 2) + new_vref = AC_PINCTL_VREF_HIZ; + else + return 0; + + if (new_vref != stac92xx_vref_get(codec, nid)) { + error = stac92xx_vref_set(codec, nid, new_vref); + return error; + } + + return 0; +} + +static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2]; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + if (kcontrol->private_value == spec->line_switch) + texts[0] = "Line In"; + else + texts[0] = "Mic In"; + texts[1] = "Line Out"; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->value.enumerated.items = 2; + uinfo->count = 1; + + if (uinfo->value.enumerated.item >= 2) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - int io_idx = kcontrol-> private_value & 0xff; + hda_nid_t nid = kcontrol->private_value; + int io_idx = (nid == spec->mic_switch) ? 1 : 0; - ucontrol->value.integer.value[0] = spec->io_switch[io_idx]; + ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx]; return 0; } @@ -2722,9 +2760,9 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value >> 8; - int io_idx = kcontrol-> private_value & 0xff; - unsigned short val = !!ucontrol->value.integer.value[0]; + hda_nid_t nid = kcontrol->private_value; + int io_idx = (nid == spec->mic_switch) ? 1 : 0; + unsigned short val = !!ucontrol->value.enumerated.item[0]; spec->io_switch[io_idx] = val; @@ -2733,7 +2771,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ else { unsigned int pinctl = AC_PINCTL_IN_EN; if (io_idx) /* set VREF for mic */ - pinctl |= stac92xx_get_vref(codec, nid); + pinctl |= stac92xx_get_default_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } @@ -2891,6 +2929,34 @@ static struct snd_kcontrol_new stac_input_src_temp = { .put = stac92xx_mux_enum_put, }; +static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, + hda_nid_t nid, int idx) +{ + int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int control = 0; + struct sigmatel_spec *spec = codec->spec; + char name[22]; + + if (!((get_defcfg_connect(def_conf)) & AC_JACK_PORT_FIXED)) { + if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD + && nid == spec->line_switch) + control = STAC_CTL_WIDGET_IO_SWITCH; + else if (snd_hda_query_pin_caps(codec, nid) + & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT)) + control = STAC_CTL_WIDGET_DC_BIAS; + else if (nid == spec->mic_switch) + control = STAC_CTL_WIDGET_IO_SWITCH; + } + + if (control) { + strcpy(name, auto_pin_cfg_labels[idx]); + return stac92xx_add_control(codec->spec, control, + strcat(name, " Jack Mode"), nid); + } + + return 0; +} + static int stac92xx_add_input_source(struct sigmatel_spec *spec) { struct snd_kcontrol_new *knew; @@ -3253,7 +3319,9 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid; int err; + int idx; err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, spec->multiout.dac_nids, @@ -3270,20 +3338,13 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } - if (spec->line_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, - "Line In as Output Switch", - spec->line_switch << 8); - if (err < 0) - return err; - } - - if (spec->mic_switch) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_DC_BIAS, - "Mic Jack Mode", - spec->mic_switch); - if (err < 0) - return err; + for (idx = AUTO_PIN_MIC; idx <= AUTO_PIN_FRONT_LINE; idx++) { + nid = cfg->input_pins[idx]; + if (nid) { + err = stac92xx_add_jack_mode_control(codec, nid, idx); + if (err < 0) + return err; + } } return 0; @@ -4193,7 +4254,7 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int pinctl, conf; if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) { /* for mic pins, force to initialize */ - pinctl = stac92xx_get_vref(codec, nid); + pinctl = stac92xx_get_default_vref(codec, nid); pinctl |= AC_PINCTL_IN_EN; stac92xx_auto_set_pinctl(codec, nid, pinctl); } else { -- cgit v1.1 From 0e4835c198e7dd9d814de25a17a1d6682107c394 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 25 May 2009 16:44:20 +0200 Subject: ALSA: hda-intel: improve initialization for ALC262_HP_BPC model Fix issues for 3 generations of HP workstations. The modest modifications do the following: 1. Change the second MIC from device 3 to device 1 2. Init the "boost" values to "0" by default From: John Brown Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 41 ++++++++++++++++++++++++++++------------- 1 file changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index add920a..d0786e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10639,31 +10639,46 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 }, /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */ + /* Input mixer1: only unmute Mic */ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, /* Input mixer3 */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -- cgit v1.1 From eaf1ac8bb58888e0773c0b81dfedb9d7c0123a1d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Jun 2009 14:06:40 +0300 Subject: ASoC: TWL4030: Check the interface format for 4 channel mode In addition to the operating mode check, also check the codec's interface format in case of four channel mode. If the codec is not in TDM (DSP_A) mode, return with error. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df474a5..c53c7ca 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1608,9 +1608,15 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* If the substream has 4 channel, do the necessary setup */ if (params_channels(params) == 4) { - /* Safety check: are we in the correct operating mode? */ - if ((twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & - TWL4030_OPTION_1)) + u8 format, mode; + + format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + + /* Safety check: are we in the correct operating mode and + * the interface is in TDM mode? */ + if ((mode & TWL4030_OPTION_1) && + ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM)) twl4030_tdm_enable(codec, substream->stream, 1); else return -EINVAL; -- cgit v1.1 From 92b9de8342c6ad0d930333851190ed25b88b190c Mon Sep 17 00:00:00 2001 From: Kacper Szczesniak Date: Tue, 2 Jun 2009 00:55:19 +0200 Subject: ALSA: hda - Macbook[Pro] 5 6ch support this is a patch against current snapshot that adds: 6 channels support for the MB5 model Signed-off-by: Kacper Szczesniak Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 80 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 65 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0786e9..12d6015 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6266,6 +6266,34 @@ static struct hda_channel_mode alc885_mbp_6ch_modes[2] = { { 6, alc885_mbp_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_mb5_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_mb5_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b @@ -6310,10 +6338,14 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = { }; static struct snd_kcontrol_new alc885_mb5_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -6322,6 +6354,7 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT), { } /* end */ }; + static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6551,22 +6584,39 @@ static struct hda_verb alc882_macpro_init_verbs[] = { /* Macbook 5,1 */ static struct hda_verb alc885_mb5_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Front mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* LineOut mixer */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin: output 0 (0x0d) */ + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* HP Pin: output 0 (0x0c) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -6986,13 +7036,13 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc885_mbp3_init_hook, }, [ALC885_MB5] = { - .mixers = { alc885_mb5_mixer }, + .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mb5_init_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .channel_mode = alc885_mb5_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, -- cgit v1.1 From d22142aa1b23d64d01f87a94b5756ff6cbd1022f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Ozan=20=C3=87a=C4=9Flayan?= Date: Mon, 1 Jun 2009 23:13:17 +0300 Subject: ALSA: hda - fix audio on LG R510 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently, LG R510 is only able to produce sound on headphones, the internal speakers are not working. The user tested and confirmed that with model=Dell headphones, internal speakers and the microphone are working flawlessly. Tested-by: Serdar Soytetir Signed-off-by: Ozan Çağlayan Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12d6015..56c2f84 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12111,6 +12111,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), -- cgit v1.1 From 8871e5b91518a47284b6bc2603b44dbc79c85446 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 01:02:50 +0200 Subject: ALSA: hda - Reorder and clean-up ALC268 quirk table Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some entries using SND_PCI_QUIRK_MASK(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56c2f84..ad6cc42 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12111,17 +12111,16 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), - SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), - SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series", + ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), - SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff64, "TOSHIBA L305", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), + SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", + ALC268_TOSHIBA), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), - SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), + SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL), {} }; -- cgit v1.1 From 8dd783304e6d0f7c2830365d63f75f08aa343e10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 01:16:07 +0200 Subject: ALSA: hda - Add codec bus reset and verb-retry at critical errors Some machines machine cause a severe CORB/RIRB stall in certain weird conditions, such as PA access at the start up together with fglrx driver. This seems unable to be recovered without the controller reset. This patch allows the bus controller reset at critical errors so that the communication gets recovered again. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 32 ++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 6 +++++- sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++++++++++------ 3 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d1d5fb9..aa0e1c1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -165,28 +165,29 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, unsigned int *res) { struct hda_bus *bus = codec->bus; - int err, repeated = 0; + int err; if (res) *res = -1; + again: snd_hda_power_up(codec); mutex_lock(&bus->cmd_mutex); - again: err = bus->ops.command(bus, cmd); - if (!err) { - if (res) { - *res = bus->ops.get_response(bus); - if (*res == -1 && bus->rirb_error) { - if (repeated++ < 1) { - snd_printd(KERN_WARNING "hda_codec: " - "Trying verb 0x%08x again\n", cmd); - goto again; - } - } - } - } + if (!err && res) + *res = bus->ops.get_response(bus); mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); + if (res && *res == -1 && bus->rirb_error) { + if (bus->response_reset) { + snd_printd("hda_codec: resetting BUS due to " + "fatal communication error\n"); + bus->ops.bus_reset(bus); + } + goto again; + } + /* clear reset-flag when the communication gets recovered */ + if (!err) + bus->response_reset = 0; return err; } @@ -3894,11 +3895,10 @@ EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); /** * snd_hda_suspend - suspend the codecs * @bus: the HDA bus - * @state: suspsend state * * Returns 0 if successful. */ -int snd_hda_suspend(struct hda_bus *bus, pm_message_t state) +int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c5bd40f..f5fa5d1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -574,6 +574,8 @@ struct hda_bus_ops { /* attach a PCM stream */ int (*attach_pcm)(struct hda_bus *bus, struct hda_codec *codec, struct hda_pcm *pcm); + /* reset bus for retry verb */ + void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ void (*pm_notify)(struct hda_bus *bus); @@ -624,6 +626,8 @@ struct hda_bus { unsigned int needs_damn_long_delay :1; unsigned int shutdown :1; /* being unloaded */ unsigned int rirb_error:1; /* error in codec communication */ + unsigned int response_reset:1; /* controller was reset */ + unsigned int in_reset:1; /* during reset operation */ }; /* @@ -907,7 +911,7 @@ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); * power management */ #ifdef CONFIG_PM -int snd_hda_suspend(struct hda_bus *bus, pm_message_t state); +int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b063d0e..44f9a0a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -661,14 +661,23 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) return -1; } - snd_printk(KERN_ERR SFX "azx_get_response timeout (ERROR): " - "last cmd=0x%08x\n", chip->last_cmd); - /* re-initialize CORB/RIRB */ - spin_lock_irq(&chip->reg_lock); + /* a fatal communication error; need either to reset or to fallback + * to the single_cmd mode + */ bus->rirb_error = 1; + if (!bus->response_reset && !bus->in_reset) { + bus->response_reset = 1; + return -1; /* give a chance to retry */ + } + + snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " + "switching to single_cmd mode: last cmd=0x%08x\n", + chip->last_cmd); + chip->single_cmd = 1; + bus->response_reset = 0; + /* re-initialize CORB/RIRB */ azx_free_cmd_io(chip); azx_init_cmd_io(chip); - spin_unlock_irq(&chip->reg_lock); return -1; } @@ -709,6 +718,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val) struct azx *chip = bus->private_data; int timeout = 50; + bus->rirb_error = 0; while (timeout--) { /* check ICB busy bit */ if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) { @@ -1247,6 +1257,23 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, struct hda_pcm *cpcm); static void azx_stop_chip(struct azx *chip); +static void azx_bus_reset(struct hda_bus *bus) +{ + struct azx *chip = bus->private_data; + int i; + + bus->in_reset = 1; + azx_stop_chip(chip); + azx_init_chip(chip); + if (chip->initialized) { + for (i = 0; i < AZX_MAX_PCMS; i++) + snd_pcm_suspend_all(chip->pcm[i]); + snd_hda_suspend(chip->bus); + snd_hda_resume(chip->bus); + } + bus->in_reset = 0; +} + /* * Codec initialization */ @@ -1270,6 +1297,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, bus_temp.ops.command = azx_send_cmd; bus_temp.ops.get_response = azx_get_response; bus_temp.ops.attach_pcm = azx_attach_pcm_stream; + bus_temp.ops.bus_reset = azx_bus_reset; #ifdef CONFIG_SND_HDA_POWER_SAVE bus_temp.power_save = &power_save; bus_temp.ops.pm_notify = azx_power_notify; @@ -1997,7 +2025,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) - snd_hda_suspend(chip->bus, state); + snd_hda_suspend(chip->bus); azx_stop_chip(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); -- cgit v1.1 From b20f3b834673be9ead83a3c6f07fa3881d1a990f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 01:20:22 +0200 Subject: ALSA: hda - Limit codec-verb retry to limited hardwares The reset of a BUS controller during operations is somehow risky and shouldn't be done inevitably for devices that have apparently no such codec-communication problems. This patch adds the check of the hardware and limits the bus-reset capability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 12 ++---------- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/patch_sigmatel.c | 9 +++++++++ 4 files changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index aa0e1c1..562403a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -214,11 +214,6 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_read); -/* Define the below to send and receive verbs synchronously. - * If you often get any codec communication errors, this is worth to try. - */ -/* #define SND_HDA_SUPPORT_SYNC_WRITE */ - /** * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec @@ -235,12 +230,9 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); -#ifdef SND_HDA_SUPPORT_SYNC_WRITE unsigned int res; - return codec_exec_verb(codec, cmd, &res); -#else - return codec_exec_verb(codec, cmd, NULL); -#endif + return codec_exec_verb(codec, cmd, + codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index f5fa5d1..cad79ef 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -624,6 +624,9 @@ struct hda_bus { /* misc op flags */ unsigned int needs_damn_long_delay :1; + unsigned int allow_bus_reset:1; /* allow bus reset at fatal error */ + unsigned int sync_write:1; /* sync after verb write */ + /* status for codec/controller */ unsigned int shutdown :1; /* being unloaded */ unsigned int rirb_error:1; /* error in codec communication */ unsigned int response_reset:1; /* controller was reset */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 44f9a0a..9f44645 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -665,7 +665,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus) * to the single_cmd mode */ bus->rirb_error = 1; - if (!bus->response_reset && !bus->in_reset) { + if (bus->allow_bus_reset && !bus->response_reset && !bus->in_reset) { bus->response_reset = 1; return -1; /* give a chance to retry */ } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 48f4a36..42f944b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5375,6 +5375,15 @@ again: if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); + /* Some HP machines seem to have unstable codec communications + * especially with ATI fglrx driver. For recovering from the + * CORB/RIRB stall, allow the BUS reset and keep always sync + */ + if (spec->board_config == STAC_HP_DV5) { + codec->bus->sync_write = 1; + codec->bus->allow_bus_reset = 1; + } + spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- cgit v1.1 From 8a4259bf89d23bfd58d87e275ef6da29cea6b3c5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 08:40:51 +0200 Subject: ALSA: ctxfi - Fix Oops at mmapping Replace a spinlock with a mutex protecting the vm block list at mmap / munmap calls, which caused Oops like below: BUG: sleeping function called from invalid context at mm/slub.c:1599 in_atomic(): 0, irqs_disabled(): 1, pid: 32065, name: xine Pid: 32065, comm: xine Tainted: P 2.6.29.4-75.fc10.x86_64 #1 Call Trace: [] __might_sleep+0x105/0x10a [] kmem_cache_alloc+0x32/0xe2 [] ct_vm_map+0xfa/0x19e [snd_ctxfi] [] ct_map_audio_buffer+0x4c/0x76 [snd_ctxfi] [] atc_pcm_playback_prepare+0x1d7/0x2a8 [snd_ctxfi] [] ? up_read+0x9/0xb [] ? __up_read+0x7c/0x87 [] ct_pcm_playback_prepare+0x39/0x60 [snd_ctxfi] [] snd_pcm_do_prepare+0x16/0x28 [snd_pcm] [] snd_pcm_action_single+0x2d/0x5b [snd_pcm] [] snd_pcm_action_nonatomic+0x52/0x6a [snd_pcm] [] snd_pcm_common_ioctl1+0x404/0xc79 [snd_pcm] [] ? alloc_pages_current+0xb9/0xc2 [] ? new_slab+0x1a5/0x1cb [] ? vma_prio_tree_insert+0x23/0xc1 [] snd_pcm_playback_ioctl1+0x213/0x230 [snd_pcm] [] ? mmap_region+0x397/0x4c9 [] snd_pcm_playback_ioctl+0x2e/0x36 [snd_pcm] [] vfs_ioctl+0x2a/0x78 [] do_vfs_ioctl+0x462/0x4a2 [] ? default_spin_lock_flags+0x9/0xe [] ? trace_hardirqs_off_thunk+0x3a/0x6c [] sys_ioctl+0x55/0x77 [] system_call_fastpath+0x16/0x1b Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 11 ----------- sound/pci/ctxfi/ctatc.h | 1 - sound/pci/ctxfi/ctvmem.c | 14 +++++++++++--- sound/pci/ctxfi/ctvmem.h | 3 ++- 4 files changed, 13 insertions(+), 16 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index ead104e..1a4bb35 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -119,7 +119,6 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm); static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - unsigned long flags; struct snd_pcm_runtime *runtime; struct ct_vm *vm; @@ -129,9 +128,7 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) runtime = apcm->substream->runtime; vm = atc->vm; - spin_lock_irqsave(&atc->vm_lock, flags); apcm->vm_block = vm->map(vm, runtime->dma_area, runtime->dma_bytes); - spin_unlock_irqrestore(&atc->vm_lock, flags); if (NULL == apcm->vm_block) return -ENOENT; @@ -141,7 +138,6 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - unsigned long flags; struct ct_vm *vm; if (NULL == apcm->vm_block) @@ -149,9 +145,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) vm = atc->vm; - spin_lock_irqsave(&atc->vm_lock, flags); vm->unmap(vm, apcm->vm_block); - spin_unlock_irqrestore(&atc->vm_lock, flags); apcm->vm_block = NULL; } @@ -161,9 +155,7 @@ static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index) struct ct_vm *vm; void *kvirt_addr; unsigned long phys_addr; - unsigned long flags; - spin_lock_irqsave(&atc->vm_lock, flags); vm = atc->vm; kvirt_addr = vm->get_ptp_virt(vm, index); if (kvirt_addr == NULL) @@ -171,8 +163,6 @@ static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index) else phys_addr = virt_to_phys(kvirt_addr); - spin_unlock_irqrestore(&atc->vm_lock, flags); - return phys_addr; } @@ -1562,7 +1552,6 @@ int ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc_set_ops(atc); spin_lock_init(&atc->atc_lock); - spin_lock_init(&atc->vm_lock); /* Find card model */ err = atc_identify_card(atc); diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 286c993..a7b0ec2 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -101,7 +101,6 @@ struct ct_atc { unsigned long (*get_ptp_phys)(struct ct_atc *atc, int index); spinlock_t atc_lock; - spinlock_t vm_lock; int (*pcm_playback_prepare)(struct ct_atc *atc, struct ct_atc_pcm *apcm); diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index cecf77e..363b67e 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -35,25 +35,27 @@ get_vm_block(struct ct_vm *vm, unsigned int size) struct ct_vm_block *block = NULL, *entry = NULL; struct list_head *pos = NULL; + mutex_lock(&vm->lock); list_for_each(pos, &vm->unused) { entry = list_entry(pos, struct ct_vm_block, list); if (entry->size >= size) break; /* found a block that is big enough */ } if (pos == &vm->unused) - return NULL; + goto out; if (entry->size == size) { /* Move the vm node from unused list to used list directly */ list_del(&entry->list); list_add(&entry->list, &vm->used); vm->size -= size; - return entry; + block = entry; + goto out; } block = kzalloc(sizeof(*block), GFP_KERNEL); if (NULL == block) - return NULL; + goto out; block->addr = entry->addr; block->size = size; @@ -62,6 +64,8 @@ get_vm_block(struct ct_vm *vm, unsigned int size) entry->size -= size; vm->size -= size; + out: + mutex_unlock(&vm->lock); return block; } @@ -70,6 +74,7 @@ static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block) struct ct_vm_block *entry = NULL, *pre_ent = NULL; struct list_head *pos = NULL, *pre = NULL; + mutex_lock(&vm->lock); list_del(&block->list); vm->size += block->size; @@ -106,6 +111,7 @@ static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block) pos = pre; pre = pos->prev; } + mutex_unlock(&vm->lock); } /* Map host addr (kmalloced/vmalloced) to device logical addr. */ @@ -191,6 +197,8 @@ int ct_vm_create(struct ct_vm **rvm) if (NULL == vm) return -ENOMEM; + mutex_init(&vm->lock); + /* Allocate page table pages */ for (i = 0; i < CT_PTP_NUM; i++) { vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL); diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index 4eb5bdd..618952e 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -20,7 +20,7 @@ #define CT_PTP_NUM 1 /* num of device page table pages */ -#include +#include #include struct ct_vm_block { @@ -35,6 +35,7 @@ struct ct_vm { unsigned int size; /* Available addr space in bytes */ struct list_head unused; /* List of unused blocks */ struct list_head used; /* List of used blocks */ + struct mutex lock; /* Map host addr (kmalloced/vmalloced) to device logical addr. */ struct ct_vm_block *(*map)(struct ct_vm *, void *host_addr, int size); -- cgit v1.1 From 67fbf880631bb4493ad8d23f25562abdf09dc01d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 09:18:26 +0200 Subject: ALSA: ctxfi - Fix a typo in MODULE_LICENSE A space has to be put between GPL and v2. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/xfi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index e31e29e..bf232e7 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -18,7 +18,7 @@ MODULE_AUTHOR("Creative Technology Ltd"); MODULE_DESCRIPTION("X-Fi driver version 1.03"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); static unsigned int reference_rate = 48000; -- cgit v1.1 From 9318dce5038f193f46091b80c61928395a4139b7 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 1 Jun 2009 21:36:23 +0200 Subject: ALSA: snd_usb_caiaq: clean whitespaces Cosmetic changes only, no code change. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 88 ++++++++++++++++++++++++------------------------ sound/usb/caiaq/device.c | 60 ++++++++++++++++----------------- sound/usb/caiaq/midi.c | 24 ++++++------- 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- cgit v1.1 From d3873a1be9f2d497e9ff013e4a83a2a4d9f1d22d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 1 Jun 2009 21:36:25 +0200 Subject: ALSA: snd_usb_caiaq: use strlcpy Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 9be0f2e..d06d7fc 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -432,8 +432,8 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) if (c) *c = '\0'; - strcpy(card->driver, MODNAME); - strcpy(card->shortname, dev->product_name); + strlcpy(card->driver, MODNAME, sizeof(card->driver)); + strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); len = snprintf(card->longname, sizeof(card->longname), "%s %s (serial %s, ", -- cgit v1.1 From 1a1df6f0434fc35c9bf6ca25f9c5115713d77291 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 1 Jun 2009 21:36:26 +0200 Subject: ALSA: snd_usb_caiaq: give better longname The serial number is of no interest in the longname, remove it. This gives space for the usb path information which is more informative. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 25 ++++++------------------- sound/usb/caiaq/device.h | 1 - 2 files changed, 6 insertions(+), 20 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index d06d7fc..15052e4 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -382,10 +382,10 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) static int __devinit init_card(struct snd_usb_caiaqdev *dev) { - char *c; + char usbpath[32]; struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; - int err, len; + int err; if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); @@ -424,27 +424,14 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, - dev->serial, CAIAQ_USB_STR_LEN); - - /* terminate serial string at first white space occurence */ - c = strchr(dev->serial, ' '); - if (c) - *c = '\0'; - strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); - len = snprintf(card->longname, sizeof(card->longname), - "%s %s (serial %s, ", - dev->vendor_name, dev->product_name, dev->serial); - - if (len < sizeof(card->longname) - 2) - len += usb_make_path(usb_dev, card->longname + len, - sizeof(card->longname) - len); + usb_make_path(usb_dev, usbpath, sizeof(usbpath)); + snprintf(card->longname, sizeof(card->longname), + "%s %s (%s)", + dev->vendor_name, dev->product_name, usbpath); - card->longname[len++] = ')'; - card->longname[len] = '\0'; setup_card(dev); return 0; } diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 4cce1ad..ece7351 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -81,7 +81,6 @@ struct snd_usb_caiaqdev { char vendor_name[CAIAQ_USB_STR_LEN]; char product_name[CAIAQ_USB_STR_LEN]; - char serial[CAIAQ_USB_STR_LEN]; int n_streams, n_audio_in, n_audio_out; int streaming, first_packet, output_running; -- cgit v1.1 From 3b315d70b094e8b439358756a9084438fd7a71c2 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Tue, 2 Jun 2009 10:54:19 +0200 Subject: ALSA: hda - Acer Aspire 8930G support Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 99 ++++++++++++++++++++++++++-- 2 files changed, 95 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 818d06f..29c6125 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -139,6 +139,7 @@ ALC883/888 acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) acer-aspire Acer Aspire 9810 acer-aspire-4930g Acer Aspire 4930G + acer-aspire-8930g Acer Aspire 8930G medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ad6cc42..2bbb9e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -223,6 +223,7 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -313,6 +314,8 @@ struct alc_spec { const struct hda_channel_mode *channel_mode; int num_channel_mode; int need_dac_fix; + int const_channel_count; + int ext_channel_count; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -368,6 +371,7 @@ struct alc_config_preset { unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; int need_dac_fix; + int const_channel_count; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -462,7 +466,7 @@ static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - spec->multiout.max_channels); + spec->ext_channel_count); } static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, @@ -472,9 +476,12 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; + &spec->ext_channel_count); + if (err >= 0 && !spec->const_channel_count) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } return err; } @@ -854,8 +861,13 @@ static void setup_preset(struct alc_spec *spec, spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; - spec->multiout.max_channels = spec->channel_mode[0].channels; + if (preset->const_channel_count) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; @@ -1456,6 +1468,48 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { { } }; +/* + * ALC888 Acer Aspire 8930G model + */ + +static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal Front to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Internal Rear to Rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect Internal CLFE to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect HP out to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable all DACs */ +/* DAC DISABLE/MUTE 1? */ +/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, +/* DAC DISABLE/MUTE 2? */ +/* some bit here disables the other DACs. Init=0x4900 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, +/* Enable amplifiers */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } +}; + static struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { @@ -1508,6 +1562,17 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x1b; + alc_automute_amp(codec); +} + /* * ALC880 3-stack model * @@ -8758,6 +8823,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8790,6 +8856,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", @@ -9009,6 +9077,27 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_8930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_8930g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_8930g_init_hook, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer }, -- cgit v1.1 From 601e1cc5df940b59e71c947726640811897d30df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 11:37:01 +0200 Subject: ALSA: ca0106 - Add missing registrations of vmaster controls Although the vmaster controls are created, they aren't registered thus they don't appear in the real world. Added the missing snd_ctl_add() calls. Signed-off-by: Takashi Iwai Cc: --- sound/pci/ca0106/ca0106_mixer.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index c111efe..f143f71 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -841,6 +841,9 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) snd_ca0106_master_db_scale); if (!vmaster) return -ENOMEM; + err = snd_ctl_add(card, vmaster); + if (err < 0) + return err; add_slaves(card, vmaster, slave_vols); if (emu->details->spi_dac == 1) { @@ -848,6 +851,9 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) NULL); if (!vmaster) return -ENOMEM; + err = snd_ctl_add(card, vmaster); + if (err < 0) + return err; add_slaves(card, vmaster, slave_sws); } return 0; -- cgit v1.1 From 10a8ebbb08c4b08292598947bbe534e04d6ee705 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Jun 2009 12:02:38 +0200 Subject: ALSA: Core - add snd_card_set_id() function Introduce snd_card_set_id() function to allow lowlevel drivers to set default identification name for card slot. The function checks also for identification name collisions and tries to create unique name. Also, the snd_card_create() function is simplified, because this new function is used. As bonus, proper name collision checks are evaluated at the card create time. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/core.h | 1 + sound/core/init.c | 62 +++++++++++++++++++++++++++++++++------------------- 2 files changed, 40 insertions(+), 23 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..0e8ae10 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -313,6 +313,7 @@ struct snd_card *snd_card_new(int idx, const char *id, int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); +void snd_card_set_id(struct snd_card *card, const char *id); int snd_card_register(struct snd_card *card); int snd_card_info_init(void); int snd_card_info_done(void); diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..6557dd8 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -152,15 +152,8 @@ int snd_card_create(int idx, const char *xid, card = kzalloc(sizeof(*card) + extra_size, GFP_KERNEL); if (!card) return -ENOMEM; - if (xid) { - if (!snd_info_check_reserved_words(xid)) { - snd_printk(KERN_ERR - "given id string '%s' is reserved.\n", xid); - err = -EBUSY; - goto __error; - } - strlcpy(card->id, xid, sizeof(card->id)); - } + if (xid) + snd_card_set_id(card, xid); err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) { @@ -483,22 +476,39 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -static void choose_default_id(struct snd_card *card) +/** + * snd_card_set_id - set card identification name + * @card: soundcard structure + * @nid: new identification string + * + * This function sets the card identification and checks for name + * collisions. + */ +void snd_card_set_id(struct snd_card *card, const char *nid) { int i, len, idx_flag = 0, loops = SNDRV_CARDS; - char *id, *spos; + const char *spos, *src; + char *id; - id = spos = card->shortname; - while (*id != '\0') { - if (*id == ' ') - spos = id + 1; - id++; + /* check if user specified own card->id */ + if (card->id[0] != '\0') + return; + if (nid == NULL) { + id = card->shortname; + spos = src = id; + while (*id != '\0') { + if (*id == ' ') + spos = id + 1; + id++; + } + } else { + spos = src = nid; } id = card->id; while (*spos != '\0' && !isalnum(*spos)) spos++; if (isdigit(*spos)) - *id++ = isalpha(card->shortname[0]) ? card->shortname[0] : 'D'; + *id++ = isalpha(src[0]) ? src[0] : 'D'; while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) { if (isalnum(*spos)) *id++ = *spos; @@ -513,16 +523,20 @@ static void choose_default_id(struct snd_card *card) while (1) { if (loops-- == 0) { - snd_printk(KERN_ERR "unable to choose default card id (%s)\n", id); + snd_printk(KERN_ERR "unable to set card id (%s)\n", id); strcpy(card->id, card->proc_root->name); return; } if (!snd_info_check_reserved_words(id)) goto __change; + mutex_lock(&snd_card_mutex); for (i = 0; i < snd_ecards_limit; i++) { - if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) + if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) { + mutex_unlock(&snd_card_mutex); goto __change; + } } + mutex_unlock(&snd_card_mutex); break; __change: @@ -539,14 +553,16 @@ static void choose_default_id(struct snd_card *card) spos = id + len - 2; if ((size_t)len <= sizeof(card->id) - 2) spos++; - *spos++ = '_'; - *spos++ = '1'; - *spos++ = '\0'; + *(char *)spos++ = '_'; + *(char *)spos++ = '1'; + *(char *)spos++ = '\0'; idx_flag++; } } } +EXPORT_SYMBOL(snd_card_set_id); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, @@ -641,7 +657,7 @@ int snd_card_register(struct snd_card *card) return 0; } if (card->id[0] == '\0') - choose_default_id(card); + snd_card_set_id(card, NULL); snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -- cgit v1.1 From bafeee5b1f8d32cbf791c322b40a6fa91d8ccf7a Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 2 Jun 2009 12:36:39 +0200 Subject: ALSA: snd_usb_caiaq: give better shortname If not passed as module option, provide an own card ID with the newly introduced snd_set_card_id() call. This will prevent ALSA from calling choose_default_name() which only takes the last part of a name containing whitespaces. This for example caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very descriptive. The implementation now takes the short name and removes all whitespaces from it which is much nicer. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 21 +++++++++++++++++++-- 1 file changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 15052e4..43bd0db 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -382,10 +382,10 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) static int __devinit init_card(struct snd_usb_caiaqdev *dev) { - char usbpath[32]; + char *c, usbpath[32]; struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; - int err; + int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); @@ -427,6 +427,23 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + /* if the id was not passed as module option, fill it with a shortened + * version of the product string which does not contain any + * whitespaces */ + + if (*card->id == '\0') { + char id[sizeof(card->id)]; + + memset(id, 0, sizeof(id)); + + for (c = card->shortname, len = 0; + *c && len < sizeof(card->id); c++) + if (*c != ' ') + id[len++] = *c; + + snd_card_set_id(card, id); + } + usb_make_path(usb_dev, usbpath, sizeof(usbpath)); snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", -- cgit v1.1 From c6e24d4db824d277303201811e88626778c59999 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 2 Jun 2009 12:36:40 +0200 Subject: ALSA: snd_usb_caiaq: bump version number Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 43bd0db..b9a2b31 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.15"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- cgit v1.1 From 9a83b7453c2c4db145666b653abe9d9f410d18a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 14:20:48 +0200 Subject: ALSA: Remove invalid GENERIC_MIX PCM sublass SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playback devices, but ca0106 and emu10k1x don't support it (unlike emu10k1). We shouldn't set that flag to avoid confusion. Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 1 - sound/pci/emu10k1/emu10k1x.c | 1 - 2 files changed, 2 deletions(-) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index bfac30f..57b992a 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1319,7 +1319,6 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device) } pcm->info_flags = 0; - pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; strcpy(pcm->name, "CA0106"); for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 1970f0e..4d3ad79 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -858,7 +858,6 @@ static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct s } pcm->info_flags = 0; - pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; switch(device) { case 0: strcpy(pcm->name, "EMU10K1X Front"); -- cgit v1.1 From 822fa19b5c23746577687a0ec48eae0ec1cd22a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 14:12:17 +0200 Subject: ALSA: ALSA: ctxfi - Release PCM resources at each prepare call The prepare callback can be called multiple times, thus it needs to release and acquire the resource again by itself at the second or later call. Simply add pcm_release_resources() at the beginning of each prepare callback in ctatc.c. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 1a4bb35..e14ed71 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -254,6 +254,9 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) return 0; } + /* first release old resources */ + atc->pcm_release_resources(atc, apcm); + /* Get SRC resource */ desc.multi = apcm->substream->runtime->channels; desc.msr = atc->msr; @@ -496,6 +499,9 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) int n_srcimp = 0, n_amixer = 0, n_srcc = 0, n_sum = 0; struct src_node_conf_t src_node_conf[2] = {{0} }; + /* first release old resources */ + atc->pcm_release_resources(atc, apcm); + /* The numbers of converting SRCs and SRCIMPs should be determined * by pitch value. */ @@ -767,6 +773,9 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc, int n_amixer = apcm->substream->runtime->channels, i = 0; unsigned int pitch = 0, rsr = atc->pll_rate; + /* first release old resources */ + atc->pcm_release_resources(atc, apcm); + /* Get SRC resource */ desc.multi = apcm->substream->runtime->channels; desc.msr = 1; -- cgit v1.1 From 6585db943aade186d38eaab2720c18887b94c875 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 14:17:27 +0200 Subject: ALSA: ctxfi - Fix surround mixer names We usually pick up "Surround" mixer for the rear output, and "Side" for the extra surround. Fix the channel mapping to follow it. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctmixer.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index b795076..177c46e 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -168,7 +168,7 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = { }, [MIXER_WAVES_P] = { .ctl = 1, - .name = "Surround Playback Volume", + .name = "Side Playback Volume", }, [MIXER_WAVEC_P] = { .ctl = 1, @@ -176,7 +176,7 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = { }, [MIXER_WAVER_P] = { .ctl = 1, - .name = "Rear Playback Volume", + .name = "Surround Playback Volume", }, [MIXER_PCM_C_S] = { @@ -213,7 +213,7 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = { }, [MIXER_WAVES_P_S] = { .ctl = 1, - .name = "Surround Playback Switch", + .name = "Side Playback Switch", }, [MIXER_WAVEC_P_S] = { .ctl = 1, @@ -221,7 +221,7 @@ ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = { }, [MIXER_WAVER_P_S] = { .ctl = 1, - .name = "Rear Playback Switch", + .name = "Surround Playback Switch", }, [MIXER_DIGITAL_IO_S] = { .ctl = 0, -- cgit v1.1 From 8372d4980fbc2e403f0dc5457441c8c6b7c04915 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 14:27:56 +0200 Subject: ALSA: ctxfi - Fix PCM device naming PCM names for surround streams should be also fixed as well as the mixer element names. Also, a bit clean up for PCM name setup. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 8 ++++---- sound/pci/ctxfi/ctatc.h | 4 ++-- sound/pci/ctxfi/ctpcm.c | 6 ++---- 3 files changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index e14ed71..675dd4c 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -72,15 +72,15 @@ static struct { [FRONT] = { .create = ct_alsa_pcm_create, .destroy = NULL, .public_name = "Front/WaveIn"}, - [REAR] = { .create = ct_alsa_pcm_create, + [SURROUND] = { .create = ct_alsa_pcm_create, .destroy = NULL, - .public_name = "Rear"}, + .public_name = "Surround"}, [CLFE] = { .create = ct_alsa_pcm_create, .destroy = NULL, .public_name = "Center/LFE"}, - [SURROUND] = { .create = ct_alsa_pcm_create, + [SIDE] = { .create = ct_alsa_pcm_create, .destroy = NULL, - .public_name = "Surround"}, + .public_name = "Side"}, [IEC958] = { .create = ct_alsa_pcm_create, .destroy = NULL, .public_name = "IEC958 Non-audio"}, diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index a7b0ec2..b86d12c 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -29,9 +29,9 @@ enum CTALSADEVS { /* Types of alsa devices */ FRONT, - REAR, - CLFE, SURROUND, + CLFE, + SIDE, IEC958, MIXER, NUM_CTALSADEVS /* This should always be the last */ diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index a64cb0e..756d8b2 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -469,12 +469,10 @@ int ct_alsa_pcm_create(struct ct_atc *atc, struct snd_pcm *pcm; int err; int playback_count, capture_count; - char name[128]; - strncpy(name, device_name, sizeof(name)); playback_count = (IEC958 == device) ? 1 : 8; capture_count = (FRONT == device) ? 1 : 0; - err = snd_pcm_new(atc->card, name, device, + err = snd_pcm_new(atc->card, "ctxfi", device, playback_count, capture_count, &pcm); if (err < 0) { printk(KERN_ERR "ctxfi: snd_pcm_new failed!! Err=%d\n", err); @@ -484,7 +482,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, pcm->private_data = atc; pcm->info_flags = 0; pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; - strcpy(pcm->name, device_name); + strlcpy(pcm->name, device_name, sizeof(pcm->name)); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ct_pcm_playback_ops); -- cgit v1.1 From d2b9b96c516d4d61663d92ab4ad4f15ca0134ef2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 14:39:05 +0200 Subject: ALSA: ctxfi - Fix supported PCM formats The device seems supporting only U8, S16, S24_3LE, S32. Other linear formats result in bad outputs. Also, added the support for 32bit float format, which wasn't listed in the original code. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 5 ++--- sound/pci/ctxfi/ctpcm.c | 15 +++++---------- 2 files changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 675dd4c..268ecc4 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -170,16 +170,15 @@ static unsigned int convert_format(snd_pcm_format_t snd_format) { switch (snd_format) { case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: return SRC_SF_U8; case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_U16_LE: return SRC_SF_S16; case SNDRV_PCM_FORMAT_S24_3LE: return SRC_SF_S24; - case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: return SRC_SF_S32; + case SNDRV_PCM_FORMAT_FLOAT_LE: + return SRC_SF_F32; default: printk(KERN_ERR "ctxfi: not recognized snd format is %d \n", snd_format); diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 756d8b2..26d86dc 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -26,12 +26,10 @@ static struct snd_pcm_hardware ct_pcm_playback_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S24_3LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_FLOAT_LE), .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000), .rate_min = 8000, @@ -52,8 +50,7 @@ static struct snd_pcm_hardware ct_spdif_passthru_playback_hw = { SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE), - .formats = (SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE), + .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_32000), @@ -77,12 +74,10 @@ static struct snd_pcm_hardware ct_pcm_capture_hw = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP_VALID), .formats = (SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S24_3LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_FLOAT_LE), .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_96000), .rate_min = 8000, -- cgit v1.1 From cd391e206f486955e216a61bd9ebcb0e142122e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 15:04:29 +0200 Subject: ALSA: ctxfi - Remove PAGE_SIZE limitation Remove the limitation of PAGE_SIZE to be 4k by defining the own page size and macros for 4k. 8kb page size could be natively supported, but it's disabled right now for simplicity. Also, clean up using upper_32_bits() macro. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 - sound/pci/ctxfi/cthw20k1.c | 16 ++++++---------- sound/pci/ctxfi/cthw20k2.c | 17 ++++------------- sound/pci/ctxfi/ctvmem.c | 25 ++++++++++++------------- sound/pci/ctxfi/ctvmem.h | 8 ++++++++ 5 files changed, 30 insertions(+), 37 deletions(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 2d7fef5..3a7640f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -277,7 +277,6 @@ config SND_CS5535AUDIO config SND_CTXFI tristate "Creative Sound Blaster X-Fi" - depends on X86 select SND_PCM help If you want to use soundcards based on Creative Sound Blastr X-Fi diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 44283bd..b7b8e6f 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1249,19 +1249,15 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) } trnctl = 0x13; /* 32-bit, 4k-size page */ -#if BITS_PER_LONG == 64 - ptp_phys_low = info->vm_pgt_phys & ((1UL<<32)-1); - ptp_phys_high = (info->vm_pgt_phys>>32) & ((1UL<<32)-1); - trnctl |= (1<<2); -#elif BITS_PER_LONG == 32 - ptp_phys_low = info->vm_pgt_phys & (~0UL); - ptp_phys_high = 0; -#else -# error "Unknown BITS_PER_LONG!" -#endif + ptp_phys_low = (u32)info->vm_pgt_phys; + ptp_phys_high = upper_32_bits(info->vm_pgt_phys); + if (sizeof(void *) == 8) /* 64bit address */ + trnctl |= (1 << 2); +#if 0 /* Only 4k h/w pages for simplicitiy */ #if PAGE_SIZE == 8192 trnctl |= (1<<5); #endif +#endif hw_write_20kx(hw, PTPALX, ptp_phys_low); hw_write_20kx(hw, PTPAHX, ptp_phys_high); hw_write_20kx(hw, TRNCTL, trnctl); diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 7c24c2c..3497287 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1203,19 +1203,10 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) } vmctl = 0x80000C0F; /* 32-bit, 4k-size page */ -#if BITS_PER_LONG == 64 - ptp_phys_low = info->vm_pgt_phys & ((1UL<<32)-1); - ptp_phys_high = (info->vm_pgt_phys>>32) & ((1UL<<32)-1); - vmctl |= (3<<8); -#elif BITS_PER_LONG == 32 - ptp_phys_low = info->vm_pgt_phys & (~0UL); - ptp_phys_high = 0; -#else -# error "Unknown BITS_PER_LONG!" -#endif -#if PAGE_SIZE == 8192 -# error "Don't support 8k-page!" -#endif + ptp_phys_low = (u32)info->vm_pgt_phys; + ptp_phys_high = upper_32_bits(info->vm_pgt_phys); + if (sizeof(void *) == 8) /* 64bit address */ + vmctl |= (3 << 8); /* Write page table physical address to all PTPAL registers */ for (i = 0; i < 64; i++) { hw_write_20kx(hw, VMEM_PTPAL+(16*i), ptp_phys_low); diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 363b67e..74a0362 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -18,12 +18,11 @@ #include "ctvmem.h" #include #include -#include /* for PAGE_SIZE macro definition */ #include #include -#define CT_PTES_PER_PAGE (PAGE_SIZE / sizeof(void *)) -#define CT_ADDRS_PER_PAGE (CT_PTES_PER_PAGE * PAGE_SIZE) +#define CT_PTES_PER_PAGE (CT_PAGE_SIZE / sizeof(void *)) +#define CT_ADDRS_PER_PAGE (CT_PTES_PER_PAGE * CT_PAGE_SIZE) /* * * Find or create vm block based on requested @size. @@ -138,24 +137,24 @@ ct_vm_map(struct ct_vm *vm, void *host_addr, int size) return NULL; } - start_phys = (virt_to_phys(host_addr) & PAGE_MASK); - pages = (PAGE_ALIGN(virt_to_phys(host_addr) + size) - - start_phys) >> PAGE_SHIFT; + start_phys = (virt_to_phys(host_addr) & CT_PAGE_MASK); + pages = (CT_PAGE_ALIGN(virt_to_phys(host_addr) + size) + - start_phys) >> CT_PAGE_SHIFT; ptp = vm->ptp[0]; - block = get_vm_block(vm, (pages << PAGE_SHIFT)); + block = get_vm_block(vm, (pages << CT_PAGE_SHIFT)); if (block == NULL) { printk(KERN_ERR "ctxfi: No virtual memory block that is big " "enough to allocate!\n"); return NULL; } - pte_start = (block->addr >> PAGE_SHIFT); + pte_start = (block->addr >> CT_PAGE_SHIFT); for (i = 0; i < pages; i++) - ptp[pte_start+i] = start_phys + (i << PAGE_SHIFT); + ptp[pte_start+i] = start_phys + (i << CT_PAGE_SHIFT); - block->addr += (virt_to_phys(host_addr) & (~PAGE_MASK)); + block->addr += (virt_to_phys(host_addr) & (~CT_PAGE_MASK)); block->size = size; return block; @@ -164,9 +163,9 @@ ct_vm_map(struct ct_vm *vm, void *host_addr, int size) static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block) { /* do unmapping */ - block->size = ((block->addr + block->size + PAGE_SIZE - 1) - & PAGE_MASK) - (block->addr & PAGE_MASK); - block->addr &= PAGE_MASK; + block->size = ((block->addr + block->size + CT_PAGE_SIZE - 1) + & CT_PAGE_MASK) - (block->addr & CT_PAGE_MASK); + block->addr &= CT_PAGE_MASK; put_vm_block(vm, block); } diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index 618952e..17d2d37 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -23,6 +23,14 @@ #include #include +/* The chip can handle the page table of 4k pages + * (emu20k1 can handle even 8k pages, but we don't use it right now) + */ +#define CT_PAGE_SIZE 4096 +#define CT_PAGE_SHIFT 12 +#define CT_PAGE_MASK (~(PAGE_SIZE - 1)) +#define CT_PAGE_ALIGN(addr) ALIGN(addr, CT_PAGE_SIZE) + struct ct_vm_block { unsigned int addr; /* starting logical addr of this block */ unsigned int size; /* size of this device virtual mem block */ -- cgit v1.1 From c76157d9286ed598c241c212aa5a3c6e5107bd82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 15:26:19 +0200 Subject: ALSA: ctxfi - Support SG-buffers Use SG-buffers instead of contiguous pages. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/ctxfi/ctpcm.c | 4 +++- sound/pci/ctxfi/ctvmem.c | 56 +++++++++++++++++++----------------------------- sound/pci/ctxfi/ctvmem.h | 5 ++++- 4 files changed, 30 insertions(+), 37 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 268ecc4..6849475 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -128,7 +128,7 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) runtime = apcm->substream->runtime; vm = atc->vm; - apcm->vm_block = vm->map(vm, runtime->dma_area, runtime->dma_bytes); + apcm->vm_block = vm->map(vm, apcm->substream, runtime->dma_bytes); if (NULL == apcm->vm_block) return -ENOENT; diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 26d86dc..52ddf19 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -442,6 +442,7 @@ static struct snd_pcm_ops ct_pcm_playback_ops = { .prepare = ct_pcm_playback_prepare, .trigger = ct_pcm_playback_trigger, .pointer = ct_pcm_playback_pointer, + .page = snd_pcm_sgbuf_ops_page, }; /* PCM operators for capture */ @@ -454,6 +455,7 @@ static struct snd_pcm_ops ct_pcm_capture_ops = { .prepare = ct_pcm_capture_prepare, .trigger = ct_pcm_capture_trigger, .pointer = ct_pcm_capture_pointer, + .page = snd_pcm_sgbuf_ops_page, }; /* Create ALSA pcm device */ @@ -485,7 +487,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &ct_pcm_capture_ops); - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(atc->pci), 128*1024, 128*1024); return 0; diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 74a0362..b7f8e58 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -19,7 +19,7 @@ #include #include #include -#include +#include #define CT_PTES_PER_PAGE (CT_PAGE_SIZE / sizeof(void *)) #define CT_ADDRS_PER_PAGE (CT_PTES_PER_PAGE * CT_PAGE_SIZE) @@ -34,6 +34,13 @@ get_vm_block(struct ct_vm *vm, unsigned int size) struct ct_vm_block *block = NULL, *entry = NULL; struct list_head *pos = NULL; + size = CT_PAGE_ALIGN(size); + if (size > vm->size) { + printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " + "memory space available!\n"); + return NULL; + } + mutex_lock(&vm->lock); list_for_each(pos, &vm->unused) { entry = list_entry(pos, struct ct_vm_block, list); @@ -73,6 +80,8 @@ static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block) struct ct_vm_block *entry = NULL, *pre_ent = NULL; struct list_head *pos = NULL, *pre = NULL; + block->size = CT_PAGE_ALIGN(block->size); + mutex_lock(&vm->lock); list_del(&block->list); vm->size += block->size; @@ -115,57 +124,36 @@ static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block) /* Map host addr (kmalloced/vmalloced) to device logical addr. */ static struct ct_vm_block * -ct_vm_map(struct ct_vm *vm, void *host_addr, int size) +ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size) { - struct ct_vm_block *block = NULL; - unsigned long pte_start; - unsigned long i; - unsigned long pages; - unsigned long start_phys; + struct ct_vm_block *block; + unsigned int pte_start; + unsigned i, pages; unsigned long *ptp; - /* do mapping */ - if ((unsigned long)host_addr >= VMALLOC_START) { - printk(KERN_ERR "ctxfi: " - "Fail! Not support vmalloced addr now!\n"); - return NULL; - } - - if (size > vm->size) { - printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " - "memory space available!\n"); - return NULL; - } - - start_phys = (virt_to_phys(host_addr) & CT_PAGE_MASK); - pages = (CT_PAGE_ALIGN(virt_to_phys(host_addr) + size) - - start_phys) >> CT_PAGE_SHIFT; - - ptp = vm->ptp[0]; - - block = get_vm_block(vm, (pages << CT_PAGE_SHIFT)); + block = get_vm_block(vm, size); if (block == NULL) { printk(KERN_ERR "ctxfi: No virtual memory block that is big " "enough to allocate!\n"); return NULL; } + ptp = vm->ptp[0]; pte_start = (block->addr >> CT_PAGE_SHIFT); - for (i = 0; i < pages; i++) - ptp[pte_start+i] = start_phys + (i << CT_PAGE_SHIFT); + pages = block->size >> CT_PAGE_SHIFT; + for (i = 0; i < pages; i++) { + unsigned long addr; + addr = snd_pcm_sgbuf_get_addr(substream, i << CT_PAGE_SHIFT); + ptp[pte_start + i] = addr; + } - block->addr += (virt_to_phys(host_addr) & (~CT_PAGE_MASK)); block->size = size; - return block; } static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block) { /* do unmapping */ - block->size = ((block->addr + block->size + CT_PAGE_SIZE - 1) - & CT_PAGE_MASK) - (block->addr & CT_PAGE_MASK); - block->addr &= CT_PAGE_MASK; put_vm_block(vm, block); } diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index 17d2d37..01e4fd0 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -37,6 +37,8 @@ struct ct_vm_block { struct list_head list; }; +struct snd_pcm_substream; + /* Virtual memory management object for card device */ struct ct_vm { void *ptp[CT_PTP_NUM]; /* Device page table pages */ @@ -46,7 +48,8 @@ struct ct_vm { struct mutex lock; /* Map host addr (kmalloced/vmalloced) to device logical addr. */ - struct ct_vm_block *(*map)(struct ct_vm *, void *host_addr, int size); + struct ct_vm_block *(*map)(struct ct_vm *, struct snd_pcm_substream *, + int size); /* Unmap device logical addr area. */ void (*unmap)(struct ct_vm *, struct ct_vm_block *block); void *(*get_ptp_virt)(struct ct_vm *vm, int index); -- cgit v1.1 From eeaf100d25cefdbe9dc40d6e0f2025411474972a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 16:06:02 +0200 Subject: ALSA: ca0106 - Add missing card->mixername field setup Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_mixer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index f143f71..7724252 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -856,6 +856,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) return err; add_slaves(card, vmaster, slave_sws); } + + strcpy(card->mixername, "CA0106"); return 0; } -- cgit v1.1 From 3f08a0e4ab1ce65bb882f6425ff482e5db025f78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Jun 2009 17:39:52 +0200 Subject: ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'99 Signed-off-by: Takashi Iwai --- sound/pci/bt87x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index ce3f2e9..24585c6 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -810,6 +810,8 @@ static struct pci_device_id snd_bt87x_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC), /* Voodoo TV 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC), + /* Askey Computer Corp. MagicTView'99 */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x144f, 0x3000, GENERIC), /* AVerMedia Studio No. 103, 203, ...? */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, AVPHONE98), /* Prolink PixelView PV-M4900 */ -- cgit v1.1 From ca85b6ba59b69b7b5adcc64a98bd2478f73b2542 Mon Sep 17 00:00:00 2001 From: Andrea Borgia Date: Tue, 2 Jun 2009 19:21:17 +0200 Subject: ALSA: usb-audio - errata corrige for quirk Cut'n'paste mistake, whose likely result was nothing at all. Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC". Signed-off-by: Andrea Borgia Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 58f24f5..f0f7624 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1986,7 +1986,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + USB_DEVICE(0x0ccd, 0x0028), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "TerraTec", .product_name = "Aureon 5.1 MkII", -- cgit v1.1 From 5c9b6e9e618868ac66d92c81b70ad57d82033d4e Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Wed, 3 Jun 2009 15:35:19 +1000 Subject: ALSA: sound/ppc: update annotations of serveral functions [I am not sure if this is the correct approach as I don't know if any of this actual hardware or drivers are really hot pluggable.] Gets rid of these build warnings: WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_new(). If .snd_pmac_new is only used by .snd_pmac_probe then annotate .snd_pmac_new with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_burgundy_init(). If .snd_pmac_burgundy_init is only used by .snd_pmac_probe then annotate .snd_pmac_burgundy_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_daca_init(). If .snd_pmac_daca_init is only used by .snd_pmac_probe then annotate .snd_pmac_daca_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_tumbler_init(). If .snd_pmac_tumbler_init is only used by .snd_pmac_probe then annotate .snd_pmac_tumbler_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_tumbler_post_init(). If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe then annotate .snd_pmac_tumbler_post_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_awacs_init(). If .snd_pmac_awacs_init is only used by .snd_pmac_probe then annotate .snd_pmac_awacs_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_pcm_new(). If .snd_pmac_pcm_new is only used by .snd_pmac_probe then annotate .snd_pmac_pcm_new with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_attach_beep(). If .snd_pmac_attach_beep is only used by .snd_pmac_probe then annotate .snd_pmac_attach_beep with a matching annotation. Signed-off-by: Stephen Rothwell Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 2 +- sound/ppc/beep.c | 2 +- sound/ppc/burgundy.c | 2 +- sound/ppc/daca.c | 2 +- sound/ppc/keywest.c | 4 ++-- sound/ppc/pmac.c | 8 ++++---- sound/ppc/tumbler.c | 4 ++-- 7 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 80df9b1..0d2d62d 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -872,7 +872,7 @@ static void snd_pmac_awacs_update_automute(struct snd_pmac *chip, int do_notify) /* * initialize chip */ -int __init +int __devinit snd_pmac_awacs_init(struct snd_pmac *chip) { int pm7500 = IS_PM7500; diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c index 89f5c32..a9d3507 100644 --- a/sound/ppc/beep.c +++ b/sound/ppc/beep.c @@ -215,7 +215,7 @@ static struct snd_kcontrol_new snd_pmac_beep_mixer = { }; /* Initialize beep stuff */ -int __init snd_pmac_attach_beep(struct snd_pmac *chip) +int __devinit snd_pmac_attach_beep(struct snd_pmac *chip) { struct pmac_beep *beep; struct input_dev *input_dev; diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 45a7629..aba3814 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -618,7 +618,7 @@ static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_noti /* * initialize burgundy */ -int __init snd_pmac_burgundy_init(struct snd_pmac *chip) +int __devinit snd_pmac_burgundy_init(struct snd_pmac *chip) { int imac = machine_is_compatible("iMac"); int i, err; diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index f8d478c..24200b7 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -244,7 +244,7 @@ static void daca_cleanup(struct snd_pmac *chip) } /* exported */ -int __init snd_pmac_daca_init(struct snd_pmac *chip) +int __devinit snd_pmac_daca_init(struct snd_pmac *chip) { int i, err; struct pmac_daca *mix; diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index a5afb26..15518e6 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -109,7 +109,7 @@ void snd_pmac_keywest_cleanup(struct pmac_keywest *i2c) } } -int __init snd_pmac_tumbler_post_init(void) +int __devinit snd_pmac_tumbler_post_init(void) { int err; @@ -124,7 +124,7 @@ int __init snd_pmac_tumbler_post_init(void) } /* exported */ -int __init snd_pmac_keywest_init(struct pmac_keywest *i2c) +int __devinit snd_pmac_keywest_init(struct pmac_keywest *i2c) { int err; diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 9b4e9c3..dfea116 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -702,7 +702,7 @@ static struct snd_pcm_ops snd_pmac_capture_ops = { .pointer = snd_pmac_capture_pointer, }; -int __init snd_pmac_pcm_new(struct snd_pmac *chip) +int __devinit snd_pmac_pcm_new(struct snd_pmac *chip) { struct snd_pcm *pcm; int err; @@ -934,7 +934,7 @@ static void __init detect_byte_swap(struct snd_pmac *chip) /* * detect a sound chip */ -static int __init snd_pmac_detect(struct snd_pmac *chip) +static int __devinit snd_pmac_detect(struct snd_pmac *chip) { struct device_node *sound; struct device_node *dn; @@ -1158,7 +1158,7 @@ static struct snd_kcontrol_new auto_mute_controls[] __initdata = { }, }; -int __init snd_pmac_add_automute(struct snd_pmac *chip) +int __devinit snd_pmac_add_automute(struct snd_pmac *chip) { int err; chip->auto_mute = 1; @@ -1175,7 +1175,7 @@ int __init snd_pmac_add_automute(struct snd_pmac *chip) /* * create and detect a pmac chip record */ -int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) +int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) { struct snd_pmac *chip; struct device_node *np; diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 40222fc..adc8341 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1269,7 +1269,7 @@ static void tumbler_resume(struct snd_pmac *chip) #endif /* initialize tumbler */ -static int __init tumbler_init(struct snd_pmac *chip) +static int __devinit tumbler_init(struct snd_pmac *chip) { int irq; struct pmac_tumbler *mix = chip->mixer_data; @@ -1339,7 +1339,7 @@ static void tumbler_cleanup(struct snd_pmac *chip) } /* exported */ -int __init snd_pmac_tumbler_init(struct snd_pmac *chip) +int __devinit snd_pmac_tumbler_init(struct snd_pmac *chip) { int i, err; struct pmac_tumbler *mix; -- cgit v1.1 From 3e1e0a5dd539f83438078759c8642c5dd7c24cb6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 Jun 2009 08:13:15 +0200 Subject: ALSA: powermac - Replace the rest of __init* All __initdata should be __devinitdata as platform device is hotpluggable. Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 52 ++++++++++++++++++++++++++-------------------------- sound/ppc/burgundy.c | 20 ++++++++++---------- sound/ppc/pmac.c | 4 ++-- sound/ppc/tumbler.c | 12 ++++++------ 4 files changed, 44 insertions(+), 44 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 0d2d62d..2cc0eda 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -477,7 +477,7 @@ static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol, #define AMP_CH_SPK 0 #define AMP_CH_HD 1 -static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PC Speaker Playback Volume", .info = snd_pmac_awacs_info_volume_amp, @@ -514,7 +514,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __initdata = { }, }; -static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Playback Switch", .info = snd_pmac_boolean_stereo_info, @@ -523,7 +523,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __initdata = { .private_value = AMP_CH_HD, }; -static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PC Speaker Playback Switch", .info = snd_pmac_boolean_stereo_info, @@ -595,46 +595,46 @@ static int snd_pmac_screamer_mic_boost_put(struct snd_kcontrol *kcontrol, /* * lists of mixer elements */ -static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __devinitdata = { AWACS_SWITCH("Master Capture Switch", 1, SHIFT_LOOPTHRU, 0), AWACS_VOLUME("Master Capture Volume", 0, 4, 0), /* AWACS_SWITCH("Unknown Playback Switch", 6, SHIFT_PAROUT0, 0), */ }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __devinitdata = { AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __devinitdata = { AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __devinitdata = { AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers_g4agp[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers_g4agp[] __devinitdata = { AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), AWACS_VOLUME("Master Playback Volume", 5, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), }; -static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __devinitdata = { AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), }; -static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __devinitdata = { AWACS_VOLUME("Headphone Playback Volume", 2, 6, 1), }; -static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __devinitdata = { AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; @@ -642,34 +642,34 @@ static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { /* FIXME: is this correct order? * screamer (powerbook G3 pismo) seems to have different bits... */ -static struct snd_kcontrol_new snd_pmac_awacs_mixers2[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mixers2[] __devinitdata = { AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_LINE, 0), AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_MIC, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __devinitdata = { AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_LINE, 0), }; -static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __devinitdata = { AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; -static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_master_sw __devinitdata = AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1); -static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __devinitdata = AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1); -static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __devinitdata = AWACS_SWITCH("Headphone Playback Switch", 1, SHIFT_HDMUTE, 1); -static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __devinitdata = { AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Mic Boost Capture Volume", .info = snd_pmac_screamer_mic_boost_info, @@ -678,34 +678,34 @@ static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = { }, }; -static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __devinitdata = { AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __initdata = +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __devinitdata = { AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), AWACS_SWITCH("CD Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __initdata = +static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __devinitdata = { AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0), AWACS_SWITCH("Mic Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0), }; -static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = { +static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __devinitdata = { AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), }; -static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __devinitdata = AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); -static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __devinitdata = AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); -static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __initdata = +static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __devinitdata = AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index aba3814..bb80770 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -468,7 +468,7 @@ static int snd_pmac_burgundy_put_switch_b(struct snd_kcontrol *kcontrol, /* * Burgundy mixers */ -static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = { +static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __devinitdata = { BURGUNDY_VOLUME_W("Master Playback Volume", 0, MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8), BURGUNDY_VOLUME_W("CD Capture Volume", 0, @@ -496,7 +496,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = { */ BURGUNDY_SWITCH_B("PCM Capture Switch", 0, MASK_ADDR_BURGUNDY_HOSTIFEH, 0x01, 0, 0) }; -static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = { +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __devinitdata = { BURGUNDY_VOLUME_W("Line in Capture Volume", 0, MASK_ADDR_BURGUNDY_VOLLINE, 16), BURGUNDY_VOLUME_W("Mic Capture Volume", 0, @@ -522,7 +522,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = { BURGUNDY_SWITCH_B("Mic Boost Capture Switch", 0, MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) }; -static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = { +static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __devinitdata = { BURGUNDY_VOLUME_W("Line in Capture Volume", 0, MASK_ADDR_BURGUNDY_VOLMIC, 16), BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, @@ -538,33 +538,33 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = { /* BURGUNDY_SWITCH_B("Line in Boost Capture Switch", 0, * MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) */ }; -static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __devinitdata = BURGUNDY_SWITCH_B("Master Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_LEFT | BURGUNDY_LINEOUT_LEFT | BURGUNDY_HP_LEFT, BURGUNDY_OUTPUT_RIGHT | BURGUNDY_LINEOUT_RIGHT | BURGUNDY_HP_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __devinitdata = BURGUNDY_SWITCH_B("Master Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_INTERN | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __devinitdata = BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __devinitdata = BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_INTERN, 0, 0); -static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __devinitdata = BURGUNDY_SWITCH_B("Line out Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_LINEOUT_LEFT, BURGUNDY_LINEOUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __devinitdata = BURGUNDY_SWITCH_B("Line out Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); -static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __initdata = +static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __devinitdata = BURGUNDY_SWITCH_B("Headphone Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_HP_LEFT, BURGUNDY_HP_RIGHT, 1); diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index dfea116..7bc492e 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -908,7 +908,7 @@ static int snd_pmac_dev_free(struct snd_device *device) * check the machine support byteswap (little-endian) */ -static void __init detect_byte_swap(struct snd_pmac *chip) +static void __devinit detect_byte_swap(struct snd_pmac *chip) { struct device_node *mio; @@ -1143,7 +1143,7 @@ static int pmac_hp_detect_get(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new auto_mute_controls[] __initdata = { +static struct snd_kcontrol_new auto_mute_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Auto Mute Switch", .info = snd_pmac_boolean_mono_info, diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index adc8341..08e584d 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -838,7 +838,7 @@ static int snapper_put_capture_source(struct snd_kcontrol *kcontrol, /* */ -static struct snd_kcontrol_new tumbler_mixers[] __initdata = { +static struct snd_kcontrol_new tumbler_mixers[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .info = tumbler_info_master_volume, @@ -862,7 +862,7 @@ static struct snd_kcontrol_new tumbler_mixers[] __initdata = { }, }; -static struct snd_kcontrol_new snapper_mixers[] __initdata = { +static struct snd_kcontrol_new snapper_mixers[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .info = tumbler_info_master_volume, @@ -895,7 +895,7 @@ static struct snd_kcontrol_new snapper_mixers[] __initdata = { }, }; -static struct snd_kcontrol_new tumbler_hp_sw __initdata = { +static struct snd_kcontrol_new tumbler_hp_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Playback Switch", .info = snd_pmac_boolean_mono_info, @@ -903,7 +903,7 @@ static struct snd_kcontrol_new tumbler_hp_sw __initdata = { .put = tumbler_put_mute_switch, .private_value = TUMBLER_MUTE_HP, }; -static struct snd_kcontrol_new tumbler_speaker_sw __initdata = { +static struct snd_kcontrol_new tumbler_speaker_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "PC Speaker Playback Switch", .info = snd_pmac_boolean_mono_info, @@ -911,7 +911,7 @@ static struct snd_kcontrol_new tumbler_speaker_sw __initdata = { .put = tumbler_put_mute_switch, .private_value = TUMBLER_MUTE_AMP, }; -static struct snd_kcontrol_new tumbler_lineout_sw __initdata = { +static struct snd_kcontrol_new tumbler_lineout_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line Out Playback Switch", .info = snd_pmac_boolean_mono_info, @@ -919,7 +919,7 @@ static struct snd_kcontrol_new tumbler_lineout_sw __initdata = { .put = tumbler_put_mute_switch, .private_value = TUMBLER_MUTE_LINE, }; -static struct snd_kcontrol_new tumbler_drc_sw __initdata = { +static struct snd_kcontrol_new tumbler_drc_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "DRC Switch", .info = snd_pmac_boolean_mono_info, -- cgit v1.1 From 2552a710f4b991136c650bf2a6d1b81f27f6273e Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Tue, 2 Jun 2009 00:18:53 -0400 Subject: ASoC: SSM2602: remove unsupported sample rates Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 87f606c..d6af069 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -497,11 +497,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.1 From 80d5bd93143439aff77fd246f5d06570b7a4641e Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Tue, 2 Jun 2009 00:18:56 -0400 Subject: ASoC: Blackfin: set the transfer size according the ac97_frame size Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Bryan Wu Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-sport.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index b7953c8..469ce7f 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -190,7 +190,7 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport) desc = get_dma_next_desc_ptr(sport->dma_rx_chan); /* Copy the descriptor which will be damaged to backup */ temp_desc = *desc; - desc->x_count = 0xa; + desc->x_count = sport->dummy_count / 2; desc->y_count = 0; desc->next_desc_addr = sport->dummy_rx_desc; local_irq_restore(flags); @@ -309,7 +309,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) desc = get_dma_next_desc_ptr(sport->dma_tx_chan); /* Store the descriptor which will be damaged */ temp_desc = *desc; - desc->x_count = 0xa; + desc->x_count = sport->dummy_count / 2; desc->y_count = 0; desc->next_desc_addr = sport->dummy_tx_desc; local_irq_restore(flags); -- cgit v1.1 From cf485da15a3b507c7dab42337639e4f4025d3373 Mon Sep 17 00:00:00 2001 From: Sonic Zhang Date: Tue, 2 Jun 2009 00:18:57 -0400 Subject: ASoC: Blackfin: document how anomaly 05000250 is handled Signed-off-by: Sonic Zhang Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 8a935f2..b1ed423 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -31,6 +31,15 @@ #include "bf5xx-sport.h" #include "bf5xx-ac97.h" +/* Anomaly notes: + * 05000250 - AD1980 is running in TDM mode and RFS/TFS are generated by SPORT + * contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1, + * while the max AC97 data size is 13*16. The DIV is always larger + * than data size. AD73311 and ad2602 are not running in TDM mode. + * AD1836 and AD73322 depend on external RFS/TFS only. So, this + * anomaly does not affect blackfin sound drivers. +*/ + static int *cmd_count; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; -- cgit v1.1 From f692fce0cf8625b6cc8678e802fb0e2e657b1ca6 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Tue, 2 Jun 2009 00:18:54 -0400 Subject: ASoC: SSM2602: assign last substream to the master when shutting down Fixes crash when shutting down. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 25 ++++++++++++++++--------- 1 file changed, 16 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d6af069..1fc4c8e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -336,15 +336,17 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, master_runtime->sample_bits, master_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); + if (master_runtime->rate != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + if (master_runtime->sample_bits != 0) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); ssm2602->slave_substream = substream; } else @@ -372,6 +374,11 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); -- cgit v1.1 From 872c78202c58d26596e25743791ee81a7d24abad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Jun 2009 20:43:29 +0100 Subject: ALSA: Fix double locking of card list in snd_card_register() The introduction of snd_card_set_id() added a lock on the card list to the old choose_default_id() function when using it to implement the new API call. This lock is needed to allow us to walk the list and check to see if our new name is a duplicate. Unfortunately this causes a lockup when called from snd_card_register() (in cases where no ID is supplied for the card) since the card list is already locked there. Fix this fairly hideously by factoring out the implementation and using a flag to indicate if the lock should be held. A better fix would probably be to refactor snd_card_register() to move the _set_id() outside the locking region but I can't immediately see anything I can convince myself is safe. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/core/init.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6557dd8..a578d05 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -476,15 +476,8 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -/** - * snd_card_set_id - set card identification name - * @card: soundcard structure - * @nid: new identification string - * - * This function sets the card identification and checks for name - * collisions. - */ -void snd_card_set_id(struct snd_card *card, const char *nid) +static void snd_card_set_id_internal(struct snd_card *card, const char *nid, + int do_locking) { int i, len, idx_flag = 0, loops = SNDRV_CARDS; const char *spos, *src; @@ -529,14 +522,16 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } if (!snd_info_check_reserved_words(id)) goto __change; - mutex_lock(&snd_card_mutex); + if (do_locking) + mutex_lock(&snd_card_mutex); for (i = 0; i < snd_ecards_limit; i++) { if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) { mutex_unlock(&snd_card_mutex); goto __change; } } - mutex_unlock(&snd_card_mutex); + if (do_locking) + mutex_unlock(&snd_card_mutex); break; __change: @@ -561,6 +556,18 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } } +/** + * snd_card_set_id - set card identification name + * @card: soundcard structure + * @nid: new identification string + * + * This function sets the card identification and checks for name + * collisions. + */ +void snd_card_set_id(struct snd_card *card, const char *nid) +{ + snd_card_set_id_internal(card, nid, 1); +} EXPORT_SYMBOL(snd_card_set_id); #ifndef CONFIG_SYSFS_DEPRECATED @@ -657,7 +664,7 @@ int snd_card_register(struct snd_card *card) return 0; } if (card->id[0] == '\0') - snd_card_set_id(card, NULL); + snd_card_set_id_internal(card, NULL, 0); snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -- cgit v1.1 From 13be1bf1467e3186a4a9d1b1276cc4bd31e472ea Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Thu, 4 Jun 2009 01:53:21 +0200 Subject: ALSA: burgundy: timeout message is off by one. Timeout message is off by one. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/ppc/burgundy.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 45a7629..d2c0a4e 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -46,12 +46,12 @@ snd_pmac_burgundy_extend_wait(struct snd_pmac *chip) timeout = 50; while (!(in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #1\n"); timeout = 50; while ((in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #2\n"); } -- cgit v1.1 From 018df41861475595a51d327b83fb5830462f7a53 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Thu, 4 Jun 2009 00:13:40 +0200 Subject: ALSA: hda - More Aspire 8930G fixes Enable all three capture channels, including the missing nid 7 which is the only one capable of capturing DMIC input Enable Headphone amp for the HP jack. This causes a volume boost for headphones, but does not cause any noticeable effect for light loads like other amps, so there is no need to make it configurable. Add Input Mix capture mux setting to capture the output of the playback input mux (that is, what goes out the speakers except for PCM) Hack another coef register because the stereo DMIC for some reason produces a nonstandard sum/difference signal. I found a bit to make it just use the sum signal for both channels, which makes it behave like a standard mono microphone. The stereo is useless anyway (they're 1cm apart). Tested working: Three capture channels, mic in, line in, DMIC. Tested not working: CD. Not sure why, might be unconnected in the actual hardware or a CD drive issue. Also looked at SPDIF. It appears to work (emitter lights up inside the HP out jack) but I lack a proper miniTOSLINK cable to test it. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 54 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2bbb9e4..8699b7f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1469,10 +1469,10 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { }; /* - * ALC888 Acer Aspire 8930G model + * ALC889 Acer Aspire 8930G model */ -static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Unselect Front Mic by default in input mixer 3 */ @@ -1492,7 +1492,7 @@ static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect HP out to Front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Enable all DACs */ @@ -1507,6 +1507,17 @@ static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { /* Enable amplifiers */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, +/* DMIC fix + * This laptop has a stereo digital microphone. The mics are only 1cm apart + * which makes the stereo useless. However, either the mic or the ALC889 + * makes the signal become a difference/sum signal instead of standard + * stereo, which is annoying. So instead we flip this bit which makes the + * codec replicate the sum signal to both channels, turning it into a + * normal mono mic. + */ +/* DMIC_CONTROL? Init value = 0x0001 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0003}, { } }; @@ -1531,6 +1542,38 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { } }; +static struct hda_input_mux alc889_capture_sources[3] = { + /* Digital mic only available on first "ADC" */ + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Front Mic", 0xb }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1562,7 +1605,7 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } -static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9081,22 +9124,22 @@ static struct alc_config_preset alc883_presets[] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .const_channel_count = 6, .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_2_capture_sources, + ARRAY_SIZE(alc889_capture_sources), + .input_mux = alc889_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_8930g_init_hook, + .init_hook = alc889_acer_aspire_8930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, -- cgit v1.1 From 5fdc18d938c99399b73fe894bc24cb9400a1a2ee Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 4 Jun 2009 00:12:18 +0200 Subject: ALSA: Core - clean up snd_card_set_id* calls and remove possible id collision Move locking outside snd_card_set_id_internal() function and rename it to snd_card_set_id_no_lock() for better function description. User defined id is just copied to card structure at allocation time. The real unique id procedure is called in snd_card_register() to ensure real atomicity. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/init.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index a578d05..d5d40d7 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -153,7 +153,7 @@ int snd_card_create(int idx, const char *xid, if (!card) return -ENOMEM; if (xid) - snd_card_set_id(card, xid); + strlcpy(card->id, xid, sizeof(card->id)); err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) { @@ -476,16 +476,12 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -static void snd_card_set_id_internal(struct snd_card *card, const char *nid, - int do_locking) +static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid) { int i, len, idx_flag = 0, loops = SNDRV_CARDS; const char *spos, *src; char *id; - /* check if user specified own card->id */ - if (card->id[0] != '\0') - return; if (nid == NULL) { id = card->shortname; spos = src = id; @@ -522,16 +518,10 @@ static void snd_card_set_id_internal(struct snd_card *card, const char *nid, } if (!snd_info_check_reserved_words(id)) goto __change; - if (do_locking) - mutex_lock(&snd_card_mutex); for (i = 0; i < snd_ecards_limit; i++) { - if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) { - mutex_unlock(&snd_card_mutex); + if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) goto __change; - } } - if (do_locking) - mutex_unlock(&snd_card_mutex); break; __change: @@ -566,7 +556,12 @@ static void snd_card_set_id_internal(struct snd_card *card, const char *nid, */ void snd_card_set_id(struct snd_card *card, const char *nid) { - snd_card_set_id_internal(card, nid, 1); + /* check if user specified own card->id */ + if (card->id[0] != '\0') + return; + mutex_lock(&snd_card_mutex); + snd_card_set_id_no_lock(card, nid); + mutex_unlock(&snd_card_mutex); } EXPORT_SYMBOL(snd_card_set_id); @@ -663,8 +658,7 @@ int snd_card_register(struct snd_card *card) mutex_unlock(&snd_card_mutex); return 0; } - if (card->id[0] == '\0') - snd_card_set_id_internal(card, NULL, 0); + snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id); snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -- cgit v1.1 From d08664fdb50795b29cf70b0269ea02f7248e76c3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jun 2009 09:58:18 +0200 Subject: ASoC: Fix build error in twl4030.c Fix the (likely cut-n-paste) error by commit 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9, which causes the error below: sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache': sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function) Signed-off-by: Takashi Iwai --- sound/soc/codecs/twl4030.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c53c7ca..4dbb853 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -145,6 +145,7 @@ struct twl4030_priv { static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { + u8 *cache = codec->reg_cache; if (reg >= TWL4030_CACHEREGNUM) return -EIO; -- cgit v1.1 From 82ced6fd28653ab456c3e5b25e9ef3c1c96cd6e9 Mon Sep 17 00:00:00 2001 From: Jean Delvare Date: Thu, 4 Jun 2009 10:46:43 +0200 Subject: ALSA: Add missing __devexit_p() markers 3 ISA sound drivers lack their __devexit_p() markers, which would cause build failures when the kernel is built without hotplug support. Signed-off-by: Jean Delvare Cc: Kyle McMartin Signed-off-by: Takashi Iwai --- sound/isa/es1688/es1688.c | 2 +- sound/isa/gus/gusextreme.c | 2 +- sound/parisc/harmony.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 442b081..07df201 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -193,7 +193,7 @@ static int __devexit snd_es1688_remove(struct device *dev, unsigned int n) static struct isa_driver snd_es1688_driver = { .match = snd_es1688_match, .probe = snd_es1688_probe, - .remove = snd_es1688_remove, + .remove = __devexit_p(snd_es1688_remove), #if 0 /* FIXME */ .suspend = snd_es1688_suspend, .resume = snd_es1688_resume, diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 180a8de..65e4b18 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -348,7 +348,7 @@ static int __devexit snd_gusextreme_remove(struct device *dev, unsigned int n) static struct isa_driver snd_gusextreme_driver = { .match = snd_gusextreme_match, .probe = snd_gusextreme_probe, - .remove = snd_gusextreme_remove, + .remove = __devexit_p(snd_gusextreme_remove), #if 0 /* FIXME */ .suspend = snd_gusextreme_suspend, .resume = snd_gusextreme_resume, diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 6055fd6..610c439 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -1020,7 +1020,7 @@ static struct parisc_driver snd_harmony_driver = { .name = "harmony", .id_table = snd_harmony_devtable, .probe = snd_harmony_probe, - .remove = snd_harmony_remove, + .remove = __devexit_p(snd_harmony_remove), }; static int __init -- cgit v1.1 From 65f759831179bb8922f2a91a989487161d476a94 Mon Sep 17 00:00:00 2001 From: Alexander Beregalov Date: Thu, 4 Jun 2009 13:46:16 +0400 Subject: ALSA: hda_intel: fix build error when !PM Fix this build error when CONFIG_PM is not set: ound/pci/hda/hda_intel.c: In function 'azx_bus_reset': sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all' sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend' sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume' Signed-off-by: Alexander Beregalov Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f44645..4e9ea70 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1260,17 +1260,20 @@ static void azx_stop_chip(struct azx *chip); static void azx_bus_reset(struct hda_bus *bus) { struct azx *chip = bus->private_data; - int i; bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip); +#ifdef CONFIG_PM if (chip->initialized) { + int i; + for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); } +#endif bus->in_reset = 0; } -- cgit v1.1 From e3509ff0fb9df53e45cd68488e3b463a80455db7 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 3 Jun 2009 17:44:49 +0200 Subject: ASoC: fix NULL pointer dereference in soc_suspend() In case the initalization of an soc_device failed, there is no codec associated with it. soc_suspend() will still dereference the pointer and cause an Ooops when entering the sleep mode. This happens on our board with a multi-target kernel image when booted on a machine without audio circuits. This patch makes the code bail out very early in this special case. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4aa8e2d..3f44150 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -628,6 +628,12 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = card->codec; int i; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (!codec) + return 0; + /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ -- cgit v1.1 From 6d74b86d3c0f9cfa949566a862aaad840e393249 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 09:26:41 +0200 Subject: ALSA: ctxfi - Allow 64bit DMA emu20kx chips support 64bit address PTE. Allow the DMA bit mask to accept 64bit address, too. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k1.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index b7b8e6f..1da1f82 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -15,8 +15,6 @@ * */ -#include "cthw20k1.h" -#include "ct20k1reg.h" #include #include #include @@ -26,8 +24,14 @@ #include #include #include +#include "cthw20k1.h" +#include "ct20k1reg.h" -#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bits */ +#if BITS_PER_LONG == 32 +#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */ +#else +#define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */ +#endif struct hw20k1 { struct hw hw; -- cgit v1.1 From 42a0b31827e4c555efebda7d347cf4ea6b82913a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 09:29:22 +0200 Subject: ALSA: ctxfi - Fix endian-dependent codes The UAA-mode check in hwct20k1.c is implemented with the endian-dependent codes. Fix to be more portable (and readable). Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k1.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 1da1f82..fd5f454 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1785,6 +1785,8 @@ static int hw_have_digit_io_switch(struct hw *hw) || ((subsys_id & 0xf000) == 0x6000)); } +#define CTLBITS(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d)) + #define UAA_CFG_PWRSTATUS 0x44 #define UAA_CFG_SPACE_FLAG 0xA0 #define UAA_CORE_CHANGE 0x3FFC @@ -1792,12 +1794,18 @@ static int uaa_to_xfi(struct pci_dev *pci) { unsigned int bar0, bar1, bar2, bar3, bar4, bar5; unsigned int cmd, irq, cl_size, l_timer, pwr; - unsigned int CTLA, CTLZ, CTLL, CTLX, CTL_, CTLF, CTLi; unsigned int is_uaa = 0; unsigned int data[4] = {0}; unsigned int io_base; void *mem_base; int i = 0; + const u32 CTLX = CTLBITS('C', 'T', 'L', 'X'); + const u32 CTL_ = CTLBITS('C', 'T', 'L', '-'); + const u32 CTLF = CTLBITS('C', 'T', 'L', 'F'); + const u32 CTLi = CTLBITS('C', 'T', 'L', 'i'); + const u32 CTLA = CTLBITS('C', 'T', 'L', 'A'); + const u32 CTLZ = CTLBITS('C', 'T', 'L', 'Z'); + const u32 CTLL = CTLBITS('C', 'T', 'L', 'L'); /* By default, Hendrix card UAA Bar0 should be using memory... */ io_base = pci_resource_start(pci, 0); @@ -1805,14 +1813,6 @@ static int uaa_to_xfi(struct pci_dev *pci) if (NULL == mem_base) return -ENOENT; - CTLX = ___constant_swab32(*((unsigned int *)"CTLX")); - CTL_ = ___constant_swab32(*((unsigned int *)"CTL-")); - CTLF = ___constant_swab32(*((unsigned int *)"CTLF")); - CTLi = ___constant_swab32(*((unsigned int *)"CTLi")); - CTLA = ___constant_swab32(*((unsigned int *)"CTLA")); - CTLZ = ___constant_swab32(*((unsigned int *)"CTLZ")); - CTLL = ___constant_swab32(*((unsigned int *)"CTLL")); - /* Read current mode from Mode Change Register */ for (i = 0; i < 4; i++) data[i] = readl(mem_base + UAA_CORE_CHANGE); -- cgit v1.1 From 3e1647c5b54a91a7182e121cfe569e6f0bf167ec Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Guido=20G=C3=BCnther?= Date: Fri, 5 Jun 2009 00:47:26 +0200 Subject: ALSA: support Sony Vaio TT MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit with BIOS probing only we offer a non functional headphone swith and volume slider. Signed-off-by: Guido Günther Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 45 ++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 29c6125..54b0805 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -159,6 +159,7 @@ ALC883/888 3stack-6ch-intel Intel DG33* boards asus-p5q ASUS P5Q-EM boards mb31 MacBook 3,1 + sony-vaio-tt Sony VAIO TT auto auto-config reading BIOS (default) ALC861/660 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8699b7f..1ab1b92 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -243,6 +243,7 @@ enum { ALC888_ASUS_EEE1601, ALC889A_MB31, ALC1200_ASUS_P5Q, + ALC883_SONY_VAIO_TT, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -8134,6 +8135,16 @@ static struct snd_kcontrol_new alc889A_mb31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_vaiott_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + static struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -8410,6 +8421,17 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; +static struct hda_verb alc883_vaiott_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static void alc888_3st_hp_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8669,6 +8691,16 @@ static void alc888_lenovo_sky_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc883_vaiott_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x17; + alc_automute_amp(codec); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -8884,6 +8916,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC889A_MB31] = "mb31", + [ALC883_SONY_VAIO_TT] = "sony-vaio-tt", [ALC883_AUTO] = "auto", }; @@ -8978,6 +9011,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), {} }; @@ -9373,6 +9407,17 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc889A_mb31_unsol_event, .init_hook = alc889A_mb31_automute, }, + [ALC883_SONY_VAIO_TT] = { + .mixers = { alc883_vaiott_mixer }, + .init_verbs = { alc883_init_verbs, alc883_vaiott_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc883_vaiott_init_hook, + }, }; -- cgit v1.1 From 6bc5874a1ddf98ac0fe6c4eab7d286c11cb1c748 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 12:15:51 +0200 Subject: ALSA: ctxfi - Fix previous fix for 64bit DMA Remove unneeded substitution to 32bit int to make it really working. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k1.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index fd5f454..e530a6d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1883,18 +1883,17 @@ static int hw_card_start(struct hw *hw) int err = 0; struct pci_dev *pci = hw->pci; u16 subsys_id = 0; - unsigned int dma_mask = 0; err = pci_enable_device(pci); if (err < 0) return err; /* Set DMA transfer mask */ - dma_mask = CT_XFI_DMA_MASK; - if (pci_set_dma_mask(pci, dma_mask) < 0 || - pci_set_consistent_dma_mask(pci, dma_mask) < 0) { + if (pci_set_dma_mask(pci, CT_XFI_DMA_MASK) < 0 || + pci_set_consistent_dma_mask(pci, CT_XFI_DMA_MASK) < 0) { printk(KERN_ERR "architecture does not support PCI " - "busmaster DMA with mask 0x%x\n", dma_mask); + "busmaster DMA with mask 0x%llx\n", + CT_XFI_DMA_MASK); err = -ENXIO; goto error1; } -- cgit v1.1 From b7bbf876087e0e2c0ba723a8398083c9a9ac1dfd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 16:11:07 +0200 Subject: ALSA: ctxfi - Use native timer interrupt on emu20k1 emu20k1 has a native timer interrupt based on the audio clock, which is more accurate than the system timer (from the synchronization POV). This patch adds the code to handle this with multiple streams. The system timer is still used on emu20k2, and can be used also for emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/Makefile | 2 +- sound/pci/ctxfi/ct20k1reg.h | 2 + sound/pci/ctxfi/ctatc.c | 21 ++- sound/pci/ctxfi/ctatc.h | 9 +- sound/pci/ctxfi/cthardware.h | 19 ++ sound/pci/ctxfi/cthw20k1.c | 43 ++++- sound/pci/ctxfi/ctpcm.c | 106 ++--------- sound/pci/ctxfi/cttimer.c | 417 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/ctxfi/cttimer.h | 29 +++ 9 files changed, 543 insertions(+), 105 deletions(-) create mode 100644 sound/pci/ctxfi/cttimer.c create mode 100644 sound/pci/ctxfi/cttimer.h diff --git a/sound/pci/ctxfi/Makefile b/sound/pci/ctxfi/Makefile index 2904323..15075f8 100644 --- a/sound/pci/ctxfi/Makefile +++ b/sound/pci/ctxfi/Makefile @@ -1,5 +1,5 @@ snd-ctxfi-objs := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \ - ctsrc.o ctamixer.o ctdaio.o ctimap.o cthardware.o \ + ctsrc.o ctamixer.o ctdaio.o ctimap.o cthardware.o cttimer.o \ cthw20k2.o cthw20k1.o obj-$(CONFIG_SND_CTXFI) += snd-ctxfi.o diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h index c62e677..f2e34e3 100644 --- a/sound/pci/ctxfi/ct20k1reg.h +++ b/sound/pci/ctxfi/ct20k1reg.h @@ -589,6 +589,8 @@ #define WC 0x1C6000 #define TIMR 0x1C6004 +# define TIMR_IE (1<<15) +# define TIMR_IP (1<<14) #define GIP 0x1C6010 #define GIE 0x1C6014 diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 6849475..10b7419 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -22,6 +22,7 @@ #include "ctsrc.h" #include "ctamixer.h" #include "ctdaio.h" +#include "cttimer.h" #include #include #include @@ -307,6 +308,8 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) src = apcm->src; } + ct_timer_prepare(apcm->timer); + return 0; error1: @@ -389,6 +392,7 @@ static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) src->ops->set_state(src, SRC_STATE_INIT); src->ops->commit_write(src); + ct_timer_start(apcm->timer); return 0; } @@ -397,6 +401,8 @@ static int atc_pcm_stop(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct src *src = NULL; int i = 0; + ct_timer_stop(apcm->timer); + src = apcm->src; src->ops->set_bm(src, 0); src->ops->set_state(src, SRC_STATE_OFF); @@ -701,6 +707,8 @@ static int atc_pcm_capture_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) } } + ct_timer_prepare(apcm->timer); + return 0; } @@ -749,6 +757,7 @@ static int atc_pcm_capture_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) /* Enable relevant SRCs synchronously */ src_mgr->commit_write(src_mgr); + ct_timer_start(apcm->timer); return 0; } @@ -906,6 +915,8 @@ spdif_passthru_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) dao->ops->set_right_input(dao, &amixer->rsc); spin_unlock_irqrestore(&atc->atc_lock, flags); + ct_timer_prepare(apcm->timer); + return 0; } @@ -1100,6 +1111,11 @@ static int ct_atc_destroy(struct ct_atc *atc) if (NULL == atc) return 0; + if (atc->timer) { + ct_timer_free(atc->timer); + atc->timer = NULL; + } + /* Stop hardware and disable all interrupts */ if (NULL != atc->hw) ((struct hw *)atc->hw)->card_stop(atc->hw); @@ -1586,6 +1602,10 @@ int ct_atc_create(struct snd_card *card, struct pci_dev *pci, /* Build topology */ atc_connect_resources(atc); + atc->timer = ct_timer_new(atc); + if (!atc->timer) + goto error1; + atc->create_alsa_devs = ct_create_alsa_devs; err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); @@ -1602,4 +1622,3 @@ error1: printk(KERN_ERR "ctxfi: Something wrong!!!\n"); return err; } - diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index b86d12c..a3f9b1b 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -59,16 +59,15 @@ struct ct_atc_chip_details { }; struct ct_atc; +struct ct_timer; +struct ct_timer_instance; /* alsa pcm stream descriptor */ struct ct_atc_pcm { struct snd_pcm_substream *substream; void (*interrupt)(struct ct_atc_pcm *apcm); + struct ct_timer_instance *timer; unsigned int started:1; - unsigned int stop_timer:1; - struct timer_list timer; - spinlock_t timer_lock; - unsigned int position; /* Only mono and interleaved modes are supported now. */ struct ct_vm_block *vm_block; @@ -144,6 +143,8 @@ struct ct_atc { unsigned char n_src; unsigned char n_srcimp; unsigned char n_pcm; + + struct ct_timer *timer; }; diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index b0512df..35350cf 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -145,6 +145,12 @@ struct hw { int (*daio_mgr_set_imapaddr)(void *blk, unsigned int addr); int (*daio_mgr_commit_write)(struct hw *hw, void *blk); + int (*set_timer_irq)(struct hw *hw, int enable); + int (*set_timer_tick)(struct hw *hw, unsigned int tick); + + void (*irq_callback)(void *data, unsigned int bit); + void *irq_callback_data; + struct pci_dev *pci; /* the pci kernel structure of this card */ int irq; unsigned long io_base; @@ -157,4 +163,17 @@ int destroy_hw_obj(struct hw *hw); unsigned int get_field(unsigned int data, unsigned int field); void set_field(unsigned int *data, unsigned int field, unsigned int value); +/* IRQ bits */ +#define PLL_INT (1 << 10) /* PLL input-clock out-of-range */ +#define FI_INT (1 << 9) /* forced interrupt */ +#define IT_INT (1 << 8) /* timer interrupt */ +#define PCI_INT (1 << 7) /* PCI bus error pending */ +#define URT_INT (1 << 6) /* UART Tx/Rx */ +#define GPI_INT (1 << 5) /* GPI pin */ +#define MIX_INT (1 << 4) /* mixer parameter segment FIFO channels */ +#define DAI_INT (1 << 3) /* DAI (SR-tracker or SPDIF-receiver) */ +#define TP_INT (1 << 2) /* transport priority queue */ +#define DSP_INT (1 << 1) /* DSP */ +#define SRC_INT (1 << 0) /* SRC channels */ + #endif /* CTHARDWARE_H */ diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index e530a6d..550b30a 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1171,6 +1171,21 @@ static int daio_mgr_put_ctrl_blk(void *blk) return 0; } +/* Timer interrupt */ +static int set_timer_irq(struct hw *hw, int enable) +{ + hw_write_20kx(hw, GIE, enable ? IT_INT : 0); + return 0; +} + +static int set_timer_tick(struct hw *hw, unsigned int ticks) +{ + if (ticks) + ticks |= TIMR_IE | TIMR_IP; + hw_write_20kx(hw, TIMR, ticks); + return 0; +} + /* Card hardware initialization block */ struct dac_conf { unsigned int msr; /* master sample rate in rsrs */ @@ -1878,6 +1893,22 @@ static int uaa_to_xfi(struct pci_dev *pci) return 0; } +static irqreturn_t ct_20k1_interrupt(int irq, void *dev_id) +{ + struct hw *hw = dev_id; + unsigned int status; + + status = hw_read_20kx(hw, GIP); + if (!status) + return IRQ_NONE; + + if (hw->irq_callback) + hw->irq_callback(hw->irq_callback_data, status); + + hw_write_20kx(hw, GIP, status); + return IRQ_HANDLED; +} + static int hw_card_start(struct hw *hw) { int err = 0; @@ -1914,12 +1945,13 @@ static int hw_card_start(struct hw *hw) hw->io_base = pci_resource_start(pci, 0); } - /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED, - atc->chip_details->nm_card, hw))) { + err = request_irq(pci->irq, ct_20k1_interrupt, IRQF_SHARED, + "ctxfi", hw); + if (err < 0) { + printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq); goto error2; } hw->irq = pci->irq; - */ pci_set_master(pci); @@ -1936,6 +1968,8 @@ error1: static int hw_card_stop(struct hw *hw) { /* TODO: Disable interrupt and so on... */ + if (hw->irq >= 0) + synchronize_irq(hw->irq); return 0; } @@ -2215,6 +2249,9 @@ int create_20k1_hw_obj(struct hw **rhw) hw->daio_mgr_set_imapaddr = daio_mgr_set_imapaddr; hw->daio_mgr_commit_write = daio_mgr_commit_write; + hw->set_timer_irq = set_timer_irq; + hw->set_timer_tick = set_timer_tick; + *rhw = hw; return 0; diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 52ddf19..32b742d 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -16,6 +16,7 @@ */ #include "ctpcm.h" +#include "cttimer.h" #include /* Hardware descriptions for playback */ @@ -108,6 +109,7 @@ static void ct_atc_pcm_free_substream(struct snd_pcm_runtime *runtime) struct ct_atc *atc = snd_pcm_substream_chip(apcm->substream); atc->pcm_release_resources(atc, apcm); + ct_timer_instance_free(apcm->timer); kfree(apcm); runtime->private_data = NULL; } @@ -124,8 +126,6 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream) if (NULL == apcm) return -ENOMEM; - spin_lock_init(&apcm->timer_lock); - apcm->stop_timer = 0; apcm->substream = substream; apcm->interrupt = ct_atc_pcm_interrupt; runtime->private_data = apcm; @@ -153,6 +153,10 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream) return err; } + apcm->timer = ct_timer_instance_new(atc->timer, apcm); + if (!apcm->timer) + return -ENOMEM; + return 0; } @@ -182,89 +186,6 @@ static int ct_pcm_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static void ct_pcm_timer_callback(unsigned long data) -{ - struct ct_atc_pcm *apcm = (struct ct_atc_pcm *)data; - struct snd_pcm_substream *substream = apcm->substream; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int period_size = runtime->period_size; - unsigned int buffer_size = runtime->buffer_size; - unsigned long flags; - unsigned int position = 0, dist = 0, interval = 0; - - position = substream->ops->pointer(substream); - dist = (position + buffer_size - apcm->position) % buffer_size; - if ((dist >= period_size) || - (position/period_size != apcm->position/period_size)) { - apcm->interrupt(apcm); - apcm->position = position; - } - /* Add extra HZ*5/1000 to avoid overrun issue when recording - * at 8kHz in 8-bit format or at 88kHz in 24-bit format. */ - interval = ((period_size - (position % period_size)) - * HZ + (runtime->rate - 1)) / runtime->rate + HZ * 5 / 1000; - spin_lock_irqsave(&apcm->timer_lock, flags); - apcm->timer.expires = jiffies + interval; - if (!apcm->stop_timer) - add_timer(&apcm->timer); - - spin_unlock_irqrestore(&apcm->timer_lock, flags); -} - -static int ct_pcm_timer_prepare(struct ct_atc_pcm *apcm) -{ - unsigned long flags; - - spin_lock_irqsave(&apcm->timer_lock, flags); - if (timer_pending(&apcm->timer)) { - /* The timer has already been started. */ - spin_unlock_irqrestore(&apcm->timer_lock, flags); - return 0; - } - - init_timer(&apcm->timer); - apcm->timer.data = (unsigned long)apcm; - apcm->timer.function = ct_pcm_timer_callback; - spin_unlock_irqrestore(&apcm->timer_lock, flags); - apcm->position = 0; - - return 0; -} - -static int ct_pcm_timer_start(struct ct_atc_pcm *apcm) -{ - struct snd_pcm_runtime *runtime = apcm->substream->runtime; - unsigned long flags; - - spin_lock_irqsave(&apcm->timer_lock, flags); - if (timer_pending(&apcm->timer)) { - /* The timer has already been started. */ - spin_unlock_irqrestore(&apcm->timer_lock, flags); - return 0; - } - - apcm->timer.expires = jiffies + (runtime->period_size * HZ + - (runtime->rate - 1)) / runtime->rate; - apcm->stop_timer = 0; - add_timer(&apcm->timer); - spin_unlock_irqrestore(&apcm->timer_lock, flags); - - return 0; -} - -static int ct_pcm_timer_stop(struct ct_atc_pcm *apcm) -{ - unsigned long flags; - - spin_lock_irqsave(&apcm->timer_lock, flags); - apcm->stop_timer = 1; - del_timer(&apcm->timer); - spin_unlock_irqrestore(&apcm->timer_lock, flags); - - try_to_del_timer_sync(&apcm->timer); - - return 0; -} static int ct_pcm_playback_prepare(struct snd_pcm_substream *substream) { @@ -283,8 +204,6 @@ static int ct_pcm_playback_prepare(struct snd_pcm_substream *substream) return err; } - ct_pcm_timer_prepare(apcm); - return 0; } @@ -300,12 +219,10 @@ ct_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: atc->pcm_playback_start(atc, apcm); - ct_pcm_timer_start(apcm); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ct_pcm_timer_stop(apcm); atc->pcm_playback_stop(atc, apcm); break; default: @@ -341,9 +258,7 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream) if (NULL == apcm) return -ENOMEM; - spin_lock_init(&apcm->timer_lock); apcm->started = 0; - apcm->stop_timer = 0; apcm->substream = substream; apcm->interrupt = ct_atc_pcm_interrupt; runtime->private_data = apcm; @@ -365,6 +280,10 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream) return err; } + apcm->timer = ct_timer_instance_new(atc->timer, apcm); + if (!apcm->timer) + return -ENOMEM; + return 0; } @@ -388,8 +307,6 @@ static int ct_pcm_capture_prepare(struct snd_pcm_substream *substream) return err; } - ct_pcm_timer_prepare(apcm); - return 0; } @@ -403,14 +320,11 @@ ct_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: atc->pcm_capture_start(atc, apcm); - ct_pcm_timer_start(apcm); break; case SNDRV_PCM_TRIGGER_STOP: - ct_pcm_timer_stop(apcm); atc->pcm_capture_stop(atc, apcm); break; default: - ct_pcm_timer_stop(apcm); atc->pcm_capture_stop(atc, apcm); break; } diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c new file mode 100644 index 0000000..3acb26d --- /dev/null +++ b/sound/pci/ctxfi/cttimer.c @@ -0,0 +1,417 @@ +/* + * PCM timer handling on ctxfi + * + * This source file is released under GPL v2 license (no other versions). + * See the COPYING file included in the main directory of this source + * distribution for the license terms and conditions. + */ + +#include +#include +#include +#include "ctatc.h" +#include "cthardware.h" +#include "cttimer.h" + +struct ct_timer_ops { + void (*init)(struct ct_timer_instance *); + void (*prepare)(struct ct_timer_instance *); + void (*start)(struct ct_timer_instance *); + void (*stop)(struct ct_timer_instance *); + void (*free_instance)(struct ct_timer_instance *); + void (*interrupt)(struct ct_timer *); + void (*free_global)(struct ct_timer *); +}; + +/* timer instance -- assigned to each PCM stream */ +struct ct_timer_instance { + spinlock_t lock; + struct ct_timer *timer_base; + struct ct_atc_pcm *apcm; + struct snd_pcm_substream *substream; + struct timer_list timer; + struct list_head instance_list; + struct list_head running_list; + unsigned int position; + unsigned int frag_count; + unsigned int running:1; + unsigned int need_update:1; +}; + +/* timer instance manager */ +struct ct_timer { + spinlock_t lock; /* global timer lock (for xfitimer) */ + spinlock_t list_lock; /* lock for instance list */ + struct ct_atc *atc; + struct ct_timer_ops *ops; + struct list_head instance_head; + struct list_head running_head; + unsigned int irq_handling:1; /* in IRQ handling */ + unsigned int reprogram:1; /* need to reprogram the internval */ + unsigned int running:1; /* global timer running */ +}; + + +/* + * system-timer-based updates + */ + +static void ct_systimer_callback(unsigned long data) +{ + struct ct_timer_instance *ti = (struct ct_timer_instance *)data; + struct snd_pcm_substream *substream = ti->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct ct_atc_pcm *apcm = ti->apcm; + unsigned int period_size = runtime->period_size; + unsigned int buffer_size = runtime->buffer_size; + unsigned long flags; + unsigned int position, dist, interval; + + position = substream->ops->pointer(substream); + dist = (position + buffer_size - ti->position) % buffer_size; + if (dist >= period_size || + position / period_size != ti->position / period_size) { + apcm->interrupt(apcm); + ti->position = position; + } + /* Add extra HZ*5/1000 to avoid overrun issue when recording + * at 8kHz in 8-bit format or at 88kHz in 24-bit format. */ + interval = ((period_size - (position % period_size)) + * HZ + (runtime->rate - 1)) / runtime->rate + HZ * 5 / 1000; + spin_lock_irqsave(&ti->lock, flags); + if (ti->running) + mod_timer(&ti->timer, jiffies + interval); + spin_unlock_irqrestore(&ti->lock, flags); +} + +static void ct_systimer_init(struct ct_timer_instance *ti) +{ + setup_timer(&ti->timer, ct_systimer_callback, + (unsigned long)ti); +} + +static void ct_systimer_start(struct ct_timer_instance *ti) +{ + struct snd_pcm_runtime *runtime = ti->substream->runtime; + unsigned long flags; + + spin_lock_irqsave(&ti->lock, flags); + ti->running = 1; + mod_timer(&ti->timer, + jiffies + (runtime->period_size * HZ + + (runtime->rate - 1)) / runtime->rate); + spin_unlock_irqrestore(&ti->lock, flags); +} + +static void ct_systimer_stop(struct ct_timer_instance *ti) +{ + unsigned long flags; + + spin_lock_irqsave(&ti->lock, flags); + ti->running = 0; + del_timer(&ti->timer); + spin_unlock_irqrestore(&ti->lock, flags); +} + +static void ct_systimer_prepare(struct ct_timer_instance *ti) +{ + ct_systimer_stop(ti); + try_to_del_timer_sync(&ti->timer); +} + +#define ct_systimer_free ct_systimer_prepare + +static struct ct_timer_ops ct_systimer_ops = { + .init = ct_systimer_init, + .free_instance = ct_systimer_free, + .prepare = ct_systimer_prepare, + .start = ct_systimer_start, + .stop = ct_systimer_stop, +}; + + +/* + * Handling multiple streams using a global emu20k1 timer irq + */ + +#define CT_TIMER_FREQ 48000 +#define MAX_TICKS ((1 << 13) - 1) + +static void ct_xfitimer_irq_rearm(struct ct_timer *atimer, int ticks) +{ + struct hw *hw = atimer->atc->hw; + if (ticks > MAX_TICKS) + ticks = MAX_TICKS; + hw->set_timer_tick(hw, ticks); + if (!atimer->running) + hw->set_timer_irq(hw, 1); + atimer->running = 1; +} + +static void ct_xfitimer_irq_stop(struct ct_timer *atimer) +{ + if (atimer->running) { + struct hw *hw = atimer->atc->hw; + hw->set_timer_irq(hw, 0); + hw->set_timer_tick(hw, 0); + atimer->running = 0; + } +} + +/* + * reprogram the timer interval; + * checks the running instance list and determines the next timer interval. + * also updates the each stream position, returns the number of streams + * to call snd_pcm_period_elapsed() appropriately + * + * call this inside the lock and irq disabled + */ +static int ct_xfitimer_reprogram(struct ct_timer *atimer) +{ + struct ct_timer_instance *ti; + int min_intr = -1; + int updates = 0; + + list_for_each_entry(ti, &atimer->running_head, running_list) { + struct snd_pcm_runtime *runtime; + unsigned int pos, diff; + int intr; + runtime = ti->substream->runtime; + pos = ti->substream->ops->pointer(ti->substream); + if (pos < ti->position) + diff = runtime->buffer_size - ti->position + pos; + else + diff = pos - ti->position; + ti->position = pos; + while (diff >= ti->frag_count) { + ti->frag_count += runtime->period_size; + ti->need_update = 1; + updates++; + } + ti->frag_count -= diff; + intr = div_u64((u64)ti->frag_count * CT_TIMER_FREQ, + runtime->rate); + if (min_intr < 0 || intr < min_intr) + min_intr = intr; + } + + if (min_intr > 0) + ct_xfitimer_irq_rearm(atimer, min_intr); + else + ct_xfitimer_irq_stop(atimer); + + atimer->reprogram = 0; /* clear flag */ + return updates; +} + +/* look through the instance list and call period_elapsed if needed */ +static void ct_xfitimer_check_period(struct ct_timer *atimer) +{ + struct ct_timer_instance *ti; + unsigned long flags; + + spin_lock_irqsave(&atimer->list_lock, flags); + list_for_each_entry(ti, &atimer->instance_head, instance_list) { + if (ti->need_update) { + ti->need_update = 0; + ti->apcm->interrupt(ti->apcm); + } + } + spin_unlock_irqrestore(&atimer->list_lock, flags); +} + +/* Handle timer-interrupt */ +static void ct_xfitimer_callback(struct ct_timer *atimer) +{ + int update; + unsigned long flags; + + spin_lock_irqsave(&atimer->lock, flags); + atimer->irq_handling = 1; + do { + update = ct_xfitimer_reprogram(atimer); + spin_unlock(&atimer->lock); + if (update) + ct_xfitimer_check_period(atimer); + spin_lock(&atimer->lock); + } while (atimer->reprogram); + atimer->irq_handling = 0; + spin_unlock_irqrestore(&atimer->lock, flags); +} + +static void ct_xfitimer_prepare(struct ct_timer_instance *ti) +{ + ti->frag_count = ti->substream->runtime->period_size; + ti->need_update = 0; +} + + +/* start/stop the timer */ +static void ct_xfitimer_update(struct ct_timer *atimer) +{ + unsigned long flags; + int update; + + if (atimer->irq_handling) { + /* reached from IRQ handler; let it handle later */ + atimer->reprogram = 1; + return; + } + + spin_lock_irqsave(&atimer->lock, flags); + ct_xfitimer_irq_stop(atimer); + update = ct_xfitimer_reprogram(atimer); + spin_unlock_irqrestore(&atimer->lock, flags); + if (update) + ct_xfitimer_check_period(atimer); +} + +static void ct_xfitimer_start(struct ct_timer_instance *ti) +{ + struct ct_timer *atimer = ti->timer_base; + unsigned long flags; + + spin_lock_irqsave(&atimer->lock, flags); + list_add(&ti->running_list, &atimer->running_head); + spin_unlock_irqrestore(&atimer->lock, flags); + ct_xfitimer_update(atimer); +} + +static void ct_xfitimer_stop(struct ct_timer_instance *ti) +{ + struct ct_timer *atimer = ti->timer_base; + unsigned long flags; + + spin_lock_irqsave(&atimer->lock, flags); + list_del_init(&ti->running_list); + ti->need_update = 0; + spin_unlock_irqrestore(&atimer->lock, flags); + ct_xfitimer_update(atimer); +} + +static void ct_xfitimer_free_global(struct ct_timer *atimer) +{ + ct_xfitimer_irq_stop(atimer); +} + +static struct ct_timer_ops ct_xfitimer_ops = { + .prepare = ct_xfitimer_prepare, + .start = ct_xfitimer_start, + .stop = ct_xfitimer_stop, + .interrupt = ct_xfitimer_callback, + .free_global = ct_xfitimer_free_global, +}; + +/* + * timer instance + */ + +struct ct_timer_instance * +ct_timer_instance_new(struct ct_timer *atimer, struct ct_atc_pcm *apcm) +{ + struct ct_timer_instance *ti; + + ti = kzalloc(sizeof(*ti), GFP_KERNEL); + if (!ti) + return NULL; + spin_lock_init(&ti->lock); + INIT_LIST_HEAD(&ti->instance_list); + INIT_LIST_HEAD(&ti->running_list); + ti->timer_base = atimer; + ti->apcm = apcm; + ti->substream = apcm->substream; + if (atimer->ops->init) + atimer->ops->init(ti); + + spin_lock_irq(&atimer->list_lock); + list_add(&ti->instance_list, &atimer->instance_head); + spin_unlock_irq(&atimer->list_lock); + + return ti; +} + +void ct_timer_prepare(struct ct_timer_instance *ti) +{ + if (ti->timer_base->ops->prepare) + ti->timer_base->ops->prepare(ti); + ti->position = 0; + ti->running = 0; +} + +void ct_timer_start(struct ct_timer_instance *ti) +{ + struct ct_timer *atimer = ti->timer_base; + atimer->ops->start(ti); +} + +void ct_timer_stop(struct ct_timer_instance *ti) +{ + struct ct_timer *atimer = ti->timer_base; + atimer->ops->stop(ti); +} + +void ct_timer_instance_free(struct ct_timer_instance *ti) +{ + struct ct_timer *atimer = ti->timer_base; + + atimer->ops->stop(ti); /* to be sure */ + if (atimer->ops->free_instance) + atimer->ops->free_instance(ti); + + spin_lock_irq(&atimer->list_lock); + list_del(&ti->instance_list); + spin_unlock_irq(&atimer->list_lock); + + kfree(ti); +} + +/* + * timer manager + */ + +#define USE_SYSTEM_TIMER 0 + +static void ct_timer_interrupt(void *data, unsigned int status) +{ + struct ct_timer *timer = data; + + /* Interval timer interrupt */ + if ((status & IT_INT) && timer->ops->interrupt) + timer->ops->interrupt(timer); +} + +struct ct_timer *ct_timer_new(struct ct_atc *atc) +{ + struct ct_timer *atimer; + struct hw *hw; + + atimer = kzalloc(sizeof(*atimer), GFP_KERNEL); + if (!atimer) + return NULL; + spin_lock_init(&atimer->lock); + spin_lock_init(&atimer->list_lock); + INIT_LIST_HEAD(&atimer->instance_head); + INIT_LIST_HEAD(&atimer->running_head); + atimer->atc = atc; + hw = atc->hw; + if (!USE_SYSTEM_TIMER && hw->set_timer_irq) { + printk(KERN_INFO "ctxfi: Use xfi-native timer\n"); + atimer->ops = &ct_xfitimer_ops; + hw->irq_callback_data = atimer; + hw->irq_callback = ct_timer_interrupt; + } else { + printk(KERN_INFO "ctxfi: Use system timer\n"); + atimer->ops = &ct_systimer_ops; + } + return atimer; +} + +void ct_timer_free(struct ct_timer *atimer) +{ + struct hw *hw = atimer->atc->hw; + hw->irq_callback = NULL; + if (atimer->ops->free_global) + atimer->ops->free_global(atimer); + kfree(atimer); +} + diff --git a/sound/pci/ctxfi/cttimer.h b/sound/pci/ctxfi/cttimer.h new file mode 100644 index 0000000..9793482 --- /dev/null +++ b/sound/pci/ctxfi/cttimer.h @@ -0,0 +1,29 @@ +/* + * Timer handling + */ + +#ifndef __CTTIMER_H +#define __CTTIMER_H + +#include +#include +#include + +struct snd_pcm_substream; +struct ct_atc; +struct ct_atc_pcm; + +struct ct_timer; +struct ct_timer_instance; + +struct ct_timer *ct_timer_new(struct ct_atc *atc); +void ct_timer_free(struct ct_timer *atimer); + +struct ct_timer_instance * +ct_timer_instance_new(struct ct_timer *atimer, struct ct_atc_pcm *apcm); +void ct_timer_instance_free(struct ct_timer_instance *ti); +void ct_timer_start(struct ct_timer_instance *ti); +void ct_timer_stop(struct ct_timer_instance *ti); +void ct_timer_prepare(struct ct_timer_instance *ti); + +#endif /* __CTTIMER_H */ -- cgit v1.1 From 775ffa1d3e5a550dd2c9d947d773021c61531b36 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 16:12:16 +0200 Subject: ALSA: ctxfi - Set periods_min to 2 Set 2 to minimal periods of playback pcm setups, too. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctpcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 32b742d..a0bd31c 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -40,7 +40,7 @@ static struct snd_pcm_hardware ct_pcm_playback_hw = { .buffer_bytes_max = (128*1024), .period_bytes_min = (64), .period_bytes_max = (128*1024), - .periods_min = 1, + .periods_min = 2, .periods_max = 1024, .fifo_size = 0, }; @@ -62,7 +62,7 @@ static struct snd_pcm_hardware ct_spdif_passthru_playback_hw = { .buffer_bytes_max = (128*1024), .period_bytes_min = (64), .period_bytes_max = (128*1024), - .periods_min = 1, + .periods_min = 2, .periods_max = 1024, .fifo_size = 0, }; -- cgit v1.1 From 2a36f67f8c81f0babda0e811c760b7bfa971010b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 16:34:10 +0200 Subject: ALSA: ctxfi - Clean up / optimize - Use static tables instead of assigining each funciton pointer - Add __devinit* to appropriate places; pcm, mixer and timer cannot be marked because they are kept in the function table that lives long - Move create_alsa_devs function out of struct ct_atc to mark it __devinit Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 82 ++++++++--------- sound/pci/ctxfi/ctatc.h | 3 +- sound/pci/ctxfi/cthardware.c | 4 +- sound/pci/ctxfi/cthw20k1.c | 205 +++++++++++++++++++++---------------------- sound/pci/ctxfi/cthw20k2.c | 195 ++++++++++++++++++++-------------------- sound/pci/ctxfi/xfi.c | 2 +- 6 files changed, 243 insertions(+), 248 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 10b7419..9b13245 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1202,7 +1202,7 @@ static int atc_dev_free(struct snd_device *dev) return ct_atc_destroy(atc); } -static int atc_identify_card(struct ct_atc *atc) +static int __devinit atc_identify_card(struct ct_atc *atc) { u16 subsys; u8 revision; @@ -1243,7 +1243,7 @@ static int atc_identify_card(struct ct_atc *atc) return 0; } -static int ct_create_alsa_devs(struct ct_atc *atc) +int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc) { enum CTALSADEVS i; struct hw *hw = atc->hw; @@ -1277,7 +1277,7 @@ static int ct_create_alsa_devs(struct ct_atc *atc) return 0; } -static int atc_create_hw_devs(struct ct_atc *atc) +static int __devinit atc_create_hw_devs(struct ct_atc *atc) { struct hw *hw = NULL; struct card_conf info = {0}; @@ -1313,7 +1313,7 @@ static int atc_create_hw_devs(struct ct_atc *atc) return 0; } -static int atc_get_resources(struct ct_atc *atc) +static int __devinit atc_get_resources(struct ct_atc *atc) { struct daio_desc da_desc = {0}; struct daio_mgr *daio_mgr = NULL; @@ -1423,7 +1423,7 @@ static int atc_get_resources(struct ct_atc *atc) return 0; } -static void +static void __devinit atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, struct src **srcs, struct srcimp **srcimps) { @@ -1462,7 +1462,7 @@ atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, src_mgr->commit_write(src_mgr); /* Synchronously enable SRCs */ } -static void atc_connect_resources(struct ct_atc *atc) +static void __devinit atc_connect_resources(struct ct_atc *atc) { struct dai *dai = NULL; struct dao *dao = NULL; @@ -1508,37 +1508,35 @@ static void atc_connect_resources(struct ct_atc *atc) } } -static void atc_set_ops(struct ct_atc *atc) -{ - /* Set operations */ - atc->map_audio_buffer = ct_map_audio_buffer; - atc->unmap_audio_buffer = ct_unmap_audio_buffer; - atc->pcm_playback_prepare = atc_pcm_playback_prepare; - atc->pcm_release_resources = atc_pcm_release_resources; - atc->pcm_playback_start = atc_pcm_playback_start; - atc->pcm_playback_stop = atc_pcm_stop; - atc->pcm_playback_position = atc_pcm_playback_position; - atc->pcm_capture_prepare = atc_pcm_capture_prepare; - atc->pcm_capture_start = atc_pcm_capture_start; - atc->pcm_capture_stop = atc_pcm_stop; - atc->pcm_capture_position = atc_pcm_capture_position; - atc->spdif_passthru_playback_prepare = spdif_passthru_playback_prepare; - atc->get_ptp_phys = atc_get_ptp_phys; - atc->select_line_in = atc_select_line_in; - atc->select_mic_in = atc_select_mic_in; - atc->select_digit_io = atc_select_digit_io; - atc->line_front_unmute = atc_line_front_unmute; - atc->line_surround_unmute = atc_line_surround_unmute; - atc->line_clfe_unmute = atc_line_clfe_unmute; - atc->line_rear_unmute = atc_line_rear_unmute; - atc->line_in_unmute = atc_line_in_unmute; - atc->spdif_out_unmute = atc_spdif_out_unmute; - atc->spdif_in_unmute = atc_spdif_in_unmute; - atc->spdif_out_get_status = atc_spdif_out_get_status; - atc->spdif_out_set_status = atc_spdif_out_set_status; - atc->spdif_out_passthru = atc_spdif_out_passthru; - atc->have_digit_io_switch = atc_have_digit_io_switch; -} +static struct ct_atc atc_preset __devinitdata = { + .map_audio_buffer = ct_map_audio_buffer, + .unmap_audio_buffer = ct_unmap_audio_buffer, + .pcm_playback_prepare = atc_pcm_playback_prepare, + .pcm_release_resources = atc_pcm_release_resources, + .pcm_playback_start = atc_pcm_playback_start, + .pcm_playback_stop = atc_pcm_stop, + .pcm_playback_position = atc_pcm_playback_position, + .pcm_capture_prepare = atc_pcm_capture_prepare, + .pcm_capture_start = atc_pcm_capture_start, + .pcm_capture_stop = atc_pcm_stop, + .pcm_capture_position = atc_pcm_capture_position, + .spdif_passthru_playback_prepare = spdif_passthru_playback_prepare, + .get_ptp_phys = atc_get_ptp_phys, + .select_line_in = atc_select_line_in, + .select_mic_in = atc_select_mic_in, + .select_digit_io = atc_select_digit_io, + .line_front_unmute = atc_line_front_unmute, + .line_surround_unmute = atc_line_surround_unmute, + .line_clfe_unmute = atc_line_clfe_unmute, + .line_rear_unmute = atc_line_rear_unmute, + .line_in_unmute = atc_line_in_unmute, + .spdif_out_unmute = atc_spdif_out_unmute, + .spdif_in_unmute = atc_spdif_in_unmute, + .spdif_out_get_status = atc_spdif_out_get_status, + .spdif_out_set_status = atc_spdif_out_set_status, + .spdif_out_passthru = atc_spdif_out_passthru, + .have_digit_io_switch = atc_have_digit_io_switch, +}; /** * ct_atc_create - create and initialize a hardware manager @@ -1552,7 +1550,7 @@ static void atc_set_ops(struct ct_atc *atc) * Returns 0 if suceeds, or negative error code if fails. */ -int ct_atc_create(struct snd_card *card, struct pci_dev *pci, +int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, struct ct_atc **ratc) { struct ct_atc *atc = NULL; @@ -1567,14 +1565,14 @@ int ct_atc_create(struct snd_card *card, struct pci_dev *pci, if (NULL == atc) return -ENOMEM; + /* Set operations */ + *atc = atc_preset; + atc->card = card; atc->pci = pci; atc->rsr = rsr; atc->msr = msr; - /* Set operations */ - atc_set_ops(atc); - spin_lock_init(&atc->atc_lock); /* Find card model */ @@ -1606,8 +1604,6 @@ int ct_atc_create(struct snd_card *card, struct pci_dev *pci, if (!atc->timer) goto error1; - atc->create_alsa_devs = ct_create_alsa_devs; - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops); if (err < 0) goto error1; diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index a3f9b1b..04459aa 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -91,8 +91,6 @@ struct ct_atc { const struct ct_atc_chip_details *chip_details; enum CTCARDS model; - /* Create all alsa devices */ - int (*create_alsa_devs)(struct ct_atc *atc); struct ct_vm *vm; /* device virtual memory manager for this card */ int (*map_audio_buffer)(struct ct_atc *atc, struct ct_atc_pcm *apcm); @@ -151,5 +149,6 @@ struct ct_atc { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, struct ct_atc **ratc); +int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); #endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c index 8e58860..53d1aca 100644 --- a/sound/pci/ctxfi/cthardware.c +++ b/sound/pci/ctxfi/cthardware.c @@ -20,7 +20,7 @@ #include "cthw20k2.h" #include -static enum CHIPTYP get_chip_type(struct hw *hw) +static enum CHIPTYP __devinitdata get_chip_type(struct hw *hw) { enum CHIPTYP type = ATCNONE; @@ -39,7 +39,7 @@ static enum CHIPTYP get_chip_type(struct hw *hw) return type; } -int create_hw_obj(struct pci_dev *pci, struct hw **rhw) +int __devinit create_hw_obj(struct pci_dev *pci, struct hw **rhw) { int err = 0; diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 550b30a..df565c1 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -2138,9 +2138,107 @@ static void hw_write_pci(struct hw *hw, u32 reg, u32 data) &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); } -int create_20k1_hw_obj(struct hw **rhw) +static struct hw ct20k1_preset __devinitdata = { + .irq = -1, + + .card_init = hw_card_init, + .card_stop = hw_card_stop, + .pll_init = hw_pll_init, + .is_adc_source_selected = hw_is_adc_input_selected, + .select_adc_source = hw_adc_input_select, + .have_digit_io_switch = hw_have_digit_io_switch, + + .src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk, + .src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk, + .src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk, + .src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk, + .src_set_state = src_set_state, + .src_set_bm = src_set_bm, + .src_set_rsr = src_set_rsr, + .src_set_sf = src_set_sf, + .src_set_wr = src_set_wr, + .src_set_pm = src_set_pm, + .src_set_rom = src_set_rom, + .src_set_vo = src_set_vo, + .src_set_st = src_set_st, + .src_set_ie = src_set_ie, + .src_set_ilsz = src_set_ilsz, + .src_set_bp = src_set_bp, + .src_set_cisz = src_set_cisz, + .src_set_ca = src_set_ca, + .src_set_sa = src_set_sa, + .src_set_la = src_set_la, + .src_set_pitch = src_set_pitch, + .src_set_dirty = src_set_dirty, + .src_set_clear_zbufs = src_set_clear_zbufs, + .src_set_dirty_all = src_set_dirty_all, + .src_commit_write = src_commit_write, + .src_get_ca = src_get_ca, + .src_get_dirty = src_get_dirty, + .src_dirty_conj_mask = src_dirty_conj_mask, + .src_mgr_enbs_src = src_mgr_enbs_src, + .src_mgr_enb_src = src_mgr_enb_src, + .src_mgr_dsb_src = src_mgr_dsb_src, + .src_mgr_commit_write = src_mgr_commit_write, + + .srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk, + .srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk, + .srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc, + .srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser, + .srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt, + .srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr, + .srcimp_mgr_commit_write = srcimp_mgr_commit_write, + + .amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk, + .amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk, + .amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk, + .amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk, + .amixer_set_mode = amixer_set_mode, + .amixer_set_iv = amixer_set_iv, + .amixer_set_x = amixer_set_x, + .amixer_set_y = amixer_set_y, + .amixer_set_sadr = amixer_set_sadr, + .amixer_set_se = amixer_set_se, + .amixer_set_dirty = amixer_set_dirty, + .amixer_set_dirty_all = amixer_set_dirty_all, + .amixer_commit_write = amixer_commit_write, + .amixer_get_y = amixer_get_y, + .amixer_get_dirty = amixer_get_dirty, + + .dai_get_ctrl_blk = dai_get_ctrl_blk, + .dai_put_ctrl_blk = dai_put_ctrl_blk, + .dai_srt_set_srco = dai_srt_set_srcr, + .dai_srt_set_srcm = dai_srt_set_srcl, + .dai_srt_set_rsr = dai_srt_set_rsr, + .dai_srt_set_drat = dai_srt_set_drat, + .dai_srt_set_ec = dai_srt_set_ec, + .dai_srt_set_et = dai_srt_set_et, + .dai_commit_write = dai_commit_write, + + .dao_get_ctrl_blk = dao_get_ctrl_blk, + .dao_put_ctrl_blk = dao_put_ctrl_blk, + .dao_set_spos = dao_set_spos, + .dao_commit_write = dao_commit_write, + .dao_get_spos = dao_get_spos, + + .daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk, + .daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk, + .daio_mgr_enb_dai = daio_mgr_enb_dai, + .daio_mgr_dsb_dai = daio_mgr_dsb_dai, + .daio_mgr_enb_dao = daio_mgr_enb_dao, + .daio_mgr_dsb_dao = daio_mgr_dsb_dao, + .daio_mgr_dao_init = daio_mgr_dao_init, + .daio_mgr_set_imaparc = daio_mgr_set_imaparc, + .daio_mgr_set_imapnxt = daio_mgr_set_imapnxt, + .daio_mgr_set_imapaddr = daio_mgr_set_imapaddr, + .daio_mgr_commit_write = daio_mgr_commit_write, + + .set_timer_irq = set_timer_irq, + .set_timer_tick = set_timer_tick, +}; + +int __devinit create_20k1_hw_obj(struct hw **rhw) { - struct hw *hw; struct hw20k1 *hw20k1; *rhw = NULL; @@ -2151,108 +2249,9 @@ int create_20k1_hw_obj(struct hw **rhw) spin_lock_init(&hw20k1->reg_20k1_lock); spin_lock_init(&hw20k1->reg_pci_lock); - hw = &hw20k1->hw; + hw20k1->hw = ct20k1_preset; - hw->io_base = 0; - hw->mem_base = (unsigned long)NULL; - hw->irq = -1; - - hw->card_init = hw_card_init; - hw->card_stop = hw_card_stop; - hw->pll_init = hw_pll_init; - hw->is_adc_source_selected = hw_is_adc_input_selected; - hw->select_adc_source = hw_adc_input_select; - hw->have_digit_io_switch = hw_have_digit_io_switch; - - hw->src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk; - hw->src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk; - hw->src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk; - hw->src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk; - hw->src_set_state = src_set_state; - hw->src_set_bm = src_set_bm; - hw->src_set_rsr = src_set_rsr; - hw->src_set_sf = src_set_sf; - hw->src_set_wr = src_set_wr; - hw->src_set_pm = src_set_pm; - hw->src_set_rom = src_set_rom; - hw->src_set_vo = src_set_vo; - hw->src_set_st = src_set_st; - hw->src_set_ie = src_set_ie; - hw->src_set_ilsz = src_set_ilsz; - hw->src_set_bp = src_set_bp; - hw->src_set_cisz = src_set_cisz; - hw->src_set_ca = src_set_ca; - hw->src_set_sa = src_set_sa; - hw->src_set_la = src_set_la; - hw->src_set_pitch = src_set_pitch; - hw->src_set_dirty = src_set_dirty; - hw->src_set_clear_zbufs = src_set_clear_zbufs; - hw->src_set_dirty_all = src_set_dirty_all; - hw->src_commit_write = src_commit_write; - hw->src_get_ca = src_get_ca; - hw->src_get_dirty = src_get_dirty; - hw->src_dirty_conj_mask = src_dirty_conj_mask; - hw->src_mgr_enbs_src = src_mgr_enbs_src; - hw->src_mgr_enb_src = src_mgr_enb_src; - hw->src_mgr_dsb_src = src_mgr_dsb_src; - hw->src_mgr_commit_write = src_mgr_commit_write; - - hw->srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk; - hw->srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk; - hw->srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc; - hw->srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser; - hw->srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt; - hw->srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr; - hw->srcimp_mgr_commit_write = srcimp_mgr_commit_write; - - hw->amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk; - hw->amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk; - hw->amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk; - hw->amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk; - hw->amixer_set_mode = amixer_set_mode; - hw->amixer_set_iv = amixer_set_iv; - hw->amixer_set_x = amixer_set_x; - hw->amixer_set_y = amixer_set_y; - hw->amixer_set_sadr = amixer_set_sadr; - hw->amixer_set_se = amixer_set_se; - hw->amixer_set_dirty = amixer_set_dirty; - hw->amixer_set_dirty_all = amixer_set_dirty_all; - hw->amixer_commit_write = amixer_commit_write; - hw->amixer_get_y = amixer_get_y; - hw->amixer_get_dirty = amixer_get_dirty; - - hw->dai_get_ctrl_blk = dai_get_ctrl_blk; - hw->dai_put_ctrl_blk = dai_put_ctrl_blk; - hw->dai_srt_set_srco = dai_srt_set_srcr; - hw->dai_srt_set_srcm = dai_srt_set_srcl; - hw->dai_srt_set_rsr = dai_srt_set_rsr; - hw->dai_srt_set_drat = dai_srt_set_drat; - hw->dai_srt_set_ec = dai_srt_set_ec; - hw->dai_srt_set_et = dai_srt_set_et; - hw->dai_commit_write = dai_commit_write; - - hw->dao_get_ctrl_blk = dao_get_ctrl_blk; - hw->dao_put_ctrl_blk = dao_put_ctrl_blk; - hw->dao_set_spos = dao_set_spos; - hw->dao_commit_write = dao_commit_write; - hw->dao_get_spos = dao_get_spos; - - hw->daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk; - hw->daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk; - hw->daio_mgr_enb_dai = daio_mgr_enb_dai; - hw->daio_mgr_dsb_dai = daio_mgr_dsb_dai; - hw->daio_mgr_enb_dao = daio_mgr_enb_dao; - hw->daio_mgr_dsb_dao = daio_mgr_dsb_dao; - hw->daio_mgr_dao_init = daio_mgr_dao_init; - hw->daio_mgr_set_imaparc = daio_mgr_set_imaparc; - hw->daio_mgr_set_imapnxt = daio_mgr_set_imapnxt; - hw->daio_mgr_set_imapaddr = daio_mgr_set_imapaddr; - hw->daio_mgr_commit_write = daio_mgr_commit_write; - - hw->set_timer_irq = set_timer_irq; - hw->set_timer_tick = set_timer_tick; - - *rhw = hw; + *rhw = &hw20k1->hw; return 0; } diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 3497287..041199f 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -2006,7 +2006,103 @@ static void hw_write_20kx(struct hw *hw, u32 reg, u32 data) writel(data, (void *)(hw->mem_base + reg)); } -int create_20k2_hw_obj(struct hw **rhw) +static struct hw ct20k2_preset __devinitdata = { + .irq = -1, + + .card_init = hw_card_init, + .card_stop = hw_card_stop, + .pll_init = hw_pll_init, + .is_adc_source_selected = hw_is_adc_input_selected, + .select_adc_source = hw_adc_input_select, + .have_digit_io_switch = hw_have_digit_io_switch, + + .src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk, + .src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk, + .src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk, + .src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk, + .src_set_state = src_set_state, + .src_set_bm = src_set_bm, + .src_set_rsr = src_set_rsr, + .src_set_sf = src_set_sf, + .src_set_wr = src_set_wr, + .src_set_pm = src_set_pm, + .src_set_rom = src_set_rom, + .src_set_vo = src_set_vo, + .src_set_st = src_set_st, + .src_set_ie = src_set_ie, + .src_set_ilsz = src_set_ilsz, + .src_set_bp = src_set_bp, + .src_set_cisz = src_set_cisz, + .src_set_ca = src_set_ca, + .src_set_sa = src_set_sa, + .src_set_la = src_set_la, + .src_set_pitch = src_set_pitch, + .src_set_dirty = src_set_dirty, + .src_set_clear_zbufs = src_set_clear_zbufs, + .src_set_dirty_all = src_set_dirty_all, + .src_commit_write = src_commit_write, + .src_get_ca = src_get_ca, + .src_get_dirty = src_get_dirty, + .src_dirty_conj_mask = src_dirty_conj_mask, + .src_mgr_enbs_src = src_mgr_enbs_src, + .src_mgr_enb_src = src_mgr_enb_src, + .src_mgr_dsb_src = src_mgr_dsb_src, + .src_mgr_commit_write = src_mgr_commit_write, + + .srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk, + .srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk, + .srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc, + .srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser, + .srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt, + .srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr, + .srcimp_mgr_commit_write = srcimp_mgr_commit_write, + + .amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk, + .amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk, + .amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk, + .amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk, + .amixer_set_mode = amixer_set_mode, + .amixer_set_iv = amixer_set_iv, + .amixer_set_x = amixer_set_x, + .amixer_set_y = amixer_set_y, + .amixer_set_sadr = amixer_set_sadr, + .amixer_set_se = amixer_set_se, + .amixer_set_dirty = amixer_set_dirty, + .amixer_set_dirty_all = amixer_set_dirty_all, + .amixer_commit_write = amixer_commit_write, + .amixer_get_y = amixer_get_y, + .amixer_get_dirty = amixer_get_dirty, + + .dai_get_ctrl_blk = dai_get_ctrl_blk, + .dai_put_ctrl_blk = dai_put_ctrl_blk, + .dai_srt_set_srco = dai_srt_set_srco, + .dai_srt_set_srcm = dai_srt_set_srcm, + .dai_srt_set_rsr = dai_srt_set_rsr, + .dai_srt_set_drat = dai_srt_set_drat, + .dai_srt_set_ec = dai_srt_set_ec, + .dai_srt_set_et = dai_srt_set_et, + .dai_commit_write = dai_commit_write, + + .dao_get_ctrl_blk = dao_get_ctrl_blk, + .dao_put_ctrl_blk = dao_put_ctrl_blk, + .dao_set_spos = dao_set_spos, + .dao_commit_write = dao_commit_write, + .dao_get_spos = dao_get_spos, + + .daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk, + .daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk, + .daio_mgr_enb_dai = daio_mgr_enb_dai, + .daio_mgr_dsb_dai = daio_mgr_dsb_dai, + .daio_mgr_enb_dao = daio_mgr_enb_dao, + .daio_mgr_dsb_dao = daio_mgr_dsb_dao, + .daio_mgr_dao_init = daio_mgr_dao_init, + .daio_mgr_set_imaparc = daio_mgr_set_imaparc, + .daio_mgr_set_imapnxt = daio_mgr_set_imapnxt, + .daio_mgr_set_imapaddr = daio_mgr_set_imapaddr, + .daio_mgr_commit_write = daio_mgr_commit_write, +}; + +int __devinit create_20k2_hw_obj(struct hw **rhw) { struct hw *hw; @@ -2015,102 +2111,7 @@ int create_20k2_hw_obj(struct hw **rhw) if (NULL == hw) return -ENOMEM; - hw->io_base = 0; - hw->mem_base = (unsigned long)NULL; - hw->irq = -1; - - hw->card_init = hw_card_init; - hw->card_stop = hw_card_stop; - hw->pll_init = hw_pll_init; - hw->is_adc_source_selected = hw_is_adc_input_selected; - hw->select_adc_source = hw_adc_input_select; - hw->have_digit_io_switch = hw_have_digit_io_switch; - - hw->src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk; - hw->src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk; - hw->src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk; - hw->src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk; - hw->src_set_state = src_set_state; - hw->src_set_bm = src_set_bm; - hw->src_set_rsr = src_set_rsr; - hw->src_set_sf = src_set_sf; - hw->src_set_wr = src_set_wr; - hw->src_set_pm = src_set_pm; - hw->src_set_rom = src_set_rom; - hw->src_set_vo = src_set_vo; - hw->src_set_st = src_set_st; - hw->src_set_ie = src_set_ie; - hw->src_set_ilsz = src_set_ilsz; - hw->src_set_bp = src_set_bp; - hw->src_set_cisz = src_set_cisz; - hw->src_set_ca = src_set_ca; - hw->src_set_sa = src_set_sa; - hw->src_set_la = src_set_la; - hw->src_set_pitch = src_set_pitch; - hw->src_set_dirty = src_set_dirty; - hw->src_set_clear_zbufs = src_set_clear_zbufs; - hw->src_set_dirty_all = src_set_dirty_all; - hw->src_commit_write = src_commit_write; - hw->src_get_ca = src_get_ca; - hw->src_get_dirty = src_get_dirty; - hw->src_dirty_conj_mask = src_dirty_conj_mask; - hw->src_mgr_enbs_src = src_mgr_enbs_src; - hw->src_mgr_enb_src = src_mgr_enb_src; - hw->src_mgr_dsb_src = src_mgr_dsb_src; - hw->src_mgr_commit_write = src_mgr_commit_write; - - hw->srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk; - hw->srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk; - hw->srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc; - hw->srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser; - hw->srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt; - hw->srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr; - hw->srcimp_mgr_commit_write = srcimp_mgr_commit_write; - - hw->amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk; - hw->amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk; - hw->amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk; - hw->amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk; - hw->amixer_set_mode = amixer_set_mode; - hw->amixer_set_iv = amixer_set_iv; - hw->amixer_set_x = amixer_set_x; - hw->amixer_set_y = amixer_set_y; - hw->amixer_set_sadr = amixer_set_sadr; - hw->amixer_set_se = amixer_set_se; - hw->amixer_set_dirty = amixer_set_dirty; - hw->amixer_set_dirty_all = amixer_set_dirty_all; - hw->amixer_commit_write = amixer_commit_write; - hw->amixer_get_y = amixer_get_y; - hw->amixer_get_dirty = amixer_get_dirty; - - hw->dai_get_ctrl_blk = dai_get_ctrl_blk; - hw->dai_put_ctrl_blk = dai_put_ctrl_blk; - hw->dai_srt_set_srco = dai_srt_set_srco; - hw->dai_srt_set_srcm = dai_srt_set_srcm; - hw->dai_srt_set_rsr = dai_srt_set_rsr; - hw->dai_srt_set_drat = dai_srt_set_drat; - hw->dai_srt_set_ec = dai_srt_set_ec; - hw->dai_srt_set_et = dai_srt_set_et; - hw->dai_commit_write = dai_commit_write; - - hw->dao_get_ctrl_blk = dao_get_ctrl_blk; - hw->dao_put_ctrl_blk = dao_put_ctrl_blk; - hw->dao_set_spos = dao_set_spos; - hw->dao_commit_write = dao_commit_write; - hw->dao_get_spos = dao_get_spos; - - hw->daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk; - hw->daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk; - hw->daio_mgr_enb_dai = daio_mgr_enb_dai; - hw->daio_mgr_dsb_dai = daio_mgr_dsb_dai; - hw->daio_mgr_enb_dao = daio_mgr_enb_dao; - hw->daio_mgr_dsb_dao = daio_mgr_dsb_dao; - hw->daio_mgr_dao_init = daio_mgr_dao_init; - hw->daio_mgr_set_imaparc = daio_mgr_set_imaparc; - hw->daio_mgr_set_imapnxt = daio_mgr_set_imapnxt; - hw->daio_mgr_set_imapaddr = daio_mgr_set_imapaddr; - hw->daio_mgr_commit_write = daio_mgr_commit_write; - + *hw = ct20k2_preset; *rhw = hw; return 0; diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index bf232e7..279dac6 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -86,7 +86,7 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) card->private_data = atc; /* Create alsa devices supported by this card */ - err = atc->create_alsa_devs(atc); + err = ct_atc_create_alsa_devs(atc); if (err < 0) goto error; -- cgit v1.1 From 032abb519c23000f8a0fae78bb460047f9129270 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 16:37:19 +0200 Subject: ALSA: ctxfi - Set device 0 for mixer control elements Mixer control elements are usually assigned to device 0. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctmixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 177c46e..fac783f 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -1097,7 +1097,7 @@ int ct_alsa_mix_create(struct ct_atc *atc, int err = 0; /* Create snd kcontrol instances on demand */ - vol_ctl.device = swh_ctl.device = device; + /* vol_ctl.device = swh_ctl.device = device; */ /* better w/ device 0 */ err = ct_mixer_kcontrols_create((struct ct_mixer *)atc->mixer); if (err) return err; -- cgit v1.1 From 3f7440a6b771169e1f11fa582e53a4259b682809 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 17:40:04 +0200 Subject: ALSA: Clean up 64bit division functions Replace the house-made div64_32() with the standard div_u64*() functions. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 74 ----------------------------------------------- sound/core/oss/pcm_oss.c | 5 ++-- sound/core/pcm_lib.c | 3 +- sound/pci/rme9652/hdsp.c | 7 ++--- sound/pci/rme9652/hdspm.c | 4 +-- 5 files changed, 9 insertions(+), 84 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c172968..0caf71e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -486,80 +486,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream); void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); -#if BITS_PER_LONG >= 64 - -static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem) -{ - *rem = *n % div; - *n /= div; -} - -#elif defined(i386) - -static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem) -{ - u_int32_t low, high; - low = *n & 0xffffffff; - high = *n >> 32; - if (high) { - u_int32_t high1 = high % div; - high /= div; - asm("divl %2":"=a" (low), "=d" (*rem):"rm" (div), "a" (low), "d" (high1)); - *n = (u_int64_t)high << 32 | low; - } else { - *n = low / div; - *rem = low % div; - } -} -#else - -static inline void divl(u_int32_t high, u_int32_t low, - u_int32_t div, - u_int32_t *q, u_int32_t *r) -{ - u_int64_t n = (u_int64_t)high << 32 | low; - u_int64_t d = (u_int64_t)div << 31; - u_int32_t q1 = 0; - int c = 32; - while (n > 0xffffffffU) { - q1 <<= 1; - if (n >= d) { - n -= d; - q1 |= 1; - } - d >>= 1; - c--; - } - q1 <<= c; - if (n) { - low = n; - *q = q1 | (low / div); - *r = low % div; - } else { - *r = 0; - *q = q1; - } - return; -} - -static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem) -{ - u_int32_t low, high; - low = *n & 0xffffffff; - high = *n >> 32; - if (high) { - u_int32_t high1 = high % div; - u_int32_t low1 = low; - high /= div; - divl(high1, low1, div, &low, rem); - *n = (u_int64_t)high << 32 | low; - } else { - *n = low / div; - *rem = low % div; - } -} -#endif - /* * PCM library */ diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index dda000b..dbe406b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -31,6 +31,7 @@ #include #include #include +#include #include #include #include @@ -617,9 +618,7 @@ static long snd_pcm_oss_bytes(struct snd_pcm_substream *substream, long frames) #else { u64 bsize = (u64)runtime->oss.buffer_bytes * (u64)bytes; - u32 rem; - div64_32(&bsize, buffer_size, &rem); - return (long)bsize; + return div_u64(bsize, buffer_size); } #endif } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d659995..a748287 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -22,6 +22,7 @@ #include #include +#include #include #include #include @@ -452,7 +453,7 @@ static inline unsigned int muldiv32(unsigned int a, unsigned int b, *r = 0; return UINT_MAX; } - div64_32(&n, c, r); + n = div_u64_rem(n, c, r); if (n >= UINT_MAX) { *r = 0; return UINT_MAX; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 314e735..bcfdbb5 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include @@ -1047,7 +1048,6 @@ static int hdsp_set_interrupt_interval(struct hdsp *s, unsigned int frames) static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) { u64 n; - u32 r; if (rate >= 112000) rate /= 4; @@ -1055,7 +1055,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) rate /= 2; n = DDS_NUMERATOR; - div64_32(&n, rate, &r); + n = div_u64(n, rate); /* n should be less than 2^32 for being written to FREQ register */ snd_BUG_ON(n >> 32); /* HDSP_freqReg and HDSP_resetPointer are the same, so keep the DDS @@ -3097,7 +3097,6 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn static int hdsp_dds_offset(struct hdsp *hdsp) { u64 n; - u32 r; unsigned int dds_value = hdsp->dds_value; int system_sample_rate = hdsp->system_sample_rate; @@ -3109,7 +3108,7 @@ static int hdsp_dds_offset(struct hdsp *hdsp) * dds_value = n / rate * rate = n / dds_value */ - div64_32(&n, dds_value, &r); + n = div_u64(n, dds_value); if (system_sample_rate >= 112000) n *= 4; else if (system_sample_rate >= 56000) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bac2dc0..0dce331 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include @@ -831,7 +832,6 @@ static int hdspm_set_interrupt_interval(struct hdspm * s, unsigned int frames) static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) { u64 n; - u32 r; if (rate >= 112000) rate /= 4; @@ -844,7 +844,7 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate) */ /* n = 104857600000000ULL; */ /* = 2^20 * 10^8 */ n = 110100480000000ULL; /* Value checked for AES32 and MADI */ - div64_32(&n, rate, &r); + n = div_u64(n, rate); /* n should be less than 2^32 for being written to FREQ register */ snd_BUG_ON(n >> 32); hdspm_write(hdspm, HDSPM_freqReg, (u32)n); -- cgit v1.1 From 28cd4aa43de2b6d3b1e3385d450bfb31cbe8d72a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 17:58:00 +0200 Subject: ALSA: ctxfi - Add missing inclusion of linux/math64.h Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cttimer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 3acb26d..ceda74e 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -7,6 +7,7 @@ */ #include +#include #include #include #include "ctatc.h" -- cgit v1.1 From ccff4b15e0847223de0a481f5b7fa5ef902cf3bd Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Fri, 5 Jun 2009 19:15:58 -0700 Subject: ASoC: codec tlv320aic23 fix bogus divide by 0 message Some code analyzer software mistakenly gives divide by 0 error messages for these lines. This patch will end its confusion. Signed-off-by: Troy Kisky Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9fcbb9c..0b8dcb5 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = { * Every divisor is a factor of 11*12 */ #define SR_MULT (11*12) -#define A(x) (x) ? (SR_MULT/x) : 0 +#define A(x) (SR_MULT/x) static const unsigned char sr_adc_mult_table[] = { - A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), - A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) + A(2), A(2), A(12), A(12), 0, 0, A(3), A(1), + A(2), A(2), A(11), A(11), 0, 0, 0, A(1) }; static const unsigned char sr_dac_mult_table[] = { - A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), - A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) + A(2), A(12), A(2), A(12), 0, 0, A(3), A(1), + A(2), A(11), A(2), A(11), 0, 0, 0, A(1) }; static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, -- cgit v1.1 From 74b8f955a73d20b1e22403fd1ef85834fbf38d98 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 6 Jun 2009 11:26:15 +0100 Subject: ASoC: Apostrophe patrol Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 39a63f9..21c6907 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -12,7 +12,7 @@ * Features: * o Changes power status of internal codec blocks depending on the * dynamic configuration of codec internal audio paths and active - * DAC's/ADC's. + * DACs/ADCs. * o Platform power domain - can support external components i.e. amps and * mic/meadphone insertion events. * o Automatic Mic Bias support @@ -220,7 +220,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } } -/* connect mux widget to it's interconnecting audio paths */ +/* connect mux widget to its interconnecting audio paths */ static int dapm_connect_mux(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, struct snd_soc_dapm_path *path, const char *control_name, @@ -243,7 +243,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, return -ENODEV; } -/* connect mixer widget to it's interconnecting audio paths */ +/* connect mixer widget to its interconnecting audio paths */ static int dapm_connect_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, struct snd_soc_dapm_path *path, const char *control_name) @@ -1797,7 +1797,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); * @codec: SoC codec * @pin: pin name * - * Enables input/output pin and it's parents or children widgets iff there is + * Enables input/output pin and its parents or children widgets iff there is * a valid audio route and active audio stream. * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. @@ -1813,7 +1813,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * @codec: SoC codec * @pin: pin name * - * Disables input/output pin and it's parents or children widgets. + * Disables input/output pin and its parents or children widgets. * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -- cgit v1.1 From d86bf92313bfd47885a92c7de63bde392d585f95 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sat, 6 Jun 2009 18:32:06 +0200 Subject: ALSA: pcm - Fix a typo in hw_ptr update check Fix a typo in the commit 13f040f9e55d41e92e485389123654971e03b819 ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed which causes obvious problems with PA. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index dd9126b..bf34603 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -371,7 +371,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); - if (runtime->status->hw_ptr != new_hw_ptr) + if (runtime->status->hw_ptr == new_hw_ptr) return 0; runtime->hw_ptr_base = hw_base; -- cgit v1.1 From ad0b0822f98ef547e2461ce463e4233bad7848a8 Mon Sep 17 00:00:00 2001 From: "Figo.zhang" Date: Sun, 7 Jun 2009 13:37:27 +0800 Subject: ALSA: sgio2audio.c: clean up checking vfree() does it's own 'NULL' check,so no need for check before calling it. Signed-off-by: Figo.zhang Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 66f3b48..e497525 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -619,8 +619,7 @@ static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, /* hw_free callback */ static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) { - if (substream->runtime->dma_area) - vfree(substream->runtime->dma_area); + vfree(substream->runtime->dma_area); substream->runtime->dma_area = NULL; return 0; } -- cgit v1.1 From ab1863fc9bc18c806338564124b1e5e7e3ef53d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 7 Jun 2009 12:09:17 +0200 Subject: ALSA: pcm - Fix update of runtime->hw_ptr_interrupt The commit 13f040f9e55d41e92e485389123654971e03b819 made another regression, the missing update of runtime->hw_ptr_interrupt. Since this field is only checked in snd_pcmupdate__hw_ptr_interrupt(), not in snd_pcm_update_hw_ptr(), it must be updated before the hw_ptr change check. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index bf34603..adb306f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -299,6 +299,8 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size; } + runtime->hw_ptr_interrupt = hw_ptr_interrupt; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); @@ -309,7 +311,6 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_interrupt = hw_ptr_interrupt; if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); -- cgit v1.1 From 54de6bc8b2437f642844cecb8d183df2368ffceb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 10:21:07 +0200 Subject: ALSA: ctxfi - Optimize the native timer handling using wc counter Optimize the timer update routine to look up wall clock once instead of checking the position of each stream at each timer update. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthardware.h | 1 + sound/pci/ctxfi/cthw20k1.c | 6 ++++ sound/pci/ctxfi/cttimer.c | 76 ++++++++++++++++++++++++++++---------------- 3 files changed, 55 insertions(+), 28 deletions(-) diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 35350cf..8f11644 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -147,6 +147,7 @@ struct hw { int (*set_timer_irq)(struct hw *hw, int enable); int (*set_timer_tick)(struct hw *hw, unsigned int tick); + unsigned int (*get_wc)(struct hw *hw); void (*irq_callback)(void *data, unsigned int bit); void *irq_callback_data; diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index df565c1..b165466 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1186,6 +1186,11 @@ static int set_timer_tick(struct hw *hw, unsigned int ticks) return 0; } +static unsigned int get_wc(struct hw *hw) +{ + return hw_read_20kx(hw, WC); +} + /* Card hardware initialization block */ struct dac_conf { unsigned int msr; /* master sample rate in rsrs */ @@ -2235,6 +2240,7 @@ static struct hw ct20k1_preset __devinitdata = { .set_timer_irq = set_timer_irq, .set_timer_tick = set_timer_tick, + .get_wc = get_wc, }; int __devinit create_20k1_hw_obj(struct hw **rhw) diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index ceda74e..ec869a4 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -47,6 +47,7 @@ struct ct_timer { struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; + unsigned int wc; /* current wallclock */ unsigned int irq_handling:1; /* in IRQ handling */ unsigned int reprogram:1; /* need to reprogram the internval */ unsigned int running:1; /* global timer running */ @@ -136,6 +137,7 @@ static struct ct_timer_ops ct_systimer_ops = { */ #define CT_TIMER_FREQ 48000 +#define MIN_TICKS 1 #define MAX_TICKS ((1 << 13) - 1) static void ct_xfitimer_irq_rearm(struct ct_timer *atimer, int ticks) @@ -159,6 +161,12 @@ static void ct_xfitimer_irq_stop(struct ct_timer *atimer) } } +static inline unsigned int ct_xfitimer_get_wc(struct ct_timer *atimer) +{ + struct hw *hw = atimer->atc->hw; + return hw->get_wc(hw); +} + /* * reprogram the timer interval; * checks the running instance list and determines the next timer interval. @@ -170,37 +178,46 @@ static void ct_xfitimer_irq_stop(struct ct_timer *atimer) static int ct_xfitimer_reprogram(struct ct_timer *atimer) { struct ct_timer_instance *ti; - int min_intr = -1; + unsigned int min_intr = (unsigned int)-1; int updates = 0; + unsigned int wc, diff; + if (list_empty(&atimer->running_head)) { + ct_xfitimer_irq_stop(atimer); + atimer->reprogram = 0; /* clear flag */ + return 0; + } + + wc = ct_xfitimer_get_wc(atimer); + diff = wc - atimer->wc; + atimer->wc = wc; list_for_each_entry(ti, &atimer->running_head, running_list) { - struct snd_pcm_runtime *runtime; - unsigned int pos, diff; - int intr; - runtime = ti->substream->runtime; - pos = ti->substream->ops->pointer(ti->substream); - if (pos < ti->position) - diff = runtime->buffer_size - ti->position + pos; - else - diff = pos - ti->position; - ti->position = pos; - while (diff >= ti->frag_count) { - ti->frag_count += runtime->period_size; - ti->need_update = 1; - updates++; + if (ti->frag_count > diff) + ti->frag_count -= diff; + else { + unsigned int pos; + unsigned int period_size, rate; + + period_size = ti->substream->runtime->period_size; + rate = ti->substream->runtime->rate; + pos = ti->substream->ops->pointer(ti->substream); + if (pos / period_size != ti->position / period_size) { + ti->need_update = 1; + ti->position = pos; + updates++; + } + pos %= period_size; + pos = period_size - pos; + ti->frag_count = div_u64((u64)pos * CT_TIMER_FREQ + + rate - 1, rate); } - ti->frag_count -= diff; - intr = div_u64((u64)ti->frag_count * CT_TIMER_FREQ, - runtime->rate); - if (min_intr < 0 || intr < min_intr) - min_intr = intr; + if (ti->frag_count < min_intr) + min_intr = ti->frag_count; } - if (min_intr > 0) - ct_xfitimer_irq_rearm(atimer, min_intr); - else - ct_xfitimer_irq_stop(atimer); - + if (min_intr < MIN_TICKS) + min_intr = MIN_TICKS; + ct_xfitimer_irq_rearm(atimer, min_intr); atimer->reprogram = 0; /* clear flag */ return updates; } @@ -253,13 +270,14 @@ static void ct_xfitimer_update(struct ct_timer *atimer) unsigned long flags; int update; + spin_lock_irqsave(&atimer->lock, flags); if (atimer->irq_handling) { /* reached from IRQ handler; let it handle later */ atimer->reprogram = 1; + spin_unlock_irqrestore(&atimer->lock, flags); return; } - spin_lock_irqsave(&atimer->lock, flags); ct_xfitimer_irq_stop(atimer); update = ct_xfitimer_reprogram(atimer); spin_unlock_irqrestore(&atimer->lock, flags); @@ -273,6 +291,8 @@ static void ct_xfitimer_start(struct ct_timer_instance *ti) unsigned long flags; spin_lock_irqsave(&atimer->lock, flags); + if (list_empty(&ti->running_list)) + atimer->wc = ct_xfitimer_get_wc(atimer); list_add(&ti->running_list, &atimer->running_head); spin_unlock_irqrestore(&atimer->lock, flags); ct_xfitimer_update(atimer); @@ -396,12 +416,12 @@ struct ct_timer *ct_timer_new(struct ct_atc *atc) atimer->atc = atc; hw = atc->hw; if (!USE_SYSTEM_TIMER && hw->set_timer_irq) { - printk(KERN_INFO "ctxfi: Use xfi-native timer\n"); + snd_printd(KERN_INFO "ctxfi: Use xfi-native timer\n"); atimer->ops = &ct_xfitimer_ops; hw->irq_callback_data = atimer; hw->irq_callback = ct_timer_interrupt; } else { - printk(KERN_INFO "ctxfi: Use system timer\n"); + snd_printd(KERN_INFO "ctxfi: Use system timer\n"); atimer->ops = &ct_systimer_ops; } return atimer; -- cgit v1.1 From c4865679df5453d1bc94c534f4d94a364c0546df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 12:57:17 +0200 Subject: ALSA: ca0106 - Fix master volume scale The master volume dB scale was wrongly defined as 0.50dB setp while it must be 0.25dB step. Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 7724252..c8c6f43 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -739,7 +739,7 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch } while (0) static __devinitdata -DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 50, 1); +DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1); static char *slave_vols[] __devinitdata = { "Analog Front Playback Volume", -- cgit v1.1 From d436dd063be605dc29f17b2cb0b99a852db89bed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 13:50:18 +0200 Subject: ALSA: ctxfi - Make volume controls more intuitive Change the volume control to dB scale (as the raw data seems so). Also added the TLV dB-scale information. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctmixer.c | 49 +++++++++++++++++++++++++++++++---------------- 1 file changed, 32 insertions(+), 17 deletions(-) diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index fac783f..796156e 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -18,11 +18,12 @@ #include "ctmixer.h" #include "ctamixer.h" +#include #include #include #include #include -#include +#include enum CT_SUM_CTL { SUM_IN_F, @@ -292,6 +293,7 @@ set_switch_state(struct ct_mixer *mixer, mixer->switch_state &= ~(0x1 << (type - SWH_MIXER_START)); } +#if 0 /* not used */ /* Map integer value ranging from 0 to 65535 to 14-bit float value ranging * from 2^-6 to (1+1023/1024) */ static unsigned int uint16_to_float14(unsigned int x) @@ -331,6 +333,12 @@ static unsigned int float14_to_uint16(unsigned int x) return x; } +#endif /* not used */ + +#define VOL_SCALE 0x1c +#define VOL_MAX 0x100 + +static const DECLARE_TLV_DB_SCALE(ct_vol_db_scale, -6400, 25, 1); static int ct_alsa_mix_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -338,8 +346,7 @@ static int ct_alsa_mix_volume_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 43690; - uinfo->value.integer.step = 128; + uinfo->value.integer.max = VOL_MAX; return 0; } @@ -349,15 +356,18 @@ static int ct_alsa_mix_volume_get(struct snd_kcontrol *kcontrol, { struct ct_atc *atc = snd_kcontrol_chip(kcontrol); enum CT_AMIXER_CTL type = get_amixer_index(kcontrol->private_value); - struct amixer *amixer = NULL; - int i = 0; + struct amixer *amixer; + int i, val; for (i = 0; i < 2; i++) { amixer = ((struct ct_mixer *)atc->mixer)-> amixers[type*CHN_NUM+i]; - /* Convert 14-bit float-point scale to 16-bit integer volume */ - ucontrol->value.integer.value[i] = - (float14_to_uint16(amixer->ops->get_scale(amixer)) & 0xffff); + val = amixer->ops->get_scale(amixer) / VOL_SCALE; + if (val < 0) + val = 0; + else if (val > VOL_MAX) + val = VOL_MAX; + ucontrol->value.integer.value[i] = val; } return 0; @@ -369,16 +379,19 @@ static int ct_alsa_mix_volume_put(struct snd_kcontrol *kcontrol, struct ct_atc *atc = snd_kcontrol_chip(kcontrol); struct ct_mixer *mixer = atc->mixer; enum CT_AMIXER_CTL type = get_amixer_index(kcontrol->private_value); - struct amixer *amixer = NULL; - int i = 0, j = 0, change = 0, val = 0; + struct amixer *amixer; + int i, j, val, oval, change = 0; for (i = 0; i < 2; i++) { - /* Convert 16-bit integer volume to 14-bit float-point scale */ - val = (ucontrol->value.integer.value[i] & 0xffff); + val = ucontrol->value.integer.value[i]; + if (val < 0) + val = 0; + else if (val > VOL_MAX) + val = VOL_MAX; + val *= VOL_SCALE; amixer = mixer->amixers[type*CHN_NUM+i]; - if ((float14_to_uint16(amixer->ops->get_scale(amixer)) & 0xff80) - != (val & 0xff80)) { - val = uint16_to_float14(val); + oval = amixer->ops->get_scale(amixer); + if (val != oval) { amixer->ops->set_scale(amixer, val); amixer->ops->commit_write(amixer); change = 1; @@ -398,11 +411,13 @@ static int ct_alsa_mix_volume_put(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new vol_ctl = { - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = ct_alsa_mix_volume_info, .get = ct_alsa_mix_volume_get, - .put = ct_alsa_mix_volume_put + .put = ct_alsa_mix_volume_put, + .tlv = { .p = ct_vol_db_scale }, }; static void -- cgit v1.1 From 4836ac655410e7f126d316b0be062b38746f7529 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 14:49:26 +0200 Subject: ALSA: ctxfi - Fix DMA mask for emu20k2 chip Allow 64bit DMA mask for emu20k2 chip, too. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k2.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 041199f..edbfb48 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -26,7 +26,11 @@ #include #include -#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bits */ +#if BITS_PER_LONG == 32 +#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */ +#else +#define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */ +#endif static u32 hw_read_20kx(struct hw *hw, u32 reg); static void hw_write_20kx(struct hw *hw, u32 reg, u32 data); @@ -1834,18 +1838,16 @@ static int hw_card_start(struct hw *hw) int err = 0; struct pci_dev *pci = hw->pci; unsigned int gctl; - unsigned int dma_mask = 0; err = pci_enable_device(pci); if (err < 0) return err; /* Set DMA transfer mask */ - dma_mask = CT_XFI_DMA_MASK; - if (pci_set_dma_mask(pci, dma_mask) < 0 || - pci_set_consistent_dma_mask(pci, dma_mask) < 0) { + if (pci_set_dma_mask(pci, CT_XFI_DMA_MASK) < 0 || + pci_set_consistent_dma_mask(pci, CT_XFI_DMA_MASK) < 0) { printk(KERN_ERR "ctxfi: architecture does not support PCI " - "busmaster DMA with mask 0x%x\n", dma_mask); + "busmaster DMA with mask 0x%llx\n", CT_XFI_DMA_MASK); err = -ENXIO; goto error1; } -- cgit v1.1 From 514eef9c2a711b4c24b97bb456d39695a6fe1775 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 14:57:57 +0200 Subject: ALSA: ctxfi - Remove useless initializations and cast Remove useless variable initializations and cast at the beginning of functions. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctamixer.c | 62 ++++++++-------- sound/pci/ctxfi/ctatc.c | 164 ++++++++++++++++++++++--------------------- sound/pci/ctxfi/ctdaio.c | 40 +++++------ sound/pci/ctxfi/cthardware.c | 6 +- sound/pci/ctxfi/cthw20k1.c | 82 +++++++++++----------- sound/pci/ctxfi/cthw20k2.c | 52 +++++++------- sound/pci/ctxfi/ctimap.c | 4 +- sound/pci/ctxfi/ctmixer.c | 60 ++++++++-------- sound/pci/ctxfi/ctresource.c | 6 +- sound/pci/ctxfi/ctsrc.c | 110 ++++++++++++++--------------- sound/pci/ctxfi/ctvmem.c | 12 ++-- 11 files changed, 301 insertions(+), 297 deletions(-) diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index 859e996..a1db51b3 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -58,9 +58,9 @@ static struct rsc_ops amixer_basic_rsc_ops = { static int amixer_set_input(struct amixer *amixer, struct rsc *rsc) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; hw->amixer_set_mode(amixer->rsc.ctrl_blk, AMIXER_Y_IMMEDIATE); amixer->input = rsc; if (NULL == rsc) @@ -75,9 +75,9 @@ static int amixer_set_input(struct amixer *amixer, struct rsc *rsc) /* y is a 14-bit immediate constant */ static int amixer_set_y(struct amixer *amixer, unsigned int y) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; hw->amixer_set_y(amixer->rsc.ctrl_blk, y); return 0; @@ -85,9 +85,9 @@ static int amixer_set_y(struct amixer *amixer, unsigned int y) static int amixer_set_invalid_squash(struct amixer *amixer, unsigned int iv) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; hw->amixer_set_iv(amixer->rsc.ctrl_blk, iv); return 0; @@ -95,9 +95,9 @@ static int amixer_set_invalid_squash(struct amixer *amixer, unsigned int iv) static int amixer_set_sum(struct amixer *amixer, struct sum *sum) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; amixer->sum = sum; if (NULL == sum) { hw->amixer_set_se(amixer->rsc.ctrl_blk, 0); @@ -112,13 +112,13 @@ static int amixer_set_sum(struct amixer *amixer, struct sum *sum) static int amixer_commit_write(struct amixer *amixer) { - struct hw *hw = NULL; - unsigned int index = 0; - int i = 0; - struct rsc *input = NULL; - struct sum *sum = NULL; + struct hw *hw; + unsigned int index; + int i; + struct rsc *input; + struct sum *sum; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; input = amixer->input; sum = amixer->sum; @@ -158,10 +158,10 @@ static int amixer_commit_write(struct amixer *amixer) static int amixer_commit_raw_write(struct amixer *amixer) { - struct hw *hw = NULL; - unsigned int index = 0; + struct hw *hw; + unsigned int index; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; index = amixer->rsc.ops->output_slot(&amixer->rsc); hw->amixer_commit_write(hw, index, amixer->rsc.ctrl_blk); @@ -170,9 +170,9 @@ static int amixer_commit_raw_write(struct amixer *amixer) static int amixer_get_y(struct amixer *amixer) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)amixer->rsc.hw; + hw = amixer->rsc.hw; return hw->amixer_get_y(amixer->rsc.ctrl_blk); } @@ -201,7 +201,7 @@ static int amixer_rsc_init(struct amixer *amixer, const struct amixer_desc *desc, struct amixer_mgr *mgr) { - int err = 0; + int err; err = rsc_init(&amixer->rsc, amixer->idx[0], AMIXER, desc->msr, mgr->mgr.hw); @@ -233,9 +233,9 @@ static int get_amixer_rsc(struct amixer_mgr *mgr, const struct amixer_desc *desc, struct amixer **ramixer) { - int err = 0, i = 0; - unsigned int idx = 0; - struct amixer *amixer = NULL; + int err, i; + unsigned int idx; + struct amixer *amixer; unsigned long flags; *ramixer = NULL; @@ -284,7 +284,7 @@ error: static int put_amixer_rsc(struct amixer_mgr *mgr, struct amixer *amixer) { unsigned long flags; - int i = 0; + int i; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < amixer->rsc.msr; i++) @@ -299,7 +299,7 @@ static int put_amixer_rsc(struct amixer_mgr *mgr, struct amixer *amixer) int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr) { - int err = 0; + int err; struct amixer_mgr *amixer_mgr; *ramixer_mgr = NULL; @@ -367,7 +367,7 @@ static int sum_rsc_init(struct sum *sum, const struct sum_desc *desc, struct sum_mgr *mgr) { - int err = 0; + int err; err = rsc_init(&sum->rsc, sum->idx[0], SUM, desc->msr, mgr->mgr.hw); if (err) @@ -388,9 +388,9 @@ static int get_sum_rsc(struct sum_mgr *mgr, const struct sum_desc *desc, struct sum **rsum) { - int err = 0, i = 0; - unsigned int idx = 0; - struct sum *sum = NULL; + int err, i; + unsigned int idx; + struct sum *sum; unsigned long flags; *rsum = NULL; @@ -438,7 +438,7 @@ error: static int put_sum_rsc(struct sum_mgr *mgr, struct sum *sum) { unsigned long flags; - int i = 0; + int i; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < sum->rsc.msr; i++) @@ -453,7 +453,7 @@ static int put_sum_rsc(struct sum_mgr *mgr, struct sum *sum) int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr) { - int err = 0; + int err; struct sum_mgr *sum_mgr; *rsum_mgr = NULL; diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 9b13245..7898a37 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -190,8 +190,8 @@ static unsigned int convert_format(snd_pcm_format_t snd_format) static unsigned int atc_get_pitch(unsigned int input_rate, unsigned int output_rate) { - unsigned int pitch = 0; - int b = 0; + unsigned int pitch; + int b; /* get pitch and convert to fixed-point 8.24 format. */ pitch = (input_rate / output_rate) << 24; @@ -241,12 +241,12 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; struct src_desc desc = {0}; struct amixer_desc mix_dsc = {0}; - struct src *src = NULL; - struct amixer *amixer = NULL; - int err = 0; + struct src *src; + struct amixer *amixer; + int err; int n_amixer = apcm->substream->runtime->channels, i = 0; int device = apcm->substream->pcm->device; - unsigned int pitch = 0; + unsigned int pitch; unsigned long flags; if (NULL != apcm->src) { @@ -324,8 +324,8 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct srcimp_mgr *srcimp_mgr = atc->rsc_mgrs[SRCIMP]; struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; struct sum_mgr *sum_mgr = atc->rsc_mgrs[SUM]; - struct srcimp *srcimp = NULL; - int i = 0; + struct srcimp *srcimp; + int i; if (NULL != apcm->srcimps) { for (i = 0; i < apcm->n_srcimp; i++) { @@ -377,7 +377,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - unsigned int max_cisz = 0; + unsigned int max_cisz; struct src *src = apcm->src; max_cisz = src->multi * src->rsc.msr; @@ -398,8 +398,8 @@ static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) static int atc_pcm_stop(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - struct src *src = NULL; - int i = 0; + struct src *src; + int i; ct_timer_stop(apcm->timer); @@ -426,8 +426,8 @@ static int atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) { struct src *src = apcm->src; - u32 size = 0, max_cisz = 0; - int position = 0; + u32 size, max_cisz; + int position; position = src->ops->get_ca(src); @@ -449,7 +449,7 @@ struct src_node_conf_t { static void setup_src_node_conf(struct ct_atc *atc, struct ct_atc_pcm *apcm, struct src_node_conf_t *conf, int *n_srcc) { - unsigned int pitch = 0; + unsigned int pitch; /* get pitch and convert to fixed-point 8.24 format. */ pitch = atc_get_pitch((atc->rsr * atc->msr), @@ -494,14 +494,14 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; struct sum_mgr *sum_mgr = atc->rsc_mgrs[SUM]; struct src_desc src_dsc = {0}; - struct src *src = NULL; + struct src *src; struct srcimp_desc srcimp_dsc = {0}; - struct srcimp *srcimp = NULL; + struct srcimp *srcimp; struct amixer_desc mix_dsc = {0}; struct sum_desc sum_dsc = {0}; - unsigned int pitch = 0; - int multi = 0, err = 0, i = 0; - int n_srcimp = 0, n_amixer = 0, n_srcc = 0, n_sum = 0; + unsigned int pitch; + int multi, err, i; + int n_srcimp, n_amixer, n_srcc, n_sum; struct src_node_conf_t src_node_conf[2] = {{0} }; /* first release old resources */ @@ -518,8 +518,8 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) setup_src_node_conf(atc, apcm, src_node_conf, &n_srcc); n_sum = (1 == multi) ? 1 : 0; - n_amixer += n_sum * 2 + n_srcc; - n_srcimp += n_srcc; + n_amixer = n_sum * 2 + n_srcc; + n_srcimp = n_srcc; if ((multi > 1) && (0x8000000 >= pitch)) { /* Need extra AMIXERs and SRCIMPs for special treatment * of interleaved recording of conjugate channels */ @@ -633,14 +633,14 @@ error1: static int atc_pcm_capture_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - struct src *src = NULL; - struct amixer *amixer = NULL; - struct srcimp *srcimp = NULL; + struct src *src; + struct amixer *amixer; + struct srcimp *srcimp; struct ct_mixer *mixer = atc->mixer; - struct sum *mono = NULL; + struct sum *mono; struct rsc *out_ports[8] = {NULL}; - int err = 0, i = 0, j = 0, n_sum = 0, multi = 0; - unsigned int pitch = 0; + int err, i, j, n_sum, multi; + unsigned int pitch; int mix_base = 0, imp_base = 0; if (NULL != apcm->src) { @@ -714,9 +714,9 @@ static int atc_pcm_capture_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) static int atc_pcm_capture_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - struct src *src = NULL; + struct src *src; struct src_mgr *src_mgr = atc->rsc_mgrs[SRC]; - int i = 0, multi = 0; + int i, multi; if (apcm->started) return 0; @@ -776,10 +776,10 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc, struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER]; struct src_desc desc = {0}; struct amixer_desc mix_dsc = {0}; - struct src *src = NULL; - int err = 0; - int n_amixer = apcm->substream->runtime->channels, i = 0; - unsigned int pitch = 0, rsr = atc->pll_rate; + struct src *src; + int err; + int n_amixer = apcm->substream->runtime->channels, i; + unsigned int pitch, rsr = atc->pll_rate; /* first release old resources */ atc->pcm_release_resources(atc, apcm); @@ -832,15 +832,24 @@ error1: return err; } +static int atc_pll_init(struct ct_atc *atc, int rate) +{ + struct hw *hw = atc->hw; + int err; + err = hw->pll_init(hw, rate); + atc->pll_rate = err ? 0 : rate; + return err; +} + static int spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm *apcm) { struct dao *dao = container_of(atc->daios[SPDIFOO], struct dao, daio); unsigned long flags; unsigned int rate = apcm->substream->runtime->rate; - unsigned int status = 0; - int err = 0; - unsigned char iec958_con_fs = 0; + unsigned int status; + int err; + unsigned char iec958_con_fs; switch (rate) { case 48000: @@ -864,10 +873,8 @@ spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm *apcm) dao->ops->set_spos(dao, status); dao->ops->commit_write(dao); } - if ((rate != atc->pll_rate) && (32000 != rate)) { - err = ((struct hw *)atc->hw)->pll_init(atc->hw, rate); - atc->pll_rate = err ? 0 : rate; - } + if ((rate != atc->pll_rate) && (32000 != rate)) + err = atc_pll_init(atc, rate); spin_unlock_irqrestore(&atc->atc_lock, flags); return err; @@ -876,11 +883,11 @@ spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm *apcm) static int spdif_passthru_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) { - struct src *src = NULL; - struct amixer *amixer = NULL; - struct dao *dao = NULL; - int err = 0; - int i = 0; + struct src *src; + struct amixer *amixer; + struct dao *dao; + int err; + int i; unsigned long flags; if (NULL != apcm->src) @@ -924,7 +931,7 @@ static int atc_select_line_in(struct ct_atc *atc) { struct hw *hw = atc->hw; struct ct_mixer *mixer = atc->mixer; - struct src *src = NULL; + struct src *src; if (hw->is_adc_source_selected(hw, ADC_LINEIN)) return 0; @@ -946,7 +953,7 @@ static int atc_select_mic_in(struct ct_atc *atc) { struct hw *hw = atc->hw; struct ct_mixer *mixer = atc->mixer; - struct src *src = NULL; + struct src *src; if (hw->is_adc_source_selected(hw, ADC_MICIN)) return 0; @@ -1063,8 +1070,8 @@ static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state) { unsigned long flags; struct dao_desc da_dsc = {0}; - struct dao *dao = NULL; - int err = 0; + struct dao *dao; + int err; struct ct_mixer *mixer = atc->mixer; struct rsc *rscs[2] = {NULL}; unsigned int spos = 0; @@ -1082,11 +1089,8 @@ static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state) dao->ops->set_left_input(dao, rscs[0]); dao->ops->set_right_input(dao, rscs[1]); /* Restore PLL to atc->rsr if needed. */ - if (atc->pll_rate != atc->rsr) { - err = ((struct hw *)atc->hw)->pll_init(atc->hw, - atc->rsr); - atc->pll_rate = err ? 0 : atc->rsr; - } + if (atc->pll_rate != atc->rsr) + err = atc_pll_init(atc, atc->rsr); } dao->ops->set_spos(dao, spos); dao->ops->commit_write(dao); @@ -1097,15 +1101,15 @@ static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state) static int ct_atc_destroy(struct ct_atc *atc) { - struct daio_mgr *daio_mgr = NULL; - struct dao *dao = NULL; - struct dai *dai = NULL; - struct daio *daio = NULL; - struct sum_mgr *sum_mgr = NULL; - struct src_mgr *src_mgr = NULL; - struct srcimp_mgr *srcimp_mgr = NULL; - struct srcimp *srcimp = NULL; - struct ct_mixer *mixer = NULL; + struct daio_mgr *daio_mgr; + struct dao *dao; + struct dai *dai; + struct daio *daio; + struct sum_mgr *sum_mgr; + struct src_mgr *src_mgr; + struct srcimp_mgr *srcimp_mgr; + struct srcimp *srcimp; + struct ct_mixer *mixer; int i = 0; if (NULL == atc) @@ -1279,9 +1283,9 @@ int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc) static int __devinit atc_create_hw_devs(struct ct_atc *atc) { - struct hw *hw = NULL; + struct hw *hw; struct card_conf info = {0}; - int i = 0, err = 0; + int i, err; err = create_hw_obj(atc->pci, &hw); if (err) { @@ -1316,14 +1320,14 @@ static int __devinit atc_create_hw_devs(struct ct_atc *atc) static int __devinit atc_get_resources(struct ct_atc *atc) { struct daio_desc da_desc = {0}; - struct daio_mgr *daio_mgr = NULL; + struct daio_mgr *daio_mgr; struct src_desc src_dsc = {0}; - struct src_mgr *src_mgr = NULL; + struct src_mgr *src_mgr; struct srcimp_desc srcimp_dsc = {0}; - struct srcimp_mgr *srcimp_mgr = NULL; + struct srcimp_mgr *srcimp_mgr; struct sum_desc sum_dsc = {0}; - struct sum_mgr *sum_mgr = NULL; - int err = 0, i = 0; + struct sum_mgr *sum_mgr; + int err, i; unsigned short subsys_id; atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); @@ -1428,8 +1432,8 @@ atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, struct src **srcs, struct srcimp **srcimps) { struct rsc *rscs[2] = {NULL}; - struct src *src = NULL; - struct srcimp *srcimp = NULL; + struct src *src; + struct srcimp *srcimp; int i = 0; rscs[0] = &dai->daio.rscl; @@ -1464,13 +1468,13 @@ atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, static void __devinit atc_connect_resources(struct ct_atc *atc) { - struct dai *dai = NULL; - struct dao *dao = NULL; - struct src *src = NULL; - struct sum *sum = NULL; - struct ct_mixer *mixer = NULL; + struct dai *dai; + struct dao *dao; + struct src *src; + struct sum *sum; + struct ct_mixer *mixer; struct rsc *rscs[2] = {NULL}; - int i = 0, j = 0; + int i, j; mixer = atc->mixer; @@ -1553,11 +1557,11 @@ static struct ct_atc atc_preset __devinitdata = { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, struct ct_atc **ratc) { - struct ct_atc *atc = NULL; + struct ct_atc *atc; static struct snd_device_ops ops = { .dev_free = atc_dev_free, }; - int err = 0; + int err; *ratc = NULL; diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index a2aea39..befead4 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -168,9 +168,9 @@ static int dao_commit_write(struct dao *dao) static int dao_set_left_input(struct dao *dao, struct rsc *input) { - struct imapper *entry = NULL; + struct imapper *entry; struct daio *daio = &dao->daio; - int i = 0; + int i; entry = kzalloc((sizeof(*entry) * daio->rscl.msr), GFP_KERNEL); if (NULL == entry) @@ -196,9 +196,9 @@ static int dao_set_left_input(struct dao *dao, struct rsc *input) static int dao_set_right_input(struct dao *dao, struct rsc *input) { - struct imapper *entry = NULL; + struct imapper *entry; struct daio *daio = &dao->daio; - int i = 0; + int i; entry = kzalloc((sizeof(*entry) * daio->rscr.msr), GFP_KERNEL); if (NULL == entry) @@ -224,9 +224,9 @@ static int dao_set_right_input(struct dao *dao, struct rsc *input) static int dao_clear_left_input(struct dao *dao) { - struct imapper *entry = NULL; + struct imapper *entry; struct daio *daio = &dao->daio; - int i = 0; + int i; if (NULL == dao->imappers[0]) return 0; @@ -248,9 +248,9 @@ static int dao_clear_left_input(struct dao *dao) static int dao_clear_right_input(struct dao *dao) { - struct imapper *entry = NULL; + struct imapper *entry; struct daio *daio = &dao->daio; - int i = 0; + int i; if (NULL == dao->imappers[daio->rscl.msr]) return 0; @@ -299,7 +299,7 @@ static int dai_set_srt_srcr(struct dai *dai, struct rsc *src) static int dai_set_srt_msr(struct dai *dai, unsigned int msr) { - unsigned int rsr = 0; + unsigned int rsr; for (rsr = 0; msr > 1; msr >>= 1) rsr++; @@ -340,8 +340,8 @@ static int daio_rsc_init(struct daio *daio, const struct daio_desc *desc, void *hw) { - int err = 0; - unsigned int idx_l = 0, idx_r = 0; + int err; + unsigned int idx_l, idx_r; switch (((struct hw *)hw)->get_chip_type(hw)) { case ATC20K1: @@ -400,8 +400,8 @@ static int dao_rsc_init(struct dao *dao, struct daio_mgr *mgr) { struct hw *hw = mgr->mgr.hw; - unsigned int conf = 0; - int err = 0; + unsigned int conf; + int err; err = daio_rsc_init(&dao->daio, desc, mgr->mgr.hw); if (err) @@ -423,7 +423,7 @@ static int dao_rsc_init(struct dao *dao, daio_device_index(dao->daio.type, hw)); hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk); - conf |= (desc->msr & 0x7) | (desc->passthru << 3); + conf = (desc->msr & 0x7) | (desc->passthru << 3); hw->daio_mgr_dao_init(mgr->mgr.ctrl_blk, daio_device_index(dao->daio.type, hw), conf); hw->daio_mgr_enb_dao(mgr->mgr.ctrl_blk, @@ -475,9 +475,9 @@ static int dai_rsc_init(struct dai *dai, const struct daio_desc *desc, struct daio_mgr *mgr) { - int err = 0; + int err; struct hw *hw = mgr->mgr.hw; - unsigned int rsr = 0, msr = 0; + unsigned int rsr, msr; err = daio_rsc_init(&dai->daio, desc, mgr->mgr.hw); if (err) @@ -536,7 +536,7 @@ static int get_daio_rsc(struct daio_mgr *mgr, const struct daio_desc *desc, struct daio **rdaio) { - int err = 0; + int err; struct dai *dai = NULL; struct dao *dao = NULL; unsigned long flags; @@ -660,7 +660,7 @@ static int daio_map_op(void *data, struct imapper *entry) static int daio_imap_add(struct daio_mgr *mgr, struct imapper *entry) { unsigned long flags; - int err = 0; + int err; spin_lock_irqsave(&mgr->imap_lock, flags); if ((0 == entry->addr) && (mgr->init_imap_added)) { @@ -677,7 +677,7 @@ static int daio_imap_add(struct daio_mgr *mgr, struct imapper *entry) static int daio_imap_delete(struct daio_mgr *mgr, struct imapper *entry) { unsigned long flags; - int err = 0; + int err; spin_lock_irqsave(&mgr->imap_lock, flags); err = input_mapper_delete(&mgr->imappers, entry, daio_map_op, mgr); @@ -701,7 +701,7 @@ static int daio_mgr_commit_write(struct daio_mgr *mgr) int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr) { - int err = 0, i = 0; + int err, i; struct daio_mgr *daio_mgr; struct imapper *entry; diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c index 53d1aca..5ec6813 100644 --- a/sound/pci/ctxfi/cthardware.c +++ b/sound/pci/ctxfi/cthardware.c @@ -22,7 +22,7 @@ static enum CHIPTYP __devinitdata get_chip_type(struct hw *hw) { - enum CHIPTYP type = ATCNONE; + enum CHIPTYP type; switch (hw->pci->device) { case 0x0005: /* 20k1 device */ @@ -41,7 +41,7 @@ static enum CHIPTYP __devinitdata get_chip_type(struct hw *hw) int __devinit create_hw_obj(struct pci_dev *pci, struct hw **rhw) { - int err = 0; + int err; switch (pci->device) { case 0x0005: /* 20k1 device */ @@ -65,7 +65,7 @@ int __devinit create_hw_obj(struct pci_dev *pci, struct hw **rhw) int destroy_hw_obj(struct hw *hw) { - int err = 0; + int err; switch (hw->pci->device) { case 0x0005: /* 20k1 device */ diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index b165466..38b87b6 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -369,7 +369,7 @@ static unsigned int src_param_pitch_mixer(unsigned int src_idx) static int src_commit_write(struct hw *hw, unsigned int idx, void *blk) { struct src_rsc_ctrl_blk *ctl = blk; - int i = 0; + int i; if (ctl->dirty.bf.czbfs) { /* Clear Z-Buffer registers */ @@ -468,8 +468,8 @@ static int src_mgr_dsb_src(void *blk, unsigned int idx) static int src_mgr_commit_write(struct hw *hw, void *blk) { struct src_mgr_ctrl_blk *ctl = blk; - int i = 0; - unsigned int ret = 0; + int i; + unsigned int ret; if (ctl->dirty.bf.enbsa) { do { @@ -1108,7 +1108,7 @@ static int daio_mgr_set_imapaddr(void *blk, unsigned int addr) static int daio_mgr_commit_write(struct hw *hw, void *blk) { struct daio_mgr_ctrl_blk *ctl = blk; - int i = 0; + int i; if (ctl->dirty.bf.i2sictl || ctl->dirty.bf.i2soctl) { for (i = 0; i < 4; i++) { @@ -1212,8 +1212,8 @@ struct trn_conf { static int hw_daio_init(struct hw *hw, const struct daio_conf *info) { - u32 i2sorg = 0; - u32 spdorg = 0; + u32 i2sorg; + u32 spdorg; /* Read I2S CTL. Keep original value. */ /*i2sorg = hw_read_20kx(hw, I2SCTL);*/ @@ -1263,8 +1263,8 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info) /* TRANSPORT operations */ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) { - u32 trnctl = 0; - unsigned long ptp_phys_low = 0, ptp_phys_high = 0; + u32 trnctl; + u32 ptp_phys_low, ptp_phys_high; /* Set up device page table */ if ((~0UL) == info->vm_pgt_phys) { @@ -1316,7 +1316,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) static int hw_pll_init(struct hw *hw, unsigned int rsr) { unsigned int pllctl; - int i = 0; + int i; pllctl = (48000 == rsr) ? 0x1480a001 : 0x1480a731; for (i = 0; i < 3; i++) { @@ -1384,7 +1384,7 @@ static void i2c_lock(struct hw *hw) static void i2c_write(struct hw *hw, u32 device, u32 addr, u32 data) { - unsigned int ret = 0; + unsigned int ret; do { ret = hw_read_pci(hw, 0xEC); @@ -1397,9 +1397,9 @@ static void i2c_write(struct hw *hw, u32 device, u32 addr, u32 data) static int hw_reset_dac(struct hw *hw) { - u32 i = 0; - u16 gpioorg = 0; - unsigned int ret = 0; + u32 i; + u16 gpioorg; + unsigned int ret; if (i2c_unlock(hw)) return -1; @@ -1430,10 +1430,10 @@ static int hw_reset_dac(struct hw *hw) static int hw_dac_init(struct hw *hw, const struct dac_conf *info) { - u32 data = 0; - u16 gpioorg = 0; - u16 subsys_id = 0; - unsigned int ret = 0; + u32 data; + u16 gpioorg; + u16 subsys_id; + unsigned int ret; pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { @@ -1494,13 +1494,12 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info) static int is_adc_input_selected_SB055x(struct hw *hw, enum ADCSRC type) { - u32 data = 0; - return data; + return 0; } static int is_adc_input_selected_SBx(struct hw *hw, enum ADCSRC type) { - u32 data = 0; + u32 data; data = hw_read_20kx(hw, GPIO); switch (type) { @@ -1521,7 +1520,7 @@ static int is_adc_input_selected_SBx(struct hw *hw, enum ADCSRC type) static int is_adc_input_selected_hendrix(struct hw *hw, enum ADCSRC type) { - u32 data = 0; + u32 data; data = hw_read_20kx(hw, GPIO); switch (type) { @@ -1539,7 +1538,7 @@ static int is_adc_input_selected_hendrix(struct hw *hw, enum ADCSRC type) static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) { - u16 subsys_id = 0; + u16 subsys_id; pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { @@ -1559,7 +1558,7 @@ static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) static int adc_input_select_SB055x(struct hw *hw, enum ADCSRC type, unsigned char boost) { - u32 data = 0; + u32 data; /* * check and set the following GPIO bits accordingly @@ -1599,9 +1598,9 @@ adc_input_select_SB055x(struct hw *hw, enum ADCSRC type, unsigned char boost) static int adc_input_select_SBx(struct hw *hw, enum ADCSRC type, unsigned char boost) { - u32 data = 0; - u32 i2c_data = 0; - unsigned int ret = 0; + u32 data; + u32 i2c_data; + unsigned int ret; if (i2c_unlock(hw)) return -1; @@ -1649,9 +1648,9 @@ adc_input_select_SBx(struct hw *hw, enum ADCSRC type, unsigned char boost) static int adc_input_select_hendrix(struct hw *hw, enum ADCSRC type, unsigned char boost) { - u32 data = 0; - u32 i2c_data = 0; - unsigned int ret = 0; + u32 data; + u32 i2c_data; + unsigned int ret; if (i2c_unlock(hw)) return -1; @@ -1693,7 +1692,7 @@ adc_input_select_hendrix(struct hw *hw, enum ADCSRC type, unsigned char boost) static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) { - u16 subsys_id = 0; + u16 subsys_id; pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { @@ -1719,8 +1718,8 @@ static int adc_init_SBx(struct hw *hw, int input, int mic20db) { u16 gpioorg; u16 input_source; - u32 adcdata = 0; - unsigned int ret = 0; + u32 adcdata; + unsigned int ret; input_source = 0x100; /* default to analog */ switch (input) { @@ -1742,6 +1741,7 @@ static int adc_init_SBx(struct hw *hw, int input, int mic20db) input_source = 0x0; /* set to Digital */ break; default: + adcdata = 0x0; break; } @@ -1781,8 +1781,8 @@ static int adc_init_SBx(struct hw *hw, int input, int mic20db) static int hw_adc_init(struct hw *hw, const struct adc_conf *info) { - int err = 0; - u16 subsys_id = 0; + int err; + u16 subsys_id; pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { @@ -1797,7 +1797,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) static int hw_have_digit_io_switch(struct hw *hw) { - u16 subsys_id = 0; + u16 subsys_id; pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); /* SB073x and Vista compatible cards have no digit IO switch */ @@ -1814,11 +1814,11 @@ static int uaa_to_xfi(struct pci_dev *pci) { unsigned int bar0, bar1, bar2, bar3, bar4, bar5; unsigned int cmd, irq, cl_size, l_timer, pwr; - unsigned int is_uaa = 0; + unsigned int is_uaa; unsigned int data[4] = {0}; unsigned int io_base; void *mem_base; - int i = 0; + int i; const u32 CTLX = CTLBITS('C', 'T', 'L', 'X'); const u32 CTL_ = CTLBITS('C', 'T', 'L', '-'); const u32 CTLF = CTLBITS('C', 'T', 'L', 'F'); @@ -1916,9 +1916,9 @@ static irqreturn_t ct_20k1_interrupt(int irq, void *dev_id) static int hw_card_start(struct hw *hw) { - int err = 0; + int err; struct pci_dev *pci = hw->pci; - u16 subsys_id = 0; + u16 subsys_id; err = pci_enable_device(pci); if (err < 0) @@ -2004,8 +2004,8 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) { int err; unsigned int gctl; - u16 subsys_id = 0; - u32 data = 0; + u16 subsys_id; + u32 data; struct dac_conf dac_info = {0}; struct adc_conf adc_info = {0}; struct daio_conf daio_info = {0}; diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index edbfb48..7d6dcba 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -168,7 +168,7 @@ static int src_get_rsc_ctrl_blk(void **rblk) static int src_put_rsc_ctrl_blk(void *blk) { - kfree((struct src_rsc_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -359,7 +359,7 @@ static unsigned int src_param_pitch_mixer(unsigned int src_idx) static int src_commit_write(struct hw *hw, unsigned int idx, void *blk) { struct src_rsc_ctrl_blk *ctl = blk; - int i = 0; + int i; if (ctl->dirty.bf.czbfs) { /* Clear Z-Buffer registers */ @@ -458,8 +458,8 @@ static int src_mgr_dsb_src(void *blk, unsigned int idx) static int src_mgr_commit_write(struct hw *hw, void *blk) { struct src_mgr_ctrl_blk *ctl = blk; - int i = 0; - unsigned int ret = 0; + int i; + unsigned int ret; if (ctl->dirty.bf.enbsa) { do { @@ -494,7 +494,7 @@ static int src_mgr_get_ctrl_blk(void **rblk) static int src_mgr_put_ctrl_blk(void *blk) { - kfree((struct src_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -515,7 +515,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk) static int srcimp_mgr_put_ctrl_blk(void *blk) { - kfree((struct srcimp_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -704,7 +704,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk) static int amixer_rsc_put_ctrl_blk(void *blk) { - kfree((struct amixer_rsc_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -893,7 +893,7 @@ static int dai_get_ctrl_blk(void **rblk) static int dai_put_ctrl_blk(void *blk) { - kfree((struct dai_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -943,7 +943,7 @@ static int dao_get_ctrl_blk(void **rblk) static int dao_put_ctrl_blk(void *blk) { - kfree((struct dao_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -1051,8 +1051,8 @@ static int daio_mgr_set_imapaddr(void *blk, unsigned int addr) static int daio_mgr_commit_write(struct hw *hw, void *blk) { struct daio_mgr_ctrl_blk *ctl = blk; - unsigned int data = 0; - int i = 0; + unsigned int data; + int i; for (i = 0; i < 8; i++) { if ((ctl->dirty.bf.atxctl & (0x1 << i))) { @@ -1080,7 +1080,7 @@ static int daio_mgr_commit_write(struct hw *hw, void *blk) static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) { struct daio_mgr_ctrl_blk *blk; - int i = 0; + int i; *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); @@ -1099,7 +1099,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) static int daio_mgr_put_ctrl_blk(void *blk) { - kfree((struct daio_mgr_ctrl_blk *)blk); + kfree(blk); return 0; } @@ -1125,7 +1125,7 @@ struct trn_conf { static int hw_daio_init(struct hw *hw, const struct daio_conf *info) { - u32 dwData = 0; + u32 dwData; int i; /* Program I2S with proper sample rate and enable the correct I2S @@ -1195,9 +1195,9 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info) /* TRANSPORT operations */ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) { - u32 vmctl = 0, data = 0; - unsigned long ptp_phys_low = 0, ptp_phys_high = 0; - int i = 0; + u32 vmctl, data; + u32 ptp_phys_low, ptp_phys_high; + int i; /* Set up device page table */ if ((~0UL) == info->vm_pgt_phys) { @@ -1433,7 +1433,7 @@ static int I2CLockChip(struct hw *hw) static int I2CInit(struct hw *hw, u8 bDeviceID, u8 bAddressSize, u8 bDataSize) { - int err = 0; + int err; unsigned int RegI2CStatus; unsigned int RegI2CAddress; @@ -1481,7 +1481,7 @@ static int I2CUninit(struct hw *hw) static int I2CWaitDataReady(struct hw *hw) { int i = 0x400000; - unsigned int ret = 0; + unsigned int ret; do { ret = hw_read_20kx(hw, I2C_IF_STATUS); @@ -1541,9 +1541,9 @@ static int I2CWrite(struct hw *hw, u16 wAddress, u32 dwData) static int hw_dac_init(struct hw *hw, const struct dac_conf *info) { - int err = 0; - u32 dwData = 0; - int i = 0; + int err; + u32 dwData; + int i; struct REGS_CS4382 cs4382_Read = {0}; struct REGS_CS4382 cs4382_Def = { 0x00000001, /* Mode Control 1 */ @@ -1696,7 +1696,7 @@ End: static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) { - u32 data = 0; + u32 data; data = hw_read_20kx(hw, GPIO_DATA); switch (type) { @@ -1714,7 +1714,7 @@ static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) { - u32 data = 0; + u32 data; data = hw_read_20kx(hw, GPIO_DATA); switch (type) { @@ -1747,8 +1747,8 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) static int hw_adc_init(struct hw *hw, const struct adc_conf *info) { - int err = 0; - u32 dwMux = 2, dwData = 0, dwCtl = 0; + int err; + u32 dwMux = 2, dwData, dwCtl; /* Set ADC reset bit as output */ dwData = hw_read_20kx(hw, GPIO_CTRL); diff --git a/sound/pci/ctxfi/ctimap.c b/sound/pci/ctxfi/ctimap.c index d34eacd..0b73368 100644 --- a/sound/pci/ctxfi/ctimap.c +++ b/sound/pci/ctxfi/ctimap.c @@ -99,8 +99,8 @@ int input_mapper_delete(struct list_head *mappers, struct imapper *entry, void free_input_mapper_list(struct list_head *head) { - struct imapper *entry = NULL; - struct list_head *pos = NULL; + struct imapper *entry; + struct list_head *pos; while (!list_empty(head)) { pos = head->next; diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 796156e..666722d 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -298,7 +298,7 @@ set_switch_state(struct ct_mixer *mixer, * from 2^-6 to (1+1023/1024) */ static unsigned int uint16_to_float14(unsigned int x) { - unsigned int i = 0; + unsigned int i; if (x < 17) return 0; @@ -318,7 +318,7 @@ static unsigned int uint16_to_float14(unsigned int x) static unsigned int float14_to_uint16(unsigned int x) { - unsigned int e = 0; + unsigned int e; if (!x) return x; @@ -491,7 +491,7 @@ static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol, struct ct_atc *atc = snd_kcontrol_chip(kcontrol); struct ct_mixer *mixer = atc->mixer; enum CTALSA_MIXER_CTL type = kcontrol->private_value; - int state = 0; + int state; state = ucontrol->value.integer.value[0]; if (get_switch_state(mixer, type) == state) @@ -574,7 +574,7 @@ static int ct_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct ct_atc *atc = snd_kcontrol_chip(kcontrol); - unsigned int status = 0; + unsigned int status; atc->spdif_out_get_status(atc, &status); ucontrol->value.iec958.status[0] = (status >> 0) & 0xff; @@ -589,8 +589,8 @@ static int ct_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct ct_atc *atc = snd_kcontrol_chip(kcontrol); - int change = 1; - unsigned int status = 0, old_status = 0; + int change; + unsigned int status, old_status; status = (ucontrol->value.iec958.status[0] << 0) | (ucontrol->value.iec958.status[1] << 8) | @@ -641,8 +641,8 @@ static struct snd_kcontrol_new iec958_ctl = { static int ct_mixer_kcontrol_new(struct ct_mixer *mixer, struct snd_kcontrol_new *new) { - struct snd_kcontrol *kctl = NULL; - int err = 0; + struct snd_kcontrol *kctl; + int err; kctl = snd_ctl_new1(new, mixer->atc); if (NULL == kctl) @@ -669,9 +669,9 @@ ct_mixer_kcontrol_new(struct ct_mixer *mixer, struct snd_kcontrol_new *new) static int ct_mixer_kcontrols_create(struct ct_mixer *mixer) { - enum CTALSA_MIXER_CTL type = 0; + enum CTALSA_MIXER_CTL type; struct ct_atc *atc = mixer->atc; - int err = 0; + int err; /* Create snd kcontrol instances on demand */ for (type = VOL_MIXER_START; type <= VOL_MIXER_END; type++) { @@ -733,9 +733,9 @@ static int ct_mixer_kcontrols_create(struct ct_mixer *mixer) static void ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type) { - struct amixer *amix_d = NULL; - struct sum *sum_c = NULL; - int i = 0; + struct amixer *amix_d; + struct sum *sum_c; + int i; for (i = 0; i < 2; i++) { amix_d = mixer->amixers[type*CHN_NUM+i]; @@ -748,8 +748,8 @@ ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type) static void ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type) { - struct amixer *amix_d = NULL; - int i = 0; + struct amixer *amix_d; + int i; for (i = 0; i < 2; i++) { amix_d = mixer->amixers[type*CHN_NUM+i]; @@ -760,14 +760,14 @@ ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type) static int ct_mixer_get_resources(struct ct_mixer *mixer) { - struct sum_mgr *sum_mgr = NULL; - struct sum *sum = NULL; + struct sum_mgr *sum_mgr; + struct sum *sum; struct sum_desc sum_desc = {0}; - struct amixer_mgr *amixer_mgr = NULL; - struct amixer *amixer = NULL; + struct amixer_mgr *amixer_mgr; + struct amixer *amixer; struct amixer_desc am_desc = {0}; - int err = 0; - int i = 0; + int err; + int i; /* Allocate sum resources for mixer obj */ sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM]; @@ -822,8 +822,8 @@ error1: static int ct_mixer_get_mem(struct ct_mixer **rmixer) { - struct ct_mixer *mixer = NULL; - int err = 0; + struct ct_mixer *mixer; + int err; *rmixer = NULL; /* Allocate mem for mixer obj */ @@ -855,9 +855,9 @@ error1: static int ct_mixer_topology_build(struct ct_mixer *mixer) { - struct sum *sum = NULL; - struct amixer *amix_d = NULL, *amix_s = NULL; - enum CT_AMIXER_CTL i = 0, j = 0; + struct sum *sum; + struct amixer *amix_d, *amix_s; + enum CT_AMIXER_CTL i, j; /* Build topology from destination to source */ @@ -1044,7 +1044,7 @@ int ct_mixer_destroy(struct ct_mixer *mixer) struct sum_mgr *sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM]; struct amixer_mgr *amixer_mgr = (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER]; - struct amixer *amixer = NULL; + struct amixer *amixer; int i = 0; /* Release amixer resources */ @@ -1071,8 +1071,8 @@ int ct_mixer_destroy(struct ct_mixer *mixer) int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer) { - struct ct_mixer *mixer = NULL; - int err = 0; + struct ct_mixer *mixer; + int err; *rmixer = NULL; @@ -1109,7 +1109,7 @@ int ct_alsa_mix_create(struct ct_atc *atc, enum CTALSADEVS device, const char *device_name) { - int err = 0; + int err; /* Create snd kcontrol instances on demand */ /* vol_ctl.device = swh_ctl.device = device; */ /* better w/ device 0 */ diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index da21a71..889c495 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -27,7 +27,7 @@ static int get_resource(u8 *rscs, unsigned int amount, unsigned int multi, unsigned int *ridx) { - int i = 0, j = 0, k = 0, n = 0; + int i, j, k, n; /* Check whether there are sufficient resources to meet request. */ for (i = 0, n = multi; i < amount; i++) { @@ -61,7 +61,7 @@ get_resource(u8 *rscs, unsigned int amount, static int put_resource(u8 *rscs, unsigned int multi, unsigned int idx) { - unsigned int i = 0, j = 0, k = 0, n = 0; + unsigned int i, j, k, n; /* Mark the contiguous bits in resource bit-map as used */ for (n = multi, i = idx; n > 0; n--) { @@ -76,7 +76,7 @@ static int put_resource(u8 *rscs, unsigned int multi, unsigned int idx) int mgr_get_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int *ridx) { - int err = 0; + int err; if (n > mgr->avail) return -ENOENT; diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index 77e118c..e1c145d 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -37,9 +37,9 @@ static int (*src_default_config[3])(struct src *) = { static int src_set_state(struct src *src, unsigned int state) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_state(src->rsc.ctrl_blk, state); return 0; @@ -47,9 +47,9 @@ static int src_set_state(struct src *src, unsigned int state) static int src_set_bm(struct src *src, unsigned int bm) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_bm(src->rsc.ctrl_blk, bm); return 0; @@ -57,9 +57,9 @@ static int src_set_bm(struct src *src, unsigned int bm) static int src_set_sf(struct src *src, unsigned int sf) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_sf(src->rsc.ctrl_blk, sf); return 0; @@ -67,9 +67,9 @@ static int src_set_sf(struct src *src, unsigned int sf) static int src_set_pm(struct src *src, unsigned int pm) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_pm(src->rsc.ctrl_blk, pm); return 0; @@ -77,9 +77,9 @@ static int src_set_pm(struct src *src, unsigned int pm) static int src_set_rom(struct src *src, unsigned int rom) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_rom(src->rsc.ctrl_blk, rom); return 0; @@ -87,9 +87,9 @@ static int src_set_rom(struct src *src, unsigned int rom) static int src_set_vo(struct src *src, unsigned int vo) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_vo(src->rsc.ctrl_blk, vo); return 0; @@ -97,9 +97,9 @@ static int src_set_vo(struct src *src, unsigned int vo) static int src_set_st(struct src *src, unsigned int st) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_st(src->rsc.ctrl_blk, st); return 0; @@ -107,9 +107,9 @@ static int src_set_st(struct src *src, unsigned int st) static int src_set_bp(struct src *src, unsigned int bp) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_bp(src->rsc.ctrl_blk, bp); return 0; @@ -117,9 +117,9 @@ static int src_set_bp(struct src *src, unsigned int bp) static int src_set_cisz(struct src *src, unsigned int cisz) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_cisz(src->rsc.ctrl_blk, cisz); return 0; @@ -127,9 +127,9 @@ static int src_set_cisz(struct src *src, unsigned int cisz) static int src_set_ca(struct src *src, unsigned int ca) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_ca(src->rsc.ctrl_blk, ca); return 0; @@ -137,9 +137,9 @@ static int src_set_ca(struct src *src, unsigned int ca) static int src_set_sa(struct src *src, unsigned int sa) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_sa(src->rsc.ctrl_blk, sa); return 0; @@ -147,9 +147,9 @@ static int src_set_sa(struct src *src, unsigned int sa) static int src_set_la(struct src *src, unsigned int la) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_la(src->rsc.ctrl_blk, la); return 0; @@ -157,9 +157,9 @@ static int src_set_la(struct src *src, unsigned int la) static int src_set_pitch(struct src *src, unsigned int pitch) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_pitch(src->rsc.ctrl_blk, pitch); return 0; @@ -167,9 +167,9 @@ static int src_set_pitch(struct src *src, unsigned int pitch) static int src_set_clear_zbufs(struct src *src) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1); return 0; @@ -177,11 +177,11 @@ static int src_set_clear_zbufs(struct src *src) static int src_commit_write(struct src *src) { - struct hw *hw = NULL; - int i = 0; + struct hw *hw; + int i; unsigned int dirty = 0; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; src->rsc.ops->master(&src->rsc); if (src->rsc.msr > 1) { /* Save dirty flags for conjugate resource programming */ @@ -207,9 +207,9 @@ static int src_commit_write(struct src *src) static int src_get_ca(struct src *src) { - struct hw *hw = NULL; + struct hw *hw; - hw = (struct hw *)src->rsc.hw; + hw = src->rsc.hw; return hw->src_get_ca(hw, src->rsc.ops->index(&src->rsc), src->rsc.ctrl_blk); } @@ -229,7 +229,7 @@ static struct src *src_next_interleave(struct src *src) static int src_default_config_memrd(struct src *src) { struct hw *hw = src->rsc.hw; - unsigned int rsr = 0, msr = 0; + unsigned int rsr, msr; hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF); hw->src_set_bm(src->rsc.ctrl_blk, 1); @@ -297,7 +297,7 @@ static int src_default_config_memwr(struct src *src) static int src_default_config_arcrw(struct src *src) { struct hw *hw = src->rsc.hw; - unsigned int rsr = 0, msr = 0; + unsigned int rsr, msr; unsigned int dirty; hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF); @@ -360,8 +360,8 @@ static int src_rsc_init(struct src *src, u32 idx, const struct src_desc *desc, struct src_mgr *mgr) { - int err = 0; - int i = 0, n = 0; + int err; + int i, n; struct src *p; n = (MEMRD == desc->mode) ? desc->multi : 1; @@ -395,7 +395,7 @@ error1: static int src_rsc_uninit(struct src *src, struct src_mgr *mgr) { - int i = 0, n = 0; + int i, n; struct src *p; n = (MEMRD == src->mode) ? src->multi : 1; @@ -416,8 +416,8 @@ static int get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc) { unsigned int idx = SRC_RESOURCE_NUM; - int err = 0; - struct src *src = NULL; + int err; + struct src *src; unsigned long flags; *rsrc = NULL; @@ -489,7 +489,7 @@ static int put_src_rsc(struct src_mgr *mgr, struct src *src) static int src_enable_s(struct src_mgr *mgr, struct src *src) { struct hw *hw = mgr->mgr.hw; - int i = 0; + int i; src->rsc.ops->master(&src->rsc); for (i = 0; i < src->rsc.msr; i++) { @@ -505,7 +505,7 @@ static int src_enable_s(struct src_mgr *mgr, struct src *src) static int src_enable(struct src_mgr *mgr, struct src *src) { struct hw *hw = mgr->mgr.hw; - int i = 0; + int i; src->rsc.ops->master(&src->rsc); for (i = 0; i < src->rsc.msr; i++) { @@ -521,7 +521,7 @@ static int src_enable(struct src_mgr *mgr, struct src *src) static int src_disable(struct src_mgr *mgr, struct src *src) { struct hw *hw = mgr->mgr.hw; - int i = 0; + int i; src->rsc.ops->master(&src->rsc); for (i = 0; i < src->rsc.msr; i++) { @@ -545,7 +545,7 @@ static int src_mgr_commit_write(struct src_mgr *mgr) int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr) { - int err = 0, i = 0; + int err, i; struct src_mgr *src_mgr; *rsrc_mgr = NULL; @@ -618,8 +618,8 @@ static struct rsc_ops srcimp_basic_rsc_ops = { static int srcimp_map(struct srcimp *srcimp, struct src *src, struct rsc *input) { - struct imapper *entry = NULL; - int i = 0; + struct imapper *entry; + int i; srcimp->rsc.ops->master(&srcimp->rsc); src->rsc.ops->master(&src->rsc); @@ -646,7 +646,7 @@ static int srcimp_map(struct srcimp *srcimp, struct src *src, struct rsc *input) static int srcimp_unmap(struct srcimp *srcimp) { - int i = 0; + int i; /* Program master and conjugate resources */ for (i = 0; i < srcimp->rsc.msr; i++) { @@ -669,7 +669,7 @@ static int srcimp_rsc_init(struct srcimp *srcimp, const struct srcimp_desc *desc, struct srcimp_mgr *mgr) { - int err = 0; + int err; err = rsc_init(&srcimp->rsc, srcimp->idx[0], SRCIMP, desc->msr, mgr->mgr.hw); @@ -715,9 +715,9 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr, const struct srcimp_desc *desc, struct srcimp **rsrcimp) { - int err = 0, i = 0; - unsigned int idx = 0; - struct srcimp *srcimp = NULL; + int err, i; + unsigned int idx; + struct srcimp *srcimp; unsigned long flags; *rsrcimp = NULL; @@ -765,7 +765,7 @@ error1: static int put_srcimp_rsc(struct srcimp_mgr *mgr, struct srcimp *srcimp) { unsigned long flags; - int i = 0; + int i; spin_lock_irqsave(&mgr->mgr_lock, flags); for (i = 0; i < srcimp->rsc.msr; i++) @@ -795,7 +795,7 @@ static int srcimp_map_op(void *data, struct imapper *entry) static int srcimp_imap_add(struct srcimp_mgr *mgr, struct imapper *entry) { unsigned long flags; - int err = 0; + int err; spin_lock_irqsave(&mgr->imap_lock, flags); if ((0 == entry->addr) && (mgr->init_imap_added)) { @@ -812,7 +812,7 @@ static int srcimp_imap_add(struct srcimp_mgr *mgr, struct imapper *entry) static int srcimp_imap_delete(struct srcimp_mgr *mgr, struct imapper *entry) { unsigned long flags; - int err = 0; + int err; spin_lock_irqsave(&mgr->imap_lock, flags); err = input_mapper_delete(&mgr->imappers, entry, srcimp_map_op, mgr); @@ -828,7 +828,7 @@ static int srcimp_imap_delete(struct srcimp_mgr *mgr, struct imapper *entry) int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr) { - int err = 0; + int err; struct srcimp_mgr *srcimp_mgr; struct imapper *entry; diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index b7f8e58..67665a7 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -31,8 +31,8 @@ static struct ct_vm_block * get_vm_block(struct ct_vm *vm, unsigned int size) { - struct ct_vm_block *block = NULL, *entry = NULL; - struct list_head *pos = NULL; + struct ct_vm_block *block = NULL, *entry; + struct list_head *pos; size = CT_PAGE_ALIGN(size); if (size > vm->size) { @@ -77,8 +77,8 @@ get_vm_block(struct ct_vm *vm, unsigned int size) static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block) { - struct ct_vm_block *entry = NULL, *pre_ent = NULL; - struct list_head *pos = NULL, *pre = NULL; + struct ct_vm_block *entry, *pre_ent; + struct list_head *pos, *pre; block->size = CT_PAGE_ALIGN(block->size); @@ -223,8 +223,8 @@ int ct_vm_create(struct ct_vm **rvm) void ct_vm_destroy(struct ct_vm *vm) { int i; - struct list_head *pos = NULL; - struct ct_vm_block *entry = NULL; + struct list_head *pos; + struct ct_vm_block *entry; /* free used and unused list nodes */ while (!list_empty(&vm->used)) { -- cgit v1.1 From af8500bbbd18438495d2f91ad07bda49fff3b770 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 15:07:46 +0200 Subject: ALSA: ctxfi - Fix possible buffer pointer overrun Fix possible buffer pointer overruns. Back to zero when it's equal or over the buffer size. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctpcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index a0bd31c..870fa17 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -243,6 +243,8 @@ ct_pcm_playback_pointer(struct snd_pcm_substream *substream) /* Read out playback position */ position = atc->pcm_playback_position(atc, apcm); position = bytes_to_frames(runtime, position); + if (position >= runtime->buffer_size) + position = 0; return position; } @@ -343,6 +345,8 @@ ct_pcm_capture_pointer(struct snd_pcm_substream *substream) /* Read out playback position */ position = atc->pcm_capture_position(atc, apcm); position = bytes_to_frames(runtime, position); + if (position >= runtime->buffer_size) + position = 0; return position; } -- cgit v1.1 From d362af62ed98f58c64a2b3dd58c79d25ad181b0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 15:31:22 +0200 Subject: ALSA: ctxfi - Fix / clean up hw20k2 chip code - Clean up Hungarian coding style - Don't use static variables for I2C information; this unables to use multiple instances. Now they are stored in struct hw20k2 fields. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cthw20k2.c | 346 +++++++++++++++++++++++---------------------- 1 file changed, 177 insertions(+), 169 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 7d6dcba..4493a51 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -15,8 +15,6 @@ * */ -#include "cthw20k2.h" -#include "ct20k2reg.h" #include #include #include @@ -25,6 +23,8 @@ #include #include #include +#include "cthw20k2.h" +#include "ct20k2reg.h" #if BITS_PER_LONG == 32 #define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */ @@ -32,6 +32,14 @@ #define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */ #endif +struct hw20k2 { + struct hw hw; + /* for i2c */ + unsigned char dev_id; + unsigned char addr_size; + unsigned char data_size; +}; + static u32 hw_read_20kx(struct hw *hw, u32 reg); static void hw_write_20kx(struct hw *hw, u32 reg, u32 data); @@ -1125,7 +1133,7 @@ struct trn_conf { static int hw_daio_init(struct hw *hw, const struct daio_conf *info) { - u32 dwData; + u32 data; int i; /* Program I2S with proper sample rate and enable the correct I2S @@ -1156,12 +1164,12 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info) if (i <= 3) { /* 1st 3 channels are SPDIFs (SB0960) */ if (i == 3) - dwData = 0x1001001; + data = 0x1001001; else - dwData = 0x1000001; + data = 0x1000001; - hw_write_20kx(hw, (AUDIO_IO_TX_CTL+(0x40*i)), dwData); - hw_write_20kx(hw, (AUDIO_IO_RX_CTL+(0x40*i)), dwData); + hw_write_20kx(hw, (AUDIO_IO_TX_CTL+(0x40*i)), data); + hw_write_20kx(hw, (AUDIO_IO_RX_CTL+(0x40*i)), data); /* Initialize the SPDIF Out Channel status registers. * The value specified here is based on the typical @@ -1179,13 +1187,13 @@ static int hw_daio_init(struct hw *hw, const struct daio_conf *info) hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_H+(0x40*i), 0x0B); } else { /* Next 5 channels are I2S (SB0960) */ - dwData = 0x11; - hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), dwData); + data = 0x11; + hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), data); if (2 == info->msr) { /* Four channels per sample period */ - dwData |= 0x1000; + data |= 0x1000; } - hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), dwData); + hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), data); } } @@ -1377,34 +1385,32 @@ static int hw_auto_init(struct hw *hw) #define I2C_ADDRESS_PTAD 0x0000FFFF #define I2C_ADDRESS_SLAD 0x007F0000 -struct REGS_CS4382 { - u32 dwModeControl_1; - u32 dwModeControl_2; - u32 dwModeControl_3; +struct regs_cs4382 { + u32 mode_control_1; + u32 mode_control_2; + u32 mode_control_3; - u32 dwFilterControl; - u32 dwInvertControl; + u32 filter_control; + u32 invert_control; - u32 dwMixControl_P1; - u32 dwVolControl_A1; - u32 dwVolControl_B1; + u32 mix_control_P1; + u32 vol_control_A1; + u32 vol_control_B1; - u32 dwMixControl_P2; - u32 dwVolControl_A2; - u32 dwVolControl_B2; + u32 mix_control_P2; + u32 vol_control_A2; + u32 vol_control_B2; - u32 dwMixControl_P3; - u32 dwVolControl_A3; - u32 dwVolControl_B3; + u32 mix_control_P3; + u32 vol_control_A3; + u32 vol_control_B3; - u32 dwMixControl_P4; - u32 dwVolControl_A4; - u32 dwVolControl_B4; + u32 mix_control_P4; + u32 vol_control_A4; + u32 vol_control_B4; }; -static u8 m_bAddressSize, m_bDataSize, m_bDeviceID; - -static int I2CUnlockFullAccess(struct hw *hw) +static int hw20k2_i2c_unlock_full_access(struct hw *hw) { u8 UnlockKeySequence_FLASH_FULLACCESS_MODE[2] = {0xB3, 0xD4}; @@ -1420,7 +1426,7 @@ static int I2CUnlockFullAccess(struct hw *hw) return -1; } -static int I2CLockChip(struct hw *hw) +static int hw20k2_i2c_lock_chip(struct hw *hw) { /* Write twice */ hw_write_20kx(hw, I2C_IF_WLOCK, STATE_LOCKED); @@ -1431,54 +1437,55 @@ static int I2CLockChip(struct hw *hw) return -1; } -static int I2CInit(struct hw *hw, u8 bDeviceID, u8 bAddressSize, u8 bDataSize) +static int hw20k2_i2c_init(struct hw *hw, u8 dev_id, u8 addr_size, u8 data_size) { + struct hw20k2 *hw20k2 = (struct hw20k2 *)hw; int err; - unsigned int RegI2CStatus; - unsigned int RegI2CAddress; + unsigned int i2c_status; + unsigned int i2c_addr; - err = I2CUnlockFullAccess(hw); + err = hw20k2_i2c_unlock_full_access(hw); if (err < 0) return err; - m_bAddressSize = bAddressSize; - m_bDataSize = bDataSize; - m_bDeviceID = bDeviceID; + hw20k2->addr_size = addr_size; + hw20k2->data_size = data_size; + hw20k2->dev_id = dev_id; - RegI2CAddress = 0; - set_field(&RegI2CAddress, I2C_ADDRESS_SLAD, bDeviceID); + i2c_addr = 0; + set_field(&i2c_addr, I2C_ADDRESS_SLAD, dev_id); - hw_write_20kx(hw, I2C_IF_ADDRESS, RegI2CAddress); + hw_write_20kx(hw, I2C_IF_ADDRESS, i2c_addr); - RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + i2c_status = hw_read_20kx(hw, I2C_IF_STATUS); - set_field(&RegI2CStatus, I2C_STATUS_DCM, 1); /* Direct control mode */ + set_field(&i2c_status, I2C_STATUS_DCM, 1); /* Direct control mode */ - hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); + hw_write_20kx(hw, I2C_IF_STATUS, i2c_status); return 0; } -static int I2CUninit(struct hw *hw) +static int hw20k2_i2c_uninit(struct hw *hw) { - unsigned int RegI2CStatus; - unsigned int RegI2CAddress; + unsigned int i2c_status; + unsigned int i2c_addr; - RegI2CAddress = 0; - set_field(&RegI2CAddress, I2C_ADDRESS_SLAD, 0x57); /* I2C id */ + i2c_addr = 0; + set_field(&i2c_addr, I2C_ADDRESS_SLAD, 0x57); /* I2C id */ - hw_write_20kx(hw, I2C_IF_ADDRESS, RegI2CAddress); + hw_write_20kx(hw, I2C_IF_ADDRESS, i2c_addr); - RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + i2c_status = hw_read_20kx(hw, I2C_IF_STATUS); - set_field(&RegI2CStatus, I2C_STATUS_DCM, 0); /* I2C mode */ + set_field(&i2c_status, I2C_STATUS_DCM, 0); /* I2C mode */ - hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); + hw_write_20kx(hw, I2C_IF_STATUS, i2c_status); - return I2CLockChip(hw); + return hw20k2_i2c_lock_chip(hw); } -static int I2CWaitDataReady(struct hw *hw) +static int hw20k2_i2c_wait_data_ready(struct hw *hw) { int i = 0x400000; unsigned int ret; @@ -1490,51 +1497,53 @@ static int I2CWaitDataReady(struct hw *hw) return i; } -static int I2CRead(struct hw *hw, u16 wAddress, u32 *pdwData) +static int hw20k2_i2c_read(struct hw *hw, u16 addr, u32 *datap) { - unsigned int RegI2CStatus; + struct hw20k2 *hw20k2 = (struct hw20k2 *)hw; + unsigned int i2c_status; - RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); - set_field(&RegI2CStatus, I2C_STATUS_BC, - (4 == m_bAddressSize) ? 0 : m_bAddressSize); - hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); - if (!I2CWaitDataReady(hw)) + i2c_status = hw_read_20kx(hw, I2C_IF_STATUS); + set_field(&i2c_status, I2C_STATUS_BC, + (4 == hw20k2->addr_size) ? 0 : hw20k2->addr_size); + hw_write_20kx(hw, I2C_IF_STATUS, i2c_status); + if (!hw20k2_i2c_wait_data_ready(hw)) return -1; - hw_write_20kx(hw, I2C_IF_WDATA, (u32)wAddress); - if (!I2CWaitDataReady(hw)) + hw_write_20kx(hw, I2C_IF_WDATA, addr); + if (!hw20k2_i2c_wait_data_ready(hw)) return -1; /* Force a read operation */ hw_write_20kx(hw, I2C_IF_RDATA, 0); - if (!I2CWaitDataReady(hw)) + if (!hw20k2_i2c_wait_data_ready(hw)) return -1; - *pdwData = hw_read_20kx(hw, I2C_IF_RDATA); + *datap = hw_read_20kx(hw, I2C_IF_RDATA); return 0; } -static int I2CWrite(struct hw *hw, u16 wAddress, u32 dwData) +static int hw20k2_i2c_write(struct hw *hw, u16 addr, u32 data) { - unsigned int dwI2CData = (dwData << (m_bAddressSize * 8)) | wAddress; - unsigned int RegI2CStatus; + struct hw20k2 *hw20k2 = (struct hw20k2 *)hw; + unsigned int i2c_data = (data << (hw20k2->addr_size * 8)) | addr; + unsigned int i2c_status; - RegI2CStatus = hw_read_20kx(hw, I2C_IF_STATUS); + i2c_status = hw_read_20kx(hw, I2C_IF_STATUS); - set_field(&RegI2CStatus, I2C_STATUS_BC, - (4 == (m_bAddressSize + m_bDataSize)) ? - 0 : (m_bAddressSize + m_bDataSize)); + set_field(&i2c_status, I2C_STATUS_BC, + (4 == (hw20k2->addr_size + hw20k2->data_size)) ? + 0 : (hw20k2->addr_size + hw20k2->data_size)); - hw_write_20kx(hw, I2C_IF_STATUS, RegI2CStatus); - I2CWaitDataReady(hw); + hw_write_20kx(hw, I2C_IF_STATUS, i2c_status); + hw20k2_i2c_wait_data_ready(hw); /* Dummy write to trigger the write oprtation */ hw_write_20kx(hw, I2C_IF_WDATA, 0); - I2CWaitDataReady(hw); + hw20k2_i2c_wait_data_ready(hw); /* This is the real data */ - hw_write_20kx(hw, I2C_IF_WDATA, dwI2CData); - I2CWaitDataReady(hw); + hw_write_20kx(hw, I2C_IF_WDATA, i2c_data); + hw20k2_i2c_wait_data_ready(hw); return 0; } @@ -1542,10 +1551,10 @@ static int I2CWrite(struct hw *hw, u16 wAddress, u32 dwData) static int hw_dac_init(struct hw *hw, const struct dac_conf *info) { int err; - u32 dwData; + u32 data; int i; - struct REGS_CS4382 cs4382_Read = {0}; - struct REGS_CS4382 cs4382_Def = { + struct regs_cs4382 cs_read = {0}; + struct regs_cs4382 cs_def = { 0x00000001, /* Mode Control 1 */ 0x00000000, /* Mode Control 2 */ 0x00000084, /* Mode Control 3 */ @@ -1566,87 +1575,86 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info) }; /* Set DAC reset bit as output */ - dwData = hw_read_20kx(hw, GPIO_CTRL); - dwData |= 0x02; - hw_write_20kx(hw, GPIO_CTRL, dwData); + data = hw_read_20kx(hw, GPIO_CTRL); + data |= 0x02; + hw_write_20kx(hw, GPIO_CTRL, data); - err = I2CInit(hw, 0x18, 1, 1); + err = hw20k2_i2c_init(hw, 0x18, 1, 1); if (err < 0) goto End; for (i = 0; i < 2; i++) { /* Reset DAC twice just in-case the chip * didn't initialized properly */ - dwData = hw_read_20kx(hw, GPIO_DATA); + data = hw_read_20kx(hw, GPIO_DATA); /* GPIO data bit 1 */ - dwData &= 0xFFFFFFFD; - hw_write_20kx(hw, GPIO_DATA, dwData); + data &= 0xFFFFFFFD; + hw_write_20kx(hw, GPIO_DATA, data); mdelay(10); - dwData |= 0x2; - hw_write_20kx(hw, GPIO_DATA, dwData); + data |= 0x2; + hw_write_20kx(hw, GPIO_DATA, data); mdelay(50); /* Reset the 2nd time */ - dwData &= 0xFFFFFFFD; - hw_write_20kx(hw, GPIO_DATA, dwData); + data &= 0xFFFFFFFD; + hw_write_20kx(hw, GPIO_DATA, data); mdelay(10); - dwData |= 0x2; - hw_write_20kx(hw, GPIO_DATA, dwData); + data |= 0x2; + hw_write_20kx(hw, GPIO_DATA, data); mdelay(50); - if (I2CRead(hw, CS4382_MC1, &cs4382_Read.dwModeControl_1)) + if (hw20k2_i2c_read(hw, CS4382_MC1, &cs_read.mode_control_1)) continue; - if (I2CRead(hw, CS4382_MC2, &cs4382_Read.dwModeControl_2)) + if (hw20k2_i2c_read(hw, CS4382_MC2, &cs_read.mode_control_2)) continue; - if (I2CRead(hw, CS4382_MC3, &cs4382_Read.dwModeControl_3)) + if (hw20k2_i2c_read(hw, CS4382_MC3, &cs_read.mode_control_3)) continue; - if (I2CRead(hw, CS4382_FC, &cs4382_Read.dwFilterControl)) + if (hw20k2_i2c_read(hw, CS4382_FC, &cs_read.filter_control)) continue; - if (I2CRead(hw, CS4382_IC, &cs4382_Read.dwInvertControl)) + if (hw20k2_i2c_read(hw, CS4382_IC, &cs_read.invert_control)) continue; - if (I2CRead(hw, CS4382_XC1, &cs4382_Read.dwMixControl_P1)) + if (hw20k2_i2c_read(hw, CS4382_XC1, &cs_read.mix_control_P1)) continue; - if (I2CRead(hw, CS4382_VCA1, &cs4382_Read.dwVolControl_A1)) + if (hw20k2_i2c_read(hw, CS4382_VCA1, &cs_read.vol_control_A1)) continue; - if (I2CRead(hw, CS4382_VCB1, &cs4382_Read.dwVolControl_B1)) + if (hw20k2_i2c_read(hw, CS4382_VCB1, &cs_read.vol_control_B1)) continue; - if (I2CRead(hw, CS4382_XC2, &cs4382_Read.dwMixControl_P2)) + if (hw20k2_i2c_read(hw, CS4382_XC2, &cs_read.mix_control_P2)) continue; - if (I2CRead(hw, CS4382_VCA2, &cs4382_Read.dwVolControl_A2)) + if (hw20k2_i2c_read(hw, CS4382_VCA2, &cs_read.vol_control_A2)) continue; - if (I2CRead(hw, CS4382_VCB2, &cs4382_Read.dwVolControl_B2)) + if (hw20k2_i2c_read(hw, CS4382_VCB2, &cs_read.vol_control_B2)) continue; - if (I2CRead(hw, CS4382_XC3, &cs4382_Read.dwMixControl_P3)) + if (hw20k2_i2c_read(hw, CS4382_XC3, &cs_read.mix_control_P3)) continue; - if (I2CRead(hw, CS4382_VCA3, &cs4382_Read.dwVolControl_A3)) + if (hw20k2_i2c_read(hw, CS4382_VCA3, &cs_read.vol_control_A3)) continue; - if (I2CRead(hw, CS4382_VCB3, &cs4382_Read.dwVolControl_B3)) + if (hw20k2_i2c_read(hw, CS4382_VCB3, &cs_read.vol_control_B3)) continue; - if (I2CRead(hw, CS4382_XC4, &cs4382_Read.dwMixControl_P4)) + if (hw20k2_i2c_read(hw, CS4382_XC4, &cs_read.mix_control_P4)) continue; - if (I2CRead(hw, CS4382_VCA4, &cs4382_Read.dwVolControl_A4)) + if (hw20k2_i2c_read(hw, CS4382_VCA4, &cs_read.vol_control_A4)) continue; - if (I2CRead(hw, CS4382_VCB4, &cs4382_Read.dwVolControl_B4)) + if (hw20k2_i2c_read(hw, CS4382_VCB4, &cs_read.vol_control_B4)) continue; - if (memcmp(&cs4382_Read, &cs4382_Def, - sizeof(struct REGS_CS4382))) + if (memcmp(&cs_read, &cs_def, sizeof(cs_read))) continue; else break; @@ -1657,29 +1665,29 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info) /* Note: Every I2C write must have some delay. * This is not a requirement but the delay works here... */ - I2CWrite(hw, CS4382_MC1, 0x80); - I2CWrite(hw, CS4382_MC2, 0x10); + hw20k2_i2c_write(hw, CS4382_MC1, 0x80); + hw20k2_i2c_write(hw, CS4382_MC2, 0x10); if (1 == info->msr) { - I2CWrite(hw, CS4382_XC1, 0x24); - I2CWrite(hw, CS4382_XC2, 0x24); - I2CWrite(hw, CS4382_XC3, 0x24); - I2CWrite(hw, CS4382_XC4, 0x24); + hw20k2_i2c_write(hw, CS4382_XC1, 0x24); + hw20k2_i2c_write(hw, CS4382_XC2, 0x24); + hw20k2_i2c_write(hw, CS4382_XC3, 0x24); + hw20k2_i2c_write(hw, CS4382_XC4, 0x24); } else if (2 == info->msr) { - I2CWrite(hw, CS4382_XC1, 0x25); - I2CWrite(hw, CS4382_XC2, 0x25); - I2CWrite(hw, CS4382_XC3, 0x25); - I2CWrite(hw, CS4382_XC4, 0x25); + hw20k2_i2c_write(hw, CS4382_XC1, 0x25); + hw20k2_i2c_write(hw, CS4382_XC2, 0x25); + hw20k2_i2c_write(hw, CS4382_XC3, 0x25); + hw20k2_i2c_write(hw, CS4382_XC4, 0x25); } else { - I2CWrite(hw, CS4382_XC1, 0x26); - I2CWrite(hw, CS4382_XC2, 0x26); - I2CWrite(hw, CS4382_XC3, 0x26); - I2CWrite(hw, CS4382_XC4, 0x26); + hw20k2_i2c_write(hw, CS4382_XC1, 0x26); + hw20k2_i2c_write(hw, CS4382_XC2, 0x26); + hw20k2_i2c_write(hw, CS4382_XC3, 0x26); + hw20k2_i2c_write(hw, CS4382_XC4, 0x26); } return 0; End: - I2CUninit(hw); + hw20k2_i2c_uninit(hw); return -1; } @@ -1721,21 +1729,21 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) case ADC_MICIN: data |= (0x1 << 14); hw_write_20kx(hw, GPIO_DATA, data); - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), MAKE_WM8775_DATA(0x101)); /* Mic-in */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ break; case ADC_LINEIN: data &= ~(0x1 << 14); hw_write_20kx(hw, GPIO_DATA, data); - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102), MAKE_WM8775_DATA(0x102)); /* Line-in */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF), MAKE_WM8775_DATA(0xCF)); /* No boost */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), MAKE_WM8775_DATA(0xCF)); /* No boost */ break; default: @@ -1748,34 +1756,34 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) static int hw_adc_init(struct hw *hw, const struct adc_conf *info) { int err; - u32 dwMux = 2, dwData, dwCtl; + u32 mux = 2, data, ctl; /* Set ADC reset bit as output */ - dwData = hw_read_20kx(hw, GPIO_CTRL); - dwData |= (0x1 << 15); - hw_write_20kx(hw, GPIO_CTRL, dwData); + data = hw_read_20kx(hw, GPIO_CTRL); + data |= (0x1 << 15); + hw_write_20kx(hw, GPIO_CTRL, data); /* Initialize I2C */ - err = I2CInit(hw, 0x1A, 1, 1); + err = hw20k2_i2c_init(hw, 0x1A, 1, 1); if (err < 0) { printk(KERN_ALERT "ctxfi: Failure to acquire I2C!!!\n"); goto error; } /* Make ADC in normal operation */ - dwData = hw_read_20kx(hw, GPIO_DATA); - dwData &= ~(0x1 << 15); + data = hw_read_20kx(hw, GPIO_DATA); + data &= ~(0x1 << 15); mdelay(10); - dwData |= (0x1 << 15); - hw_write_20kx(hw, GPIO_DATA, dwData); + data |= (0x1 << 15); + hw_write_20kx(hw, GPIO_DATA, data); mdelay(50); /* Set the master mode (256fs) */ if (1 == info->msr) { - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02), MAKE_WM8775_DATA(0x02)); } else if (2 == info->msr) { - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A), MAKE_WM8775_DATA(0x0A)); } else { printk(KERN_ALERT "ctxfi: Invalid master sampling " @@ -1785,35 +1793,35 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) } /* Configure GPIO bit 14 change to line-in/mic-in */ - dwCtl = hw_read_20kx(hw, GPIO_CTRL); - dwCtl |= 0x1<<14; - hw_write_20kx(hw, GPIO_CTRL, dwCtl); + ctl = hw_read_20kx(hw, GPIO_CTRL); + ctl |= 0x1 << 14; + hw_write_20kx(hw, GPIO_CTRL, ctl); /* Check using Mic-in or Line-in */ - dwData = hw_read_20kx(hw, GPIO_DATA); + data = hw_read_20kx(hw, GPIO_DATA); - if (dwMux == 1) { + if (mux == 1) { /* Configures GPIO data to select Mic-in */ - dwData |= 0x1<<14; - hw_write_20kx(hw, GPIO_DATA, dwData); + data |= 0x1 << 14; + hw_write_20kx(hw, GPIO_DATA, data); - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), MAKE_WM8775_DATA(0x101)); /* Mic-in */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ - } else if (dwMux == 2) { + } else if (mux == 2) { /* Configures GPIO data to select Line-in */ - dwData &= ~(0x1<<14); - hw_write_20kx(hw, GPIO_DATA, dwData); + data &= ~(0x1 << 14); + hw_write_20kx(hw, GPIO_DATA, data); /* Setup ADC */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102), MAKE_WM8775_DATA(0x102)); /* Line-in */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF), MAKE_WM8775_DATA(0xCF)); /* No boost */ - I2CWrite(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), + hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF), MAKE_WM8775_DATA(0xCF)); /* No boost */ } else { printk(KERN_ALERT "ctxfi: ERROR!!! Invalid input mux!!!\n"); @@ -1824,7 +1832,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) return 0; error: - I2CUninit(hw); + hw20k2_i2c_uninit(hw); return err; } @@ -2106,15 +2114,15 @@ static struct hw ct20k2_preset __devinitdata = { int __devinit create_20k2_hw_obj(struct hw **rhw) { - struct hw *hw; + struct hw20k2 *hw20k2; *rhw = NULL; - hw = kzalloc(sizeof(*hw), GFP_KERNEL); - if (NULL == hw) + hw20k2 = kzalloc(sizeof(*hw20k2), GFP_KERNEL); + if (!hw20k2) return -ENOMEM; - *hw = ct20k2_preset; - *rhw = hw; + hw20k2->hw = ct20k2_preset; + *rhw = &hw20k2->hw; return 0; } -- cgit v1.1 From 806d31d73843a018cb239fc16ba53d82f4d8dc33 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 15:50:42 +0200 Subject: ALSA: emu10k1 - Fix minimum periods for efx playback EFX playback stream should have periods_min = 2 to avoid the buffer position overflow (due to restrictions of the pcm-indirect helper). Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emupcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 78f62fd..55b83ef 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1736,7 +1736,7 @@ static struct snd_pcm_hardware snd_emu10k1_fx8010_playback = .buffer_bytes_max = (128*1024), .period_bytes_min = 1024, .period_bytes_max = (128*1024), - .periods_min = 1, + .periods_min = 2, .periods_max = 1024, .fifo_size = 0, }; -- cgit v1.1 From c00701101b82f2bc61dfc259748ec6e5288af6a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 15:58:48 +0200 Subject: ALSA: pcm - A helper function to compose PCM stream name for debug prints Use a common helper function for the PCM stream name displayed in XRUN and buffer-pointer debug prints. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index adb306f..2288fa0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -136,6 +136,16 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram dump_stack(); \ } while (0) +static void pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len) +{ + snprintf(name, len, "pcmC%dD%d%c:%d", + substream->pcm->card->number, + substream->pcm->device, + substream->stream ? 'c' : 'p', + substream->number); +} + static void xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -144,10 +154,9 @@ static void xrun(struct snd_pcm_substream *substream) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); if (xrun_debug(substream, 1)) { - snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p'); + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd(KERN_DEBUG "XRUN: %s\n", name); dump_stack_on_xrun(substream); } } @@ -163,9 +172,11 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, return pos; /* XRUN */ if (pos >= runtime->buffer_size) { if (printk_ratelimit()) { - snd_printd(KERN_ERR "BUG: stream = %i, pos = 0x%lx, " + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " "buffer size = 0x%lx, period size = 0x%lx\n", - substream->stream, pos, runtime->buffer_size, + name, pos, runtime->buffer_size, runtime->period_size); } pos = 0; -- cgit v1.1 From 64a8be74357477558183b43156c5536b642de134 Mon Sep 17 00:00:00 2001 From: David Heidelberger Date: Mon, 8 Jun 2009 16:15:18 +0200 Subject: ALSA: hda - Add 7.1 support for MSI GX620 Added 7.1 support for MSI GX620 and jack quirk. Reference: kernel bug#13430 http://bugzilla.kernel.org/show_bug.cgi?id=13430 Signed-off-by: David Heidelberger Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 +- sound/pci/hda/patch_realtek.c | 115 +++++++++++++++++++++++++-- 2 files changed, 109 insertions(+), 9 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 54b0805..de8e10a 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -143,7 +143,8 @@ ALC883/888 medion Medion Laptops medion-md2 Medion MD2 targa-dig Targa/MSI - targa-2ch-dig Targs/MSI with 2-channel + targa-2ch-dig Targa/MSI with 2-channel + targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) lenovo-101e Lenovo 101E lenovo-nb0763 Lenovo NB0763 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ab1b92..cfe4808 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -220,6 +220,7 @@ enum { ALC883_6ST_DIG, ALC883_TARGA_DIG, ALC883_TARGA_2ch_DIG, + ALC883_TARGA_8ch_DIG, ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, @@ -2722,6 +2723,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = { /* Enable GPIO mask and set output */ #define alc880_gpio1_init_verbs alc_gpio1_init_verbs #define alc880_gpio2_init_verbs alc_gpio2_init_verbs +#define alc880_gpio3_init_verbs alc_gpio3_init_verbs /* Clevo m520g init */ static struct hda_verb alc880_pin_clevo_init_verbs[] = { @@ -7719,6 +7721,73 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { { 6, alc883_3ST_ch6_init }, }; + +/* + * 2ch mode + */ +static struct hda_verb alc883_4ST_ch2_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PhIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_4ST_ch4_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_4ST_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc883_4ST_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { + { 2, alc883_4ST_ch2_init }, + { 4, alc883_4ST_ch4_init }, + { 6, alc883_4ST_ch6_init }, + { 8, alc883_4ST_ch8_init }, +}; + + /* * 2ch mode */ @@ -8355,14 +8424,24 @@ static struct hda_verb alc883_tagra_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ +/* Connect Line-Out side jack (SPDIF) to Side */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, +/* Connect Mic jack to CLFE */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect Line-in jack to Surround */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect HP out jack to Front */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -8607,8 +8686,8 @@ static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); @@ -8895,6 +8974,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_6ST_DIG] = "6stack-dig", [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", + [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", @@ -8979,6 +9059,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), @@ -9108,6 +9189,24 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_tagra_unsol_event, .init_hook = alc883_tagra_init_hook, }, + [ALC883_TARGA_8ch_DIG] = { + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_tagra_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes), + .channel_mode = alc883_4ST_8ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_tagra_unsol_event, + .init_hook = alc883_tagra_automute, + }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, /* On TravelMate laptops, GPIO 0 enables the internal speaker -- cgit v1.1 From f03ecf50534a81b06544c58a713076d59d54baf9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 16:38:17 +0200 Subject: ALSA: hda - Fix the previous tagra-8ch patch - Fix a typo in the patch - Adapted to follow the recent change for unsol event handling Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfe4808..337d2a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7726,7 +7726,7 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { * 2ch mode */ static struct hda_verb alc883_4ST_ch2_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PhIN_OUT }, + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -9205,7 +9205,7 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc883_tagra_unsol_event, - .init_hook = alc883_tagra_automute, + .init_hook = alc883_tagra_init_hook, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, -- cgit v1.1 From 9470195a9cd13e6d90221b8b5d897e9232da8d28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 18:10:32 +0200 Subject: ALSA: ctxfi - Clean up probe routines Clean up probe routines and model detection routines so that the driver won't call and check the PCI subsystem id at each time. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 132 +++++++++++++++++++------------------------ sound/pci/ctxfi/ctatc.h | 17 ++---- sound/pci/ctxfi/ctdaio.c | 6 +- sound/pci/ctxfi/cthardware.c | 31 +++------- sound/pci/ctxfi/cthardware.h | 20 ++++++- sound/pci/ctxfi/cthw20k1.c | 88 +++++++++++------------------ sound/pci/ctxfi/xfi.c | 15 +++-- 7 files changed, 135 insertions(+), 174 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 7898a37..002a70e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -39,29 +39,40 @@ | (0x10 << 16) \ | ((IEC958_AES3_CON_FS_48000) << 24)) -static const struct ct_atc_chip_sub_details atc_sub_details[NUM_CTCARDS] = { - [CTSB0760] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB0760, - .nm_model = "SB076x"}, - [CTHENDRIX] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, - .nm_model = "Hendrix"}, - [CTSB08801] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB08801, - .nm_model = "SB0880"}, - [CTSB08802] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB08802, - .nm_model = "SB0880"}, - [CTSB08803] = {.subsys = PCI_SUBDEVICE_ID_CREATIVE_SB08803, - .nm_model = "SB0880"} +static struct snd_pci_quirk __devinitdata subsys_20k1_list[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0022, "SB055x", CTSB055X), + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x002f, "SB055x", CTSB055X), + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0029, "SB073x", CTSB073X), + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0031, "SB073x", CTSB073X), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0x6000, + PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, "UAA", CTUAA), + SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_CREATIVE, + "Unknown", CT20K1_UNKNOWN), + { } /* terminator */ }; -static struct ct_atc_chip_details atc_chip_details[] = { - {.vendor = PCI_VENDOR_ID_CREATIVE, - .device = PCI_DEVICE_ID_CREATIVE_20K1, - .sub_details = NULL, - .nm_card = "X-Fi 20k1"}, - {.vendor = PCI_VENDOR_ID_CREATIVE, - .device = PCI_DEVICE_ID_CREATIVE_20K2, - .sub_details = atc_sub_details, - .nm_card = "X-Fi 20k2"}, - {} /* terminator */ +static struct snd_pci_quirk __devinitdata subsys_20k2_list[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB0760, + "SB0760", CTSB0760), + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08801, + "SB0880", CTSB0880), + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08802, + "SB0880", CTSB0880), + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08803, + "SB0880", CTSB0880), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0x6000, + PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, "UAA", CTHENDRIX), + { } /* terminator */ +}; + +static const char *ct_subsys_name[NUM_CTCARDS] = { + [CTSB055X] = "SB055x", + [CTSB073X] = "SB073x", + [CTSB0760] = "SB076x", + [CTUAA] = "UAA", + [CT20K1_UNKNOWN] = "Unknown", + [CTHENDRIX] = "Hendrix", + [CTSB0880] = "SB0880", }; static struct { @@ -1208,62 +1219,39 @@ static int atc_dev_free(struct snd_device *dev) static int __devinit atc_identify_card(struct ct_atc *atc) { - u16 subsys; - u8 revision; - struct pci_dev *pci = atc->pci; - const struct ct_atc_chip_details *d; - enum CTCARDS i; - - subsys = pci->subsystem_device; - revision = pci->revision; - atc->chip_details = NULL; - atc->model = NUM_CTCARDS; - for (d = atc_chip_details; d->vendor; d++) { - if (d->vendor != pci->vendor || d->device != pci->device) - continue; + const struct snd_pci_quirk *p; + const struct snd_pci_quirk *list; - if (NULL == d->sub_details) { - atc->chip_details = d; - break; - } - for (i = 0; i < NUM_CTCARDS; i++) { - if ((d->sub_details[i].subsys == subsys) || - (((subsys & 0x6000) == 0x6000) && - ((d->sub_details[i].subsys & 0x6000) == 0x6000))) { - atc->model = i; - break; - } - } - if (i >= NUM_CTCARDS) - continue; - - atc->chip_details = d; + switch (atc->chip_type) { + case ATC20K1: + atc->chip_name = "20K1"; + list = subsys_20k1_list; + break; + case ATC20K2: + atc->chip_name = "20K2"; + list = subsys_20k2_list; break; - /* not take revision into consideration now */ + default: + return -ENOENT; } - if (!d->vendor) + p = snd_pci_quirk_lookup(atc->pci, list); + if (!p) return -ENOENT; - + atc->model = p->value; + atc->model_name = ct_subsys_name[atc->model]; + snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n", + atc->chip_name, atc->model_name, + atc->pci->subsystem_vendor, + atc->pci->subsystem_device); return 0; } int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc) { enum CTALSADEVS i; - struct hw *hw = atc->hw; int err; - switch (hw->get_chip_type(hw)) { - case ATC20K1: - alsa_dev_funcs[MIXER].public_name = "20K1"; - break; - case ATC20K2: - alsa_dev_funcs[MIXER].public_name = "20K2"; - break; - default: - alsa_dev_funcs[MIXER].public_name = "Unknown"; - break; - } + alsa_dev_funcs[MIXER].public_name = atc->chip_name; for (i = 0; i < NUM_CTALSADEVS; i++) { if (NULL == alsa_dev_funcs[i].create) @@ -1287,7 +1275,7 @@ static int __devinit atc_create_hw_devs(struct ct_atc *atc) struct card_conf info = {0}; int i, err; - err = create_hw_obj(atc->pci, &hw); + err = create_hw_obj(atc->pci, atc->chip_type, atc->model, &hw); if (err) { printk(KERN_ERR "Failed to create hw obj!!!\n"); return err; @@ -1328,7 +1316,6 @@ static int __devinit atc_get_resources(struct ct_atc *atc) struct sum_desc sum_dsc = {0}; struct sum_mgr *sum_mgr; int err, i; - unsigned short subsys_id; atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); if (NULL == atc->daios) @@ -1359,13 +1346,10 @@ static int __devinit atc_get_resources(struct ct_atc *atc) } atc->n_daio++; } - subsys_id = atc->pci->subsystem_device; - if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { - /* SB073x cards */ + if (atc->model == CTSB073X) da_desc.type = SPDIFI1; - } else { + else da_desc.type = SPDIFIO; - } err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); if (err) { @@ -1555,7 +1539,8 @@ static struct ct_atc atc_preset __devinitdata = { */ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, - unsigned int rsr, unsigned int msr, struct ct_atc **ratc) + unsigned int rsr, unsigned int msr, + int chip_type, struct ct_atc **ratc) { struct ct_atc *atc; static struct snd_device_ops ops = { @@ -1576,6 +1561,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, atc->pci = pci; atc->rsr = rsr; atc->msr = msr; + atc->chip_type = chip_type; spin_lock_init(&atc->atc_lock); diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 04459aa..a033472 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -37,15 +37,6 @@ enum CTALSADEVS { /* Types of alsa devices */ NUM_CTALSADEVS /* This should always be the last */ }; -enum CTCARDS { - CTSB0760, - CTHENDRIX, - CTSB08801, - CTSB08802, - CTSB08803, - NUM_CTCARDS /* This should always be the last */ -}; - struct ct_atc_chip_sub_details { u16 subsys; const char *nm_model; @@ -89,8 +80,10 @@ struct ct_atc { unsigned int msr; /* master sample rate in rsr */ unsigned int pll_rate; /* current rate of Phase Lock Loop */ - const struct ct_atc_chip_details *chip_details; - enum CTCARDS model; + int chip_type; + int model; + const char *chip_name; + const char *model_name; struct ct_vm *vm; /* device virtual memory manager for this card */ int (*map_audio_buffer)(struct ct_atc *atc, struct ct_atc_pcm *apcm); @@ -147,7 +140,7 @@ struct ct_atc { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, - unsigned int rsr, unsigned int msr, + unsigned int rsr, unsigned int msr, int chip_type, struct ct_atc **ratc); int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index befead4..082e35c 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -116,7 +116,7 @@ static struct rsc_ops daio_in_rsc_ops_20k2 = { static unsigned int daio_device_index(enum DAIOTYP type, struct hw *hw) { - switch (hw->get_chip_type(hw)) { + switch (hw->chip_type) { case ATC20K1: switch (type) { case SPDIFOO: return 0; @@ -343,7 +343,7 @@ static int daio_rsc_init(struct daio *daio, int err; unsigned int idx_l, idx_r; - switch (((struct hw *)hw)->get_chip_type(hw)) { + switch (((struct hw *)hw)->chip_type) { case ATC20K1: idx_l = idx_20k1[desc->type].left; idx_r = idx_20k1[desc->type].right; @@ -367,7 +367,7 @@ static int daio_rsc_init(struct daio *daio, if (desc->type <= DAIO_OUT_MAX) { daio->rscl.ops = daio->rscr.ops = &daio_out_rsc_ops; } else { - switch (((struct hw *)hw)->get_chip_type(hw)) { + switch (((struct hw *)hw)->chip_type) { case ATC20K1: daio->rscl.ops = daio->rscr.ops = &daio_in_rsc_ops_20k1; break; diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c index 5ec6813..8e64f48 100644 --- a/sound/pci/ctxfi/cthardware.c +++ b/sound/pci/ctxfi/cthardware.c @@ -20,34 +20,16 @@ #include "cthw20k2.h" #include -static enum CHIPTYP __devinitdata get_chip_type(struct hw *hw) -{ - enum CHIPTYP type; - - switch (hw->pci->device) { - case 0x0005: /* 20k1 device */ - type = ATC20K1; - break; - case 0x000B: /* 20k2 device */ - type = ATC20K2; - break; - default: - type = ATCNONE; - break; - } - - return type; -} - -int __devinit create_hw_obj(struct pci_dev *pci, struct hw **rhw) +int __devinit create_hw_obj(struct pci_dev *pci, enum CHIPTYP chip_type, + enum CTCARDS model, struct hw **rhw) { int err; - switch (pci->device) { - case 0x0005: /* 20k1 device */ + switch (chip_type) { + case ATC20K1: err = create_20k1_hw_obj(rhw); break; - case 0x000B: /* 20k2 device */ + case ATC20K2: err = create_20k2_hw_obj(rhw); break; default: @@ -58,7 +40,8 @@ int __devinit create_hw_obj(struct pci_dev *pci, struct hw **rhw) return err; (*rhw)->pci = pci; - (*rhw)->get_chip_type = get_chip_type; + (*rhw)->chip_type = chip_type; + (*rhw)->model = model; return 0; } diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 8f11644..4a8e04f 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -27,6 +27,19 @@ enum CHIPTYP { ATCNONE }; +enum CTCARDS { + /* 20k1 models */ + CTSB055X, + CTSB073X, + CTUAA, + CT20K1_UNKNOWN, + /* 20k2 models */ + CTSB0760, + CTHENDRIX, + CTSB0880, + NUM_CTCARDS /* This should always be the last */ +}; + /* Type of input source for ADC */ enum ADCSRC{ ADC_MICIN, @@ -48,7 +61,6 @@ struct hw { int (*card_init)(struct hw *hw, struct card_conf *info); int (*card_stop)(struct hw *hw); int (*pll_init)(struct hw *hw, unsigned int rsr); - enum CHIPTYP (*get_chip_type)(struct hw *hw); int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source); int (*select_adc_source)(struct hw *hw, enum ADCSRC source); int (*have_digit_io_switch)(struct hw *hw); @@ -156,9 +168,13 @@ struct hw { int irq; unsigned long io_base; unsigned long mem_base; + + enum CHIPTYP chip_type; + enum CTCARDS model; }; -int create_hw_obj(struct pci_dev *pci, struct hw **rhw); +int create_hw_obj(struct pci_dev *pci, enum CHIPTYP chip_type, + enum CTCARDS model, struct hw **rhw); int destroy_hw_obj(struct hw *hw); unsigned int get_field(unsigned int data, unsigned int field); diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 38b87b6..5d58650 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1432,11 +1432,9 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info) { u32 data; u16 gpioorg; - u16 subsys_id; unsigned int ret; - pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); - if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { + if (hw->model == CTSB055X) { /* SB055x, unmute outputs */ gpioorg = (u16)hw_read_20kx(hw, GPIO); gpioorg &= 0xffbf; /* set GPIO6 to low */ @@ -1538,19 +1536,14 @@ static int is_adc_input_selected_hendrix(struct hw *hw, enum ADCSRC type) static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) { - u16 subsys_id; - - pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); - if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { - /* SB055x cards */ + switch (hw->model) { + case CTSB055X: return is_adc_input_selected_SB055x(hw, type); - } else if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { - /* SB073x cards */ + case CTSB073X: return is_adc_input_selected_hendrix(hw, type); - } else if ((subsys_id & 0xf000) == 0x6000) { - /* Vista compatible cards */ + case CTHENDRIX: return is_adc_input_selected_hendrix(hw, type); - } else { + default: return is_adc_input_selected_SBx(hw, type); } } @@ -1692,20 +1685,17 @@ adc_input_select_hendrix(struct hw *hw, enum ADCSRC type, unsigned char boost) static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) { - u16 subsys_id; - - pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); - if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { - /* SB055x cards */ - return adc_input_select_SB055x(hw, type, (ADC_MICIN == type)); - } else if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { - /* SB073x cards */ - return adc_input_select_hendrix(hw, type, (ADC_MICIN == type)); - } else if ((subsys_id & 0xf000) == 0x6000) { - /* Vista compatible cards */ - return adc_input_select_hendrix(hw, type, (ADC_MICIN == type)); - } else { - return adc_input_select_SBx(hw, type, (ADC_MICIN == type)); + int state = type == ADC_MICIN; + + switch (hw->model) { + case CTSB055X: + return adc_input_select_SB055x(hw, type, state); + case CTSB073X: + return adc_input_select_hendrix(hw, type, state); + case CTHENDRIX: + return adc_input_select_hendrix(hw, type, state); + default: + return adc_input_select_SBx(hw, type, state); } } @@ -1781,28 +1771,16 @@ static int adc_init_SBx(struct hw *hw, int input, int mic20db) static int hw_adc_init(struct hw *hw, const struct adc_conf *info) { - int err; - u16 subsys_id; - - pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); - if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { - /* Sb055x card */ - err = adc_init_SB055x(hw, info->input, info->mic20db); - } else { - err = adc_init_SBx(hw, info->input, info->mic20db); - } - - return err; + if (hw->model == CTSB055X) + return adc_init_SB055x(hw, info->input, info->mic20db); + else + return adc_init_SBx(hw, info->input, info->mic20db); } static int hw_have_digit_io_switch(struct hw *hw) { - u16 subsys_id; - - pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); /* SB073x and Vista compatible cards have no digit IO switch */ - return !((subsys_id == 0x0029) || (subsys_id == 0x0031) - || ((subsys_id & 0xf000) == 0x6000)); + return !(hw->model == CTSB073X || hw->model == CTHENDRIX); } #define CTLBITS(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d)) @@ -1918,7 +1896,6 @@ static int hw_card_start(struct hw *hw) { int err; struct pci_dev *pci = hw->pci; - u16 subsys_id; err = pci_enable_device(pci); if (err < 0) @@ -1939,8 +1916,7 @@ static int hw_card_start(struct hw *hw) goto error1; /* Switch to X-Fi mode from UAA mode if neeeded */ - pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsys_id); - if ((0x5 == pci->device) && (0x6000 == (subsys_id & 0x6000))) { + if (hw->model == CTHENDRIX) { err = uaa_to_xfi(pci); if (err) goto error2; @@ -2004,7 +1980,6 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) { int err; unsigned int gctl; - u16 subsys_id; u32 data; struct dac_conf dac_info = {0}; struct adc_conf adc_info = {0}; @@ -2044,19 +2019,20 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) hw_write_20kx(hw, SRCIP, 0); mdelay(30); - pci_read_config_word(hw->pci, PCI_SUBSYSTEM_ID, &subsys_id); /* Detect the card ID and configure GPIO accordingly. */ - if ((subsys_id == 0x0022) || (subsys_id == 0x002F)) { - /* SB055x cards */ + switch (hw->model) { + case CTSB055X: hw_write_20kx(hw, GPIOCTL, 0x13fe); - } else if ((subsys_id == 0x0029) || (subsys_id == 0x0031)) { - /* SB073x cards */ + break; + case CTSB073X: hw_write_20kx(hw, GPIOCTL, 0x00e6); - } else if ((subsys_id & 0xf000) == 0x6000) { - /* Vista compatible cards */ + break; + case CTHENDRIX: /* Vista compatible cards */ hw_write_20kx(hw, GPIOCTL, 0x00c2); - } else { + break; + default: hw_write_20kx(hw, GPIOCTL, 0x01e6); + break; } trn_info.vm_pgt_phys = info->vm_pgt_phys; diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 279dac6..2d3dd89 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -15,6 +15,7 @@ #include #include #include "ctatc.h" +#include "cthardware.h" MODULE_AUTHOR("Creative Technology Ltd"); MODULE_DESCRIPTION("X-Fi driver version 1.03"); @@ -41,8 +42,12 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); static struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ - { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1) }, - { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K2) }, + { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), + .driver_data = ATC20K1, + }, + { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K2), + .driver_data = ATC20K2, + }, { 0, } }; MODULE_DEVICE_TABLE(pci, ct_pci_dev_ids); @@ -79,7 +84,8 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) "1 and 2, Value 2 is assumed.\n"); multiple = 2; } - err = ct_atc_create(card, pci, reference_rate, multiple, &atc); + err = ct_atc_create(card, pci, reference_rate, multiple, + pci_id->driver_data, &atc); if (err < 0) goto error; @@ -92,7 +98,8 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) strcpy(card->driver, "SB-XFi"); strcpy(card->shortname, "Creative X-Fi"); - strcpy(card->longname, "Creative ALSA Driver X-Fi"); + snprintf(card->longname, sizeof(card->longname), "%s %s %s", + card->shortname, atc->chip_name, atc->model_name); err = snd_card_register(card); if (err < 0) -- cgit v1.1 From 09521d2e3edd0bf02b66e5b8c13f1559f2d6958a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 18:29:38 +0200 Subject: ALSA: ctxfi - Fix wrong model id for UAA CTUAA should be checked instead of CTHENDRIX. The latter is for 20k2 chip. Also, fixed the detection of UAA/HENDRIX models by fixing the mask bits. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 9 +++++---- sound/pci/ctxfi/cthw20k1.c | 10 +++++----- 2 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 002a70e..4e25b24 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -44,8 +44,8 @@ static struct snd_pci_quirk __devinitdata subsys_20k1_list[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x002f, "SB055x", CTSB055X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0029, "SB073x", CTSB073X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0031, "SB073x", CTSB073X), - SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0x6000, - PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, "UAA", CTUAA), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0xf000, 0x6000, + "UAA", CTUAA), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_CREATIVE, "Unknown", CT20K1_UNKNOWN), { } /* terminator */ @@ -60,8 +60,9 @@ static struct snd_pci_quirk __devinitdata subsys_20k2_list[] = { "SB0880", CTSB0880), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08803, "SB0880", CTSB0880), - SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0x6000, - PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, "UAA", CTHENDRIX), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0xf000, + PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, "HENDRIX", + CTHENDRIX), { } /* terminator */ }; diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 5d58650..cb69d9d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1541,7 +1541,7 @@ static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) return is_adc_input_selected_SB055x(hw, type); case CTSB073X: return is_adc_input_selected_hendrix(hw, type); - case CTHENDRIX: + case CTUAA: return is_adc_input_selected_hendrix(hw, type); default: return is_adc_input_selected_SBx(hw, type); @@ -1692,7 +1692,7 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) return adc_input_select_SB055x(hw, type, state); case CTSB073X: return adc_input_select_hendrix(hw, type, state); - case CTHENDRIX: + case CTUAA: return adc_input_select_hendrix(hw, type, state); default: return adc_input_select_SBx(hw, type, state); @@ -1780,7 +1780,7 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) static int hw_have_digit_io_switch(struct hw *hw) { /* SB073x and Vista compatible cards have no digit IO switch */ - return !(hw->model == CTSB073X || hw->model == CTHENDRIX); + return !(hw->model == CTSB073X || hw->model == CTUAA); } #define CTLBITS(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d)) @@ -1916,7 +1916,7 @@ static int hw_card_start(struct hw *hw) goto error1; /* Switch to X-Fi mode from UAA mode if neeeded */ - if (hw->model == CTHENDRIX) { + if (hw->model == CTUAA) { err = uaa_to_xfi(pci); if (err) goto error2; @@ -2027,7 +2027,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) case CTSB073X: hw_write_20kx(hw, GPIOCTL, 0x00e6); break; - case CTHENDRIX: /* Vista compatible cards */ + case CTUAA: hw_write_20kx(hw, GPIOCTL, 0x00c2); break; default: -- cgit v1.1 From 92a43793a908f395dc687e6c5fc90d63f999d6d5 Mon Sep 17 00:00:00 2001 From: Dan Allongo Date: Mon, 8 Jun 2009 11:21:52 -0400 Subject: ALSA: usb - Add boot quirk for C-Media 6206 USB Audio Added boot quirk for C-Media CM6206 device in snd_usb_audio_probe. The function snd_usb_cm6206_boot_quirk sets up six internal 16-bit registers in order to initialize the device. Values for the registers came from sniffing USB traffic under Windows since only four of the six are documented in the datasheet for CM106 and some reserved bits were also being set. [Minor coding-style fixes by tiwai] Signed-off-by: Dan Allongo Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index a6b8848..e871217 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3279,6 +3279,25 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +/* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + int err, reg; + int val[] = {0x200c, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + + for (reg = 0; reg < ARRAY_SIZE(val); reg++) { + err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); + if (err < 0) + return err; + } + + return err; +} /* * Setup quirks @@ -3565,6 +3584,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ -- cgit v1.1 From 04145f2bfbc3ac4dbb71fa085b82380444d50b4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Jun 2009 21:31:50 +0200 Subject: ALSA: ctxfi - Add use_system_timer module option Added use_system_timer module option to force to use the system timer instead of emu20k1 timer irq for debugging. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/cttimer.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index ec869a4..779c6c3 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -8,12 +8,17 @@ #include #include +#include #include #include #include "ctatc.h" #include "cthardware.h" #include "cttimer.h" +static int use_system_timer; +MODULE_PARM_DESC(use_system_timer, "Foce to use system-timer"); +module_param(use_system_timer, bool, S_IRUGO); + struct ct_timer_ops { void (*init)(struct ct_timer_instance *); void (*prepare)(struct ct_timer_instance *); @@ -390,8 +395,6 @@ void ct_timer_instance_free(struct ct_timer_instance *ti) * timer manager */ -#define USE_SYSTEM_TIMER 0 - static void ct_timer_interrupt(void *data, unsigned int status) { struct ct_timer *timer = data; @@ -415,7 +418,7 @@ struct ct_timer *ct_timer_new(struct ct_atc *atc) INIT_LIST_HEAD(&atimer->running_head); atimer->atc = atc; hw = atc->hw; - if (!USE_SYSTEM_TIMER && hw->set_timer_irq) { + if (!use_system_timer && hw->set_timer_irq) { snd_printd(KERN_INFO "ctxfi: Use xfi-native timer\n"); atimer->ops = &ct_xfitimer_ops; hw->irq_callback_data = atimer; -- cgit v1.1 From c399f3be82e1249221a7f410855ab1aa747fa82b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Jun 2009 08:16:20 +0200 Subject: ALSA: ctxfi - Add missing start check in atc_pcm_playback_start() Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 4e25b24..799eb98 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -392,6 +392,10 @@ static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm) unsigned int max_cisz; struct src *src = apcm->src; + if (apcm->started) + return 0; + apcm->started = 1; + max_cisz = src->multi * src->rsc.msr; max_cisz = 0x80 * (max_cisz < 8 ? max_cisz : 8); -- cgit v1.1 From 5242bc7613311aa1a3d5ed41e9cf81015b65563f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Jun 2009 08:17:14 +0200 Subject: ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callbacks The SRC instances may not exist when PCM pointer callback is called at the state before initialization is finished. Add the NULL check just to be sure. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 799eb98..e54006e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -445,6 +445,8 @@ atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) u32 size, max_cisz; int position; + if (!src) + return 0; position = src->ops->get_ca(src); size = apcm->vm_block->size; @@ -782,6 +784,8 @@ atc_pcm_capture_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) { struct src *src = apcm->src; + if (!src) + return 0; return src->ops->get_ca(src) - apcm->vm_block->addr; } -- cgit v1.1 From a5990dc5b96f537618b0f057c8723a6a0b0cdc74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Jun 2009 08:19:02 +0200 Subject: ALSA: ctxfi - Clear PCM resources at hw_params and hw_free Currently the PCM resources are allocated only once and ever in prepare callback, assuming that the PCM parameters are never changed. But it's not true. This patch adds the call of atc->pcm_release_resources() at hw_params and hw_free callbacks to assure that the PCM setup is done correctly for each h/w parameter changes. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 4 ++-- sound/pci/ctxfi/ctpcm.c | 16 +++++++++++++++- 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index e54006e..80fb2ba 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -523,7 +523,7 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct src_node_conf_t src_node_conf[2] = {{0} }; /* first release old resources */ - atc->pcm_release_resources(atc, apcm); + atc_pcm_release_resources(atc, apcm); /* The numbers of converting SRCs and SRCIMPs should be determined * by pitch value. */ @@ -802,7 +802,7 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc, unsigned int pitch, rsr = atc->pll_rate; /* first release old resources */ - atc->pcm_release_resources(atc, apcm); + atc_pcm_release_resources(atc, apcm); /* Get SRC resource */ desc.multi = apcm->substream->runtime->channels; diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 870fa17..9e5c0c4 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -176,12 +176,26 @@ static int ct_pcm_playback_close(struct snd_pcm_substream *substream) static int ct_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(substream, + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct ct_atc_pcm *apcm = substream->runtime->private_data; + int err; + + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + if (err < 0) + return err; + /* clear previous resources */ + atc->pcm_release_resources(atc, apcm); + return err; } static int ct_pcm_hw_free(struct snd_pcm_substream *substream) { + struct ct_atc *atc = snd_pcm_substream_chip(substream); + struct ct_atc_pcm *apcm = substream->runtime->private_data; + + /* clear previous resources */ + atc->pcm_release_resources(atc, apcm); /* Free snd-allocated pages */ return snd_pcm_lib_free_pages(substream); } -- cgit v1.1 From 236e6723bedb483b2ebf73ada01e2c853c5cac01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jun 2009 13:55:34 +0100 Subject: ASoC: Fix lm4857 control Commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf changed semantics of private_value member of kcontrol. This resulted in inability to control amplifier and subsequently in very low output volume. Tested-by: Johannes Schauer Signed-off-by: Paul Fertser Signed-off-by: Mark Brown --- sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 289fadf..906709e 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -345,9 +345,11 @@ static void lm4857_write_regs(void) static int lm4857_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int reg = kcontrol->private_value & 0xFF; - int shift = (kcontrol->private_value >> 8) & 0x0F; - int mask = (kcontrol->private_value >> 16) & 0xFF; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; pr_debug("Entered %s\n", __func__); @@ -358,9 +360,11 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, static int lm4857_set_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int reg = kcontrol->private_value & 0xFF; - int shift = (kcontrol->private_value >> 8) & 0x0F; - int mask = (kcontrol->private_value >> 16) & 0xFF; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; if (((lm4857_regs[reg] >> shift) & mask) == ucontrol->value.integer.value[0]) -- cgit v1.1 From 112ac808eb8a953dd356bbbc8322fdd6861e2c75 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 10 Jun 2009 16:39:01 +0200 Subject: ALSA: sound/ps3: Fix checkpatch issues Signed-off-by: Geert Uytterhoeven Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 32 +++++++++++++++----------------- 1 file changed, 15 insertions(+), 17 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index f361c26..d660b0f 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -165,8 +165,7 @@ static const struct snd_pcm_hardware snd_ps3_pcm_hw = { .fifo_size = PS3_AUDIO_FIFO_SIZE }; -static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = -{ +static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = { .open = snd_ps3_pcm_open, .close = snd_ps3_pcm_close, .prepare = snd_ps3_pcm_prepare, @@ -183,7 +182,7 @@ static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card, int dma_ch, done, retries, stop_forced = 0; uint32_t status; - for (dma_ch = 0; dma_ch < 8; dma_ch ++) { + for (dma_ch = 0; dma_ch < 8; dma_ch++) { retries = count; do { status = read_reg(PS3_AUDIO_KICK(dma_ch)) & @@ -259,9 +258,7 @@ static void snd_ps3_kick_dma(struct snd_ps3_card_info *card) /* * convert virtual addr to ioif bus addr. */ -static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, - void * paddr, - int ch) +static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, void *paddr, int ch) { return card->dma_start_bus_addr[ch] + (paddr - card->dma_start_vaddr[ch]); @@ -321,7 +318,7 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, spin_lock_irqsave(&card->dma_lock, irqsave); for (ch = 0; ch < 2; ch++) { start_vaddr = card->dma_next_transfer_vaddr[0]; - for (stage = 0; stage < fill_stages; stage ++) { + for (stage = 0; stage < fill_stages; stage++) { dma_ch = stage * 2 + ch; if (silent) dma_addr = card->null_buffer_start_dma_addr; @@ -619,7 +616,7 @@ static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card) PS3_AUDIO_AO_3WMCTRL_ASOEN(2) | PS3_AUDIO_AO_3WMCTRL_ASOEN(3)), 0); - wmb(); /* ensure the hardware sees the change */ + wmb(); /* ensure the hardware sees the change */ /* wait for actually stopped */ retries = 1000; while ((read_reg(PS3_AUDIO_AO_3WMCTRL) & @@ -798,20 +795,20 @@ static struct snd_kcontrol_new spdif_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_default_get, .put = snd_ps3_spdif_default_put, @@ -1020,11 +1017,12 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) * its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2 * PAGE_SIZE is enogh */ - if (!(the_card.null_buffer_start_vaddr = - dma_alloc_coherent(&the_card.ps3_dev->core, - PAGE_SIZE, - &the_card.null_buffer_start_dma_addr, - GFP_KERNEL))) { + the_card.null_buffer_start_vaddr = + dma_alloc_coherent(&the_card.ps3_dev->core, + PAGE_SIZE, + &the_card.null_buffer_start_dma_addr, + GFP_KERNEL); + if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); goto clean_preallocate; } @@ -1148,7 +1146,7 @@ static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL : SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); snd_ps3_kick_dma(card); - card->silent --; + card->silent--; } else { snd_ps3_program_dma(card, (underflow_occured) ? -- cgit v1.1 From cb6492e4a4e68281358510f0ffe2b0c4972ec166 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 10 Jun 2009 16:39:02 +0200 Subject: ALSA: sound/ps3: Restructure driver source Sort includes, and reorder code so we can kill the forward declarations No functional changes Signed-off-by: Geert Uytterhoeven Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 753 ++++++++++++++++++++++++---------------------------- 1 file changed, 354 insertions(+), 399 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index d660b0f..cd9109a 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -18,81 +18,31 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include +#include #include -#include -#include #include +#include +#include + +#include +#include #include #include -#include -#include #include +#include #include -#include -#include -#include -#include + #include +#include #include #include #include -#include "snd_ps3_reg.h" #include "snd_ps3.h" - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("PS3 sound driver"); -MODULE_AUTHOR("Sony Computer Entertainment Inc."); - -/* module entries */ -static int __init snd_ps3_init(void); -static void __exit snd_ps3_exit(void); - -/* ALSA snd driver ops */ -static int snd_ps3_pcm_open(struct snd_pcm_substream *substream); -static int snd_ps3_pcm_close(struct snd_pcm_substream *substream); -static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream); -static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, - int cmd); -static snd_pcm_uframes_t snd_ps3_pcm_pointer(struct snd_pcm_substream - *substream); -static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params); -static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream); - - -/* ps3_system_bus_driver entries */ -static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev); -static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev); - -/* address setup */ -static int snd_ps3_map_mmio(void); -static void snd_ps3_unmap_mmio(void); -static int snd_ps3_allocate_irq(void); -static void snd_ps3_free_irq(void); -static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start); - -/* interrupt handler */ -static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id); - - -/* set sampling rate/format */ -static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream); -/* take effect parameter change */ -static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card); -/* initialize avsetting and take it effect */ -static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card); -/* setup dma */ -static int snd_ps3_program_dma(struct snd_ps3_card_info *card, - enum snd_ps3_dma_filltype filltype); -static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card); - -static dma_addr_t v_to_bus(struct snd_ps3_card_info *, void *vaddr, int ch); +#include "snd_ps3_reg.h" -module_init(snd_ps3_init); -module_exit(snd_ps3_exit); - /* * global */ @@ -165,17 +115,6 @@ static const struct snd_pcm_hardware snd_ps3_pcm_hw = { .fifo_size = PS3_AUDIO_FIFO_SIZE }; -static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = { - .open = snd_ps3_pcm_open, - .close = snd_ps3_pcm_close, - .prepare = snd_ps3_pcm_prepare, - .ioctl = snd_pcm_lib_ioctl, - .trigger = snd_ps3_pcm_trigger, - .pointer = snd_ps3_pcm_pointer, - .hw_params = snd_ps3_pcm_hw_params, - .hw_free = snd_ps3_pcm_hw_free -}; - static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card, int count, int force_stop) { @@ -369,6 +308,71 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, } /* + * Interrupt handler + */ +static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) +{ + + uint32_t port_intr; + int underflow_occured = 0; + struct snd_ps3_card_info *card = dev_id; + + if (!card->running) { + update_reg(PS3_AUDIO_AX_IS, 0); + update_reg(PS3_AUDIO_INTR_0, 0); + return IRQ_HANDLED; + } + + port_intr = read_reg(PS3_AUDIO_AX_IS); + /* + *serial buffer empty detected (every 4 times), + *program next dma and kick it + */ + if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) { + write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0)); + if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) { + write_reg(PS3_AUDIO_AX_IS, port_intr); + underflow_occured = 1; + } + if (card->silent) { + /* we are still in silent time */ + snd_ps3_program_dma(card, + (underflow_occured) ? + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL : + SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); + snd_ps3_kick_dma(card); + card->silent--; + } else { + snd_ps3_program_dma(card, + (underflow_occured) ? + SND_PS3_DMA_FILLTYPE_FIRSTFILL : + SND_PS3_DMA_FILLTYPE_RUNNING); + snd_ps3_kick_dma(card); + snd_pcm_period_elapsed(card->substream); + } + } else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) { + write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0)); + /* + * serial out underflow, but buffer empty not detected. + * in this case, fill fifo with 0 to recover. After + * filling dummy data, serial automatically start to + * consume them and then will generate normal buffer + * empty interrupts. + * If both buffer underflow and buffer empty are occured, + * it is better to do nomal data transfer than empty one + */ + snd_ps3_program_dma(card, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + snd_ps3_program_dma(card, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + } + /* clear interrupt cause */ + return IRQ_HANDLED; +}; + +/* * audio mute on/off * mute_on : 0 output enabled * 1 mute @@ -379,6 +383,142 @@ static int snd_ps3_mute(int mute_on) } /* + * av setting + * NOTE: calling this function may generate audio interrupt. + */ +static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card) +{ + int ret, retries, i; + pr_debug("%s: start\n", __func__); + + ret = ps3av_set_audio_mode(card->avs.avs_audio_ch, + card->avs.avs_audio_rate, + card->avs.avs_audio_width, + card->avs.avs_audio_format, + card->avs.avs_audio_source); + /* + * Reset the following unwanted settings: + */ + + /* disable all 3wire buffers */ + update_mask_reg(PS3_AUDIO_AO_3WMCTRL, + ~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) | + PS3_AUDIO_AO_3WMCTRL_ASOEN(1) | + PS3_AUDIO_AO_3WMCTRL_ASOEN(2) | + PS3_AUDIO_AO_3WMCTRL_ASOEN(3)), + 0); + wmb(); /* ensure the hardware sees the change */ + /* wait for actually stopped */ + retries = 1000; + while ((read_reg(PS3_AUDIO_AO_3WMCTRL) & + (PS3_AUDIO_AO_3WMCTRL_ASORUN(0) | + PS3_AUDIO_AO_3WMCTRL_ASORUN(1) | + PS3_AUDIO_AO_3WMCTRL_ASORUN(2) | + PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) && + --retries) { + udelay(1); + } + + /* reset buffer pointer */ + for (i = 0; i < 4; i++) { + update_reg(PS3_AUDIO_AO_3WCTRL(i), + PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET); + udelay(10); + } + wmb(); /* ensure the hardware actually start resetting */ + + /* enable 3wire#0 buffer */ + update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0)); + + + /* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */ + update_mask_reg(PS3_AUDIO_AO_3WCTRL(0), + ~PS3_AUDIO_AO_3WCTRL_ASODF, + PS3_AUDIO_AO_3WCTRL_ASODF_LSB); + update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0), + ~PS3_AUDIO_AO_SPDCTRL_SPODF, + PS3_AUDIO_AO_SPDCTRL_SPODF_LSB); + /* ensure all the setting above is written back to register */ + wmb(); + /* avsetting driver altered AX_IE, caller must reset it if you want */ + pr_debug("%s: end\n", __func__); + return ret; +} + +/* + * set sampling rate according to the substream + */ +static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) +{ + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + struct snd_ps3_avsetting_info avs; + int ret; + + avs = card->avs; + + pr_debug("%s: called freq=%d width=%d\n", __func__, + substream->runtime->rate, + snd_pcm_format_width(substream->runtime->format)); + + pr_debug("%s: before freq=%d width=%d\n", __func__, + card->avs.avs_audio_rate, card->avs.avs_audio_width); + + /* sample rate */ + switch (substream->runtime->rate) { + case 44100: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K; + break; + case 48000: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K; + break; + case 88200: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K; + break; + case 96000: + avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K; + break; + default: + pr_info("%s: invalid rate %d\n", __func__, + substream->runtime->rate); + return 1; + } + + /* width */ + switch (snd_pcm_format_width(substream->runtime->format)) { + case 16: + avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; + break; + case 24: + avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24; + break; + default: + pr_info("%s: invalid width %d\n", __func__, + snd_pcm_format_width(substream->runtime->format)); + return 1; + } + + memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8); + + if (memcmp(&card->avs, &avs, sizeof(avs))) { + pr_debug("%s: after freq=%d width=%d\n", __func__, + card->avs.avs_audio_rate, card->avs.avs_audio_width); + + card->avs = avs; + snd_ps3_change_avsetting(card); + ret = 0; + } else + ret = 1; + + /* check CS non-audio bit and mute accordingly */ + if (avs.avs_cs_info[0] & 0x02) + ps3av_audio_mute_analog(1); /* mute if non-audio */ + else + ps3av_audio_mute_analog(0); + + return ret; +} + +/* * PCM operators */ static int snd_ps3_pcm_open(struct snd_pcm_substream *substream) @@ -403,6 +543,13 @@ static int snd_ps3_pcm_open(struct snd_pcm_substream *substream) return 0; }; +static int snd_ps3_pcm_close(struct snd_pcm_substream *substream) +{ + /* mute on */ + snd_ps3_mute(1); + return 0; +}; + static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -414,6 +561,13 @@ static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream, return 0; }; +static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream) +{ + int ret; + ret = snd_pcm_lib_free_pages(substream); + return ret; +}; + static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream, unsigned int delay_ms) { @@ -467,287 +621,91 @@ static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream) runtime->dma_area; card->dma_start_bus_addr[SND_PS3_CH_L] = runtime->dma_addr; - card->dma_last_transfer_vaddr[SND_PS3_CH_R] = - card->dma_next_transfer_vaddr[SND_PS3_CH_R] = - card->dma_start_vaddr[SND_PS3_CH_R] = - runtime->dma_area + (runtime->dma_bytes / 2); - card->dma_start_bus_addr[SND_PS3_CH_R] = - runtime->dma_addr + (runtime->dma_bytes / 2); - - pr_debug("%s: vaddr=%p bus=%#llx\n", __func__, - card->dma_start_vaddr[SND_PS3_CH_L], - card->dma_start_bus_addr[SND_PS3_CH_L]); - - } - spin_unlock_irqrestore(&card->dma_lock, irqsave); - - /* ensure the hardware sees the change */ - mb(); - - return 0; -}; - -static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* clear outstanding interrupts */ - update_reg(PS3_AUDIO_AX_IS, 0); - - spin_lock(&card->dma_lock); - { - card->running = 1; - } - spin_unlock(&card->dma_lock); - - snd_ps3_program_dma(card, - SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); - snd_ps3_kick_dma(card); - while (read_reg(PS3_AUDIO_KICK(7)) & - PS3_AUDIO_KICK_STATUS_MASK) { - udelay(1); - } - snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); - snd_ps3_kick_dma(card); - break; - - case SNDRV_PCM_TRIGGER_STOP: - spin_lock(&card->dma_lock); - { - card->running = 0; - } - spin_unlock(&card->dma_lock); - snd_ps3_wait_for_dma_stop(card); - break; - default: - break; - - } - - return ret; -}; - -/* - * report current pointer - */ -static snd_pcm_uframes_t snd_ps3_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); - size_t bytes; - snd_pcm_uframes_t ret; - - spin_lock(&card->dma_lock); - { - bytes = (size_t)(card->dma_last_transfer_vaddr[SND_PS3_CH_L] - - card->dma_start_vaddr[SND_PS3_CH_L]); - } - spin_unlock(&card->dma_lock); - - ret = bytes_to_frames(substream->runtime, bytes * 2); - - return ret; -}; - -static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream) -{ - int ret; - ret = snd_pcm_lib_free_pages(substream); - return ret; -}; - -static int snd_ps3_pcm_close(struct snd_pcm_substream *substream) -{ - /* mute on */ - snd_ps3_mute(1); - return 0; -}; - -static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card) -{ - /* - * avsetting driver seems to never change the followings - * so, init them here once - */ - - /* no dma interrupt needed */ - write_reg(PS3_AUDIO_INTR_EN_0, 0); - - /* use every 4 buffer empty interrupt */ - update_mask_reg(PS3_AUDIO_AX_IC, - PS3_AUDIO_AX_IC_AASOIMD_MASK, - PS3_AUDIO_AX_IC_AASOIMD_EVERY4); - - /* enable 3wire clocks */ - update_mask_reg(PS3_AUDIO_AO_3WMCTRL, - ~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED | - PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED), - 0); - update_reg(PS3_AUDIO_AO_3WMCTRL, - PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT); -} - -/* - * av setting - * NOTE: calling this function may generate audio interrupt. - */ -static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card) -{ - int ret, retries, i; - pr_debug("%s: start\n", __func__); - - ret = ps3av_set_audio_mode(card->avs.avs_audio_ch, - card->avs.avs_audio_rate, - card->avs.avs_audio_width, - card->avs.avs_audio_format, - card->avs.avs_audio_source); - /* - * Reset the following unwanted settings: - */ - - /* disable all 3wire buffers */ - update_mask_reg(PS3_AUDIO_AO_3WMCTRL, - ~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) | - PS3_AUDIO_AO_3WMCTRL_ASOEN(1) | - PS3_AUDIO_AO_3WMCTRL_ASOEN(2) | - PS3_AUDIO_AO_3WMCTRL_ASOEN(3)), - 0); - wmb(); /* ensure the hardware sees the change */ - /* wait for actually stopped */ - retries = 1000; - while ((read_reg(PS3_AUDIO_AO_3WMCTRL) & - (PS3_AUDIO_AO_3WMCTRL_ASORUN(0) | - PS3_AUDIO_AO_3WMCTRL_ASORUN(1) | - PS3_AUDIO_AO_3WMCTRL_ASORUN(2) | - PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) && - --retries) { - udelay(1); - } - - /* reset buffer pointer */ - for (i = 0; i < 4; i++) { - update_reg(PS3_AUDIO_AO_3WCTRL(i), - PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET); - udelay(10); - } - wmb(); /* ensure the hardware actually start resetting */ - - /* enable 3wire#0 buffer */ - update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0)); - - - /* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */ - update_mask_reg(PS3_AUDIO_AO_3WCTRL(0), - ~PS3_AUDIO_AO_3WCTRL_ASODF, - PS3_AUDIO_AO_3WCTRL_ASODF_LSB); - update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0), - ~PS3_AUDIO_AO_SPDCTRL_SPODF, - PS3_AUDIO_AO_SPDCTRL_SPODF_LSB); - /* ensure all the setting above is written back to register */ - wmb(); - /* avsetting driver altered AX_IE, caller must reset it if you want */ - pr_debug("%s: end\n", __func__); - return ret; -} - -static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) -{ - int ret; - pr_debug("%s: start\n", __func__); - card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2; - card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K; - card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; - card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM; - card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL; - memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8); - - ret = snd_ps3_change_avsetting(card); + card->dma_last_transfer_vaddr[SND_PS3_CH_R] = + card->dma_next_transfer_vaddr[SND_PS3_CH_R] = + card->dma_start_vaddr[SND_PS3_CH_R] = + runtime->dma_area + (runtime->dma_bytes / 2); + card->dma_start_bus_addr[SND_PS3_CH_R] = + runtime->dma_addr + (runtime->dma_bytes / 2); - snd_ps3_audio_fixup(card); + pr_debug("%s: vaddr=%p bus=%#llx\n", __func__, + card->dma_start_vaddr[SND_PS3_CH_L], + card->dma_start_bus_addr[SND_PS3_CH_L]); - /* to start to generate SPDIF signal, fill data */ - snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); - snd_ps3_kick_dma(card); - pr_debug("%s: end\n", __func__); - return ret; -} + } + spin_unlock_irqrestore(&card->dma_lock, irqsave); -/* - * set sampling rate according to the substream - */ -static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) + /* ensure the hardware sees the change */ + mb(); + + return 0; +}; + +static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) { struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); - struct snd_ps3_avsetting_info avs; - int ret; - - avs = card->avs; + int ret = 0; - pr_debug("%s: called freq=%d width=%d\n", __func__, - substream->runtime->rate, - snd_pcm_format_width(substream->runtime->format)); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* clear outstanding interrupts */ + update_reg(PS3_AUDIO_AX_IS, 0); - pr_debug("%s: before freq=%d width=%d\n", __func__, - card->avs.avs_audio_rate, card->avs.avs_audio_width); + spin_lock(&card->dma_lock); + { + card->running = 1; + } + spin_unlock(&card->dma_lock); - /* sample rate */ - switch (substream->runtime->rate) { - case 44100: - avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K; - break; - case 48000: - avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K; - break; - case 88200: - avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K; - break; - case 96000: - avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K; + snd_ps3_program_dma(card, + SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + while (read_reg(PS3_AUDIO_KICK(7)) & + PS3_AUDIO_KICK_STATUS_MASK) { + udelay(1); + } + snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); + snd_ps3_kick_dma(card); break; - default: - pr_info("%s: invalid rate %d\n", __func__, - substream->runtime->rate); - return 1; - } - /* width */ - switch (snd_pcm_format_width(substream->runtime->format)) { - case 16: - avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; - break; - case 24: - avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24; + case SNDRV_PCM_TRIGGER_STOP: + spin_lock(&card->dma_lock); + { + card->running = 0; + } + spin_unlock(&card->dma_lock); + snd_ps3_wait_for_dma_stop(card); break; default: - pr_info("%s: invalid width %d\n", __func__, - snd_pcm_format_width(substream->runtime->format)); - return 1; + break; + } - memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8); + return ret; +}; - if (memcmp(&card->avs, &avs, sizeof(avs))) { - pr_debug("%s: after freq=%d width=%d\n", __func__, - card->avs.avs_audio_rate, card->avs.avs_audio_width); +/* + * report current pointer + */ +static snd_pcm_uframes_t snd_ps3_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); + size_t bytes; + snd_pcm_uframes_t ret; - card->avs = avs; - snd_ps3_change_avsetting(card); - ret = 0; - } else - ret = 1; + spin_lock(&card->dma_lock); + { + bytes = (size_t)(card->dma_last_transfer_vaddr[SND_PS3_CH_L] - + card->dma_start_vaddr[SND_PS3_CH_L]); + } + spin_unlock(&card->dma_lock); - /* check CS non-audio bit and mute accordingly */ - if (avs.avs_cs_info[0] & 0x02) - ps3av_audio_mute_analog(1); /* mute if non-audio */ - else - ps3av_audio_mute_analog(0); + ret = bytes_to_frames(substream->runtime, bytes * 2); return ret; -} +}; /* * SPDIF status bits controls @@ -815,6 +773,17 @@ static struct snd_kcontrol_new spdif_ctls[] = { }, }; +static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = { + .open = snd_ps3_pcm_open, + .close = snd_ps3_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ps3_pcm_hw_params, + .hw_free = snd_ps3_pcm_hw_free, + .prepare = snd_ps3_pcm_prepare, + .trigger = snd_ps3_pcm_trigger, + .pointer = snd_ps3_pcm_pointer, +}; + static int snd_ps3_map_mmio(void) { @@ -912,6 +881,52 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) ret); } +static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card) +{ + /* + * avsetting driver seems to never change the followings + * so, init them here once + */ + + /* no dma interrupt needed */ + write_reg(PS3_AUDIO_INTR_EN_0, 0); + + /* use every 4 buffer empty interrupt */ + update_mask_reg(PS3_AUDIO_AX_IC, + PS3_AUDIO_AX_IC_AASOIMD_MASK, + PS3_AUDIO_AX_IC_AASOIMD_EVERY4); + + /* enable 3wire clocks */ + update_mask_reg(PS3_AUDIO_AO_3WMCTRL, + ~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED | + PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED), + 0); + update_reg(PS3_AUDIO_AO_3WMCTRL, + PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT); +} + +static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) +{ + int ret; + pr_debug("%s: start\n", __func__); + card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2; + card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K; + card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; + card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM; + card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL; + memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8); + + ret = snd_ps3_change_avsetting(card); + + snd_ps3_audio_fixup(card); + + /* to start to generate SPDIF signal, fill data */ + snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); + snd_ps3_kick_dma(card); + pr_debug("%s: end\n", __func__); + return ret; +} + static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) { int i, ret; @@ -1113,71 +1128,6 @@ static struct ps3_system_bus_driver snd_ps3_bus_driver_info = { /* - * Interrupt handler - */ -static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) -{ - - uint32_t port_intr; - int underflow_occured = 0; - struct snd_ps3_card_info *card = dev_id; - - if (!card->running) { - update_reg(PS3_AUDIO_AX_IS, 0); - update_reg(PS3_AUDIO_INTR_0, 0); - return IRQ_HANDLED; - } - - port_intr = read_reg(PS3_AUDIO_AX_IS); - /* - *serial buffer empty detected (every 4 times), - *program next dma and kick it - */ - if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) { - write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0)); - if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) { - write_reg(PS3_AUDIO_AX_IS, port_intr); - underflow_occured = 1; - } - if (card->silent) { - /* we are still in silent time */ - snd_ps3_program_dma(card, - (underflow_occured) ? - SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL : - SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); - snd_ps3_kick_dma(card); - card->silent--; - } else { - snd_ps3_program_dma(card, - (underflow_occured) ? - SND_PS3_DMA_FILLTYPE_FIRSTFILL : - SND_PS3_DMA_FILLTYPE_RUNNING); - snd_ps3_kick_dma(card); - snd_pcm_period_elapsed(card->substream); - } - } else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) { - write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0)); - /* - * serial out underflow, but buffer empty not detected. - * in this case, fill fifo with 0 to recover. After - * filling dummy data, serial automatically start to - * consume them and then will generate normal buffer - * empty interrupts. - * If both buffer underflow and buffer empty are occured, - * it is better to do nomal data transfer than empty one - */ - snd_ps3_program_dma(card, - SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); - snd_ps3_kick_dma(card); - snd_ps3_program_dma(card, - SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL); - snd_ps3_kick_dma(card); - } - /* clear interrupt cause */ - return IRQ_HANDLED; -}; - -/* * module/subsystem initialize/terminate */ static int __init snd_ps3_init(void) @@ -1195,10 +1145,15 @@ static int __init snd_ps3_init(void) return ret; } +module_init(snd_ps3_init); static void __exit snd_ps3_exit(void) { ps3_system_bus_driver_unregister(&snd_ps3_bus_driver_info); } +module_exit(snd_ps3_exit); +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("PS3 sound driver"); +MODULE_AUTHOR("Sony Computer Entertainment Inc."); MODULE_ALIAS(PS3_MODULE_ALIAS_SOUND); -- cgit v1.1 From 2233123f2776f29d3ac31df1fd1dc40c44d337a5 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 10 Jun 2009 16:39:03 +0200 Subject: ALSA: sound/ps3: Correct existing and add missing annotations probe functions should be __devinit Signed-off-by: Geert Uytterhoeven Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index cd9109a..53c81a5 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -785,7 +785,7 @@ static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = { }; -static int snd_ps3_map_mmio(void) +static int __devinit snd_ps3_map_mmio(void) { the_card.mapped_mmio_vaddr = ioremap(the_card.ps3_dev->m_region->bus_addr, @@ -807,7 +807,7 @@ static void snd_ps3_unmap_mmio(void) the_card.mapped_mmio_vaddr = NULL; } -static int snd_ps3_allocate_irq(void) +static int __devinit snd_ps3_allocate_irq(void) { int ret; u64 lpar_addr, lpar_size; @@ -865,7 +865,7 @@ static void snd_ps3_free_irq(void) ps3_irq_plug_destroy(the_card.irq_no); } -static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) +static void __devinit snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) { uint64_t val; int ret; @@ -881,7 +881,7 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) ret); } -static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card) +static void __devinit snd_ps3_audio_fixup(struct snd_ps3_card_info *card) { /* * avsetting driver seems to never change the followings @@ -905,7 +905,7 @@ static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card) PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT); } -static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) +static int __devinit snd_ps3_init_avsetting(struct snd_ps3_card_info *card) { int ret; pr_debug("%s: start\n", __func__); @@ -927,7 +927,7 @@ static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) return ret; } -static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) +static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev) { int i, ret; u64 lpar_addr, lpar_size; -- cgit v1.1 From 1f3fff7bda95b75a6be5a02c2a6902573d8c18e6 Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Wed, 10 Jun 2009 19:50:33 +0200 Subject: ALSA: use card device as parent for jack input-devices This moves the jack devices from the PCI device into the ALSA card device, which makes it easier for userspace to find all devices belonging to a specific card while granting access to logged-in users. Jack input devices from sound cards can now simply be matched with udev by doing: SUBSYSTEM="input", SUBSYSTEMS="sound", ... ls -l /sys/devices/pci0000:00/0000:00:1b.0/sound/card0 controlC0 device -> ../../../0000:00:1b.0 id input10 input11 input8 input9 number pcmC0D0c pcmC0D0p pcmC0D1p power subsystem -> ../../../../../class/sound uevent Cc: Lennart Poettering Signed-off-by: Kay Sievers Signed-off-by: Takashi Iwai --- sound/core/jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index d54d1a0..f705eec 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -63,7 +63,7 @@ static int snd_jack_dev_register(struct snd_device *device) /* Default to the sound card device. */ if (!jack->input_dev->dev.parent) - jack->input_dev->dev.parent = card->dev; + jack->input_dev->dev.parent = snd_card_get_device_link(card); err = input_register_device(jack->input_dev); if (err == 0) -- cgit v1.1 From a5c0f88678cd2fb1f649f7d366d756f2b2f97f0c Mon Sep 17 00:00:00 2001 From: Simos Xenitellis Date: Wed, 10 Jun 2009 16:33:31 +0100 Subject: ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205) A quirk is required for 8086:284b (rev 03) [Subsystem: 161f:2073]. The following has been tested with Alsa 1.0.20 (git master). Background details can be found at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4561 http://forum.ubuntu-gr.org/viewtopic.php?f=38&t=5290 Tested-by: Theodora Iliopoulou Signed-off-by: Simos Xenitellis Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 42f944b..93e47c9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -5938,6 +5940,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 }, { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 }, { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 }, + { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 }, { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 }, { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 }, { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 }, -- cgit v1.1 From 955f2d966534803ec32411086a1698170f17f962 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 10 Jun 2009 21:32:01 +0200 Subject: ALSA: snd_usb_caiaq: set mixername alsamixer and friends want the mixername to be set. Even though the driver does not exports a real mixer device, export the name doesn't harm. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index b9a2b31..2240624 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.15"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -426,6 +426,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + strlcpy(card->mixername, dev->product_name, sizeof(card->mixername)); /* if the id was not passed as module option, fill it with a shortened * version of the product string which does not contain any -- cgit v1.1 From de0525ca34aa7a65faf6902e3e00956ab80d2ede Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Thu, 11 Jun 2009 16:03:34 +0200 Subject: ALSA: lx6464es - support standard alsa module parameters trivial patch to support the alsa module parameters `index', `id' and `enable' Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 870bfc5..ccf1b38 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -43,6 +43,13 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Digigram LX6464ES interface."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable/disable specific Digigram LX6464ES soundcards."); + static const char card_name[] = "LX6464ES"; -- cgit v1.1 From 4f64e150191bfddc7f5c0768f325f747dbca1913 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jun 2009 16:14:11 +0200 Subject: ALSA: pcm - Update document about xrun_debug proc file Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Procfile.txt | 36 +++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index cfac20c..381908d 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -88,26 +88,34 @@ card*/pcm*/info substreams, etc. card*/pcm*/xrun_debug - This file appears when CONFIG_SND_DEBUG=y. - This shows the status of xrun (= buffer overrun/xrun) debug of - ALSA PCM middle layer, as an integer from 0 to 2. The value - can be changed by writing to this file, such as - - # cat 2 > /proc/asound/card0/pcm0p/xrun_debug - - When this value is greater than 0, the driver will show the - messages to kernel log when an xrun is detected. The debug - message is shown also when the invalid H/W pointer is detected - at the update of periods (usually called from the interrupt + This file appears when CONFIG_SND_DEBUG=y and + CONFIG_PCM_XRUN_DEBUG=y. + This shows the status of xrun (= buffer overrun/xrun) and + invalid PCM position debug/check of ALSA PCM middle layer. + It takes an integer value, can be changed by writing to this + file, such as + + # cat 5 > /proc/asound/card0/pcm0p/xrun_debug + + The value consists of the following bit flags: + bit 0 = Enable XRUN/jiffies debug messages + bit 1 = Show stack trace at XRUN / jiffies check + bit 2 = Enable additional jiffies check + + When the bit 0 is set, the driver will show the messages to + kernel log when an xrun is detected. The debug message is + shown also when the invalid H/W pointer is detected at the + update of periods (usually called from the interrupt handler). - When this value is greater than 1, the driver will show the - stack trace additionally. This may help the debugging. + When the bit 1 is set, the driver will show the stack trace + additionally. This may help the debugging. - Since 2.6.30, this option also enables the hwptr check using + Since 2.6.30, this option can enable the hwptr check using jiffies. This detects spontaneous invalid pointer callback values, but can be lead to too much corrections for a (mostly buggy) hardware that doesn't give smooth pointer updates. + This feature is enabled via the bit 2. card*/pcm*/sub*/info The general information of this PCM sub-stream. -- cgit v1.1