ASoC: add back generic SDP4430 board file

Just to be in sync with p-android-omap-3.0-dev.

2 files changed, 1174 insertions, 3 deletions

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 <misael.lopez@ti.com>
+ *         Liam Girdwood <lrg@ti.com>
+ *
+ * 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 <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/i2c/twl.h>
+#include <linux/regulator/consumer.h>
+#include <linux/cdc_tcxo.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <sound/soc-dsp.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+#include <plat/mcbsp.h>
+
+#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(&params->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(&params->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 <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP4430");
+MODULE_LICENSE("GPL");
+