From 84d447e30a9494d93eaff237aae0cba090c27a70 Mon Sep 17 00:00:00 2001 From: Ziyann Date: Sat, 6 Dec 2014 02:24:40 +0100 Subject: ASoC: add back generic SDP4430 board file Just to be in sync with p-android-omap-3.0-dev. --- sound/soc/omap/Kconfig | 8 +- sound/soc/omap/sdp4430.c | 1169 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 1174 insertions(+), 3 deletions(-) create mode 100644 sound/soc/omap/sdp4430.c (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 19d1f24..9b72bb0 100755 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -113,7 +113,9 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_SDP4430 tristate "SoC Audio support for Texas Instruments SDP4430 or PandaBoard" - depends on TWL4030_CORE && (MACH_TUNA || MACH_OMAP_4430SDP || MACH_OMAP4_PANDA) + depends on TWL4030_CORE && \ + (MACH_OMAP_4430SDP || MACH_OMAP4_PANDA || MACH_OMAP4_TABLET) + select SND_OMAP_SOC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_OMAP_SOC_ABE @@ -121,9 +123,8 @@ config SND_OMAP_SOC_SDP4430 select SND_SOC_DMIC select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_ABE_DSP - select SND_OMAP_SOC_MCASP - select SND_OMAP_SOC_SPDIF select SND_OMAP_SOC_VXREC + select CDC_TCXO if (MACH_OMAP_4430SDP || MACH_OMAP4_TABLET) help Say Y if you want to add support for SoC audio on Texas Instruments SDP4430 or PandaBoard. @@ -195,6 +196,7 @@ config SND_OMAP_SOC_IGEP0020 config SND_OMAP_SOC_TUNA tristate "SoC Audio support for Tuna" depends on TWL4030_CORE && MACH_TUNA + select SND_OMAP_SOC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_OMAP_SOC_ABE diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c new file mode 100644 index 0000000..8d488e8 --- /dev/null +++ b/sound/soc/omap/sdp4430.c @@ -0,0 +1,1169 @@ +/* + * sdp4430.c -- SoC audio for TI OMAP4430 SDP + * + * Author: Misael Lopez Cruz + * 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. + * + * 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 +#include +#include + +#include "omap-mcpdm.h" +#include "omap-abe.h" +#include "omap-abe-dsp.h" +#include "omap-pcm.h" +#include "omap-mcbsp.h" +#include "omap-dmic.h" +#include "../codecs/twl6040.h" + +static struct regulator *av_switch_reg; +static int twl6040_power_mode; +static int mcbsp_cfg; +static struct snd_soc_codec *twl6040_codec; + +static int sdp4430_modem_mcbsp_configure(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, int flag) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_substream *modem_substream[2]; + struct snd_soc_pcm_runtime *modem_rtd; + int channels; + + if (flag) { + modem_substream[substream->stream] = + snd_soc_get_dai_substream(rtd->card, + OMAP_ABE_BE_MM_EXT1, + substream->stream); + if (unlikely(modem_substream[substream->stream] == NULL)) + return -ENODEV; + + modem_rtd = + modem_substream[substream->stream]->private_data; + + if (!mcbsp_cfg) { + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(modem_rtd->cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + + if (unlikely(ret < 0)) { + printk(KERN_ERR "can't set Modem cpu DAI configuration\n"); + goto exit; + } else { + mcbsp_cfg = 1; + } + } + + if (params != NULL) { + /* Configure McBSP internal buffer usage */ + /* this need to be done for playback and/or record */ + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + omap_mcbsp_set_rx_threshold( + modem_rtd->cpu_dai->id, channels); + else + omap_mcbsp_set_tx_threshold( + modem_rtd->cpu_dai->id, channels); + } + } else { + mcbsp_cfg = 0; + } + +exit: + return ret; +} + +static int sdp4430_modem_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int ret; + + ret = sdp4430_modem_mcbsp_configure(substream, params, 1); + if (ret) + printk(KERN_ERR "can't set modem cpu DAI configuration\n"); + + return ret; +} + +static int sdp4430_modem_hw_free(struct snd_pcm_substream *substream) +{ + int ret; + + ret = sdp4430_modem_mcbsp_configure(substream, NULL, 0); + if (ret) + printk(KERN_ERR "can't clear modem cpu DAI configuration\n"); + + return ret; +} + +static struct snd_soc_ops sdp4430_modem_ops = { + .hw_params = sdp4430_modem_hw_params, + .hw_free = sdp4430_modem_hw_free, +}; + +static int sdp4430_mcpdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct twl6040 *twl6040 = codec->control_data; + int clk_id, freq, ret; + + /* TWL6040 supplies McPDM PAD_CLKS */ + ret = twl6040_enable(twl6040); + if (ret) { + printk(KERN_ERR "failed to enable TWL6040\n"); + return ret; + } + + if (twl6040_power_mode) { + clk_id = TWL6040_HPPLL_ID; + freq = 38400000; + + /* + * TWL6040 requires MCLK to be active as long as + * high-performance mode is in use. Glitch-free mux + * cannot tolerate MCLK gating + */ + ret = cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 1); + if (ret) { + printk(KERN_ERR "failed to enable twl6040 MCLK\n"); + goto err; + } + } else { + clk_id = TWL6040_LPPLL_ID; + freq = 32768; + } + + /* set the codec mclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, + SND_SOC_CLOCK_IN); + if (ret) { + printk(KERN_ERR "can't set codec system clock\n"); + goto err; + } + + /* low-power mode uses 32k clock, MCLK is not required */ + if (!twl6040_power_mode) { + ret = cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 0); + if (ret) + printk(KERN_ERR "failed to disable twl6040 MCLK\n"); + } + + return 0; + +err: + twl6040_disable(twl6040); + return ret; +} + +static void sdp4430_mcpdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct twl6040 *twl6040 = codec->control_data; + + /* TWL6040 supplies McPDM PAD_CLKS */ + twl6040_disable(twl6040); +} + +static struct snd_soc_ops sdp4430_mcpdm_ops = { + .startup = sdp4430_mcpdm_startup, + .shutdown = sdp4430_mcpdm_shutdown, +}; + +static int sdp4430_mcbsp_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 *cpu_dai = rtd->cpu_dai; + int ret = 0; + unsigned int channels; + + 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; + } + + if (params != NULL) { + /* Configure McBSP internal buffer usage */ + /* this need to be done for playback and/or record */ + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + omap_mcbsp_set_tx_threshold( + cpu_dai->id, channels); + else + omap_mcbsp_set_rx_threshold( + cpu_dai->id, channels); + } + + /* + * TODO: where does this clock come from (external source??) - + * do we need to enable it. + */ + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, + 64 * params_rate(params), + SND_SOC_CLOCK_IN); + if (ret < 0) + printk(KERN_ERR "can't set cpu system clock\n"); + + return ret; +} + +static struct snd_soc_ops sdp4430_mcbsp_ops = { + .hw_params = sdp4430_mcbsp_hw_params, +}; + +static int sdp4430_dmic_startup(struct snd_pcm_substream *substream) +{ + struct twl6040 *twl6040 = twl6040_codec->control_data; + /* In order for the DMIC's to use the PAD CLOCKS, the twl6040 + * must be powered up, since it supplies the clock source. + */ + return twl6040_enable(twl6040); +} + +static void sdp4430_dmic_shutdown(struct snd_pcm_substream *substream) +{ + struct twl6040 *twl6040 = twl6040_codec->control_data; + twl6040_disable(twl6040); +} + +static int sdp4430_dmic_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 *cpu_dai = rtd->cpu_dai; + int ret = 0; + + if (!rtd->dai_link->no_pcm) + ret = snd_soc_dai_set_sysclk(cpu_dai, + OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS, 24000000, + SND_SOC_CLOCK_IN); + else + ret = snd_soc_dai_set_sysclk(cpu_dai, + OMAP_DMIC_SYSCLK_PAD_CLKS, 19200000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + printk(KERN_ERR "can't set DMIC cpu system clock\n"); + return ret; + } + + if (!rtd->dai_link->no_pcm) + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_DMIC_CLKDIV, 10); + else + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_DMIC_CLKDIV, 8); + + if (ret < 0) { + printk(KERN_ERR "can't set DMIC cpu clock divider\n"); + return ret; + } + return 0; +} + +static struct snd_soc_ops sdp4430_dmic_ops = { + .startup = sdp4430_dmic_startup, + .shutdown = sdp4430_dmic_shutdown, + .hw_params = sdp4430_dmic_hw_params, +}; + +static int mcbsp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + channels->min = 2; + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int dmic_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ABE will covert the FE rate to 96k */ + rate->min = rate->max = 96000; + channels->min = channels->max = 2; + + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S32_LE); + return 0; +} + +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static int sdp4430_av_switch_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + int ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = regulator_enable(av_switch_reg); + else + ret = regulator_disable(av_switch_reg); + + return ret; +} + +static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = twl6040_power_mode; + return 0; +} + +static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (twl6040_power_mode == ucontrol->value.integer.value[0]) + return 0; + + twl6040_power_mode = ucontrol->value.integer.value[0]; + abe_dsp_set_power_mode(twl6040_power_mode); + + return 1; +} + +static const char *power_texts[] = {"Low-Power", "High-Performance"}; + +static const struct soc_enum sdp4430_enum[] = { + SOC_ENUM_SINGLE_EXT(2, power_texts), +}; + +static const struct snd_kcontrol_new sdp4430_controls[] = { + SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0], + sdp4430_get_power_mode, sdp4430_set_power_mode), +}; + +/* SDP4430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_SUPPLY("AV Switch Supply", + SND_SOC_NOPM, 0, 0, sdp4430_av_switch_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIC("Digital Mic 0", NULL), + SND_SOC_DAPM_MIC("Digital Mic 1", NULL), + SND_SOC_DAPM_MIC("Digital Mic 2", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Main Mic Bias"}, + {"SUBMIC", NULL, "Main Mic Bias"}, + {"Main Mic Bias", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "AV Switch Supply"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Earphone speaker */ + {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "Aux/FM Stereo In"}, + {"AFMR", NULL, "Aux/FM Stereo In"}, + + /* Digital Mics: DMic0, DMic1, DMic2 with bias */ + {"DMIC0", NULL, "Digital Mic1 Bias"}, + {"Digital Mic1 Bias", NULL, "Digital Mic 0"}, + + {"DMIC1", NULL, "Digital Mic1 Bias"}, + {"Digital Mic1 Bias", NULL, "Digital Mic 1"}, + + {"DMIC2", NULL, "Digital Mic1 Bias"}, + {"Digital Mic1 Bias", NULL, "Digital Mic 2"}, +}; + +static int sdp4430_set_pdm_dl1_gains(struct snd_soc_dapm_context *dapm) +{ + int output, val; + + if (snd_soc_dapm_get_pin_power(dapm, "Earphone Spk")) { + output = OMAP_ABE_DL1_EARPIECE; + } else if (snd_soc_dapm_get_pin_power(dapm, "Headset Stereophone")) { + val = snd_soc_read(twl6040_codec, TWL6040_REG_HSLCTL); + if (val & TWL6040_HSDACMODEL) + /* HSDACL in LP mode */ + output = OMAP_ABE_DL1_HEADSET_LP; + else + /* HSDACL in HP mode */ + output = OMAP_ABE_DL1_HEADSET_HP; +#if !defined(CONFIG_SND_OMAP_SOC_ABE_DL2) + } else if (snd_soc_dapm_get_pin_power(dapm, "Ext Spk")) { + output = OMAP_ABE_DL1_HANDSFREE; +#endif + } else { + output = OMAP_ABE_DL1_NO_PDM; + } + + return omap_abe_set_dl1_output(output); +} + +static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct twl6040 *twl6040 = codec->control_data; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int hsotrim, left_offset, right_offset, mode, ret; + + + /* Add SDP4430 specific controls */ + ret = snd_soc_add_controls(codec, sdp4430_controls, + ARRAY_SIZE(sdp4430_controls)); + if (ret) + return ret; + + /* Add SDP4430 specific widgets */ + ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets, + ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP4430 specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP4430 connected pins */ + snd_soc_dapm_enable_pin(dapm, "Ext Mic"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "AFML"); + snd_soc_dapm_enable_pin(dapm, "AFMR"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); + + /* allow audio paths from the audio modem to run during suspend */ + snd_soc_dapm_ignore_suspend(dapm, "Ext Mic"); + snd_soc_dapm_ignore_suspend(dapm, "Ext Spk"); + snd_soc_dapm_ignore_suspend(dapm, "AFML"); + snd_soc_dapm_ignore_suspend(dapm, "AFMR"); + snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); + snd_soc_dapm_ignore_suspend(dapm, "Headset Stereophone"); + snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 0"); + snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 1"); + snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 2"); + + ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + + if (machine_is_omap_4430sdp() || machine_is_omap_tabletblaze() + || machine_is_omap4_panda()) + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); + else + snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + + /* DC offset cancellation computation */ + hsotrim = snd_soc_read(codec, TWL6040_REG_HSOTRIM); + right_offset = (hsotrim & TWL6040_HSRO) >> TWL6040_HSRO_OFFSET; + left_offset = hsotrim & TWL6040_HSLO; + + if (twl6040_get_icrev(twl6040) < TWL6040_REV_1_3) + /* For ES under ES_1.3 HS step is 2 mV */ + mode = 2; + else + /* For ES_1.3 HS step is 1 mV */ + mode = 1; + + abe_dsp_set_hs_offset(left_offset, right_offset, mode); + + /* don't wait before switching of HS power */ + rtd->pmdown_time = 0; + + return ret; +} + +static int sdp4430_twl6040_dl2_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int hfotrim, left_offset, right_offset; + + /* DC offset cancellation computation */ + hfotrim = snd_soc_read(codec, TWL6040_REG_HFOTRIM); + right_offset = (hfotrim & TWL6040_HFRO) >> TWL6040_HFRO_OFFSET; + left_offset = hfotrim & TWL6040_HFLO; + + abe_dsp_set_hf_offset(left_offset, right_offset); + + /* don't wait before switching of HF power */ + rtd->pmdown_time = 0; + + return 0; +} + +/* SDP4430 digital microphones DAPM */ +static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Digital Mic Legacy", NULL), +}; + +static const struct snd_soc_dapm_route dmic_audio_map[] = { + {"DMic", NULL, "Digital Mic1 Bias"}, + {"Digital Mic1 Bias", NULL, "Digital Mic Legacy"}, +}; + +static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, + ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, dmic_audio_map, + ARRAY_SIZE(dmic_audio_map)); + if (ret) + return ret; + + snd_soc_dapm_enable_pin(dapm, "Digital Mic Legacy"); + + ret = snd_soc_dapm_sync(dapm); + + return ret; +} + +static int sdp4430_twl6040_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + + /* don't wait before switching of FE power */ + rtd->pmdown_time = 0; + + return 0; +} + +static int sdp4430_bt_init(struct snd_soc_pcm_runtime *rtd) +{ + + /* don't wait before switching of BT power */ + rtd->pmdown_time = 0; + + return 0; +} + +static int sdp4430_stream_event(struct snd_soc_dapm_context *dapm) +{ + /* + * set DL1 gains dynamically according to the active output + * (Headset, Earpiece) and HSDAC power mode + */ + return sdp4430_set_pdm_dl1_gains(dapm); +} + +/* TODO: make this a separate BT CODEC driver or DUMMY */ +static struct snd_soc_dai_driver dai[] = { +{ + .name = "Bluetooth", + .playback = { + .stream_name = "BT Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "BT Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +/* TODO: make this a separate FM CODEC driver or DUMMY */ +{ + .name = "FM Digital", + .playback = { + .stream_name = "FM Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "FM Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "HDMI", + .playback = { + .stream_name = "HDMI Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +}; + +struct snd_soc_dsp_link fe_media = { + .playback = true, + .capture = true, + .trigger = + {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, +}; + +struct snd_soc_dsp_link fe_media_capture = { + .capture = true, + .trigger = + {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, +}; + +struct snd_soc_dsp_link fe_tones = { + .playback = true, + .trigger = + {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, +}; + +struct snd_soc_dsp_link fe_vib = { + .playback = true, + .trigger = + {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, +}; + +struct snd_soc_dsp_link fe_modem = { + .playback = true, + .capture = true, + .trigger = + {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, +}; + +struct snd_soc_dsp_link fe_lp_media = { + .playback = true, + .trigger = + {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, +}; +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp4430_dai[] = { + +/* + * Frontend DAIs - i.e. userspace visible interfaces (ALSA PCMs) + */ + + { + .name = "SDP4430 Media", + .stream_name = "Multimedia", + + /* ABE components - MM-UL & MM_DL */ + .cpu_dai_name = "MultiMedia1", + .platform_name = "omap-pcm-audio", + + .dynamic = 1, /* BE is dynamic */ + .init = sdp4430_twl6040_fe_init, + .dsp_link = &fe_media, + }, + { + .name = "SDP4430 Media Capture", + .stream_name = "Multimedia Capture", + + /* ABE components - MM-UL2 */ + .cpu_dai_name = "MultiMedia2", + .platform_name = "omap-pcm-audio", + + .dynamic = 1, /* BE is dynamic */ + .dsp_link = &fe_media_capture, + }, + { + .name = "SDP4430 Voice", + .stream_name = "Voice", + + /* ABE components - VX-UL & VX-DL */ + .cpu_dai_name = "Voice", + .platform_name = "omap-pcm-audio", + + .dynamic = 1, /* BE is dynamic */ + .dsp_link = &fe_media, + .no_host_mode = SND_SOC_DAI_LINK_OPT_HOST, + }, + { + .name = "SDP4430 Tones Playback", + .stream_name = "Tone Playback", + + /* ABE components - TONES_DL */ + .cpu_dai_name = "Tones", + .platform_name = "omap-pcm-audio", + + .dynamic = 1, /* BE is dynamic */ + .dsp_link = &fe_tones, + }, + { + .name = "SDP4430 Vibra Playback", + .stream_name = "VIB-DL", + + /* ABE components - DMIC UL 2 */ + .cpu_dai_name = "Vibra", + .platform_name = "omap-pcm-audio", + + .dynamic = 1, /* BE is dynamic */ + .dsp_link = &fe_vib, + }, + { + .name = "SDP4430 MODEM", + .stream_name = "MODEM", + + /* ABE components - MODEM <-> McBSP2 */ + .cpu_dai_name = "MODEM", + .platform_name = "aess", + + .dynamic = 1, /* BE is dynamic */ + .init = sdp4430_twl6040_fe_init, + .dsp_link = &fe_modem, + .ops = &sdp4430_modem_ops, + .no_host_mode = SND_SOC_DAI_LINK_NO_HOST, + .ignore_suspend = 1, + }, + { + .name = "SDP4430 Media LP", + .stream_name = "Multimedia", + + /* ABE components - MM-DL (mmap) */ + .cpu_dai_name = "MultiMedia1 LP", + .platform_name = "aess", + + .dynamic = 1, /* BE is dynamic */ + .dsp_link = &fe_lp_media, + }, + { + .name = "Legacy McBSP", + .stream_name = "Multimedia", + + /* ABE components - MCBSP2 - MM-EXT */ + .cpu_dai_name = "omap-mcbsp-dai.1", + .platform_name = "omap-pcm-audio", + + /* FM */ + .codec_dai_name = "FM Digital", + + .no_codec = 1, /* TODO: have a dummy CODEC */ + .ops = &sdp4430_mcbsp_ops, + .ignore_suspend = 1, + }, + { + .name = "Legacy McPDM", + .stream_name = "Headset Playback", + + /* ABE components - DL1 */ + .cpu_dai_name = "mcpdm-dl", + .platform_name = "omap-pcm-audio", + + /* Phoenix - DL1 DAC */ + .codec_dai_name = "twl6040-dl1", + .codec_name = "twl6040-codec", + + .ops = &sdp4430_mcpdm_ops, + .ignore_suspend = 1, + }, + { + .name = "Legacy DMIC", + .stream_name = "DMIC Capture", + + /* ABE components - DMIC0 */ + .cpu_dai_name = "omap-dmic-dai-0", + .platform_name = "omap-pcm-audio", + + /* DMIC codec */ + .codec_dai_name = "dmic-hifi", + .codec_name = "dmic-codec.0", + + .init = sdp4430_dmic_init, + .ops = &sdp4430_dmic_ops, + .ignore_suspend = 1, + }, + +/* + * Backend DAIs - i.e. dynamically matched interfaces, invisible to userspace. + * Matched to above interfaces at runtime, based upon use case. + */ + + { + .name = OMAP_ABE_BE_PDM_DL1, + .stream_name = "HS Playback", + + /* ABE components - DL1 */ + .cpu_dai_name = "mcpdm-dl1", + .platform_name = "aess", + + /* Phoenix - DL1 DAC */ + .codec_dai_name = "twl6040-dl1", + .codec_name = "twl6040-codec", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .init = sdp4430_twl6040_init, + .ops = &sdp4430_mcpdm_ops, + .be_id = OMAP_ABE_DAI_PDM_DL1, + .ignore_suspend = 1, + }, + { + .name = OMAP_ABE_BE_PDM_UL1, + .stream_name = "Analog Capture", + + /* ABE components - UL1 */ + .cpu_dai_name = "mcpdm-ul1", + .platform_name = "aess", + + /* Phoenix - UL ADC */ + .codec_dai_name = "twl6040-ul", + .codec_name = "twl6040-codec", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .ops = &sdp4430_mcpdm_ops, + .be_id = OMAP_ABE_DAI_PDM_UL, + .ignore_suspend = 1, + }, + { + .name = OMAP_ABE_BE_PDM_DL2, + .stream_name = "HF Playback", + + /* ABE components - DL2 */ + .cpu_dai_name = "mcpdm-dl2", + .platform_name = "aess", + + /* Phoenix - DL2 DAC */ + .codec_dai_name = "twl6040-dl2", + .codec_name = "twl6040-codec", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .init = sdp4430_twl6040_dl2_init, + .ops = &sdp4430_mcpdm_ops, + .be_id = OMAP_ABE_DAI_PDM_DL2, + .ignore_suspend = 1, + }, + { + .name = OMAP_ABE_BE_PDM_VIB, + .stream_name = "Vibra", + + /* ABE components - VIB1 DL */ + .cpu_dai_name = "mcpdm-vib", + .platform_name = "aess", + + /* Phoenix - PDM to PWM */ + .codec_dai_name = "twl6040-vib", + .codec_name = "twl6040-codec", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .ops = &sdp4430_mcpdm_ops, + .be_id = OMAP_ABE_DAI_PDM_VIB, + }, + { + .name = OMAP_ABE_BE_BT_VX_UL, + .stream_name = "BT Capture", + + /* ABE components - MCBSP1 - BT-VX */ + .cpu_dai_name = "omap-mcbsp-dai.0", + .platform_name = "aess", + + /* Bluetooth */ + .codec_dai_name = "Bluetooth", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .no_codec = 1, /* TODO: have a dummy CODEC */ + .be_hw_params_fixup = mcbsp_be_hw_params_fixup, + .ops = &sdp4430_mcbsp_ops, + .be_id = OMAP_ABE_DAI_BT_VX, + .ignore_suspend = 1, + }, + { + .name = OMAP_ABE_BE_BT_VX_DL, + .stream_name = "BT Playback", + + /* ABE components - MCBSP1 - BT-VX */ + .cpu_dai_name = "omap-mcbsp-dai.0", + .platform_name = "aess", + + /* Bluetooth */ + .codec_dai_name = "Bluetooth", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .no_codec = 1, /* TODO: have a dummy CODEC */ + .init = sdp4430_bt_init, + .be_hw_params_fixup = mcbsp_be_hw_params_fixup, + .ops = &sdp4430_mcbsp_ops, + .be_id = OMAP_ABE_DAI_BT_VX, + .ignore_suspend = 1, + }, + { + .name = OMAP_ABE_BE_MM_EXT0, + .stream_name = "FM Playback", + + /* ABE components - MCBSP2 - MM-EXT */ + .cpu_dai_name = "omap-mcbsp-dai.1", + .platform_name = "aess", + + /* FM */ + .codec_dai_name = "FM Digital", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .no_codec = 1, /* TODO: have a dummy CODEC */ + .be_hw_params_fixup = mcbsp_be_hw_params_fixup, + .ops = &sdp4430_mcbsp_ops, + .be_id = OMAP_ABE_DAI_MM_FM, + }, + { + .name = OMAP_ABE_BE_MM_EXT1, + .stream_name = "MODEM", + + /* ABE components - MCBSP2 - MM-EXT */ + .cpu_dai_name = "omap-mcbsp-dai.1", + .platform_name = "aess", + + /* MODEM */ + .codec_dai_name = "MODEM", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .no_codec = 1, /* TODO: have a dummy CODEC */ + .be_hw_params_fixup = mcbsp_be_hw_params_fixup, + .ops = &sdp4430_mcbsp_ops, + .be_id = OMAP_ABE_DAI_MODEM, + .ignore_suspend = 1, + }, + { + .name = OMAP_ABE_BE_DMIC0, + .stream_name = "DMIC0 Capture", + + /* ABE components - DMIC UL 1 */ + .cpu_dai_name = "omap-dmic-abe-dai-0", + .platform_name = "aess", + + /* DMIC 0 */ + .codec_dai_name = "dmic-hifi", + .codec_name = "dmic-codec.0", + .ops = &sdp4430_dmic_ops, + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .be_hw_params_fixup = dmic_be_hw_params_fixup, + .be_id = OMAP_ABE_DAI_DMIC0, + }, + { + .name = OMAP_ABE_BE_DMIC1, + .stream_name = "DMIC1 Capture", + + /* ABE components - DMIC UL 1 */ + .cpu_dai_name = "omap-dmic-abe-dai-1", + .platform_name = "aess", + + /* DMIC 1 */ + .codec_dai_name = "dmic-hifi", + .codec_name = "dmic-codec.1", + .ops = &sdp4430_dmic_ops, + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .be_hw_params_fixup = dmic_be_hw_params_fixup, + .be_id = OMAP_ABE_DAI_DMIC1, + }, + { + .name = OMAP_ABE_BE_DMIC2, + .stream_name = "DMIC2 Capture", + + /* ABE components - DMIC UL 2 */ + .cpu_dai_name = "omap-dmic-abe-dai-2", + .platform_name = "aess", + + /* DMIC 2 */ + .codec_dai_name = "dmic-hifi", + .codec_name = "dmic-codec.2", + .ops = &sdp4430_dmic_ops, + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .be_hw_params_fixup = dmic_be_hw_params_fixup, + .be_id = OMAP_ABE_DAI_DMIC2, + }, + { + .name = OMAP_ABE_BE_VXREC, + .stream_name = "VXREC Capture", + + /* ABE components - VxREC */ + .cpu_dai_name = "omap-abe-vxrec-dai", + .platform_name = "aess", + + /* no codec needed */ + .codec_dai_name = "null-codec-dai", + + .no_pcm = 1, /* don't create ALSA pcm for this */ + .no_codec = 1, + .be_id = OMAP_ABE_DAI_VXREC, + .ignore_suspend = 1, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sdp4430 = { + .driver_name = "OMAP4", + .long_name = "TI OMAP4 Board", + .dai_link = sdp4430_dai, + .num_links = ARRAY_SIZE(sdp4430_dai), + .stream_event = sdp4430_stream_event, +}; + +static struct platform_device *sdp4430_snd_device; + +static int __init sdp4430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_4430sdp() && !machine_is_omap4_panda() && + !machine_is_omap_tabletblaze()) { + pr_debug("Not SDP4430, BlazeTablet or PandaBoard!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP4430 SoC init\n"); + if (machine_is_omap_4430sdp()) + snd_soc_sdp4430.name = "SDP4430"; + else if (machine_is_omap4_panda()) + snd_soc_sdp4430.name = "Panda"; + else if (machine_is_omap_tabletblaze()) + snd_soc_sdp4430.name = "Tablet44xx"; + + sdp4430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp4430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + ret = snd_soc_register_dais(&sdp4430_snd_device->dev, dai, ARRAY_SIZE(dai)); + if (ret < 0) + goto err; + platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); + + ret = platform_device_add(sdp4430_snd_device); + if (ret) + goto err_dev; + + twl6040_codec = snd_soc_card_get_codec(&snd_soc_sdp4430, + "twl6040-codec"); + if(twl6040_codec <= 0) { + printk(KERN_ERR "sdp4430: could not find `twl6040-codec`\n"); + ret = -ENODEV; + goto err_dev; + } + + av_switch_reg = regulator_get(&sdp4430_snd_device->dev, "av-switch"); + if (IS_ERR(av_switch_reg)) { + ret = PTR_ERR(av_switch_reg); + printk(KERN_ERR "couldn't get AV Switch regulator %d\n", + ret); + goto err_dev; + } + + /* Default mode is low-power, MCLK not required */ + twl6040_power_mode = 0; + cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 0); + + /* + * CDC CLK2 supplies TWL6040 MCLK, drive it from REQ2INT to + * have full control of MCLK gating + */ + cdc_tcxo_set_req_prio(CDC_TCXO_CLK2, CDC_TCXO_PRIO_REQINT); + + return ret; + +err_dev: + snd_soc_unregister_dais(&sdp4430_snd_device->dev, ARRAY_SIZE(dai)); +err: + platform_device_put(sdp4430_snd_device); + return ret; +} +module_init(sdp4430_soc_init); + +static void __exit sdp4430_soc_exit(void) +{ + regulator_put(av_switch_reg); + cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 0); + cdc_tcxo_set_req_prio(CDC_TCXO_CLK2, CDC_TCXO_PRIO_REQINT); + platform_device_unregister(sdp4430_snd_device); + snd_soc_unregister_dais(&sdp4430_snd_device->dev, ARRAY_SIZE(dai)); +} +module_exit(sdp4430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_DESCRIPTION("ALSA SoC SDP4430"); +MODULE_LICENSE("GPL"); + -- cgit v1.1