More ASoC updates

More ASoC updates

[Mark Brown]

The following changes since commit ddaea5119fd2efde7b3936d4ccfa34c4201be0d3:
  Takashi Iwai (1):
        Merge branch 'asoc-fixes' into topic/asoc

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

There's more to come on the TLV320AIC23 but at this point it's all
cleanups and it's much easier to review incremental patches.

Arun KS (3):
      ASoC: Add TLV320AIC23 codec driver
      ASoC: Add support for osk5912
      ASoC: Add DSP DAI format support to the OMAP McBSP driver

 sound/soc/codecs/Kconfig       |    5 +
 sound/soc/codecs/Makefile      |    2 +
 sound/soc/codecs/tlv320aic23.c |  670 ++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/tlv320aic23.h |  122 ++++++++
 sound/soc/omap/Kconfig         |    8 +
 sound/soc/omap/Makefile        |    2 +
 sound/soc/omap/omap-mcbsp.c    |    5 +
 sound/soc/omap/osk5912.c       |  232 ++++++++++++++
 8 files changed, 1046 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/tlv320aic23.c
 create mode 100644 sound/soc/codecs/tlv320aic23.h
 create mode 100644 sound/soc/omap/osk5912.c

[PATCH 1/3] ASoC: Add TLV320AIC23 codec driver

[Mark Brown]

From: Arun KS <arunks@mistralsolutions.com>

ASoC codec driver for TLV320AIC23 device

Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig       |    5 +
 sound/soc/codecs/Makefile      |    2 +
 sound/soc/codecs/tlv320aic23.c |  670 ++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/tlv320aic23.h |  122 ++++++++
 4 files changed, 799 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/tlv320aic23.c
 create mode 100644 sound/soc/codecs/tlv320aic23.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0507fcf..bdead2d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -7,6 +7,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4535
 	select SND_SOC_CS4270
 	select SND_SOC_SSM2602
+	select SND_SOC_TLV320AIC23
 	select SND_SOC_TLV320AIC26
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_UDA1380
@@ -62,6 +63,10 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_TLV320AIC23
+	tristate
+	depends on I2C
+
 config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support"
 	depends on SND_SOC && SPI
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 0731844..90f0a58 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-uda1380-objs := uda1380.o
@@ -25,6 +26,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 0000000..c2d35e9
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,670 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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.
+ *
+ * Notes:
+ *  The AIC23 is a driver for a low power stereo audio
+ *  codec tlv320aic23
+ *
+ *  The machine layer should disable unsupported inputs/outputs by
+ *  snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.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/tlv.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+#define AUDIO_NAME "tlv320aic23"
+#define AIC23_VERSION "0.1"
+
+struct tlv320aic23_srate_reg_info {
+	u32 sample_rate;
+	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
+	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+/*
+ * AIC23 register cache
+ */
+static const u16 tlv320aic23_reg[] = {
+	0x0097, 0x0097, 0x00F9, 0x00F9,	/* 0 */
+	0x001A, 0x0004, 0x0007, 0x0001,	/* 4 */
+	0x0020, 0x0000, 0x0000, 0x0000,	/* 8 */
+	0x0000, 0x0000, 0x0000, 0x0000,	/* 12 */
+};
+
+/*
+ * read tlv320aic23 register cache
+ */
+static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
+						      *codec, unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write tlv320aic23 register cache
+ */
+static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
+					       u8 reg, u16 value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the tlv320aic23 register space
+ */
+static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
+			     unsigned int value)
+{
+
+	u8 data;
+
+	/* TLV320AIC23 has 7 bit address and 9 bits of data
+	 * so we need to switch one data bit into reg and rest
+	 * of data into val
+	 */
+
+	if ((reg < 0 || reg > 9) && (reg != 15)) {
+		printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+		return -1;
+	}
+
+	data = (reg << 1) | (value >> 8 & 0x01);
+
+	tlv320aic23_write_reg_cache(codec, reg, value);
+
+	if (codec->hw_write(codec->control_data, data,
+			    (value & 0xff)) == 0)
+		return 0;
+
+	printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+	       value, reg);
+
+	return -EIO;
+}
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *sidetone_text[] = {"-6db", "-9db", "-12db", "-18db", "0db"};
+
+static const struct soc_enum rec_src_enum =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static const struct soc_enum tlv320aic23_rec_src =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static const struct soc_enum tlv320aic23_deemph =
+	SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+static const struct soc_enum tlv320aic23_sidetone =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 6, 5, sidetone_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+			 TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+	SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+	SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+		     TLV320AIC23_RINVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+			 TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+	SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+	SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+	SOC_ENUM("Sidetone Gain", tlv320aic23_sidetone),
+	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* add non dapm controls */
+static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
+{
+
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+
+}
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+	SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+	SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+			 &tlv320aic23_rec_src_mux_controls),
+	SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+			   &tlv320aic23_output_mixer_controls[0],
+			   ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LHPOUT"),
+	SND_SOC_DAPM_OUTPUT("RHPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+
+	SND_SOC_DAPM_INPUT("LLINEIN"),
+	SND_SOC_DAPM_INPUT("RLINEIN"),
+
+	SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* Output Mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+	/* Outputs */
+	{"RHPOUT", NULL, "Output Mixer"},
+	{"LHPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+
+	/* Inputs */
+	{"Line Input", "NULL", "LLINEIN"},
+	{"Line Input", "NULL", "RLINEIN"},
+
+	{"Mic Input", "NULL", "MICIN"},
+
+	/* input mux */
+	{"Capture Source", "Line", "Line Input"},
+	{"Capture Source", "Mic", "Mic Input"},
+	{"ADC", NULL, "Capture Source"},
+
+};
+
+/* tlv320aic23 related */
+static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
+	{4000, 0x06, 1},	/*  4000 */
+	{8000, 0x06, 0},	/*  8000 */
+	{16000, 0x0C, 1},	/* 16000 */
+	{22050, 0x11, 1},	/* 22050 */
+	{24000, 0x00, 1},	/* 24000 */
+	{32000, 0x0C, 0},	/* 32000 */
+	{44100, 0x11, 0},	/* 44100 */
+	{48000, 0x00, 0},	/* 48000 */
+	{88200, 0x1F, 0},	/* 88200 */
+	{96000, 0x0E, 0},	/* 96000 */
+};
+
+static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* set up audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int tlv320aic23_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_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 iface_reg, data;
+	u8 count = 0;
+
+	iface_reg =
+	    tlv320aic23_read_reg_cache(codec,
+				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
+
+	/* Search for the right sample rate */
+	/* Verify what happens if the rate is not supported
+	 * now it goes to 96Khz */
+	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
+	       (count < ARRAY_SIZE(srate_reg_info))) {
+		count++;
+	}
+
+	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
+		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
+		TLV320AIC23_USB_CLK_ON;
+
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface_reg |= (0x01 << 2);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface_reg |= (0x02 << 2);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface_reg |= (0x03 << 2);
+		break;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* set active */
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+	return 0;
+}
+
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* deactivate */
+	if (!codec->active) {
+		udelay(50);
+		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	}
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 reg;
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
+	if (mute)
+		reg |= TLV320AIC23_DACM_MUTE;
+
+	else
+		reg &= ~TLV320AIC23_DACM_MUTE;
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				   unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface_reg;
+
+	iface_reg =
+	    tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg |= TLV320AIC23_MS_MASTER;
+		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_reg |= TLV320AIC23_FOR_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg |= TLV320AIC23_FOR_DSP;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg |= TLV320AIC23_FOR_LJUST;
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_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;
+
+	switch (freq) {
+	case 12000000:
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
+{
+	u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* vref/mid, osc on, dac unmute */
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* everything off except vref/vmid, */
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+		tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define AIC23_RATES	SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai tlv320aic23_dai = {
+	.name = "tlv320aic23",
+	.playback = {
+		     .stream_name = "Playback",
+		     .channels_min = 2,
+		     .channels_max = 2,
+		     .rates = AIC23_RATES,
+		     .formats = AIC23_FORMATS,},
+	.capture = {
+		    .stream_name = "Capture",
+		    .channels_min = 2,
+		    .channels_max = 2,
+		    .rates = AIC23_RATES,
+		    .formats = AIC23_FORMATS,},
+	.ops = {
+		.prepare = tlv320aic23_pcm_prepare,
+		.hw_params = tlv320aic23_hw_params,
+		.shutdown = tlv320aic23_shutdown,
+		},
+	.dai_ops = {
+		    .digital_mute = tlv320aic23_mute,
+		    .set_fmt = tlv320aic23_set_dai_fmt,
+		    .set_sysclk = tlv320aic23_set_dai_sysclk,
+		    }
+};
+EXPORT_SYMBOL_GPL(tlv320aic23_dai);
+
+static int tlv320aic23_suspend(struct platform_device *pdev,
+			       pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int tlv320aic23_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u16 reg;
+
+	/* Sync reg_cache with the hardware */
+	for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+		u16 val = tlv320aic23_read_reg_cache(codec, reg);
+		tlv320aic23_write(codec, reg, val);
+	}
+
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+/*
+ * initialise the AIC23 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int tlv320aic23_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+	u16 reg;
+
+	codec->name = "tlv320aic23";
+	codec->owner = THIS_MODULE;
+	codec->read = tlv320aic23_read_reg_cache;
+	codec->write = tlv320aic23_write;
+	codec->set_bias_level = tlv320aic23_set_bias_level;
+	codec->dai = &tlv320aic23_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
+	codec->reg_cache =
+	    kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* Reset codec */
+	tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+	/* Unmute input */
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
+	tlv320aic23_write(codec, TLV320AIC23_LINVOL,
+			  (reg & (~TLV320AIC23_LIM_MUTED)) |
+			  (TLV320AIC23_LRS_ENABLED));
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
+	tlv320aic23_write(codec, TLV320AIC23_RINVOL,
+			  (reg & (~TLV320AIC23_LIM_MUTED)) |
+			  TLV320AIC23_LRS_ENABLED);
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
+	tlv320aic23_write(codec, TLV320AIC23_ANLG,
+			 (reg) & (~TLV320AIC23_BYPASS_ON) &
+			 (~TLV320AIC23_MICM_MUTED));
+
+	/* Default output volume */
+	tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
+			  TLV320AIC23_DEFAULT_OUT_VOL &
+			  TLV320AIC23_OUT_VOL_MASK);
+	tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
+			  TLV320AIC23_DEFAULT_OUT_VOL &
+			  TLV320AIC23_OUT_VOL_MASK);
+
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+	tlv320aic23_add_controls(codec);
+	tlv320aic23_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+static struct snd_soc_device *tlv320aic23_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int tlv320aic23_codec_probe(struct i2c_client *i2c,
+				   const struct i2c_device_id *i2c_id)
+{
+	struct snd_soc_device *socdev = tlv320aic23_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+		return -EINVAL;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = tlv320aic23_init(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+	put_device(&i2c->dev);
+	return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+	{"tlv320aic23", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+	.driver = {
+		   .name = "tlv320aic23",
+		   },
+	.probe = tlv320aic23_codec_probe,
+	.remove = __exit_p(tlv320aic23_i2c_remove),
+	.id_table = tlv320aic23_id,
+};
+
+#endif
+
+static int tlv320aic23_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 "AIC23 Audio Codec %s\n", AIC23_VERSION);
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	tlv320aic23_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	codec->hw_write = (hw_write_t) i2c_smbus_write_byte_data;
+	codec->hw_read = NULL;
+	ret = i2c_add_driver(&tlv320aic23_i2c_driver);
+	if (ret != 0)
+		printk(KERN_ERR "can't add i2c driver");
+#endif
+	return ret;
+}
+
+static int tlv320aic23_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec->control_data)
+		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&tlv320aic23_i2c_driver);
+#endif
+	kfree(codec->reg_cache);
+	kfree(codec);
+
+	return 0;
+}
+struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
+	.probe = tlv320aic23_probe,
+	.remove = tlv320aic23_remove,
+	.suspend = tlv320aic23_suspend,
+	.resume = tlv320aic23_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
new file mode 100644
index 0000000..79d1faf
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -0,0 +1,122 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd
+ *
+ * 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 _TLV320AIC23_H
+#define _TLV320AIC23_H
+
+/* Codec TLV320AIC23 */
+#define TLV320AIC23_LINVOL		0x00
+#define TLV320AIC23_RINVOL		0x01
+#define TLV320AIC23_LCHNVOL		0x02
+#define TLV320AIC23_RCHNVOL		0x03
+#define TLV320AIC23_ANLG		0x04
+#define TLV320AIC23_DIGT		0x05
+#define TLV320AIC23_PWR			0x06
+#define TLV320AIC23_DIGT_FMT		0x07
+#define TLV320AIC23_SRATE		0x08
+#define TLV320AIC23_ACTIVE		0x09
+#define TLV320AIC23_RESET		0x0F
+
+/* Left (right) line input volume control register */
+#define TLV320AIC23_LRS_ENABLED		0x0100
+#define TLV320AIC23_LIM_MUTED		0x0080
+#define TLV320AIC23_LIV_DEFAULT		0x0017
+#define TLV320AIC23_LIV_MAX		0x001f
+#define TLV320AIC23_LIV_MIN		0x0000
+
+/* Left (right) channel headphone volume control register */
+#define TLV320AIC23_LZC_ON		0x0080
+#define TLV320AIC23_LHV_DEFAULT		0x0079
+#define TLV320AIC23_LHV_MAX		0x007f
+#define TLV320AIC23_LHV_MIN		0x0000
+
+/* Analog audio path control register */
+#define TLV320AIC23_STA_REG(x)		((x)<<6)
+#define TLV320AIC23_STE_ENABLED		0x0020
+#define TLV320AIC23_DAC_SELECTED	0x0010
+#define TLV320AIC23_BYPASS_ON		0x0008
+#define TLV320AIC23_INSEL_MIC		0x0004
+#define TLV320AIC23_MICM_MUTED		0x0002
+#define TLV320AIC23_MICB_20DB		0x0001
+
+/* Digital audio path control register */
+#define TLV320AIC23_DACM_MUTE		0x0008
+#define TLV320AIC23_DEEMP_32K		0x0002
+#define TLV320AIC23_DEEMP_44K		0x0004
+#define TLV320AIC23_DEEMP_48K		0x0006
+#define TLV320AIC23_ADCHP_ON		0x0001
+
+/* Power control down register */
+#define TLV320AIC23_DEVICE_PWR_OFF  	0x0080
+#define TLV320AIC23_CLK_OFF		0x0040
+#define TLV320AIC23_OSC_OFF		0x0020
+#define TLV320AIC23_OUT_OFF		0x0010
+#define TLV320AIC23_DAC_OFF		0x0008
+#define TLV320AIC23_ADC_OFF		0x0004
+#define TLV320AIC23_MIC_OFF		0x0002
+#define TLV320AIC23_LINE_OFF		0x0001
+
+/* Digital audio interface register */
+#define TLV320AIC23_MS_MASTER		0x0040
+#define TLV320AIC23_LRSWAP_ON		0x0020
+#define TLV320AIC23_LRP_ON		0x0010
+#define TLV320AIC23_IWL_16		0x0000
+#define TLV320AIC23_IWL_20		0x0004
+#define TLV320AIC23_IWL_24		0x0008
+#define TLV320AIC23_IWL_32		0x000C
+#define TLV320AIC23_FOR_I2S		0x0002
+#define TLV320AIC23_FOR_DSP		0x0003
+#define TLV320AIC23_FOR_LJUST		0x0001
+
+/* Sample rate control register */
+#define TLV320AIC23_CLKOUT_HALF		0x0080
+#define TLV320AIC23_CLKIN_HALF		0x0040
+#define TLV320AIC23_BOSR_384fs		0x0002	/* BOSR_272fs in USB mode */
+#define TLV320AIC23_USB_CLK_ON		0x0001
+#define TLV320AIC23_SR_MASK             0xf
+#define TLV320AIC23_CLKOUT_SHIFT        7
+#define TLV320AIC23_CLKIN_SHIFT         6
+#define TLV320AIC23_SR_SHIFT            2
+#define TLV320AIC23_BOSR_SHIFT          1
+
+/* Digital interface register */
+#define TLV320AIC23_ACT_ON		0x0001
+
+/*
+ * AUDIO related MACROS
+ */
+
+#define TLV320AIC23_DEFAULT_OUT_VOL	0x70
+#define TLV320AIC23_DEFAULT_IN_VOLUME	0x10
+
+#define TLV320AIC23_OUT_VOL_MIN		TLV320AIC23_LHV_MIN
+#define TLV320AIC23_OUT_VOL_MAX		TLV320AIC23_LHV_MAX
+#define TLV320AIC23_OUT_VO_RANGE	(TLV320AIC23_OUT_VOL_MAX - \
+					TLV320AIC23_OUT_VOL_MIN)
+#define TLV320AIC23_OUT_VOL_MASK	TLV320AIC23_OUT_VOL_MAX
+
+#define TLV320AIC23_IN_VOL_MIN		TLV320AIC23_LIV_MIN
+#define TLV320AIC23_IN_VOL_MAX		TLV320AIC23_LIV_MAX
+#define TLV320AIC23_IN_VOL_RANGE	(TLV320AIC23_IN_VOL_MAX - \
+					TLV320AIC23_IN_VOL_MIN)
+#define TLV320AIC23_IN_VOL_MASK		TLV320AIC23_IN_VOL_MAX
+
+#define TLV320AIC23_SIDETONE_MASK	0x1c0
+#define TLV320AIC23_SIDETONE_0		0x100
+#define TLV320AIC23_SIDETONE_6		0x000
+#define TLV320AIC23_SIDETONE_9		0x040
+#define TLV320AIC23_SIDETONE_12		0x080
+#define TLV320AIC23_SIDETONE_18		0x0c0
+
+extern struct snd_soc_dai tlv320aic23_dai;
+extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
+
+#endif /* _TLV320AIC23_H */
-- 
1.5.6.5

[PATCH 2/3] ASoC: Add support for osk5912

[Mark Brown]

From: Arun KS <arunks@mistralsolutions.com>

Adding ASoC machine driver for osk5912

Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/Kconfig   |    8 ++
 sound/soc/omap/Makefile  |    2 +
 sound/soc/omap/osk5912.c |  232 ++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 242 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/omap/osk5912.c

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index aea27e7..8b7766b 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_OSK5912
+	tristate "SoC Audio support for omap osk5912"
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on osk5912.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index d8d8d58..e09d1f2 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
+snd-soc-osk5912-objs := osk5912.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 0000000..0fe7337
--- /dev/null
+++ b/sound/soc/omap/osk5912.c
@@ -0,0 +1,232 @@
+/*
+ * osk5912.c  --  SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS  <arunks@mistralsolutions.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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+	return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_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_DSP_A |
+				  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_DSP_A |
+				  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 osk_ops = {
+	.startup = osk_startup,
+	.hw_params = osk_hw_params,
+	.shutdown = osk_shutdown,
+};
+
+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 osk_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add osk5912 specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up osk5912 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 osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = osk_tlv320aic23_init,
+	.ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_osk = {
+	.name = "OSK5912",
+	.dai_link = &osk_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+	.machine = &snd_soc_machine_osk,
+	.platform = &omap_soc_platform,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+	int err;
+	u32 curRate;
+	struct device *dev;
+
+	if (!(machine_is_omap_osk()))
+		return -ENODEV;
+
+	osk_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!osk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
+	osk_snd_devdata.dev = &osk_snd_device->dev;
+	*(unsigned int *)osk_dai.cpu_dai->private_data = 0;	/* McBSP1 */
+	err = platform_device_add(osk_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &osk_snd_device->dev;
+
+	tlv320aic23_mclk = clk_get(dev, "mclk");
+	if (IS_ERR(tlv320aic23_mclk)) {
+		printk(KERN_ERR "Could not get mclk clock\n");
+		return -ENODEV;
+	}
+
+	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
+		/* MCLK is already in use */
+		printk(KERN_WARNING
+		       "MCLK in use at %d Hz. We change it to %d Hz\n",
+		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+	}
+
+	/*
+	 * Configure 12 MHz output on MCLK.
+	 */
+	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+	if (curRate != CODEC_CLOCK) {
+		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+			err = -ECANCELED;
+			goto err1;
+		}
+	}
+
+	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
+	       clk_get_usecount(tlv320aic23_mclk));
+
+	return 0;
+err1:
+	clk_put(tlv320aic23_mclk);
+	platform_device_del(osk_snd_device);
+	platform_device_put(osk_snd_device);
+
+	return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+	platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");
-- 
1.5.6.5

[PATCH 3/3] ASoC: Add DSP DAI format support to the OMAP McBSP driver

[Mark Brown]

From: Arun KS <arunks@mistralsolutions.com>

Enables DSP DAI format for McBSP in OMAP platform driver

Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap-mcbsp.c |    5 +++++
 1 files changed, 5 insertions(+), 0 deletions(-)

diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 35310e1..fb920e1 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -245,6 +245,11 @@ 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:
+		/* 0-bit data delay */
+		regs->rcr2      |= RDATDLY(0);
+		regs->xcr2      |= XDATDLY(0);
+		break;
 	default:
 		/* Unsupported data format */
 		return -EINVAL;
-- 
1.5.6.5

Re: More ASoC updates

[Takashi Iwai]

At Fri, 3 Oct 2008 09:53:38 +0100,
Mark Brown wrote:
> 
> The following changes since commit ddaea5119fd2efde7b3936d4ccfa34c4201be0d3:
>   Takashi Iwai (1):
>         Merge branch 'asoc-fixes' into topic/asoc
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled now.

> There's more to come on the TLV320AIC23 but at this point it's all
> cleanups and it's much easier to review incremental patches.

OK.  I hope these can be merged before 2.6.28 merge window.


Takashi