From d273ebe77a780d3aba1f5b86615af3b0e87b76bf Mon Sep 17 00:00:00 2001
From: jassi brar <jassisinghbrar@gmail.com>
Date: Mon, 22 Feb 2010 15:58:04 +0900
Subject: ASoC: Pass dai_link as argument to platform suspend and resume

Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel-pcm.c | 6 ++++--
 sound/soc/soc-core.c        | 4 ++--
 2 files changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 9ef6b96373f5..67a32920af22 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -415,8 +415,9 @@ static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm)
 }
 
 #ifdef CONFIG_PM
-static int atmel_pcm_suspend(struct snd_soc_dai *dai)
+static int atmel_pcm_suspend(struct snd_soc_dai_link *dai_link)
 {
+	struct snd_soc_dai *dai = dai_link->cpu_dai;
 	struct snd_pcm_runtime *runtime = dai->runtime;
 	struct atmel_runtime_data *prtd;
 	struct atmel_pcm_dma_params *params;
@@ -439,8 +440,9 @@ static int atmel_pcm_suspend(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static int atmel_pcm_resume(struct snd_soc_dai *dai)
+static int atmel_pcm_resume(struct snd_soc_dai_link *dai_link)
 {
+	struct snd_soc_dai *dai = dai_link->cpu_dai;
 	struct snd_pcm_runtime *runtime = dai->runtime;
 	struct atmel_runtime_data *prtd;
 	struct atmel_pcm_dma_params *params;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e1c0336868e1..524a85a73aca 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -854,7 +854,7 @@ static int soc_suspend(struct device *dev)
 		if (cpu_dai->suspend && !cpu_dai->ac97_control)
 			cpu_dai->suspend(cpu_dai);
 		if (platform->suspend)
-			platform->suspend(cpu_dai);
+			platform->suspend(&card->dai_link[i]);
 	}
 
 	/* close any waiting streams and save state */
@@ -943,7 +943,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		if (cpu_dai->resume && !cpu_dai->ac97_control)
 			cpu_dai->resume(cpu_dai);
 		if (platform->resume)
-			platform->resume(cpu_dai);
+			platform->resume(&card->dai_link[i]);
 	}
 
 	if (card->resume_post)
-- 
cgit v1.2.3


From 10cab262f49ee4574a20ab4c445bc4fdc41c36a8 Mon Sep 17 00:00:00 2001
From: jassi brar <jassisinghbrar@gmail.com>
Date: Mon, 22 Feb 2010 15:59:02 +0900
Subject: ASoC: Change how suspend and resume obtain the PCM runtime

Currently only the atmel driver make use of snd_soc_dai.runtime field.
If the dais are to be shared among two or more dai_links, the field
must be got rid of.
So, in atmel driver reach the substream via dai_link->pcm so as to
not depend of snd_soc_dai.runtime field.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/atmel-pcm.c | 12 ++++++++----
 1 file changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 67a32920af22..fdb255372127 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -417,8 +417,10 @@ static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm)
 #ifdef CONFIG_PM
 static int atmel_pcm_suspend(struct snd_soc_dai_link *dai_link)
 {
-	struct snd_soc_dai *dai = dai_link->cpu_dai;
-	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct snd_pcm *pcm = dai_link->pcm;
+	struct snd_pcm_str *stream = &pcm->streams[0];
+	struct snd_pcm_substream *substream = stream->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct atmel_runtime_data *prtd;
 	struct atmel_pcm_dma_params *params;
 
@@ -442,8 +444,10 @@ static int atmel_pcm_suspend(struct snd_soc_dai_link *dai_link)
 
 static int atmel_pcm_resume(struct snd_soc_dai_link *dai_link)
 {
-	struct snd_soc_dai *dai = dai_link->cpu_dai;
-	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct snd_pcm *pcm = dai_link->pcm;
+	struct snd_pcm_str *stream = &pcm->streams[0];
+	struct snd_pcm_substream *substream = stream->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct atmel_runtime_data *prtd;
 	struct atmel_pcm_dma_params *params;
 
-- 
cgit v1.2.3


From 6423c1875c87fa5ae56974ab8386d7c6110e3701 Mon Sep 17 00:00:00 2001
From: jassi brar <jassisinghbrar@gmail.com>
Date: Mon, 22 Feb 2010 16:00:20 +0900
Subject: ASoC: Remove runtime field from DAI

In order for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h | 1 -
 sound/soc/soc-core.c    | 2 --
 2 files changed, 3 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 061f16d4c878..20de0bcaa137 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -215,7 +215,6 @@ struct snd_soc_dai {
 	unsigned int symmetric_rates:1;
 
 	/* DAI runtime info */
-	struct snd_pcm_runtime *runtime;
 	struct snd_soc_codec *codec;
 	unsigned int active;
 	unsigned char pop_wait:1;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 524a85a73aca..1215a0ec2df0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -459,7 +459,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	else
 		cpu_dai->capture.active = codec_dai->capture.active = 1;
 	cpu_dai->active = codec_dai->active = 1;
-	cpu_dai->runtime = runtime;
 	card->codec->active++;
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -559,7 +558,6 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 
 	if (platform->pcm_ops->close)
 		platform->pcm_ops->close(substream);
-	cpu_dai->runtime = NULL;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* start delayed pop wq here for playback streams */
-- 
cgit v1.2.3


From 14dc5734bdac2629ed4228f3d30662bb440a3982 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Fri, 26 Feb 2010 09:12:32 +0900
Subject: ASoC: Allow mulitple usage count of codec and cpu dai

If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  2 +-
 sound/soc/soc-core.c | 31 ++++++++++++++++++-------------
 2 files changed, 19 insertions(+), 14 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 27a2ad9a6b8b..f792c1881b0a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -375,7 +375,7 @@ struct snd_soc_pcm_stream {
 	unsigned int rate_max;		/* max rate */
 	unsigned int channels_min;	/* min channels */
 	unsigned int channels_max;	/* max channels */
-	unsigned int active:1;		/* stream is in use */
+	unsigned int active;		/* num of active users of the stream */
 };
 
 /* SoC audio ops */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1215a0ec2df0..f30959cc954a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -454,11 +454,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
 		 runtime->hw.rate_max);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->playback.active = codec_dai->playback.active = 1;
-	else
-		cpu_dai->capture.active = codec_dai->capture.active = 1;
-	cpu_dai->active = codec_dai->active = 1;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		cpu_dai->playback.active++;
+		codec_dai->playback.active++;
+	} else {
+		cpu_dai->capture.active++;
+		codec_dai->capture.active++;
+	}
+	cpu_dai->active++;
+	codec_dai->active++;
 	card->codec->active++;
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -530,15 +534,16 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 
 	mutex_lock(&pcm_mutex);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->playback.active = codec_dai->playback.active = 0;
-	else
-		cpu_dai->capture.active = codec_dai->capture.active = 0;
-
-	if (codec_dai->playback.active == 0 &&
-		codec_dai->capture.active == 0) {
-		cpu_dai->active = codec_dai->active = 0;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		cpu_dai->playback.active--;
+		codec_dai->playback.active--;
+	} else {
+		cpu_dai->capture.active--;
+		codec_dai->capture.active--;
 	}
+
+	cpu_dai->active--;
+	codec_dai->active--;
 	codec->active--;
 
 	/* Muting the DAC suppresses artifacts caused during digital
-- 
cgit v1.2.3


From eeec124685164cb8f9bbf274e1d4c9b9c2f8d6c9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 23 Feb 2010 11:15:34 +0000
Subject: ASoC: Wolfson Microelectronics 1133-EV1 audio support

Initial support for audio using the 1133-EV1 audio and PMIC module for
the i.MX31ADS.  Currently only playback is supported, and the FIQ DMA
driver has performance problems at higher sample rates which cause
audible dropouts.

This driver is based heavily on an out of tree one written by Liam
Girdwood.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/imx/Kconfig      |   8 ++
 sound/soc/imx/Makefile     |   3 +
 sound/soc/imx/wm1133-ev1.c | 291 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 302 insertions(+)
 create mode 100644 sound/soc/imx/wm1133-ev1.c

(limited to 'sound/soc')

diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index c7d0fd9b7de8..c045da8ff61c 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,3 +11,11 @@ config SND_IMX_SOC
 config SND_MXC_SOC_SSI
 	tristate
 
+config SND_MXC_SOC_WM1133_EV1
+	tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
+	depends on SND_IMX_SOC && EXPERIMENTAL
+	select SND_SOC_WM8350
+	select SND_MXC_SOC_SSI
+	help
+	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
+	  PMIC board with WM8835x fitted.
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index 9f8bb92ddfcc..2d203635ac11 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -9,4 +9,7 @@ obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o
 
 # i.MX Machine Support
 snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
new file mode 100644
index 000000000000..b75fcde85e88
--- /dev/null
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -0,0 +1,291 @@
+/*
+ *  wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
+ *
+ *  Copyright (c) 2010 Wolfson Microelectronics plc
+ *  Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  Based on an earlier driver for the same hardware by Liam Girdwood.
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audmux.h>
+
+#include "imx-ssi.h"
+#include "../codecs/wm8350.h"
+
+/* There is a silicon mic on the board optionally connected via a solder pad
+ * SP1.  Define this to enable it.
+ */
+#undef USE_SIMIC
+
+struct _wm8350_audio {
+	unsigned int channels;
+	snd_pcm_format_t format;
+	unsigned int rate;
+	unsigned int sysclk;
+	unsigned int bclkdiv;
+	unsigned int clkdiv;
+	unsigned int lr_rate;
+};
+
+/* in order of power consumption per rate (lowest first) */
+static const struct _wm8350_audio wm8350_audio[] = {
+	/* 16bit mono modes */
+	{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
+	 WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
+
+	/* 16 bit stereo modes */
+	{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
+	 WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
+	 WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
+	 WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
+	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
+	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
+	 WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
+	 WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
+	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+	{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
+	 WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
+
+	/* 24bit stereo modes */
+	{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
+	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+	{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
+	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+	{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
+	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+	{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
+	 WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
+};
+
+static int wm1133_ev1_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 i, found = 0;
+	snd_pcm_format_t format = params_format(params);
+	unsigned int rate = params_rate(params);
+	unsigned int channels = params_channels(params);
+	u32 dai_format;
+
+	/* find the correct audio parameters */
+	for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
+		if (rate == wm8350_audio[i].rate &&
+		    format == wm8350_audio[i].format &&
+		    channels == wm8350_audio[i].channels) {
+			found = 1;
+			break;
+		}
+	}
+	if (!found)
+		return -EINVAL;
+
+	/* codec FLL input is 14.75 MHz from MCLK */
+	snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
+
+	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM;
+
+	/* set codec DAI configuration */
+	snd_soc_dai_set_fmt(codec_dai, dai_format);
+
+	/* set cpu DAI configuration */
+	snd_soc_dai_set_fmt(cpu_dai, dai_format);
+
+	/* TODO: The SSI driver should figure this out for us */
+	switch (channels) {
+	case 2:
+		snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+		break;
+	case 1:
+		snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set MCLK as the codec system clock for DAC and ADC */
+	snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
+			       wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
+
+	/* set codec BCLK division for sample rate */
+	snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
+			       wm8350_audio[i].bclkdiv);
+
+	/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
+	snd_soc_dai_set_clkdiv(codec_dai,
+			       WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
+	snd_soc_dai_set_clkdiv(codec_dai,
+			       WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
+
+	/* now configure DAC and ADC clocks */
+	snd_soc_dai_set_clkdiv(codec_dai,
+			       WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
+
+	snd_soc_dai_set_clkdiv(codec_dai,
+			       WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
+
+	return 0;
+}
+
+static struct snd_soc_ops wm1133_ev1_ops = {
+	.hw_params = wm1133_ev1_hw_params,
+};
+
+static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
+#ifdef USE_SIMIC
+	SND_SOC_DAPM_MIC("SiMIC", NULL),
+#endif
+	SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+/* imx32ads soc_card audio map */
+static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
+
+#ifdef USE_SIMIC
+	/* SiMIC --> IN1LN (with automatic bias) via SP1 */
+	{ "IN1LN", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "SiMIC" },
+#endif
+
+	/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
+	{ "IN1LN", NULL, "Mic Bias" },
+	{ "IN1LP", NULL, "Mic1 Jack" },
+	{ "Mic Bias", NULL, "Mic1 Jack" },
+
+	/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
+	{ "IN1RN", NULL, "Mic Bias" },
+	{ "IN1RP", NULL, "Mic1 Jack" },
+	{ "Mic Bias", NULL, "Mic1 Jack" },
+
+	/* Line in Jack --> AUX (L+R) */
+	{ "IN3R", NULL, "Line In Jack" },
+	{ "IN3L", NULL, "Line In Jack" },
+
+	/* Out1 --> Headphone Jack */
+	{ "Headphone Jack", NULL, "OUT1R" },
+	{ "Headphone Jack", NULL, "OUT1L" },
+
+	/* Out1 --> Line Out Jack */
+	{ "Line Out Jack", NULL, "OUT2R" },
+	{ "Line Out Jack", NULL, "OUT2L" },
+};
+
+static struct snd_soc_jack hp_jack;
+
+static struct snd_soc_jack_pin hp_jack_pins[] = {
+	{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
+};
+
+static int wm1133_ev1_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_card *card = codec->socdev->card;
+
+	snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets,
+				  ARRAY_SIZE(wm1133_ev1_widgets));
+
+	snd_soc_dapm_add_routes(codec, wm1133_ev1_map,
+				ARRAY_SIZE(wm1133_ev1_map));
+
+	/* Headphone jack detection */
+	snd_soc_jack_new(card, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
+	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
+			      hp_jack_pins);
+	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
+
+	return 0;
+}
+
+
+static struct snd_soc_dai_link wm1133_ev1_dai = {
+	.name = "WM1133-EV1",
+	.stream_name = "Audio",
+	.cpu_dai = &imx_ssi_pcm_dai[0],
+	.codec_dai = &wm8350_dai,
+	.init = wm1133_ev1_init,
+	.ops = &wm1133_ev1_ops,
+	.symmetric_rates = 1,
+};
+
+static struct snd_soc_card wm1133_ev1 = {
+	.name = "WM1133-EV1",
+	.platform = &imx_soc_platform,
+	.dai_link = &wm1133_ev1_dai,
+	.num_links = 1,
+};
+
+static struct snd_soc_device wm1133_ev1_snd_devdata = {
+	.card = &wm1133_ev1,
+	.codec_dev = &soc_codec_dev_wm8350,
+};
+
+static struct platform_device *wm1133_ev1_snd_device;
+
+static int __init wm1133_ev1_audio_init(void)
+{
+	int ret;
+	unsigned int ptcr, pdcr;
+
+	/* SSI0 mastered by port 5 */
+	ptcr = MXC_AUDMUX_V2_PTCR_SYN |
+		MXC_AUDMUX_V2_PTCR_TFSDIR |
+		MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
+		MXC_AUDMUX_V2_PTCR_TCLKDIR |
+		MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+	pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
+	mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
+
+	ptcr = MXC_AUDMUX_V2_PTCR_SYN;
+	pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
+	mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
+
+	wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!wm1133_ev1_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1_snd_devdata);
+	wm1133_ev1_snd_devdata.dev = &wm1133_ev1_snd_device->dev;
+	ret = platform_device_add(wm1133_ev1_snd_device);
+
+	if (ret)
+		platform_device_put(wm1133_ev1_snd_device);
+
+	return ret;
+}
+module_init(wm1133_ev1_audio_init);
+
+static void __exit wm1133_ev1_audio_exit(void)
+{
+	platform_device_unregister(wm1133_ev1_snd_device);
+}
+module_exit(wm1133_ev1_audio_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From 50831450503b828b3f8c031aa664340062afb3b0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Wed, 3 Mar 2010 15:08:04 +0200
Subject: ASoC: core: fix tailing whitespace in soc_pcm_apply_symmetry

My editor removes the tailing spaces, which causes problems when
changing the soc-core.c
Removing the space.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f30959cc954a..7b4179eca893 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -315,7 +315,7 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
 
 	if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
 	    machine->symmetric_rates) {
-		dev_dbg(card->dev, "Symmetry forces %dHz rate\n", 
+		dev_dbg(card->dev, "Symmetry forces %dHz rate\n",
 			machine->rate);
 
 		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
-- 
cgit v1.2.3


From 377b6f62effcb91c53cd7ff8709a94d72d23b6ae Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Wed, 3 Mar 2010 15:08:06 +0200
Subject: ASoC: core: soc level wrapper for pcm_pointer callback

Create a soc level wrapper for pcm_pointer callback.
This will facilitate the soc level handling of different
HW buffers in the audio path.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 19 ++++++++++++++++++-
 1 file changed, 18 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7b4179eca893..feb572c616cd 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -800,6 +800,23 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	return 0;
 }
 
+/*
+ * soc level wrapper for pointer callback
+ */
+static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
+	snd_pcm_uframes_t offset = 0;
+
+	if (platform->pcm_ops->pointer)
+		offset = platform->pcm_ops->pointer(substream);
+
+	return offset;
+}
+
 /* ASoC PCM operations */
 static struct snd_pcm_ops soc_pcm_ops = {
 	.open		= soc_pcm_open,
@@ -808,6 +825,7 @@ static struct snd_pcm_ops soc_pcm_ops = {
 	.hw_free	= soc_pcm_hw_free,
 	.prepare	= soc_pcm_prepare,
 	.trigger	= soc_pcm_trigger,
+	.pointer	= soc_pcm_pointer,
 };
 
 #ifdef CONFIG_PM
@@ -1328,7 +1346,6 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	dai_link->pcm = pcm;
 	pcm->private_data = rtd;
 	soc_pcm_ops.mmap = platform->pcm_ops->mmap;
-	soc_pcm_ops.pointer = platform->pcm_ops->pointer;
 	soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
 	soc_pcm_ops.copy = platform->pcm_ops->copy;
 	soc_pcm_ops.silence = platform->pcm_ops->silence;
-- 
cgit v1.2.3


From 258020d0882e89c1462800a70eb414b8a4fec78c Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Wed, 3 Mar 2010 15:08:07 +0200
Subject: ASoC: core: Add delay operation to snd_soc_dai_ops

The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h |  6 ++++++
 include/sound/soc.h     |  7 +++++++
 sound/soc/soc-core.c    | 18 ++++++++++++++++++
 3 files changed, 31 insertions(+)

(limited to 'sound/soc')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 20de0bcaa137..6cf76a41501e 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -182,6 +182,12 @@ struct snd_soc_dai_ops {
 		struct snd_soc_dai *);
 	int (*trigger)(struct snd_pcm_substream *, int,
 		struct snd_soc_dai *);
+	/*
+	 * For hardware based FIFO caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
 };
 
 /*
diff --git a/include/sound/soc.h b/include/sound/soc.h
index f792c1881b0a..dbfec16015de 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -470,6 +470,13 @@ struct snd_soc_platform {
 		struct snd_pcm *);
 	void (*pcm_free)(struct snd_pcm *);
 
+	/*
+	 * For platform caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+
 	/* platform stream ops */
 	struct snd_pcm_ops *pcm_ops;
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index feb572c616cd..4011ad3dc57a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -802,6 +802,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 
 /*
  * soc level wrapper for pointer callback
+ * If cpu_dai, codec_dai, platform driver has the delay callback, than
+ * the runtime->delay will be updated accordingly.
  */
 static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 {
@@ -809,11 +811,27 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = card->platform;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t offset = 0;
+	snd_pcm_sframes_t delay = 0;
 
 	if (platform->pcm_ops->pointer)
 		offset = platform->pcm_ops->pointer(substream);
 
+	if (cpu_dai->ops->delay)
+		delay += cpu_dai->ops->delay(substream, cpu_dai);
+
+	if (codec_dai->ops->delay)
+		delay += codec_dai->ops->delay(substream, codec_dai);
+
+	if (platform->delay)
+		delay += platform->delay(substream, codec_dai);
+
+	runtime->delay = delay;
+
 	return offset;
 }
 
-- 
cgit v1.2.3


From a24d62d297fad31c866eef28e3d8b691a2b14a92 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 1 Mar 2010 20:00:04 +0000
Subject: ASoC: Prettify wm8960 logging

The driver name gets used by dev_() logging so use something a bit
more idiomatic.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8960.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index d07bcc1e1c60..35171e4354a2 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -882,7 +882,7 @@ MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
 
 static struct i2c_driver wm8960_i2c_driver = {
 	.driver = {
-		.name = "WM8960 I2C Codec",
+		.name = "wm8960",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8960_i2c_probe,
-- 
cgit v1.2.3


From b6877a477d356a7c07a6c173d58c34a0a6abb086 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Mar 2010 11:43:38 +0000
Subject: ASoC: Move WM8960 platform data into include/sound

Avoids machine files having to peer into sound/soc which is a bit
rude and icky.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8960.h    | 22 ++++++++++++++++++++++
 sound/soc/codecs/wm8960.c |  1 +
 sound/soc/codecs/wm8960.h | 10 ----------
 3 files changed, 23 insertions(+), 10 deletions(-)
 create mode 100644 include/sound/wm8960.h

(limited to 'sound/soc')

diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h
new file mode 100644
index 000000000000..808a258e696c
--- /dev/null
+++ b/include/sound/wm8960.h
@@ -0,0 +1,22 @@
+/*
+ * wm8960.h  --  WM8960 Soc Audio driver platform data
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8960_PDATA_H
+#define _WM8960_PDATA_H
+
+#define WM8960_DRES_400R 0
+#define WM8960_DRES_200R 1
+#define WM8960_DRES_600R 2
+#define WM8960_DRES_150R 3
+#define WM8960_DRES_MAX  3
+
+struct wm8960_data {
+	int dres;  /* Discharge resistance for headphone outputs */
+};
+
+#endif
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 35171e4354a2..cf5cb3f73b69 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -22,6 +22,7 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
+#include <sound/wm8960.h>
 
 #include "wm8960.h"
 
diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h
index c9af56c9d9d4..d67bfe1300da 100644
--- a/sound/soc/codecs/wm8960.h
+++ b/sound/soc/codecs/wm8960.h
@@ -114,14 +114,4 @@
 extern struct snd_soc_dai wm8960_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8960;
 
-#define WM8960_DRES_400R 0
-#define WM8960_DRES_200R 1
-#define WM8960_DRES_600R 2
-#define WM8960_DRES_150R 3
-#define WM8960_DRES_MAX  3
-
-struct wm8960_data {
-	int dres;
-};
-
 #endif
-- 
cgit v1.2.3


From 913d7b4cc0d958df9f2e4bc0e6926c037d96d07e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Mar 2010 13:47:03 +0000
Subject: ASoC: Add support for WM8960 capless mode

The WM8960 headphone outputs can be run in capless mode with OUT3
used to drive a pseudo ground for the headphone drivers. In this
mode the mono mixer is not used, the mixer should be turned on
in concert with the headphone output drivers and the device bias
levels are managed differently.

Also tweak the existing bias management to remove the use of active
discharge while we're at it since that's often audible.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8960.h    |   2 +
 sound/soc/codecs/wm8960.c | 206 +++++++++++++++++++++++++++++++++++++++-------
 2 files changed, 179 insertions(+), 29 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h
index 808a258e696c..74e9a95529c5 100644
--- a/include/sound/wm8960.h
+++ b/include/sound/wm8960.h
@@ -16,6 +16,8 @@
 #define WM8960_DRES_MAX  3
 
 struct wm8960_data {
+	bool capless;  /* Headphone outputs configured in capless mode */
+
 	int dres;  /* Discharge resistance for headphone outputs */
 };
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index cf5cb3f73b69..c2960d3ec6df 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -31,8 +31,14 @@
 struct snd_soc_codec_device soc_codec_dev_wm8960;
 
 /* R25 - Power 1 */
+#define WM8960_VMID_MASK 0x180
 #define WM8960_VREF      0x40
 
+/* R26 - Power 2 */
+#define WM8960_PWR2_LOUT1	0x40
+#define WM8960_PWR2_ROUT1	0x20
+#define WM8960_PWR2_OUT3	0x02
+
 /* R28 - Anti-pop 1 */
 #define WM8960_POBCTRL   0x80
 #define WM8960_BUFDCOPEN 0x10
@@ -42,6 +48,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8960;
 
 /* R29 - Anti-pop 2 */
 #define WM8960_DISOP     0x40
+#define WM8960_DRES_MASK 0x30
 
 /*
  * wm8960 register cache
@@ -68,6 +75,9 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
 struct wm8960_priv {
 	u16 reg_cache[WM8960_CACHEREGNUM];
 	struct snd_soc_codec codec;
+	struct snd_soc_dapm_widget *lout1;
+	struct snd_soc_dapm_widget *rout1;
+	struct snd_soc_dapm_widget *out3;
 };
 
 #define wm8960_reset(c)	snd_soc_write(c, WM8960_RESET, 0)
@@ -226,10 +236,6 @@ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0,
 	&wm8960_routput_mixer[0],
 	ARRAY_SIZE(wm8960_routput_mixer)),
 
-SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
-	&wm8960_mono_out[0],
-	ARRAY_SIZE(wm8960_mono_out)),
-
 SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0),
 SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0),
 
@@ -248,6 +254,17 @@ SND_SOC_DAPM_OUTPUT("SPK_RN"),
 SND_SOC_DAPM_OUTPUT("OUT3"),
 };
 
+static const struct snd_soc_dapm_widget wm8960_dapm_widgets_out3[] = {
+SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
+	&wm8960_mono_out[0],
+	ARRAY_SIZE(wm8960_mono_out)),
+};
+
+/* Represent OUT3 as a PGA so that it gets turned on with LOUT1/ROUT1 */
+static const struct snd_soc_dapm_widget wm8960_dapm_widgets_capless[] = {
+SND_SOC_DAPM_PGA("OUT3 VMID", WM8960_POWER2, 1, 0, NULL, 0),
+};
+
 static const struct snd_soc_dapm_route audio_paths[] = {
 	{ "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" },
 	{ "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" },
@@ -278,9 +295,6 @@ static const struct snd_soc_dapm_route audio_paths[] = {
 	{ "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } ,
 	{ "Right Output Mixer", "PCM Playback Switch", "Right DAC" },
 
-	{ "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
-	{ "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
-
 	{ "LOUT1 PGA", NULL, "Left Output Mixer" },
 	{ "ROUT1 PGA", NULL, "Right Output Mixer" },
 
@@ -297,17 +311,65 @@ static const struct snd_soc_dapm_route audio_paths[] = {
 	{ "SPK_LP", NULL, "Left Speaker Output" },
 	{ "SPK_RN", NULL, "Right Speaker Output" },
 	{ "SPK_RP", NULL, "Right Speaker Output" },
+};
+
+static const struct snd_soc_dapm_route audio_paths_out3[] = {
+	{ "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
+	{ "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
 
 	{ "OUT3", NULL, "Mono Output Mixer", }
 };
 
+static const struct snd_soc_dapm_route audio_paths_capless[] = {
+	{ "HP_L", NULL, "OUT3 VMID" },
+	{ "HP_R", NULL, "OUT3 VMID" },
+
+	{ "OUT3 VMID", NULL, "Left Output Mixer" },
+	{ "OUT3 VMID", NULL, "Right Output Mixer" },
+};
+
 static int wm8960_add_widgets(struct snd_soc_codec *codec)
 {
+	struct wm8960_data *pdata = codec->dev->platform_data;
+	struct wm8960_priv *wm8960 = codec->private_data;
+	struct snd_soc_dapm_widget *w;
+
 	snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
 				  ARRAY_SIZE(wm8960_dapm_widgets));
 
 	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
 
+	/* In capless mode OUT3 is used to provide VMID for the
+	 * headphone outputs, otherwise it is used as a mono mixer.
+	 */
+	if (pdata && pdata->capless) {
+		snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_capless,
+					  ARRAY_SIZE(wm8960_dapm_widgets_capless));
+
+		snd_soc_dapm_add_routes(codec, audio_paths_capless,
+					ARRAY_SIZE(audio_paths_capless));
+	} else {
+		snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_out3,
+					  ARRAY_SIZE(wm8960_dapm_widgets_out3));
+
+		snd_soc_dapm_add_routes(codec, audio_paths_out3,
+					ARRAY_SIZE(audio_paths_out3));
+	}
+
+	/* We need to power up the headphone output stage out of
+	 * sequence for capless mode.  To save scanning the widget
+	 * list each time to find the desired power state do so now
+	 * and save the result.
+	 */
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (strcmp(w->name, "LOUT1 PGA") == 0)
+			wm8960->lout1 = w;
+		if (strcmp(w->name, "ROUT1 PGA") == 0)
+			wm8960->rout1 = w;
+		if (strcmp(w->name, "OUT3 VMID") == 0)
+			wm8960->out3 = w;
+	}
+	
 	return 0;
 }
 
@@ -408,10 +470,9 @@ static int wm8960_mute(struct snd_soc_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8960_set_bias_level(struct snd_soc_codec *codec,
-				 enum snd_soc_bias_level level)
+static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
 {
-	struct wm8960_data *pdata = codec->dev->platform_data;
 	u16 reg;
 
 	switch (level) {
@@ -430,18 +491,8 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec,
 		if (codec->bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable anti-pop features */
 			snd_soc_write(codec, WM8960_APOP1,
-				     WM8960_POBCTRL | WM8960_SOFT_ST |
-				     WM8960_BUFDCOPEN | WM8960_BUFIOEN);
-
-			/* Discharge HP output */
-			reg = WM8960_DISOP;
-			if (pdata)
-				reg |= pdata->dres << 4;
-			snd_soc_write(codec, WM8960_APOP2, reg);
-
-			msleep(400);
-
-			snd_soc_write(codec, WM8960_APOP2, 0);
+				      WM8960_POBCTRL | WM8960_SOFT_ST |
+				      WM8960_BUFDCOPEN | WM8960_BUFIOEN);
 
 			/* Enable & ramp VMID at 2x50k */
 			reg = snd_soc_read(codec, WM8960_POWER1);
@@ -472,8 +523,101 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec,
 		/* Disable VMID and VREF, let them discharge */
 		snd_soc_write(codec, WM8960_POWER1, 0);
 		msleep(600);
+		break;
+	}
+
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
+					 enum snd_soc_bias_level level)
+{
+	struct wm8960_priv *wm8960 = codec->private_data;
+	int reg;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		switch (codec->bias_level) {
+		case SND_SOC_BIAS_STANDBY:
+			/* Enable anti pop mode */
+			snd_soc_update_bits(codec, WM8960_APOP1,
+					    WM8960_POBCTRL | WM8960_SOFT_ST |
+					    WM8960_BUFDCOPEN,
+					    WM8960_POBCTRL | WM8960_SOFT_ST |
+					    WM8960_BUFDCOPEN);
+
+			/* Enable LOUT1, ROUT1 and OUT3 if they're enabled */
+			reg = 0;
+			if (wm8960->lout1 && wm8960->lout1->power)
+				reg |= WM8960_PWR2_LOUT1;
+			if (wm8960->rout1 && wm8960->rout1->power)
+				reg |= WM8960_PWR2_ROUT1;
+			if (wm8960->out3 && wm8960->out3->power)
+				reg |= WM8960_PWR2_OUT3;
+			snd_soc_update_bits(codec, WM8960_POWER2,
+					    WM8960_PWR2_LOUT1 |
+					    WM8960_PWR2_ROUT1 |
+					    WM8960_PWR2_OUT3, reg);
+
+			/* Enable VMID at 2*50k */
+			snd_soc_update_bits(codec, WM8960_POWER1,
+					    WM8960_VMID_MASK, 0x80);
+
+			/* Ramp */
+			msleep(100);
+
+			/* Enable VREF */
+			snd_soc_update_bits(codec, WM8960_POWER1,
+					    WM8960_VREF, WM8960_VREF);
+
+			msleep(100);
+			break;
+
+		case SND_SOC_BIAS_ON:
+			/* Enable anti-pop mode */
+			snd_soc_update_bits(codec, WM8960_APOP1,
+					    WM8960_POBCTRL | WM8960_SOFT_ST |
+					    WM8960_BUFDCOPEN,
+					    WM8960_POBCTRL | WM8960_SOFT_ST |
+					    WM8960_BUFDCOPEN);
+
+			/* Disable VMID and VREF */
+			snd_soc_update_bits(codec, WM8960_POWER1,
+					    WM8960_VREF | WM8960_VMID_MASK, 0);
+			break;
+
+		default:
+			break;
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		switch (codec->bias_level) {
+		case SND_SOC_BIAS_PREPARE:
+			/* Disable HP discharge */
+			snd_soc_update_bits(codec, WM8960_APOP2,
+					    WM8960_DISOP | WM8960_DRES_MASK,
+					    0);
+
+			/* Disable anti-pop features */
+			snd_soc_update_bits(codec, WM8960_APOP1,
+					    WM8960_POBCTRL | WM8960_SOFT_ST |
+					    WM8960_BUFDCOPEN,
+					    WM8960_POBCTRL | WM8960_SOFT_ST |
+					    WM8960_BUFDCOPEN);
+			break;
+
+		default:
+			break;
+		}
+		break;
 
-		snd_soc_write(codec, WM8960_APOP1, 0);
+	case SND_SOC_BIAS_OFF:
 		break;
 	}
 
@@ -663,7 +807,7 @@ static int wm8960_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
-	wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	codec->set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -682,8 +826,8 @@ static int wm8960_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8960_set_bias_level(codec, codec->suspend_bias_level);
+	codec->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	codec->set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
@@ -753,6 +897,8 @@ static int wm8960_register(struct wm8960_priv *wm8960,
 		goto err;
 	}
 
+	codec->set_bias_level = wm8960_set_bias_level_out3;
+
 	if (!pdata) {
 		dev_warn(codec->dev, "No platform data supplied\n");
 	} else {
@@ -760,6 +906,9 @@ static int wm8960_register(struct wm8960_priv *wm8960,
 			dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres);
 			pdata->dres = 0;
 		}
+
+		if (pdata->capless)
+			codec->set_bias_level = wm8960_set_bias_level_capless;
 	}
 
 	mutex_init(&codec->mutex);
@@ -770,7 +919,6 @@ static int wm8960_register(struct wm8960_priv *wm8960,
 	codec->name = "WM8960";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
-	codec->set_bias_level = wm8960_set_bias_level;
 	codec->dai = &wm8960_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = WM8960_CACHEREGNUM;
@@ -792,7 +940,7 @@ static int wm8960_register(struct wm8960_priv *wm8960,
 
 	wm8960_dai.dev = codec->dev;
 
-	wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	codec->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch the update bits */
 	reg = snd_soc_read(codec, WM8960_LINVOL);
@@ -841,7 +989,7 @@ err:
 
 static void wm8960_unregister(struct wm8960_priv *wm8960)
 {
-	wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF);
+	wm8960->codec.set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dai(&wm8960_dai);
 	snd_soc_unregister_codec(&wm8960->codec);
 	kfree(wm8960);
-- 
cgit v1.2.3


From 1ca7578043a79d74152774acee0ed6e393134d12 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 4 Mar 2010 14:54:40 +0000
Subject: ASoC: Add delay information for Samsung IISv2 DAIs

Report the current FIFO depth when delay is queried. The FIFO is only
16 frames deep so the latency will be at most a couple of miliseconds
(and we tend to end up reporting zero most of the time) but it may
help some applications.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index e994d8374fe6..aa84f4cb4d8a 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -550,6 +550,21 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
+static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream,
+					   struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+	u32 reg = readl(i2s->regs + S3C2412_IISFIC);
+	snd_pcm_sframes_t delay;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		delay = S3C2412_IISFIC_TXCOUNT(reg);
+	else
+		delay = S3C2412_IISFIC_RXCOUNT(reg);
+
+	return delay;
+}
+
 /* default table of all avaialable root fs divisors */
 static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
 
@@ -736,6 +751,10 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 	ops->set_fmt = s3c2412_i2s_set_fmt;
 	ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
 
+	/* Allow overriding by (for example) IISv4 */
+	if (!ops->delay)
+		ops->delay = s3c2412_i2s_delay,
+
 	dai->suspend = s3c2412_i2s_suspend;
 	dai->resume = s3c2412_i2s_resume;
 
-- 
cgit v1.2.3


From bc6552f4717e07e7737b5dc17883c2ff99f2a315 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 5 Mar 2010 16:27:15 +0000
Subject: ASoC: Add 16/16 registers to soc-cache

I2C only at the minute.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/soc-cache.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 83 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 5869dc3be781..bf593a834f5a 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -366,6 +366,84 @@ static int snd_soc_16_8_spi_write(void *control_data, const char *data,
 #define snd_soc_16_8_spi_write NULL
 #endif
 
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
+					   unsigned int r)
+{
+	struct i2c_msg xfer[2];
+	u16 reg = cpu_to_be16(r);
+	u16 data;
+	int ret;
+	struct i2c_client *client = codec->control_data;
+
+	/* Write register */
+	xfer[0].addr = client->addr;
+	xfer[0].flags = 0;
+	xfer[0].len = 2;
+	xfer[0].buf = (u8 *)&reg;
+
+	/* Read data */
+	xfer[1].addr = client->addr;
+	xfer[1].flags = I2C_M_RD;
+	xfer[1].len = 2;
+	xfer[1].buf = (u8 *)&data;
+
+	ret = i2c_transfer(client->adapter, xfer, 2);
+	if (ret != 2) {
+		dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+		return 0;
+	}
+
+	return be16_to_cpu(data);
+}
+#else
+#define snd_soc_16_16_read_i2c NULL
+#endif
+
+static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg >= codec->reg_cache_size ||
+	    snd_soc_codec_volatile_register(codec, reg)) {
+		if (codec->cache_only)
+			return -EINVAL;
+
+		return codec->hw_read(codec, reg);
+	}
+
+	return cache[reg];
+}
+
+static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
+			       unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	u8 data[4];
+	int ret;
+
+	data[0] = (reg >> 8) & 0xff;
+	data[1] = reg & 0xff;
+	data[2] = (value >> 8) & 0xff;
+	data[3] = value & 0xff;
+
+	if (reg < codec->reg_cache_size)
+		cache[reg] = value;
+
+	if (codec->cache_only) {
+		codec->cache_sync = 1;
+		return 0;
+	}
+
+	ret = codec->hw_write(codec->control_data, data, 4);
+	if (ret == 4)
+		return 0;
+	if (ret < 0)
+		return ret;
+	else
+		return -EIO;
+}
 
 static struct {
 	int addr_bits;
@@ -400,6 +478,11 @@ static struct {
 		.i2c_read = snd_soc_16_8_read_i2c,
 		.spi_write = snd_soc_16_8_spi_write,
 	},
+	{
+		.addr_bits = 16, .data_bits = 16,
+		.write = snd_soc_16_16_write, .read = snd_soc_16_16_read,
+		.i2c_read = snd_soc_16_16_read_i2c,
+	},
 };
 
 /**
-- 
cgit v1.2.3


From 692247196dd845eef02f26bf2959719b82e935c6 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Mar 2010 14:57:09 +0000
Subject: ASoC: Improve DAPM pop_wait delays

Currently during pop/click debug we're inserting a delay both after
every log message we generate and at explicit points in the sequence,
slowing things down even further than they need to be especially when
many writes get coalesced by the sequence generation code.

Remove the per-printk delay and ensure that we have explicit delays
where we say we want them.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/soc-dapm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 6c3351095786..86ded22e36af 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -97,7 +97,6 @@ static void pop_dbg(u32 pop_time, const char *fmt, ...)
 
 	if (pop_time) {
 		vprintk(fmt, args);
-		pop_wait(pop_time);
 	}
 
 	va_end(args);
@@ -314,8 +313,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
 		pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
 			widget->name, widget->power ? "on" : "off",
 			codec->pop_time);
-		snd_soc_write(codec, widget->reg, new);
 		pop_wait(codec->pop_time);
+		snd_soc_write(codec, widget->reg, new);
 	}
 	pr_debug("reg %x old %x new %x change %d\n", widget->reg,
 		 old, new, change);
@@ -1075,6 +1074,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 
 	pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n",
 		codec->pop_time);
+	pop_wait(codec->pop_time);
 
 	return 0;
 }
-- 
cgit v1.2.3


From 3a9d620278981f367a77a8888c0d59a70473b056 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Mon, 1 Mar 2010 10:10:43 +0900
Subject: ASoC: da7210: Add 8/12/16/24/32/48/96 kHz rate support

Signed-off-by: Phil Edworthy <Phil.Edworthy@renesas.com>
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/da7210.c | 67 +++++++++++++++++++++++++++++++++--------------
 1 file changed, 48 insertions(+), 19 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index cf2975a7294a..3bd867de597b 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -74,15 +74,14 @@
 /* INMIX_R bit fields */
 #define DA7210_IN_R_EN			(1 << 7)
 
-/* ADC_HPF bit fields */
-#define DA7210_ADC_VOICE_EN		(1 << 7)
-
 /* ADC bit fields */
 #define DA7210_ADC_L_EN			(1 << 3)
 #define DA7210_ADC_R_EN			(1 << 7)
 
-/* DAC_HPF fields */
-#define DA7210_DAC_VOICE_EN		(1 << 7)
+/* DAC/ADC HPF fields */
+#define DA7210_VOICE_F0_MASK		(0x7 << 4)
+#define DA7210_VOICE_F0_25		(1 << 4)
+#define DA7210_VOICE_EN			(1 << 7)
 
 /* DAC_SEL bit fields */
 #define DA7210_DAC_L_SRC_DAI_L		(4 << 0)
@@ -123,7 +122,15 @@
 #define DA7210_PLL_BYP			(1 << 6)
 
 /* PLL bit fields */
-#define DA7210_PLL_FS_48000		(11 << 0)
+#define DA7210_PLL_FS_MASK		(0xF << 0)
+#define DA7210_PLL_FS_8000		(0x1 << 0)
+#define DA7210_PLL_FS_12000		(0x3 << 0)
+#define DA7210_PLL_FS_16000		(0x5 << 0)
+#define DA7210_PLL_FS_24000		(0x7 << 0)
+#define DA7210_PLL_FS_32000		(0x9 << 0)
+#define DA7210_PLL_FS_48000		(0xB << 0)
+#define DA7210_PLL_FS_96000		(0xF << 0)
+
 
 #define DA7210_VERSION "0.0.1"
 
@@ -241,7 +248,8 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
 	u32 dai_cfg1;
-	u32 reg, mask;
+	u32 hpf_reg, hpf_mask, hpf_value;
+	u32 fs;
 
 	/* set DAI source to Left and Right ADC */
 	da7210_write(codec, DA7210_DAI_SRC_SEL,
@@ -265,25 +273,46 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 
 	da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1);
 
-	/* FIXME
-	 *
-	 * It support 48K only now
-	 */
+	hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ?
+		DA7210_DAC_HPF : DA7210_ADC_HPF;
+
 	switch (params_rate(params)) {
+	case 8000:
+		fs		= DA7210_PLL_FS_8000;
+		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
+		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		break;
+	case 12000:
+		fs		= DA7210_PLL_FS_12000;
+		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
+		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		break;
+	case 16000:
+		fs		= DA7210_PLL_FS_16000;
+		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
+		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		break;
+	case 32000:
+		fs		= DA7210_PLL_FS_32000;
+		hpf_mask	= DA7210_VOICE_EN;
+		hpf_value	= 0;
+		break;
 	case 48000:
-		if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) {
-			reg  = DA7210_DAC_HPF;
-			mask = DA7210_DAC_VOICE_EN;
-		} else {
-			reg  = DA7210_ADC_HPF;
-			mask = DA7210_ADC_VOICE_EN;
-		}
+		fs		= DA7210_PLL_FS_48000;
+		hpf_mask	= DA7210_VOICE_EN;
+		hpf_value	= 0;
+		break;
+	case 96000:
+		fs		= DA7210_PLL_FS_96000;
+		hpf_mask	= DA7210_VOICE_EN;
+		hpf_value	= 0;
 		break;
 	default:
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, reg, mask, 0);
+	snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value);
+	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
 
 	return 0;
 }
-- 
cgit v1.2.3


From 8d43d1bc81190ee8e4e694d805afa586339b0afc Mon Sep 17 00:00:00 2001
From: Chaithrika U S <chaithrika@ti.com>
Date: Wed, 10 Mar 2010 14:48:33 +0530
Subject: ASoC: DaVinci: Add hw_param callback for S/PDIF DIT link

On TI DM6467 EVM, S/PDIF DIT codec fails to open as it is unable to install
hardware params. This dummy codec has no set_fmt and set_sysclk implementations
and calls from the application to these functions cause errors. This patch adds
a new hardware params callback function for S/PDIF transciever codec.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/davinci/davinci-evm.c | 16 +++++++++++++++-
 1 file changed, 15 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 7ccbe6684fc2..dba6651547c1 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -81,10 +81,24 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int evm_spdif_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->dai->cpu_dai;
+
+	/* set cpu DAI configuration */
+	return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
+}
+
 static struct snd_soc_ops evm_ops = {
 	.hw_params = evm_hw_params,
 };
 
+static struct snd_soc_ops evm_spdif_ops = {
+	.hw_params = evm_spdif_hw_params,
+};
+
 /* davinci-evm machine dapm widgets */
 static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
@@ -165,7 +179,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.stream_name = "spdif",
 		.cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI],
 		.codec_dai = &dit_stub_dai,
-		.ops = &evm_ops,
+		.ops = &evm_spdif_ops,
 	},
 };
 static struct snd_soc_dai_link da8xx_evm_dai = {
-- 
cgit v1.2.3


From 4b527e2900531befd3d4df2e70c85029ea571f1a Mon Sep 17 00:00:00 2001
From: Cliff Cai <cliff.cai@analog.com>
Date: Tue, 9 Mar 2010 12:58:09 -0500
Subject: ASoC: SSM2602: add SND control for mic boost2 and default it to off

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ssm2602.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index d2ff1cde6883..942f5dc30801 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -139,6 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
 SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
 
 SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
+SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0),
 SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
 
 SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
@@ -604,8 +605,7 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 	reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V);
 	ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH);
 	/*select Line in as default input*/
-	ssm2602_write(codec, SSM2602_APANA,
-			APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC |
+	ssm2602_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
 			APANA_ENABLE_MIC_BOOST);
 	ssm2602_write(codec, SSM2602_PWR, 0);
 
-- 
cgit v1.2.3


From f0d10f5aa3955fcd4a9a7f950c66b64666589d4d Mon Sep 17 00:00:00 2001
From: Barry Song <barry.song@analog.com>
Date: Tue, 9 Mar 2010 12:58:14 -0500
Subject: ASoC: bf5xx-sport: use common SPORT code for MMR info

No point in duplicating this structure layout in each driver.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/bf5xx-sport.h | 28 +---------------------------
 1 file changed, 1 insertion(+), 27 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
index 2e63dea73e9c..a86e8cc0b2d3 100644
--- a/sound/soc/blackfin/bf5xx-sport.h
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -34,33 +34,7 @@
 #include <linux/wait.h>
 #include <linux/workqueue.h>
 #include <asm/dma.h>
-
-struct sport_register {
-	u16 tcr1;	u16 reserved0;
-	u16 tcr2;	u16 reserved1;
-	u16 tclkdiv;	u16 reserved2;
-	u16 tfsdiv;	u16 reserved3;
-	u32 tx;
-	u32 reserved_l0;
-	u32 rx;
-	u32 reserved_l1;
-	u16 rcr1;	u16 reserved4;
-	u16 rcr2;	u16 reserved5;
-	u16 rclkdiv;	u16 reserved6;
-	u16 rfsdiv;	u16 reserved7;
-	u16 stat;	u16 reserved8;
-	u16 chnl;	u16 reserved9;
-	u16 mcmc1;	u16 reserved10;
-	u16 mcmc2;	u16 reserved11;
-	u32 mtcs0;
-	u32 mtcs1;
-	u32 mtcs2;
-	u32 mtcs3;
-	u32 mrcs0;
-	u32 mrcs1;
-	u32 mrcs2;
-	u32 mrcs3;
-};
+#include <asm/bfin_sport.h>
 
 #define DESC_ELEMENT_COUNT 9
 
-- 
cgit v1.2.3


From 08226614789496dc395e3714115a3bcf2ca3baa2 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:50 +0900
Subject: ASoC: S3C: I2Sv2: Add missing semicolon

Add missing semicolon after s3c2412_i2s_delay

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index aa84f4cb4d8a..8f085087aa8c 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -753,7 +753,7 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 
 	/* Allow overriding by (for example) IISv4 */
 	if (!ops->delay)
-		ops->delay = s3c2412_i2s_delay,
+		ops->delay = s3c2412_i2s_delay;
 
 	dai->suspend = s3c2412_i2s_suspend;
 	dai->resume = s3c2412_i2s_resume;
-- 
cgit v1.2.3


From 4793d6afbd7d66d64799a9dd1881888c28c4e1d0 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:54 +0900
Subject: ASoC: S3C: I2Sv2: Unify clock source IDs

Rather than having the multiple definitions of the same clocks,
define them in one common place and refer by SoC specific names.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.h  | 4 ++++
 sound/soc/s3c24xx/s3c2412-i2s.h | 4 ++--
 sound/soc/s3c24xx/s3c64xx-i2s.h | 6 +++---
 3 files changed, 9 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
index ecf8eaaed1db..b094d3c23cbe 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.h
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -25,6 +25,10 @@
 #define S3C_I2SV2_DIV_RCLK	(2)
 #define S3C_I2SV2_DIV_PRESCALER	(3)
 
+#define S3C_I2SV2_CLKSRC_PCLK		0
+#define S3C_I2SV2_CLKSRC_AUDIOBUS	1
+#define S3C_I2SV2_CLKSRC_CDCLK		2
+
 /**
  * struct s3c_i2sv2_info - S3C I2S-V2 information
  * @dev: The parent device passed to use from the probe.
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index 92848e54be16..60cac002a830 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -21,8 +21,8 @@
 #define S3C2412_DIV_RCLK	S3C_I2SV2_DIV_RCLK
 #define S3C2412_DIV_PRESCALER	S3C_I2SV2_DIV_PRESCALER
 
-#define S3C2412_CLKSRC_PCLK	(0)
-#define S3C2412_CLKSRC_I2SCLK	(1)
+#define S3C2412_CLKSRC_PCLK	S3C_I2SV2_CLKSRC_PCLK
+#define S3C2412_CLKSRC_I2SCLK	S3C_I2SV2_CLKSRC_AUDIOBUS
 
 extern struct clk *s3c2412_get_iisclk(void);
 
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index abe7253b55fc..e350d280cd7a 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -23,9 +23,9 @@ struct clk;
 #define S3C64XX_DIV_RCLK	S3C_I2SV2_DIV_RCLK
 #define S3C64XX_DIV_PRESCALER	S3C_I2SV2_DIV_PRESCALER
 
-#define S3C64XX_CLKSRC_PCLK	(0)
-#define S3C64XX_CLKSRC_MUX	(1)
-#define S3C64XX_CLKSRC_CDCLK    (2)
+#define S3C64XX_CLKSRC_PCLK	S3C_I2SV2_CLKSRC_PCLK
+#define S3C64XX_CLKSRC_MUX	S3C_I2SV2_CLKSRC_AUDIOBUS
+#define S3C64XX_CLKSRC_CDCLK    S3C_I2SV2_CLKSRC_CDCLK
 
 extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
-- 
cgit v1.2.3


From b568f84b30f2b690e5ef704835effd09a212cd72 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:55 +0900
Subject: ASoC: s3c-i2s-v2 remove unnecessary headers

s3c-i2s-v2 remove unnecessary headers

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c | 6 ------
 1 file changed, 6 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 8f085087aa8c..65d1cddc4b3a 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -16,18 +16,12 @@
  * option) any later version.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/clk.h>
-#include <linux/kernel.h>
 #include <linux/io.h>
 
-#include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
 
 #include <plat/regs-s3c2412-iis.h>
-- 
cgit v1.2.3


From 87b7eb266c0bcdb5a0f1e670a73f2de742da459f Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:56 +0900
Subject: ASoC: s3c64xx-i2s remove unncessary headers

s3c64xx-i2s remove unncessary headers

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c64xx-i2s.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 93ed3aad1631..39563edabe28 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -12,9 +12,6 @@
  * published by the Free Software Foundation.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/device.h>
 #include <linux/clk.h>
 #include <linux/gpio.h>
 #include <linux/io.h>
-- 
cgit v1.2.3


From fa6231e1738617a864869600f51c3feb9e1081c8 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:57 +0900
Subject: ASoC: S3C64XX: I2S: Move RATE and FMT defines to header

In order for the RATE and FMT defines to be reuseable in future by the
i2sv4 driver, move the MACROs out to the header file.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c64xx-i2s.c | 9 ---------
 sound/soc/s3c24xx/s3c64xx-i2s.h | 9 +++++++++
 2 files changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 39563edabe28..65528943579b 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -127,15 +127,6 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev,
 }
 
 
-#define S3C64XX_I2S_RATES \
-	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
-	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
-	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-
-#define S3C64XX_I2S_FMTS \
-	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
-	 SNDRV_PCM_FMTBIT_S24_LE)
-
 static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
 	.set_sysclk	= s3c64xx_i2s_set_sysclk,	
 };
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index e350d280cd7a..53d2a0a0df36 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -27,6 +27,15 @@ struct clk;
 #define S3C64XX_CLKSRC_MUX	S3C_I2SV2_CLKSRC_AUDIOBUS
 #define S3C64XX_CLKSRC_CDCLK    S3C_I2SV2_CLKSRC_CDCLK
 
+#define S3C64XX_I2S_RATES \
+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define S3C64XX_I2S_FMTS \
+	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+	 SNDRV_PCM_FMTBIT_S24_LE)
+
 extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
 extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai);
-- 
cgit v1.2.3


From 51c6ab130642ed975681df843c772dda48a1d2ed Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:51 +0900
Subject: ASoC: S3C: I2Sv2: Reject immidiate register value

Towards generalizing CPU driver interface, do not accept direct field
values for the BCLK and RCLK.
The machine driver should simply request the FS-multiple and not provide
the value to be set in divide field of IISMOD.

[Confirmed by Jassi that no existing machine drivers are affected --
broonie]

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c | 68 +++++++++++++++++++-----------------------
 1 file changed, 30 insertions(+), 38 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 65d1cddc4b3a..b846f563cb50 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -463,29 +463,25 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 
 	switch (div_id) {
 	case S3C_I2SV2_DIV_BCLK:
-		if (div > 3) {
-			/* convert value to bit field */
-
-			switch (div) {
-			case 16:
-				div = S3C2412_IISMOD_BCLK_16FS;
-				break;
+		switch (div) {
+		case 16:
+			div = S3C2412_IISMOD_BCLK_16FS;
+			break;
 
-			case 32:
-				div = S3C2412_IISMOD_BCLK_32FS;
-				break;
+		case 32:
+			div = S3C2412_IISMOD_BCLK_32FS;
+			break;
 
-			case 24:
-				div = S3C2412_IISMOD_BCLK_24FS;
-				break;
+		case 24:
+			div = S3C2412_IISMOD_BCLK_24FS;
+			break;
 
-			case 48:
-				div = S3C2412_IISMOD_BCLK_48FS;
-				break;
+		case 48:
+			div = S3C2412_IISMOD_BCLK_48FS;
+			break;
 
-			default:
-				return -EINVAL;
-			}
+		default:
+			return -EINVAL;
 		}
 
 		reg = readl(i2s->regs + S3C2412_IISMOD);
@@ -496,29 +492,25 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 		break;
 
 	case S3C_I2SV2_DIV_RCLK:
-		if (div > 3) {
-			/* convert value to bit field */
-
-			switch (div) {
-			case 256:
-				div = S3C2412_IISMOD_RCLK_256FS;
-				break;
+		switch (div) {
+		case 256:
+			div = S3C2412_IISMOD_RCLK_256FS;
+			break;
 
-			case 384:
-				div = S3C2412_IISMOD_RCLK_384FS;
-				break;
+		case 384:
+			div = S3C2412_IISMOD_RCLK_384FS;
+			break;
 
-			case 512:
-				div = S3C2412_IISMOD_RCLK_512FS;
-				break;
+		case 512:
+			div = S3C2412_IISMOD_RCLK_512FS;
+			break;
 
-			case 768:
-				div = S3C2412_IISMOD_RCLK_768FS;
-				break;
+		case 768:
+			div = S3C2412_IISMOD_RCLK_768FS;
+			break;
 
-			default:
-				return -EINVAL;
-			}
+		default:
+			return -EINVAL;
 		}
 
 		reg = readl(i2s->regs + S3C2412_IISMOD);
-- 
cgit v1.2.3


From bf3288260288160e22f85bbf00b281523c14b587 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:52 +0900
Subject: ASoC: S3C64XX: I2S: Make BCLK independent of sample size

For some CPU-CODEC and source clock combination we might need to set
BCLK to N*Sample_size*LRCLK, where N may be even 3 or 4, not just 2.

We can simply remove the dependency of BCLK on sample size as there
is already a callback(S3C_I2SV2_DIV_BCLK) available to set required BCLK.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index b846f563cb50..667431c824d0 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -358,19 +358,16 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 #endif
 
 #ifdef CONFIG_PLAT_S3C64XX
-	iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK);
+	iismod &= ~S3C64XX_IISMOD_BLC_MASK;
 	/* Sample size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
-		/* 8 bit sample, 16fs BCLK */
-		iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS);
+		iismod |= S3C64XX_IISMOD_BLC_8BIT;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
-		/* 16 bit sample, 32fs BCLK */
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		/* 24 bit sample, 48fs BCLK */
-		iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS);
+		iismod |= S3C64XX_IISMOD_BLC_24BIT;
 		break;
 	}
 #endif
-- 
cgit v1.2.3


From 9c9b125736db934b98aa7e16b675767ed4fcf5ce Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Wed, 10 Mar 2010 16:48:58 +0900
Subject: ASoC: S3C: I2Sv2: Segregate hw_params callback

Towards having build for multiple SoCs segregate hw_params callback
for s3c2412 and s3c64xx.
Since, all new SoCs have s3c64xx like register map, we keep that as
default implementation if no SoC specific callback is already defined.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c  | 18 +++---------------
 sound/soc/s3c24xx/s3c2412-i2s.c | 37 +++++++++++++++++++++++++++++++++++++
 2 files changed, 40 insertions(+), 15 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 667431c824d0..c3fcb63cbf25 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -326,7 +326,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
+static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *socdai)
 {
@@ -346,18 +346,6 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	iismod = readl(i2s->regs + S3C2412_IISMOD);
 	pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
 
-#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		iismod |= S3C2412_IISMOD_8BIT;
-		break;
-	case SNDRV_PCM_FORMAT_S16_LE:
-		iismod &= ~S3C2412_IISMOD_8BIT;
-		break;
-	}
-#endif
-
-#ifdef CONFIG_PLAT_S3C64XX
 	iismod &= ~S3C64XX_IISMOD_BLC_MASK;
 	/* Sample size */
 	switch (params_format(params)) {
@@ -370,7 +358,6 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 		iismod |= S3C64XX_IISMOD_BLC_24BIT;
 		break;
 	}
-#endif
 
 	writel(iismod, i2s->regs + S3C2412_IISMOD);
 	pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
@@ -730,7 +717,8 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 	struct snd_soc_dai_ops *ops = dai->ops;
 
 	ops->trigger = s3c2412_i2s_trigger;
-	ops->hw_params = s3c2412_i2s_hw_params;
+	if (!ops->hw_params)
+		ops->hw_params = s3c_i2sv2_hw_params;
 	ops->set_fmt = s3c2412_i2s_set_fmt;
 	ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
 
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 359e59346ba2..a5b21f63261f 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -103,6 +103,10 @@ struct clk *s3c2412_get_iisclk(void)
 }
 EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
 
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
 
 static int s3c2412_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
@@ -142,6 +146,38 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
+static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *cpu_dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod;
+
+	pr_debug("Entered %s\n", __func__);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		cpu_dai->dma_data = i2s->dma_playback;
+	else
+		cpu_dai->dma_data = i2s->dma_capture;
+
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		iismod |= S3C2412_IISMOD_8BIT;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iismod &= ~S3C2412_IISMOD_8BIT;
+		break;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
+
+	return 0;
+}
+
 #define S3C2412_I2S_RATES \
 	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
@@ -149,6 +185,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 
 static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
 	.set_sysclk	= s3c2412_i2s_set_sysclk,
+	.hw_params	= s3c2412_i2s_hw_params,
 };
 
 struct snd_soc_dai s3c2412_i2s_dai = {
-- 
cgit v1.2.3


From 960b3b4b4cb6d22d55ad0ebb280dac9876937fa0 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Thu, 11 Mar 2010 11:37:44 +0900
Subject: ASoC: da7210: Add 11025/22050/44100/88200 rate support

This driver USE PLL for 11025/22050/44100/88200 rate.
To enable switching to bypass mode, PLL is always turned on.
Special thanks to Phil

Signed-off-by: Phil Edworthy <Phil.Edworthy@renesas.com>
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/da7210.c | 90 ++++++++++++++++++++++++++++++++++++++++++++---
 1 file changed, 86 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 3bd867de597b..35b881638658 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -55,8 +55,14 @@
 #define DA7210_DAI_SRC_SEL		0x25
 #define DA7210_DAI_CFG1			0x26
 #define DA7210_DAI_CFG3			0x28
+#define DA7210_PLL_DIV1			0x29
+#define DA7210_PLL_DIV2			0x2A
 #define DA7210_PLL_DIV3			0x2B
 #define DA7210_PLL			0x2C
+#define DA7210_A_HID_UNLOCK		0x8A
+#define DA7210_A_TEST_UNLOCK		0x8B
+#define DA7210_A_PLL1			0x90
+#define DA7210_A_CP_MODE		0xA7
 
 /* STARTUP1 bit fields */
 #define DA7210_SC_MST_EN		(1 << 0)
@@ -124,13 +130,17 @@
 /* PLL bit fields */
 #define DA7210_PLL_FS_MASK		(0xF << 0)
 #define DA7210_PLL_FS_8000		(0x1 << 0)
+#define DA7210_PLL_FS_11025		(0x2 << 0)
 #define DA7210_PLL_FS_12000		(0x3 << 0)
 #define DA7210_PLL_FS_16000		(0x5 << 0)
+#define DA7210_PLL_FS_22050		(0x6 << 0)
 #define DA7210_PLL_FS_24000		(0x7 << 0)
 #define DA7210_PLL_FS_32000		(0x9 << 0)
+#define DA7210_PLL_FS_44100		(0xA << 0)
 #define DA7210_PLL_FS_48000		(0xB << 0)
+#define DA7210_PLL_FS_88200		(0xE << 0)
 #define DA7210_PLL_FS_96000		(0xF << 0)
-
+#define DA7210_PLL_EN			(0x1 << 7)
 
 #define DA7210_VERSION "0.0.1"
 
@@ -249,7 +259,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = socdev->card->codec;
 	u32 dai_cfg1;
 	u32 hpf_reg, hpf_mask, hpf_value;
-	u32 fs;
+	u32 fs, bypass;
 
 	/* set DAI source to Left and Right ADC */
 	da7210_write(codec, DA7210_DAI_SRC_SEL,
@@ -281,38 +291,76 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 		fs		= DA7210_PLL_FS_8000;
 		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
 		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		bypass		= DA7210_PLL_BYP;
+		break;
+	case 11025:
+		fs		= DA7210_PLL_FS_11025;
+		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
+		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		bypass		= 0;
 		break;
 	case 12000:
 		fs		= DA7210_PLL_FS_12000;
 		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
 		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		bypass		= DA7210_PLL_BYP;
 		break;
 	case 16000:
 		fs		= DA7210_PLL_FS_16000;
 		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
 		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
+		bypass		= DA7210_PLL_BYP;
+		break;
+	case 22050:
+		fs		= DA7210_PLL_FS_22050;
+		hpf_mask	= DA7210_VOICE_EN;
+		hpf_value	= 0;
+		bypass		= 0;
 		break;
 	case 32000:
 		fs		= DA7210_PLL_FS_32000;
 		hpf_mask	= DA7210_VOICE_EN;
 		hpf_value	= 0;
+		bypass		= DA7210_PLL_BYP;
+		break;
+	case 44100:
+		fs		= DA7210_PLL_FS_44100;
+		hpf_mask	= DA7210_VOICE_EN;
+		hpf_value	= 0;
+		bypass		= 0;
 		break;
 	case 48000:
 		fs		= DA7210_PLL_FS_48000;
 		hpf_mask	= DA7210_VOICE_EN;
 		hpf_value	= 0;
+		bypass		= DA7210_PLL_BYP;
+		break;
+	case 88200:
+		fs		= DA7210_PLL_FS_88200;
+		hpf_mask	= DA7210_VOICE_EN;
+		hpf_value	= 0;
+		bypass		= 0;
 		break;
 	case 96000:
 		fs		= DA7210_PLL_FS_96000;
 		hpf_mask	= DA7210_VOICE_EN;
 		hpf_value	= 0;
+		bypass		= DA7210_PLL_BYP;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	/* Disable active mode */
+	snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
+
 	snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value);
 	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
+	snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);
+
+	/* Enable active mode */
+	snd_soc_update_bits(codec, DA7210_STARTUP1,
+			    DA7210_SC_MST_EN, DA7210_SC_MST_EN);
 
 	return 0;
 }
@@ -390,6 +438,7 @@ struct snd_soc_dai da7210_dai = {
 		.formats = DA7210_FORMATS,
 	},
 	.ops = &da7210_dai_ops,
+	.symmetric_rates = 1,
 };
 EXPORT_SYMBOL_GPL(da7210_dai);
 
@@ -444,7 +493,21 @@ static int da7210_init(struct da7210_priv *da7210)
 	/* FIXME
 	 *
 	 * This driver use fixed value here
+	 * And below settings expects MCLK = 12.288MHz
+	 *
+	 * When you select different MCLK, please check...
+	 *      DA7210_PLL_DIV1 val
+	 *      DA7210_PLL_DIV2 val
+	 *      DA7210_PLL_DIV3 val
+	 *      DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx
+	 */
+
+	/*
+	 * make sure that DA7210 use bypass mode before start up
 	 */
+	da7210_write(codec, DA7210_STARTUP1, 0);
+	da7210_write(codec, DA7210_PLL_DIV3,
+		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
 
 	/*
 	 * ADC settings
@@ -482,9 +545,28 @@ static int da7210_init(struct da7210_priv *da7210)
 	/* Diable PLL and bypass it */
 	da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
 
-	/* Bypass PLL and set MCLK freq rang to 10-20MHz */
-	da7210_write(codec, DA7210_PLL_DIV3,
+	/*
+	 * If 48kHz sound came, it use bypass mode,
+	 * and when it is 44.1kHz, it use PLL.
+	 *
+	 * This time, this driver sets PLL always ON
+	 * and controls bypass/PLL mode by switching
+	 * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit.
+	 *   see da7210_hw_params
+	 */
+	da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
+	da7210_write(codec, DA7210_PLL_DIV2, 0x99);
+	da7210_write(codec, DA7210_PLL_DIV3, 0x0A |
 		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
+	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
+
+	/* As suggested by Dialog */
+	da7210_write(codec, DA7210_A_HID_UNLOCK,	0x8B); /* unlock */
+	da7210_write(codec, DA7210_A_TEST_UNLOCK,	0xB4);
+	da7210_write(codec, DA7210_A_PLL1,		0x01);
+	da7210_write(codec, DA7210_A_CP_MODE,		0x7C);
+	da7210_write(codec, DA7210_A_HID_UNLOCK,	0x00); /* re-lock */
+	da7210_write(codec, DA7210_A_TEST_UNLOCK,	0x00);
 
 	/* Activate all enabled subsystem */
 	da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
-- 
cgit v1.2.3


From d9ad6296ec3b4a55ba25f2c5e4824be487242e1f Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Fri, 12 Mar 2010 13:38:52 +0900
Subject: ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags

For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver
should be given a chance to figure out if the dai, that set the flag, can
accomodate a rate that it does not explicitly specify but is specified
by the dai at the other end of the link.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 06c38d1502b7..eb73aab78b94 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -404,6 +404,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			codec_dai->playback.formats & cpu_dai->playback.formats;
 		runtime->hw.rates =
 			codec_dai->playback.rates & cpu_dai->playback.rates;
+		if (codec_dai->playback.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= cpu_dai->playback.rates;
+		if (cpu_dai->playback.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= codec_dai->playback.rates;
 	} else {
 		runtime->hw.rate_min =
 			max(codec_dai->capture.rate_min,
@@ -421,6 +427,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			codec_dai->capture.formats & cpu_dai->capture.formats;
 		runtime->hw.rates =
 			codec_dai->capture.rates & cpu_dai->capture.rates;
+		if (codec_dai->capture.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= cpu_dai->capture.rates;
+		if (cpu_dai->capture.rates
+			   & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
+			runtime->hw.rates |= codec_dai->capture.rates;
 	}
 
 	snd_pcm_limit_hw_rates(runtime);
-- 
cgit v1.2.3


From e155fcc23c918f5d78a35d1eb6f3f0d523f51991 Mon Sep 17 00:00:00 2001
From: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Date: Thu, 11 Mar 2010 09:32:42 -0600
Subject: ASoC: DaVinci: Voice Codec Interface

This patch adds the support for the interface needed by the DaVinci
Voice Codec CQ93VC.

Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/davinci/Kconfig        |   3 +
 sound/soc/davinci/Makefile       |   2 +
 sound/soc/davinci/davinci-vcif.c | 272 +++++++++++++++++++++++++++++++++++++++
 sound/soc/davinci/davinci-vcif.h |  28 ++++
 4 files changed, 305 insertions(+)
 create mode 100644 sound/soc/davinci/davinci-vcif.c
 create mode 100644 sound/soc/davinci/davinci-vcif.h

(limited to 'sound/soc')

diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 047ee39418c0..47e7ccecc577 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -12,6 +12,9 @@ config SND_DAVINCI_SOC_I2S
 config SND_DAVINCI_SOC_MCASP
 	tristate
 
+config SND_DAVINCI_SOC_VCIF
+	tristate
+
 config SND_DAVINCI_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
 	depends on SND_DAVINCI_SOC
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index a6939d71b988..a93679d618cd 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -2,10 +2,12 @@
 snd-soc-davinci-objs := davinci-pcm.o
 snd-soc-davinci-i2s-objs := davinci-i2s.o
 snd-soc-davinci-mcasp-objs:= davinci-mcasp.o
+snd-soc-davinci-vcif-objs:= davinci-vcif.o
 
 obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
 obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o
+obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o
 
 # DAVINCI Machine Support
 snd-soc-evm-objs := davinci-evm.o
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
new file mode 100644
index 000000000000..03f3feb0fe2f
--- /dev/null
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -0,0 +1,272 @@
+/*
+ * ALSA SoC Voice Codec Interface for TI DAVINCI processor
+ *
+ * Copyright (C) 2010 Texas Instruments.
+ *
+ * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/mfd/davinci_voicecodec.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+#include "davinci-vcif.h"
+
+#define MOD_REG_BIT(val, mask, set) do { \
+	if (set) { \
+		val |= mask; \
+	} else { \
+		val &= ~mask; \
+	} \
+} while (0)
+
+struct davinci_vcif_dev {
+	struct davinci_vc *davinci_vc;
+	struct davinci_pcm_dma_params	dma_params[2];
+};
+
+static void davinci_vcif_start(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct davinci_vcif_dev *davinci_vcif_dev =
+					rtd->dai->cpu_dai->private_data;
+	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
+	u32 w;
+
+	/* Start the sample generator and enable transmitter/receiver */
+	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1);
+	else
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1);
+
+	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
+}
+
+static void davinci_vcif_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct davinci_vcif_dev *davinci_vcif_dev =
+					rtd->dai->cpu_dai->private_data;
+	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
+	u32 w;
+
+	/* Reset transmitter/receiver and sample rate/frame sync generators */
+	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0);
+	else
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0);
+
+	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
+}
+
+static int davinci_vcif_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data;
+	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
+	struct davinci_pcm_dma_params *dma_params =
+			&davinci_vcif_dev->dma_params[substream->stream];
+	u32 w;
+
+	/* Restart the codec before setup */
+	davinci_vcif_stop(substream);
+	davinci_vcif_start(substream);
+
+	/* General line settings */
+	writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL);
+
+	writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR);
+
+	writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN);
+
+	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
+
+	/* Determine xfer data type */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
+		dma_params->data_type = 0;
+
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
+			    DAVINCI_VC_CTRL_RD_UNSIGNED |
+			    DAVINCI_VC_CTRL_WD_BITS_8 |
+			    DAVINCI_VC_CTRL_WD_UNSIGNED, 1);
+		break;
+	case SNDRV_PCM_FORMAT_S8:
+		dma_params->data_type = 1;
+
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
+			    DAVINCI_VC_CTRL_WD_BITS_8, 1);
+
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED |
+			    DAVINCI_VC_CTRL_WD_UNSIGNED, 0);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		dma_params->data_type = 2;
+
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
+			    DAVINCI_VC_CTRL_RD_UNSIGNED |
+			    DAVINCI_VC_CTRL_WD_BITS_8 |
+			    DAVINCI_VC_CTRL_WD_UNSIGNED, 0);
+		break;
+	default:
+		printk(KERN_WARNING "davinci-vcif: unsupported PCM format");
+		return -EINVAL;
+	}
+
+	dma_params->acnt  = dma_params->data_type;
+
+	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
+
+	return 0;
+}
+
+static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		davinci_vcif_start(substream);
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		davinci_vcif_stop(substream);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+#define DAVINCI_VCIF_RATES	SNDRV_PCM_RATE_8000_48000
+
+static struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+	.trigger	= davinci_vcif_trigger,
+	.hw_params	= davinci_vcif_hw_params,
+};
+
+struct snd_soc_dai davinci_vcif_dai = {
+	.name = "davinci-vcif",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = DAVINCI_VCIF_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = DAVINCI_VCIF_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &davinci_vcif_dai_ops,
+
+};
+EXPORT_SYMBOL_GPL(davinci_vcif_dai);
+
+static int davinci_vcif_probe(struct platform_device *pdev)
+{
+	struct davinci_vc *davinci_vc = platform_get_drvdata(pdev);
+	struct davinci_vcif_dev *davinci_vcif_dev;
+	int ret;
+
+	davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL);
+	if (!davinci_vc) {
+		dev_dbg(&pdev->dev,
+			"could not allocate memory for private data\n");
+		return -ENOMEM;
+	}
+
+	/* DMA tx params */
+	davinci_vcif_dev->davinci_vc = davinci_vc;
+	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel =
+					davinci_vc->davinci_vcif.dma_tx_channel;
+	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
+					davinci_vc->davinci_vcif.dma_tx_addr;
+
+	/* DMA rx params */
+	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel =
+					davinci_vc->davinci_vcif.dma_rx_channel;
+	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
+					davinci_vc->davinci_vcif.dma_rx_addr;
+
+	davinci_vcif_dai.dev = &pdev->dev;
+	davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params;
+	davinci_vcif_dai.private_data = davinci_vcif_dev;
+
+	ret = snd_soc_register_dai(&davinci_vcif_dai);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "could not register dai\n");
+		goto fail;
+	}
+
+	return 0;
+
+fail:
+	kfree(davinci_vcif_dev);
+
+	return ret;
+}
+
+static int davinci_vcif_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&davinci_vcif_dai);
+
+	return 0;
+}
+
+static struct platform_driver davinci_vcif_driver = {
+	.probe		= davinci_vcif_probe,
+	.remove		= davinci_vcif_remove,
+	.driver		= {
+		.name	= "davinci_vcif",
+		.owner	= THIS_MODULE,
+	},
+};
+
+static int __init davinci_vcif_init(void)
+{
+	return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe);
+}
+module_init(davinci_vcif_init);
+
+static void __exit davinci_vcif_exit(void)
+{
+	platform_driver_unregister(&davinci_vcif_driver);
+}
+module_exit(davinci_vcif_exit);
+
+MODULE_AUTHOR("Miguel Aguilar");
+MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h
new file mode 100644
index 000000000000..571c9948724f
--- /dev/null
+++ b/sound/soc/davinci/davinci-vcif.h
@@ -0,0 +1,28 @@
+/*
+ * ALSA SoC Voice Codec Interface for TI DAVINCI processor
+ *
+ * Copyright (C) 2010 Texas Instruments.
+ *
+ * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef _DAVINCI_VCIF_H
+#define _DAVINCI_VCIF_H
+
+extern struct snd_soc_dai davinci_vcif_dai;
+
+#endif
-- 
cgit v1.2.3


From b56e972b75a945e97818437720551859c7ef2d53 Mon Sep 17 00:00:00 2001
From: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Date: Thu, 11 Mar 2010 09:32:59 -0600
Subject: ASoC: DaVinci: CQ93VC Voice Codec

Currently the DM365 is the only SoC that includes this Voice Codec.

Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   4 +
 sound/soc/codecs/Makefile |   2 +
 sound/soc/codecs/cq93vc.c | 298 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/cq93vc.h |  29 +++++
 4 files changed, 333 insertions(+)
 create mode 100644 sound/soc/codecs/cq93vc.c
 create mode 100644 sound/soc/codecs/cq93vc.h

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1743d565e996..16c47edb8266 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
+	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_DA7210 if I2C
@@ -114,6 +115,9 @@ config SND_SOC_AK4642
 config SND_SOC_AK4671
 	tristate
 
+config SND_SOC_CQ0093VC
+	tristate
+
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index dd5ce6df6292..69817778b2bd 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
+snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
@@ -70,6 +71,7 @@ obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
new file mode 100644
index 000000000000..513297441369
--- /dev/null
+++ b/sound/soc/codecs/cq93vc.c
@@ -0,0 +1,298 @@
+/*
+ * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms
+ *
+ * Copyright (C) 2010 Texas Instruments, Inc
+ *
+ * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/mfd/davinci_voicecodec.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <mach/dm365.h>
+
+#include "cq93vc.h"
+
+static inline unsigned int cq93vc_read(struct snd_soc_codec *codec,
+						unsigned int reg)
+{
+	struct davinci_vc *davinci_vc = codec->control_data;
+
+	return readl(davinci_vc->base + reg);
+}
+
+static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg,
+		       unsigned int value)
+{
+	struct davinci_vc *davinci_vc = codec->control_data;
+
+	writel(value, davinci_vc->base + reg);
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new cq93vc_snd_controls[] = {
+	SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0),
+	SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0),
+};
+
+static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE;
+
+	if (mute)
+		cq93vc_write(codec, DAVINCI_VC_REG09,
+			     reg | DAVINCI_VC_REG09_MUTE);
+	else
+		cq93vc_write(codec, DAVINCI_VC_REG09, reg);
+
+	return 0;
+}
+
+static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct davinci_vc *davinci_vc = codec->control_data;
+
+	switch (freq) {
+	case 22579200:
+	case 27000000:
+	case 33868800:
+		davinci_vc->cq93vc.sysclk = freq;
+		return 0;
+	}
+
+	return -EINVAL;
+}
+
+static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		cq93vc_write(codec, DAVINCI_VC_REG12,
+			     DAVINCI_VC_REG12_POWER_ALL_ON);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		cq93vc_write(codec, DAVINCI_VC_REG12,
+			     DAVINCI_VC_REG12_POWER_ALL_OFF);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* force all power off */
+		cq93vc_write(codec, DAVINCI_VC_REG12,
+			     DAVINCI_VC_REG12_POWER_ALL_OFF);
+		break;
+	}
+	codec->bias_level = level;
+
+	return 0;
+}
+
+#define CQ93VC_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000)
+#define CQ93VC_FORMATS	(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
+
+static struct snd_soc_dai_ops cq93vc_dai_ops = {
+	.digital_mute	= cq93vc_mute,
+	.set_sysclk	= cq93vc_set_dai_sysclk,
+};
+
+struct snd_soc_dai cq93vc_dai = {
+	.name = "CQ93VC",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = CQ93VC_RATES,
+		.formats = CQ93VC_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = CQ93VC_RATES,
+		.formats = CQ93VC_FORMATS,},
+	.ops = &cq93vc_dai_ops,
+};
+EXPORT_SYMBOL_GPL(cq93vc_dai);
+
+static int cq93vc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	cq93vc_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+static struct snd_soc_codec *cq93vc_codec;
+
+static int cq93vc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct device *dev = &pdev->dev;
+	struct snd_soc_codec *codec;
+	int ret;
+
+	socdev->card->codec = cq93vc_codec;
+	codec = socdev->card->codec;
+
+	/* Register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(dev, "%s: failed to create pcms\n", pdev->name);
+		return ret;
+	}
+
+	/* Set controls */
+	snd_soc_add_controls(codec, cq93vc_snd_controls,
+			     ARRAY_SIZE(cq93vc_snd_controls));
+
+	/* Off, with power on */
+	cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static int cq93vc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_cq93vc = {
+	.probe = cq93vc_probe,
+	.remove = cq93vc_remove,
+	.resume = cq93vc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc);
+
+static __init int cq93vc_codec_probe(struct platform_device *pdev)
+{
+	struct davinci_vc *davinci_vc = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL) {
+		dev_dbg(davinci_vc->dev,
+			"could not allocate memory for codec data\n");
+		return -ENOMEM;
+	}
+
+	davinci_vc->cq93vc.codec = codec;
+
+	cq93vc_dai.dev = &pdev->dev;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec->dev = &pdev->dev;
+	codec->name = "CQ93VC";
+	codec->owner = THIS_MODULE;
+	codec->read = cq93vc_read;
+	codec->write = cq93vc_write;
+	codec->set_bias_level = cq93vc_set_bias_level;
+	codec->dai = &cq93vc_dai;
+	codec->num_dai = 1;
+	codec->control_data = davinci_vc;
+
+	cq93vc_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret) {
+		dev_err(davinci_vc->dev, "failed to register codec\n");
+		goto fail1;
+	}
+
+	ret = snd_soc_register_dai(&cq93vc_dai);
+	if (ret) {
+		dev_err(davinci_vc->dev, "could register dai\n");
+		goto fail2;
+	}
+	return 0;
+
+fail2:
+	snd_soc_unregister_codec(codec);
+
+fail1:
+	kfree(codec);
+	cq93vc_codec = NULL;
+
+	return ret;
+}
+
+static int __devexit cq93vc_codec_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	snd_soc_unregister_dai(&cq93vc_dai);
+	snd_soc_unregister_codec(&codec);
+
+	kfree(codec);
+	cq93vc_codec = NULL;
+
+	return 0;
+}
+
+static struct platform_driver cq93vc_codec_driver = {
+	.driver = {
+		   .name = "cq93vc",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = cq93vc_codec_probe,
+	.remove = __devexit_p(cq93vc_codec_remove),
+};
+
+static __init int cq93vc_init(void)
+{
+	return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe);
+}
+module_init(cq93vc_init);
+
+static __exit void cq93vc_exit(void)
+{
+	platform_driver_unregister(&cq93vc_codec_driver);
+}
+module_exit(cq93vc_exit);
+
+MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver");
+MODULE_AUTHOR("Miguel Aguilar");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h
new file mode 100644
index 000000000000..845b1968ef9c
--- /dev/null
+++ b/sound/soc/codecs/cq93vc.h
@@ -0,0 +1,29 @@
+/*
+ * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms
+ *
+ * Copyright (C) 2010 Texas Instruments, Inc
+ *
+ * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef _CQ93VC_H
+#define _CQ93VC_H
+
+extern struct snd_soc_dai cq93vc_dai;
+extern struct snd_soc_codec_device soc_codec_dev_cq93vc;
+
+#endif
-- 
cgit v1.2.3


From aa9b88ee80d3985ebae9567a8f4e0d07ce15b5a7 Mon Sep 17 00:00:00 2001
From: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Date: Thu, 11 Mar 2010 09:33:40 -0600
Subject: DaVinci: DM365: Voice Codec support for the DM365 EVM

The DM365 EVM has two codecs: the Audio Codec (AIC3x) and the Voice Codec,
the idea is to have both enabled in the same kernel simultaneously. However,
the current soc-core doesn't support simultaneous codecs, once that
support will have added, a patch will be posted to enable both codecs in
the DM365 EVM.

Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/arm/mach-davinci/board-dm365-evm.c |  4 +++
 sound/soc/davinci/Kconfig               | 24 ++++++++++++++++--
 sound/soc/davinci/davinci-evm.c         | 45 +++++++++++++++++++++++++++++++--
 3 files changed, 69 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c
index d15beceb632e..512174a874a1 100644
--- a/arch/arm/mach-davinci/board-dm365-evm.c
+++ b/arch/arm/mach-davinci/board-dm365-evm.c
@@ -604,7 +604,11 @@ static __init void dm365_evm_init(void)
 	/* maybe setup mmc1/etc ... _after_ mmc0 */
 	evm_init_cpld();
 
+#ifdef CONFIG_SND_DM365_AIC3X_CODEC
 	dm365_init_asp(&dm365_evm_snd_data);
+#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
+	dm365_init_vc(&dm365_evm_snd_data);
+#endif
 	dm365_init_rtc();
 	dm365_init_ks(&dm365evm_ks_data);
 
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 47e7ccecc577..6bbf001f6591 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -18,12 +18,32 @@ config SND_DAVINCI_SOC_VCIF
 config SND_DAVINCI_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
 	depends on SND_DAVINCI_SOC
-	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM  || MACH_DAVINCI_DM365_EVM
+	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM
 	select SND_DAVINCI_SOC_I2S
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on TI
-	  DaVinci DM6446 or DM355 EVM platforms.
+	  DaVinci DM6446, DM355 or DM365 EVM platforms.
+
+choice
+	prompt "DM365 codec select"
+	depends on SND_DAVINCI_SOC_EVM
+	depends on MACH_DAVINCI_DM365_EVM
+	default SND_DM365_EXTERNAL_CODEC
+
+config SND_DM365_AIC3X_CODEC
+	bool "Audio Codec - AIC3101"
+	help
+	  Say Y if you want to add support for AIC3101 audio codec
+
+config SND_DM365_VOICE_CODEC
+	bool "Voice Codec - CQ93VC"
+	select MFD_DAVINCI_VOICECODEC
+	select SND_DAVINCI_SOC_VCIF
+	select SND_SOC_CQ0093VC
+	help
+	  Say Y if you want to add support for SoC On-chip voice codec
+endchoice
 
 config  SND_DM6467_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6467 EVM"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index dba6651547c1..97f74d6a33e6 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -28,10 +28,12 @@
 #include <mach/mux.h>
 
 #include "../codecs/tlv320aic3x.h"
+#include "../codecs/cq93vc.h"
 #include "../codecs/spdif_transciever.h"
 #include "davinci-pcm.h"
 #include "davinci-i2s.h"
 #include "davinci-mcasp.h"
+#include "davinci-vcif.h"
 
 #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
 		SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
@@ -165,6 +167,22 @@ static struct snd_soc_dai_link evm_dai = {
 	.ops = &evm_ops,
 };
 
+static struct snd_soc_dai_link dm365_evm_dai = {
+#ifdef CONFIG_SND_DM365_AIC3X_CODEC
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai = &davinci_i2s_dai,
+	.codec_dai = &aic3x_dai,
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
+	.name = "Voice Codec - CQ93VC",
+	.stream_name = "CQ93",
+	.cpu_dai = &davinci_vcif_dai,
+	.codec_dai = &cq93vc_dai,
+#endif
+};
+
 static struct snd_soc_dai_link dm6467_evm_dai[] = {
 	{
 		.name = "TLV320AIC3X",
@@ -191,7 +209,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
 	.ops = &evm_ops,
 };
 
-/* davinci dm6446, dm355 or dm365 evm audio machine driver */
+/* davinci dm6446, dm355 evm audio machine driver */
 static struct snd_soc_card snd_soc_card_evm = {
 	.name = "DaVinci EVM",
 	.platform = &davinci_soc_platform,
@@ -199,6 +217,15 @@ static struct snd_soc_card snd_soc_card_evm = {
 	.num_links = 1,
 };
 
+/* davinci dm365 evm audio machine driver */
+static struct snd_soc_card dm365_snd_soc_card_evm = {
+	.name = "DaVinci DM365 EVM",
+	.platform = &davinci_soc_platform,
+	.dai_link = &dm365_evm_dai,
+	.num_links = 1,
+};
+
+
 /* davinci dm6467 evm audio machine driver */
 static struct snd_soc_card dm6467_snd_soc_card_evm = {
 	.name = "DaVinci DM6467 EVM",
@@ -230,6 +257,17 @@ static struct snd_soc_device evm_snd_devdata = {
 	.codec_data = &aic3x_setup,
 };
 
+/* evm audio subsystem */
+static struct snd_soc_device dm365_evm_snd_devdata = {
+	.card = &dm365_snd_soc_card_evm,
+#ifdef CONFIG_SND_DM365_AIC3X_CODEC
+	.codec_dev = &soc_codec_dev_aic3x,
+	.codec_data = &aic3x_setup,
+#elif defined(CONFIG_SND_DM365_VOICE_CODEC)
+	.codec_dev = &soc_codec_dev_cq93vc,
+#endif
+};
+
 /* evm audio subsystem */
 static struct snd_soc_device dm6467_evm_snd_devdata = {
 	.card = &dm6467_snd_soc_card_evm,
@@ -258,12 +296,15 @@ static int __init evm_init(void)
 	int index;
 	int ret;
 
-	if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) {
+	if (machine_is_davinci_evm()) {
 		evm_snd_dev_data = &evm_snd_devdata;
 		index = 0;
 	} else if (machine_is_davinci_dm355_evm()) {
 		evm_snd_dev_data = &evm_snd_devdata;
 		index = 1;
+	} else if (machine_is_davinci_dm365_evm()) {
+		evm_snd_dev_data = &dm365_evm_snd_devdata;
+		index = 0;
 	} else if (machine_is_davinci_dm6467_evm()) {
 		evm_snd_dev_data = &dm6467_evm_snd_devdata;
 		index = 0;
-- 
cgit v1.2.3


From c3746a07f13f8711885fd3909c03477a1b47ab82 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 11 Mar 2010 16:26:21 +0200
Subject: ASoC: tlv320dac33: Start/stop sequence change

To avoid race condition especially in FIFO modes the
sequence for enabling and disabling the codec need to
be changed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320dac33.c | 25 +++----------------------
 1 file changed, 3 insertions(+), 22 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index f9f367d29a90..e845c4b9066e 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -310,7 +310,8 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
 	if (power)
 		reg |= DAC33_PDNALLB;
 	else
-		reg &= ~DAC33_PDNALLB;
+		reg &= ~(DAC33_PDNALLB | DAC33_OSCPDNB |
+			 DAC33_DACRPDNB | DAC33_DACLPDNB);
 	dac33_write(codec, DAC33_PWR_CTRL, reg);
 }
 
@@ -634,26 +635,6 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev)
 	return IRQ_HANDLED;
 }
 
-static void dac33_shutdown(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
-	unsigned int pwr_ctrl;
-
-	/* Stop pending workqueue */
-	if (dac33->fifo_mode)
-		cancel_work_sync(&dac33->work);
-
-	mutex_lock(&dac33->mutex);
-	pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL);
-	pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB);
-	dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl);
-	mutex_unlock(&dac33->mutex);
-}
-
 static void dac33_oscwait(struct snd_soc_codec *codec)
 {
 	int timeout = 20;
@@ -751,6 +732,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 	}
 
 	mutex_lock(&dac33->mutex);
+	dac33_soft_power(codec, 0);
 	dac33_soft_power(codec, 1);
 
 	reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL);
@@ -1185,7 +1167,6 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33);
 #define DAC33_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
 
 static struct snd_soc_dai_ops dac33_dai_ops = {
-	.shutdown	= dac33_shutdown,
 	.hw_params	= dac33_hw_params,
 	.prepare	= dac33_pcm_prepare,
 	.trigger	= dac33_pcm_trigger,
-- 
cgit v1.2.3


From eeb309a8a6ff86ce462e072bc95ecbe85618505e Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 11 Mar 2010 16:26:22 +0200
Subject: ASoC: tlv320dac33: Add option for keeping the BCLK running

Platform data option for the codec to keep the BCLK clock
continuously running in FIFO modes (codec master).

OMAP3 McBSP when in slave mode needs continuous BCLK running
on the serial bus in order to operate correctly.

Since in FIFO mode the DAC33 can also shut down the BCLK clock
and enable it only when it is needed, let the platforms decide
if the CPU side needs the BCLK running or not.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/tlv320dac33-plat.h |  1 +
 sound/soc/codecs/tlv320dac33.c   | 13 +++++++++++--
 2 files changed, 12 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
index ac0665264bdf..3f428d53195b 100644
--- a/include/sound/tlv320dac33-plat.h
+++ b/include/sound/tlv320dac33-plat.h
@@ -15,6 +15,7 @@
 
 struct tlv320dac33_platform_data {
 	int power_gpio;
+	int keep_bclk;	/* Keep the BCLK running in FIFO modes */
 	u8 burst_bclkdiv;
 };
 
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index e845c4b9066e..a6f19271aaea 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -93,6 +93,8 @@ struct tlv320dac33_priv {
 	unsigned int nsample;		/* burst read amount from host */
 	u8 burst_bclkdiv;		/* BCLK divider value in burst mode */
 
+	int keep_bclk;			/* Keep the BCLK continuously running
+					 * in FIFO modes */
 	enum dac33_state state;
 };
 
@@ -803,7 +805,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		 */
 		fifoctrl_a &= ~DAC33_FBYPAS;
 		fifoctrl_a &= ~DAC33_FAUTO;
-		aictrl_b &= ~DAC33_BCLKON;
+		if (dac33->keep_bclk)
+			aictrl_b |= DAC33_BCLKON;
+		else
+			aictrl_b &= ~DAC33_BCLKON;
 		break;
 	case DAC33_FIFO_MODE7:
 		/*
@@ -814,7 +819,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		 */
 		fifoctrl_a &= ~DAC33_FBYPAS;
 		fifoctrl_a |= DAC33_FAUTO;
-		aictrl_b &= ~DAC33_BCLKON;
+		if (dac33->keep_bclk)
+			aictrl_b |= DAC33_BCLKON;
+		else
+			aictrl_b &= ~DAC33_BCLKON;
 		break;
 	default:
 		/*
@@ -1234,6 +1242,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 
 	dac33->power_gpio = pdata->power_gpio;
 	dac33->burst_bclkdiv = pdata->burst_bclkdiv;
+	dac33->keep_bclk = pdata->keep_bclk;
 	dac33->irq = client->irq;
 	dac33->nsample = NSAMPLE_MAX;
 	/* Disable FIFO use by default */
-- 
cgit v1.2.3


From 75581d2459d3e6e5e796a42b362e4abb612bc958 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Wed, 3 Mar 2010 15:08:09 +0200
Subject: ASoC: OMAP3: Report delay caused by the internal FIFO

Use the new delay calback function to report the delay through
ALSA for application caused by the internal FIFO.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap-mcbsp.c | 26 ++++++++++++++++++++++++++
 1 file changed, 26 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index e814a9591f78..2952fb046ed6 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -256,6 +256,31 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	return err;
 }
 
+static snd_pcm_sframes_t omap_mcbsp_dai_delay(
+			struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	u16 fifo_use;
+	snd_pcm_sframes_t delay;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		fifo_use = omap_mcbsp_get_tx_delay(mcbsp_data->bus_id);
+	else
+		fifo_use = omap_mcbsp_get_rx_delay(mcbsp_data->bus_id);
+
+	/*
+	 * Divide the used locations with the channel count to get the
+	 * FIFO usage in samples (don't care about partial samples in the
+	 * buffer).
+	 */
+	delay = fifo_use / substream->runtime->channels;
+
+	return delay;
+}
+
 static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 				    struct snd_pcm_hw_params *params,
 				    struct snd_soc_dai *dai)
@@ -607,6 +632,7 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
 	.startup	= omap_mcbsp_dai_startup,
 	.shutdown	= omap_mcbsp_dai_shutdown,
 	.trigger	= omap_mcbsp_dai_trigger,
+	.delay		= omap_mcbsp_dai_delay,
 	.hw_params	= omap_mcbsp_dai_hw_params,
 	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
 	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
-- 
cgit v1.2.3


From b4452d1fbf3fe30435850d2971d700e0ab3852b6 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 14 Mar 2010 22:05:40 +0000
Subject: ASoC: Remove version display from WM8750

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8750.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 475c67ac7818..c0c487501248 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -29,8 +29,6 @@
 
 #include "wm8750.h"
 
-#define WM8750_VERSION "0.12"
-
 /* codec private data */
 struct wm8750_priv {
 	unsigned int sysclk;
@@ -912,7 +910,6 @@ static int wm8750_probe(struct platform_device *pdev)
 	struct wm8750_priv *wm8750;
 	int ret;
 
-	pr_info("WM8750 Audio Codec %s", WM8750_VERSION);
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
-- 
cgit v1.2.3


From da34183e640ed380f708bf9ebcd592afda619028 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 15 Mar 2010 19:23:37 +0000
Subject: ASoC: Allow pins to be force enabled

Allow pins to be forced on regardless of their power state. This is
intended for use with microphone bias supplies which need to be
enabled in order to support microphone detection - in systems without
appropriate hardware leaving the microphone unbiased when not in use
saves power.

The force done at power check time in order to avoid disrupting other
power detection logic.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/soc-dapm.h |  3 +++
 sound/soc/soc-dapm.c     | 35 ++++++++++++++++++++++++++++++++++-
 2 files changed, 37 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 5915fc059842..d5d6ba862dfe 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -339,6 +339,8 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin);
 int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin);
 int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin);
 int snd_soc_dapm_sync(struct snd_soc_codec *codec);
+int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec,
+				  const char *pin);
 
 /* dapm widget types */
 enum snd_soc_dapm_type {
@@ -426,6 +428,7 @@ struct snd_soc_dapm_widget {
 	unsigned char new:1;			/* cnew complete */
 	unsigned char ext:1;			/* has external widgets */
 	unsigned char suspend:1;		/* was active before suspend */
+	unsigned char force:1;			/* force state */
 
 	int (*power_check)(struct snd_soc_dapm_widget *w);
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 86ded22e36af..bbb2135a0b21 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -979,7 +979,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 				break;
 
 			default:
-				power = w->power_check(w);
+				if (!w->force)
+					power = w->power_check(w);
+				else
+					power = 1;
 				if (power)
 					sys_power = 1;
 				break;
@@ -2133,6 +2136,36 @@ int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
 
+/**
+ * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Enables input/output pin regardless of any other state.  This is
+ * intended for use with microphone bias supplies used in microphone
+ * jack detection.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec, const char *pin)
+{
+	struct snd_soc_dapm_widget *w;
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (!strcmp(w->name, pin)) {
+			pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+			w->connected = 1;
+			w->force = 1;
+			return 0;
+		}
+	}
+
+	pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin);
+
 /**
  * snd_soc_dapm_disable_pin - disable pin.
  * @codec: SoC codec
-- 
cgit v1.2.3


From 73b34ead7429789f35eea147a3e185abd61c7d94 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 15 Mar 2010 17:46:02 +0000
Subject: ASoC: Add GPIO configuration support for WM8903

Allow users to pass in a default configuration for the GPIOs of
the WM8903 as platform data. This allows configuration of the pin
muxing of the device.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8903.h    | 216 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8903.c |  13 ++-
 sound/soc/codecs/wm8903.h | 195 -----------------------------------------
 3 files changed, 228 insertions(+), 196 deletions(-)
 create mode 100644 include/sound/wm8903.h

(limited to 'sound/soc')

diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
new file mode 100644
index 000000000000..00458589fe85
--- /dev/null
+++ b/include/sound/wm8903.h
@@ -0,0 +1,216 @@
+/*
+ * linux/sound/wm8903.h -- Platform data for WM8903
+ *
+ * Copyright 2010 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM8903_H
+#define __LINUX_SND_WM8903_H
+
+/* Used to enable configuration of a GPIO to all zeros */
+#define WM8903_GPIO_NO_CONFIG 0x8000
+
+/*
+ * R116 (0x74) - GPIO Control 1
+ */
+#define WM8903_GP1_FN_MASK                      0x1F00  /* GP1_FN - [12:8] */
+#define WM8903_GP1_FN_SHIFT                          8  /* GP1_FN - [12:8] */
+#define WM8903_GP1_FN_WIDTH                          5  /* GP1_FN - [12:8] */
+#define WM8903_GP1_DIR                          0x0080  /* GP1_DIR */
+#define WM8903_GP1_DIR_MASK                     0x0080  /* GP1_DIR */
+#define WM8903_GP1_DIR_SHIFT                         7  /* GP1_DIR */
+#define WM8903_GP1_DIR_WIDTH                         1  /* GP1_DIR */
+#define WM8903_GP1_OP_CFG                       0x0040  /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_MASK                  0x0040  /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_SHIFT                      6  /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_WIDTH                      1  /* GP1_OP_CFG */
+#define WM8903_GP1_IP_CFG                       0x0020  /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_MASK                  0x0020  /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_SHIFT                      5  /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_WIDTH                      1  /* GP1_IP_CFG */
+#define WM8903_GP1_LVL                          0x0010  /* GP1_LVL */
+#define WM8903_GP1_LVL_MASK                     0x0010  /* GP1_LVL */
+#define WM8903_GP1_LVL_SHIFT                         4  /* GP1_LVL */
+#define WM8903_GP1_LVL_WIDTH                         1  /* GP1_LVL */
+#define WM8903_GP1_PD                           0x0008  /* GP1_PD */
+#define WM8903_GP1_PD_MASK                      0x0008  /* GP1_PD */
+#define WM8903_GP1_PD_SHIFT                          3  /* GP1_PD */
+#define WM8903_GP1_PD_WIDTH                          1  /* GP1_PD */
+#define WM8903_GP1_PU                           0x0004  /* GP1_PU */
+#define WM8903_GP1_PU_MASK                      0x0004  /* GP1_PU */
+#define WM8903_GP1_PU_SHIFT                          2  /* GP1_PU */
+#define WM8903_GP1_PU_WIDTH                          1  /* GP1_PU */
+#define WM8903_GP1_INTMODE                      0x0002  /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_MASK                 0x0002  /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_SHIFT                     1  /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_WIDTH                     1  /* GP1_INTMODE */
+#define WM8903_GP1_DB                           0x0001  /* GP1_DB */
+#define WM8903_GP1_DB_MASK                      0x0001  /* GP1_DB */
+#define WM8903_GP1_DB_SHIFT                          0  /* GP1_DB */
+#define WM8903_GP1_DB_WIDTH                          1  /* GP1_DB */
+
+/*
+ * R117 (0x75) - GPIO Control 2
+ */
+#define WM8903_GP2_FN_MASK                      0x1F00  /* GP2_FN - [12:8] */
+#define WM8903_GP2_FN_SHIFT                          8  /* GP2_FN - [12:8] */
+#define WM8903_GP2_FN_WIDTH                          5  /* GP2_FN - [12:8] */
+#define WM8903_GP2_DIR                          0x0080  /* GP2_DIR */
+#define WM8903_GP2_DIR_MASK                     0x0080  /* GP2_DIR */
+#define WM8903_GP2_DIR_SHIFT                         7  /* GP2_DIR */
+#define WM8903_GP2_DIR_WIDTH                         1  /* GP2_DIR */
+#define WM8903_GP2_OP_CFG                       0x0040  /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_MASK                  0x0040  /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_SHIFT                      6  /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_WIDTH                      1  /* GP2_OP_CFG */
+#define WM8903_GP2_IP_CFG                       0x0020  /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_MASK                  0x0020  /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_SHIFT                      5  /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_WIDTH                      1  /* GP2_IP_CFG */
+#define WM8903_GP2_LVL                          0x0010  /* GP2_LVL */
+#define WM8903_GP2_LVL_MASK                     0x0010  /* GP2_LVL */
+#define WM8903_GP2_LVL_SHIFT                         4  /* GP2_LVL */
+#define WM8903_GP2_LVL_WIDTH                         1  /* GP2_LVL */
+#define WM8903_GP2_PD                           0x0008  /* GP2_PD */
+#define WM8903_GP2_PD_MASK                      0x0008  /* GP2_PD */
+#define WM8903_GP2_PD_SHIFT                          3  /* GP2_PD */
+#define WM8903_GP2_PD_WIDTH                          1  /* GP2_PD */
+#define WM8903_GP2_PU                           0x0004  /* GP2_PU */
+#define WM8903_GP2_PU_MASK                      0x0004  /* GP2_PU */
+#define WM8903_GP2_PU_SHIFT                          2  /* GP2_PU */
+#define WM8903_GP2_PU_WIDTH                          1  /* GP2_PU */
+#define WM8903_GP2_INTMODE                      0x0002  /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_MASK                 0x0002  /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_SHIFT                     1  /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_WIDTH                     1  /* GP2_INTMODE */
+#define WM8903_GP2_DB                           0x0001  /* GP2_DB */
+#define WM8903_GP2_DB_MASK                      0x0001  /* GP2_DB */
+#define WM8903_GP2_DB_SHIFT                          0  /* GP2_DB */
+#define WM8903_GP2_DB_WIDTH                          1  /* GP2_DB */
+
+/*
+ * R118 (0x76) - GPIO Control 3
+ */
+#define WM8903_GP3_FN_MASK                      0x1F00  /* GP3_FN - [12:8] */
+#define WM8903_GP3_FN_SHIFT                          8  /* GP3_FN - [12:8] */
+#define WM8903_GP3_FN_WIDTH                          5  /* GP3_FN - [12:8] */
+#define WM8903_GP3_DIR                          0x0080  /* GP3_DIR */
+#define WM8903_GP3_DIR_MASK                     0x0080  /* GP3_DIR */
+#define WM8903_GP3_DIR_SHIFT                         7  /* GP3_DIR */
+#define WM8903_GP3_DIR_WIDTH                         1  /* GP3_DIR */
+#define WM8903_GP3_OP_CFG                       0x0040  /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_MASK                  0x0040  /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_SHIFT                      6  /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_WIDTH                      1  /* GP3_OP_CFG */
+#define WM8903_GP3_IP_CFG                       0x0020  /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_MASK                  0x0020  /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_SHIFT                      5  /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_WIDTH                      1  /* GP3_IP_CFG */
+#define WM8903_GP3_LVL                          0x0010  /* GP3_LVL */
+#define WM8903_GP3_LVL_MASK                     0x0010  /* GP3_LVL */
+#define WM8903_GP3_LVL_SHIFT                         4  /* GP3_LVL */
+#define WM8903_GP3_LVL_WIDTH                         1  /* GP3_LVL */
+#define WM8903_GP3_PD                           0x0008  /* GP3_PD */
+#define WM8903_GP3_PD_MASK                      0x0008  /* GP3_PD */
+#define WM8903_GP3_PD_SHIFT                          3  /* GP3_PD */
+#define WM8903_GP3_PD_WIDTH                          1  /* GP3_PD */
+#define WM8903_GP3_PU                           0x0004  /* GP3_PU */
+#define WM8903_GP3_PU_MASK                      0x0004  /* GP3_PU */
+#define WM8903_GP3_PU_SHIFT                          2  /* GP3_PU */
+#define WM8903_GP3_PU_WIDTH                          1  /* GP3_PU */
+#define WM8903_GP3_INTMODE                      0x0002  /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_MASK                 0x0002  /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_SHIFT                     1  /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_WIDTH                     1  /* GP3_INTMODE */
+#define WM8903_GP3_DB                           0x0001  /* GP3_DB */
+#define WM8903_GP3_DB_MASK                      0x0001  /* GP3_DB */
+#define WM8903_GP3_DB_SHIFT                          0  /* GP3_DB */
+#define WM8903_GP3_DB_WIDTH                          1  /* GP3_DB */
+
+/*
+ * R119 (0x77) - GPIO Control 4
+ */
+#define WM8903_GP4_FN_MASK                      0x1F00  /* GP4_FN - [12:8] */
+#define WM8903_GP4_FN_SHIFT                          8  /* GP4_FN - [12:8] */
+#define WM8903_GP4_FN_WIDTH                          5  /* GP4_FN - [12:8] */
+#define WM8903_GP4_DIR                          0x0080  /* GP4_DIR */
+#define WM8903_GP4_DIR_MASK                     0x0080  /* GP4_DIR */
+#define WM8903_GP4_DIR_SHIFT                         7  /* GP4_DIR */
+#define WM8903_GP4_DIR_WIDTH                         1  /* GP4_DIR */
+#define WM8903_GP4_OP_CFG                       0x0040  /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_MASK                  0x0040  /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_SHIFT                      6  /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_WIDTH                      1  /* GP4_OP_CFG */
+#define WM8903_GP4_IP_CFG                       0x0020  /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_MASK                  0x0020  /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_SHIFT                      5  /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_WIDTH                      1  /* GP4_IP_CFG */
+#define WM8903_GP4_LVL                          0x0010  /* GP4_LVL */
+#define WM8903_GP4_LVL_MASK                     0x0010  /* GP4_LVL */
+#define WM8903_GP4_LVL_SHIFT                         4  /* GP4_LVL */
+#define WM8903_GP4_LVL_WIDTH                         1  /* GP4_LVL */
+#define WM8903_GP4_PD                           0x0008  /* GP4_PD */
+#define WM8903_GP4_PD_MASK                      0x0008  /* GP4_PD */
+#define WM8903_GP4_PD_SHIFT                          3  /* GP4_PD */
+#define WM8903_GP4_PD_WIDTH                          1  /* GP4_PD */
+#define WM8903_GP4_PU                           0x0004  /* GP4_PU */
+#define WM8903_GP4_PU_MASK                      0x0004  /* GP4_PU */
+#define WM8903_GP4_PU_SHIFT                          2  /* GP4_PU */
+#define WM8903_GP4_PU_WIDTH                          1  /* GP4_PU */
+#define WM8903_GP4_INTMODE                      0x0002  /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_MASK                 0x0002  /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_SHIFT                     1  /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_WIDTH                     1  /* GP4_INTMODE */
+#define WM8903_GP4_DB                           0x0001  /* GP4_DB */
+#define WM8903_GP4_DB_MASK                      0x0001  /* GP4_DB */
+#define WM8903_GP4_DB_SHIFT                          0  /* GP4_DB */
+#define WM8903_GP4_DB_WIDTH                          1  /* GP4_DB */
+
+/*
+ * R120 (0x78) - GPIO Control 5
+ */
+#define WM8903_GP5_FN_MASK                      0x1F00  /* GP5_FN - [12:8] */
+#define WM8903_GP5_FN_SHIFT                          8  /* GP5_FN - [12:8] */
+#define WM8903_GP5_FN_WIDTH                          5  /* GP5_FN - [12:8] */
+#define WM8903_GP5_DIR                          0x0080  /* GP5_DIR */
+#define WM8903_GP5_DIR_MASK                     0x0080  /* GP5_DIR */
+#define WM8903_GP5_DIR_SHIFT                         7  /* GP5_DIR */
+#define WM8903_GP5_DIR_WIDTH                         1  /* GP5_DIR */
+#define WM8903_GP5_OP_CFG                       0x0040  /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_MASK                  0x0040  /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_SHIFT                      6  /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_WIDTH                      1  /* GP5_OP_CFG */
+#define WM8903_GP5_IP_CFG                       0x0020  /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_MASK                  0x0020  /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_SHIFT                      5  /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_WIDTH                      1  /* GP5_IP_CFG */
+#define WM8903_GP5_LVL                          0x0010  /* GP5_LVL */
+#define WM8903_GP5_LVL_MASK                     0x0010  /* GP5_LVL */
+#define WM8903_GP5_LVL_SHIFT                         4  /* GP5_LVL */
+#define WM8903_GP5_LVL_WIDTH                         1  /* GP5_LVL */
+#define WM8903_GP5_PD                           0x0008  /* GP5_PD */
+#define WM8903_GP5_PD_MASK                      0x0008  /* GP5_PD */
+#define WM8903_GP5_PD_SHIFT                          3  /* GP5_PD */
+#define WM8903_GP5_PD_WIDTH                          1  /* GP5_PD */
+#define WM8903_GP5_PU                           0x0004  /* GP5_PU */
+#define WM8903_GP5_PU_MASK                      0x0004  /* GP5_PU */
+#define WM8903_GP5_PU_SHIFT                          2  /* GP5_PU */
+#define WM8903_GP5_PU_WIDTH                          1  /* GP5_PU */
+#define WM8903_GP5_INTMODE                      0x0002  /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_MASK                 0x0002  /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_SHIFT                     1  /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_WIDTH                     1  /* GP5_INTMODE */
+#define WM8903_GP5_DB                           0x0001  /* GP5_DB */
+#define WM8903_GP5_DB_MASK                      0x0001  /* GP5_DB */
+#define WM8903_GP5_DB_SHIFT                          0  /* GP5_DB */
+#define WM8903_GP5_DB_WIDTH                          1  /* GP5_DB */
+
+struct wm8903_platform_data {
+	u32 gpio_cfg[5];       /* Default register values for GPIO pin mux */
+};
+
+#endif
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 3595bd57c4eb..36952d77c271 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1531,7 +1531,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 {
 	struct wm8903_priv *wm8903;
 	struct snd_soc_codec *codec;
-	int ret;
+	int ret, i;
 	u16 val;
 
 	wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
@@ -1578,6 +1578,17 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 	wm8903_reset(codec);
 
+	/* Set up GPIOs */
+	if (pdata) {
+		for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
+			if (!pdata->gpio_cfg[i])
+				continue;
+
+			snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
+				      pdata->gpio_cfg[i] & 0xffff);
+		}
+	}
+
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index 0ea27e2b9963..551736cf2195 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -1134,201 +1134,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8903;
 #define WM8903_MASK_WRITE_ENA_SHIFT                  0  /* MASK_WRITE_ENA */
 #define WM8903_MASK_WRITE_ENA_WIDTH                  1  /* MASK_WRITE_ENA */
 
-/*
- * R116 (0x74) - GPIO Control 1
- */
-#define WM8903_GP1_FN_MASK                      0x1F00  /* GP1_FN - [12:8] */
-#define WM8903_GP1_FN_SHIFT                          8  /* GP1_FN - [12:8] */
-#define WM8903_GP1_FN_WIDTH                          5  /* GP1_FN - [12:8] */
-#define WM8903_GP1_DIR                          0x0080  /* GP1_DIR */
-#define WM8903_GP1_DIR_MASK                     0x0080  /* GP1_DIR */
-#define WM8903_GP1_DIR_SHIFT                         7  /* GP1_DIR */
-#define WM8903_GP1_DIR_WIDTH                         1  /* GP1_DIR */
-#define WM8903_GP1_OP_CFG                       0x0040  /* GP1_OP_CFG */
-#define WM8903_GP1_OP_CFG_MASK                  0x0040  /* GP1_OP_CFG */
-#define WM8903_GP1_OP_CFG_SHIFT                      6  /* GP1_OP_CFG */
-#define WM8903_GP1_OP_CFG_WIDTH                      1  /* GP1_OP_CFG */
-#define WM8903_GP1_IP_CFG                       0x0020  /* GP1_IP_CFG */
-#define WM8903_GP1_IP_CFG_MASK                  0x0020  /* GP1_IP_CFG */
-#define WM8903_GP1_IP_CFG_SHIFT                      5  /* GP1_IP_CFG */
-#define WM8903_GP1_IP_CFG_WIDTH                      1  /* GP1_IP_CFG */
-#define WM8903_GP1_LVL                          0x0010  /* GP1_LVL */
-#define WM8903_GP1_LVL_MASK                     0x0010  /* GP1_LVL */
-#define WM8903_GP1_LVL_SHIFT                         4  /* GP1_LVL */
-#define WM8903_GP1_LVL_WIDTH                         1  /* GP1_LVL */
-#define WM8903_GP1_PD                           0x0008  /* GP1_PD */
-#define WM8903_GP1_PD_MASK                      0x0008  /* GP1_PD */
-#define WM8903_GP1_PD_SHIFT                          3  /* GP1_PD */
-#define WM8903_GP1_PD_WIDTH                          1  /* GP1_PD */
-#define WM8903_GP1_PU                           0x0004  /* GP1_PU */
-#define WM8903_GP1_PU_MASK                      0x0004  /* GP1_PU */
-#define WM8903_GP1_PU_SHIFT                          2  /* GP1_PU */
-#define WM8903_GP1_PU_WIDTH                          1  /* GP1_PU */
-#define WM8903_GP1_INTMODE                      0x0002  /* GP1_INTMODE */
-#define WM8903_GP1_INTMODE_MASK                 0x0002  /* GP1_INTMODE */
-#define WM8903_GP1_INTMODE_SHIFT                     1  /* GP1_INTMODE */
-#define WM8903_GP1_INTMODE_WIDTH                     1  /* GP1_INTMODE */
-#define WM8903_GP1_DB                           0x0001  /* GP1_DB */
-#define WM8903_GP1_DB_MASK                      0x0001  /* GP1_DB */
-#define WM8903_GP1_DB_SHIFT                          0  /* GP1_DB */
-#define WM8903_GP1_DB_WIDTH                          1  /* GP1_DB */
-
-/*
- * R117 (0x75) - GPIO Control 2
- */
-#define WM8903_GP2_FN_MASK                      0x1F00  /* GP2_FN - [12:8] */
-#define WM8903_GP2_FN_SHIFT                          8  /* GP2_FN - [12:8] */
-#define WM8903_GP2_FN_WIDTH                          5  /* GP2_FN - [12:8] */
-#define WM8903_GP2_DIR                          0x0080  /* GP2_DIR */
-#define WM8903_GP2_DIR_MASK                     0x0080  /* GP2_DIR */
-#define WM8903_GP2_DIR_SHIFT                         7  /* GP2_DIR */
-#define WM8903_GP2_DIR_WIDTH                         1  /* GP2_DIR */
-#define WM8903_GP2_OP_CFG                       0x0040  /* GP2_OP_CFG */
-#define WM8903_GP2_OP_CFG_MASK                  0x0040  /* GP2_OP_CFG */
-#define WM8903_GP2_OP_CFG_SHIFT                      6  /* GP2_OP_CFG */
-#define WM8903_GP2_OP_CFG_WIDTH                      1  /* GP2_OP_CFG */
-#define WM8903_GP2_IP_CFG                       0x0020  /* GP2_IP_CFG */
-#define WM8903_GP2_IP_CFG_MASK                  0x0020  /* GP2_IP_CFG */
-#define WM8903_GP2_IP_CFG_SHIFT                      5  /* GP2_IP_CFG */
-#define WM8903_GP2_IP_CFG_WIDTH                      1  /* GP2_IP_CFG */
-#define WM8903_GP2_LVL                          0x0010  /* GP2_LVL */
-#define WM8903_GP2_LVL_MASK                     0x0010  /* GP2_LVL */
-#define WM8903_GP2_LVL_SHIFT                         4  /* GP2_LVL */
-#define WM8903_GP2_LVL_WIDTH                         1  /* GP2_LVL */
-#define WM8903_GP2_PD                           0x0008  /* GP2_PD */
-#define WM8903_GP2_PD_MASK                      0x0008  /* GP2_PD */
-#define WM8903_GP2_PD_SHIFT                          3  /* GP2_PD */
-#define WM8903_GP2_PD_WIDTH                          1  /* GP2_PD */
-#define WM8903_GP2_PU                           0x0004  /* GP2_PU */
-#define WM8903_GP2_PU_MASK                      0x0004  /* GP2_PU */
-#define WM8903_GP2_PU_SHIFT                          2  /* GP2_PU */
-#define WM8903_GP2_PU_WIDTH                          1  /* GP2_PU */
-#define WM8903_GP2_INTMODE                      0x0002  /* GP2_INTMODE */
-#define WM8903_GP2_INTMODE_MASK                 0x0002  /* GP2_INTMODE */
-#define WM8903_GP2_INTMODE_SHIFT                     1  /* GP2_INTMODE */
-#define WM8903_GP2_INTMODE_WIDTH                     1  /* GP2_INTMODE */
-#define WM8903_GP2_DB                           0x0001  /* GP2_DB */
-#define WM8903_GP2_DB_MASK                      0x0001  /* GP2_DB */
-#define WM8903_GP2_DB_SHIFT                          0  /* GP2_DB */
-#define WM8903_GP2_DB_WIDTH                          1  /* GP2_DB */
-
-/*
- * R118 (0x76) - GPIO Control 3
- */
-#define WM8903_GP3_FN_MASK                      0x1F00  /* GP3_FN - [12:8] */
-#define WM8903_GP3_FN_SHIFT                          8  /* GP3_FN - [12:8] */
-#define WM8903_GP3_FN_WIDTH                          5  /* GP3_FN - [12:8] */
-#define WM8903_GP3_DIR                          0x0080  /* GP3_DIR */
-#define WM8903_GP3_DIR_MASK                     0x0080  /* GP3_DIR */
-#define WM8903_GP3_DIR_SHIFT                         7  /* GP3_DIR */
-#define WM8903_GP3_DIR_WIDTH                         1  /* GP3_DIR */
-#define WM8903_GP3_OP_CFG                       0x0040  /* GP3_OP_CFG */
-#define WM8903_GP3_OP_CFG_MASK                  0x0040  /* GP3_OP_CFG */
-#define WM8903_GP3_OP_CFG_SHIFT                      6  /* GP3_OP_CFG */
-#define WM8903_GP3_OP_CFG_WIDTH                      1  /* GP3_OP_CFG */
-#define WM8903_GP3_IP_CFG                       0x0020  /* GP3_IP_CFG */
-#define WM8903_GP3_IP_CFG_MASK                  0x0020  /* GP3_IP_CFG */
-#define WM8903_GP3_IP_CFG_SHIFT                      5  /* GP3_IP_CFG */
-#define WM8903_GP3_IP_CFG_WIDTH                      1  /* GP3_IP_CFG */
-#define WM8903_GP3_LVL                          0x0010  /* GP3_LVL */
-#define WM8903_GP3_LVL_MASK                     0x0010  /* GP3_LVL */
-#define WM8903_GP3_LVL_SHIFT                         4  /* GP3_LVL */
-#define WM8903_GP3_LVL_WIDTH                         1  /* GP3_LVL */
-#define WM8903_GP3_PD                           0x0008  /* GP3_PD */
-#define WM8903_GP3_PD_MASK                      0x0008  /* GP3_PD */
-#define WM8903_GP3_PD_SHIFT                          3  /* GP3_PD */
-#define WM8903_GP3_PD_WIDTH                          1  /* GP3_PD */
-#define WM8903_GP3_PU                           0x0004  /* GP3_PU */
-#define WM8903_GP3_PU_MASK                      0x0004  /* GP3_PU */
-#define WM8903_GP3_PU_SHIFT                          2  /* GP3_PU */
-#define WM8903_GP3_PU_WIDTH                          1  /* GP3_PU */
-#define WM8903_GP3_INTMODE                      0x0002  /* GP3_INTMODE */
-#define WM8903_GP3_INTMODE_MASK                 0x0002  /* GP3_INTMODE */
-#define WM8903_GP3_INTMODE_SHIFT                     1  /* GP3_INTMODE */
-#define WM8903_GP3_INTMODE_WIDTH                     1  /* GP3_INTMODE */
-#define WM8903_GP3_DB                           0x0001  /* GP3_DB */
-#define WM8903_GP3_DB_MASK                      0x0001  /* GP3_DB */
-#define WM8903_GP3_DB_SHIFT                          0  /* GP3_DB */
-#define WM8903_GP3_DB_WIDTH                          1  /* GP3_DB */
-
-/*
- * R119 (0x77) - GPIO Control 4
- */
-#define WM8903_GP4_FN_MASK                      0x1F00  /* GP4_FN - [12:8] */
-#define WM8903_GP4_FN_SHIFT                          8  /* GP4_FN - [12:8] */
-#define WM8903_GP4_FN_WIDTH                          5  /* GP4_FN - [12:8] */
-#define WM8903_GP4_DIR                          0x0080  /* GP4_DIR */
-#define WM8903_GP4_DIR_MASK                     0x0080  /* GP4_DIR */
-#define WM8903_GP4_DIR_SHIFT                         7  /* GP4_DIR */
-#define WM8903_GP4_DIR_WIDTH                         1  /* GP4_DIR */
-#define WM8903_GP4_OP_CFG                       0x0040  /* GP4_OP_CFG */
-#define WM8903_GP4_OP_CFG_MASK                  0x0040  /* GP4_OP_CFG */
-#define WM8903_GP4_OP_CFG_SHIFT                      6  /* GP4_OP_CFG */
-#define WM8903_GP4_OP_CFG_WIDTH                      1  /* GP4_OP_CFG */
-#define WM8903_GP4_IP_CFG                       0x0020  /* GP4_IP_CFG */
-#define WM8903_GP4_IP_CFG_MASK                  0x0020  /* GP4_IP_CFG */
-#define WM8903_GP4_IP_CFG_SHIFT                      5  /* GP4_IP_CFG */
-#define WM8903_GP4_IP_CFG_WIDTH                      1  /* GP4_IP_CFG */
-#define WM8903_GP4_LVL                          0x0010  /* GP4_LVL */
-#define WM8903_GP4_LVL_MASK                     0x0010  /* GP4_LVL */
-#define WM8903_GP4_LVL_SHIFT                         4  /* GP4_LVL */
-#define WM8903_GP4_LVL_WIDTH                         1  /* GP4_LVL */
-#define WM8903_GP4_PD                           0x0008  /* GP4_PD */
-#define WM8903_GP4_PD_MASK                      0x0008  /* GP4_PD */
-#define WM8903_GP4_PD_SHIFT                          3  /* GP4_PD */
-#define WM8903_GP4_PD_WIDTH                          1  /* GP4_PD */
-#define WM8903_GP4_PU                           0x0004  /* GP4_PU */
-#define WM8903_GP4_PU_MASK                      0x0004  /* GP4_PU */
-#define WM8903_GP4_PU_SHIFT                          2  /* GP4_PU */
-#define WM8903_GP4_PU_WIDTH                          1  /* GP4_PU */
-#define WM8903_GP4_INTMODE                      0x0002  /* GP4_INTMODE */
-#define WM8903_GP4_INTMODE_MASK                 0x0002  /* GP4_INTMODE */
-#define WM8903_GP4_INTMODE_SHIFT                     1  /* GP4_INTMODE */
-#define WM8903_GP4_INTMODE_WIDTH                     1  /* GP4_INTMODE */
-#define WM8903_GP4_DB                           0x0001  /* GP4_DB */
-#define WM8903_GP4_DB_MASK                      0x0001  /* GP4_DB */
-#define WM8903_GP4_DB_SHIFT                          0  /* GP4_DB */
-#define WM8903_GP4_DB_WIDTH                          1  /* GP4_DB */
-
-/*
- * R120 (0x78) - GPIO Control 5
- */
-#define WM8903_GP5_FN_MASK                      0x1F00  /* GP5_FN - [12:8] */
-#define WM8903_GP5_FN_SHIFT                          8  /* GP5_FN - [12:8] */
-#define WM8903_GP5_FN_WIDTH                          5  /* GP5_FN - [12:8] */
-#define WM8903_GP5_DIR                          0x0080  /* GP5_DIR */
-#define WM8903_GP5_DIR_MASK                     0x0080  /* GP5_DIR */
-#define WM8903_GP5_DIR_SHIFT                         7  /* GP5_DIR */
-#define WM8903_GP5_DIR_WIDTH                         1  /* GP5_DIR */
-#define WM8903_GP5_OP_CFG                       0x0040  /* GP5_OP_CFG */
-#define WM8903_GP5_OP_CFG_MASK                  0x0040  /* GP5_OP_CFG */
-#define WM8903_GP5_OP_CFG_SHIFT                      6  /* GP5_OP_CFG */
-#define WM8903_GP5_OP_CFG_WIDTH                      1  /* GP5_OP_CFG */
-#define WM8903_GP5_IP_CFG                       0x0020  /* GP5_IP_CFG */
-#define WM8903_GP5_IP_CFG_MASK                  0x0020  /* GP5_IP_CFG */
-#define WM8903_GP5_IP_CFG_SHIFT                      5  /* GP5_IP_CFG */
-#define WM8903_GP5_IP_CFG_WIDTH                      1  /* GP5_IP_CFG */
-#define WM8903_GP5_LVL                          0x0010  /* GP5_LVL */
-#define WM8903_GP5_LVL_MASK                     0x0010  /* GP5_LVL */
-#define WM8903_GP5_LVL_SHIFT                         4  /* GP5_LVL */
-#define WM8903_GP5_LVL_WIDTH                         1  /* GP5_LVL */
-#define WM8903_GP5_PD                           0x0008  /* GP5_PD */
-#define WM8903_GP5_PD_MASK                      0x0008  /* GP5_PD */
-#define WM8903_GP5_PD_SHIFT                          3  /* GP5_PD */
-#define WM8903_GP5_PD_WIDTH                          1  /* GP5_PD */
-#define WM8903_GP5_PU                           0x0004  /* GP5_PU */
-#define WM8903_GP5_PU_MASK                      0x0004  /* GP5_PU */
-#define WM8903_GP5_PU_SHIFT                          2  /* GP5_PU */
-#define WM8903_GP5_PU_WIDTH                          1  /* GP5_PU */
-#define WM8903_GP5_INTMODE                      0x0002  /* GP5_INTMODE */
-#define WM8903_GP5_INTMODE_MASK                 0x0002  /* GP5_INTMODE */
-#define WM8903_GP5_INTMODE_SHIFT                     1  /* GP5_INTMODE */
-#define WM8903_GP5_INTMODE_WIDTH                     1  /* GP5_INTMODE */
-#define WM8903_GP5_DB                           0x0001  /* GP5_DB */
-#define WM8903_GP5_DB_MASK                      0x0001  /* GP5_DB */
-#define WM8903_GP5_DB_SHIFT                          0  /* GP5_DB */
-#define WM8903_GP5_DB_WIDTH                          1  /* GP5_DB */
-
 /*
  * R121 (0x79) - Interrupt Status 1
  */
-- 
cgit v1.2.3


From 37f88e8407f75fc6ced5cefb633c314556de3ad1 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 15 Mar 2010 18:14:34 +0000
Subject: ASoC: Initial WM8903 microphone bias and short detection

Provide support for WM8903 microphone presence and short detection
using the GPIOs to route out a logic signal suitable for handling
using snd_soc_jack_add_gpios() on the processor GPIOs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8903.h    | 29 +++++++++++++++++++++++++++++
 sound/soc/codecs/wm8903.c | 24 ++++++++++++++++--------
 sound/soc/codecs/wm8903.h | 22 ----------------------
 3 files changed, 45 insertions(+), 30 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index 00458589fe85..22b66a76febd 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -14,6 +14,28 @@
 /* Used to enable configuration of a GPIO to all zeros */
 #define WM8903_GPIO_NO_CONFIG 0x8000
 
+/*
+ * R6 (0x06) - Mic Bias Control 0
+ */
+#define WM8903_MICDET_HYST_ENA                  0x0080  /* MICDET_HYST_ENA */
+#define WM8903_MICDET_HYST_ENA_MASK             0x0080  /* MICDET_HYST_ENA */
+#define WM8903_MICDET_HYST_ENA_SHIFT                 7  /* MICDET_HYST_ENA */
+#define WM8903_MICDET_HYST_ENA_WIDTH                 1  /* MICDET_HYST_ENA */
+#define WM8903_MICDET_THR_MASK                  0x0070  /* MICDET_THR - [6:4] */
+#define WM8903_MICDET_THR_SHIFT                      4  /* MICDET_THR - [6:4] */
+#define WM8903_MICDET_THR_WIDTH                      3  /* MICDET_THR - [6:4] */
+#define WM8903_MICSHORT_THR_MASK                0x000C  /* MICSHORT_THR - [3:2] */
+#define WM8903_MICSHORT_THR_SHIFT                    2  /* MICSHORT_THR - [3:2] */
+#define WM8903_MICSHORT_THR_WIDTH                    2  /* MICSHORT_THR - [3:2] */
+#define WM8903_MICDET_ENA                       0x0002  /* MICDET_ENA */
+#define WM8903_MICDET_ENA_MASK                  0x0002  /* MICDET_ENA */
+#define WM8903_MICDET_ENA_SHIFT                      1  /* MICDET_ENA */
+#define WM8903_MICDET_ENA_WIDTH                      1  /* MICDET_ENA */
+#define WM8903_MICBIAS_ENA                      0x0001  /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_MASK                 0x0001  /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_SHIFT                     0  /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_WIDTH                     1  /* MICBIAS_ENA */
+
 /*
  * R116 (0x74) - GPIO Control 1
  */
@@ -210,6 +232,13 @@
 #define WM8903_GP5_DB_WIDTH                          1  /* GP5_DB */
 
 struct wm8903_platform_data {
+        /* Default register value for R6 (Mic bias), used to configure
+	 * microphone detection.  In conjunction with gpio_cfg this
+	 * can be used to route the microphone status signals out onto
+	 * the GPIOs for use with snd_soc_jack_add_gpios().
+	 */
+	u16 micdet_cfg;
+
 	u32 gpio_cfg[5];       /* Default register values for GPIO pin mux */
 };
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 36952d77c271..467e6c335c5a 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -11,7 +11,6 @@
  *
  * TODO:
  *  - TDM mode configuration.
- *  - Mic detect.
  *  - Digital microphone support.
  *  - Interrupt support (mic detect and sequencer).
  */
@@ -246,10 +245,10 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
 
 	BUG_ON(start > 48);
 
-	/* Enable the sequencer */
+	/* Enable the sequencer if it's not already on */
 	reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0);
-	reg[0] |= WM8903_WSEQ_ENA;
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
+	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0,
+		      reg[0] | WM8903_WSEQ_ENA);
 
 	dev_dbg(&i2c->dev, "Starting sequence at %d\n", start);
 
@@ -268,9 +267,8 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
 
 	dev_dbg(&i2c->dev, "Sequence complete\n");
 
-	/* Disable the sequencer again */
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0,
-		     reg[0] & ~WM8903_WSEQ_ENA);
+	/* Disable the sequencer again if we enabled it */
+	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
 
 	return 0;
 }
@@ -1578,7 +1576,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 	wm8903_reset(codec);
 
-	/* Set up GPIOs */
+	/* Set up GPIOs and microphone detection */
 	if (pdata) {
 		for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
 			if (!pdata->gpio_cfg[i])
@@ -1587,6 +1585,16 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 			snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
 				      pdata->gpio_cfg[i] & 0xffff);
 		}
+
+		snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
+			      pdata->micdet_cfg);
+
+		/* Microphone detection needs the WSEQ clock */
+		if (pdata->micdet_cfg)
+			snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
+					    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+
+		wm8903->mic_delay = pdata->micdet_delay;
 	}
 
 	/* power on device */
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index 551736cf2195..8f19a2413785 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -172,28 +172,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8903;
 #define WM8903_VMID_RES_250K                         3
 #define WM8903_VMID_RES_5K                           4
 
-/*
- * R6 (0x06) - Mic Bias Control 0
- */
-#define WM8903_MICDET_HYST_ENA                  0x0080  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_MASK             0x0080  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_SHIFT                 7  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_WIDTH                 1  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_THR_MASK                  0x0070  /* MICDET_THR - [6:4] */
-#define WM8903_MICDET_THR_SHIFT                      4  /* MICDET_THR - [6:4] */
-#define WM8903_MICDET_THR_WIDTH                      3  /* MICDET_THR - [6:4] */
-#define WM8903_MICSHORT_THR_MASK                0x000C  /* MICSHORT_THR - [3:2] */
-#define WM8903_MICSHORT_THR_SHIFT                    2  /* MICSHORT_THR - [3:2] */
-#define WM8903_MICSHORT_THR_WIDTH                    2  /* MICSHORT_THR - [3:2] */
-#define WM8903_MICDET_ENA                       0x0002  /* MICDET_ENA */
-#define WM8903_MICDET_ENA_MASK                  0x0002  /* MICDET_ENA */
-#define WM8903_MICDET_ENA_SHIFT                      1  /* MICDET_ENA */
-#define WM8903_MICDET_ENA_WIDTH                      1  /* MICDET_ENA */
-#define WM8903_MICBIAS_ENA                      0x0001  /* MICBIAS_ENA */
-#define WM8903_MICBIAS_ENA_MASK                 0x0001  /* MICBIAS_ENA */
-#define WM8903_MICBIAS_ENA_SHIFT                     0  /* MICBIAS_ENA */
-#define WM8903_MICBIAS_ENA_WIDTH                     1  /* MICBIAS_ENA */
-
 /*
  * R8 (0x08) - Analogue DAC 0
  */
-- 
cgit v1.2.3


From 8abd16a65d81756706016720e2cc7eeb81d06a2e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 15 Mar 2010 18:25:26 +0000
Subject: ASoC: Add WM8903 interrupt support

Currently used to detect completion of the write sequencer.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8903.h    |  2 ++
 sound/soc/codecs/wm8903.c | 66 +++++++++++++++++++++++++++++++++++++++++++----
 2 files changed, 63 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index 22b66a76febd..49bb1a844bfc 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -232,6 +232,8 @@
 #define WM8903_GP5_DB_WIDTH                          1  /* GP5_DB */
 
 struct wm8903_platform_data {
+	bool irq_active_low;   /* Set if IRQ active low, default high */
+
         /* Default register value for R6 (Mic bias), used to configure
 	 * microphone detection.  In conjunction with gpio_cfg this
 	 * can be used to route the microphone status signals out onto
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 467e6c335c5a..7aa2adbe6fba 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -12,12 +12,12 @@
  * TODO:
  *  - TDM mode configuration.
  *  - Digital microphone support.
- *  - Interrupt support (mic detect and sequencer).
  */
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/completion.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
@@ -29,6 +29,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
+#include <sound/wm8903.h>
 
 #include "wm8903.h"
 
@@ -220,6 +221,8 @@ struct wm8903_priv {
 	int playback_active;
 	int capture_active;
 
+	struct completion wseq;
+
 	struct snd_pcm_substream *master_substream;
 	struct snd_pcm_substream *slave_substream;
 };
@@ -242,6 +245,7 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
 {
 	u16 reg[5];
 	struct i2c_client *i2c = codec->control_data;
+	struct wm8903_priv *wm8903 = codec->private_data;
 
 	BUG_ON(start > 48);
 
@@ -256,11 +260,11 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
 		     start | WM8903_WSEQ_START);
 
 	/* Wait for it to complete.  If we have the interrupt wired up then
-	 * we could block waiting for an interrupt, though polling may still
-	 * be desirable for diagnostic purposes.
+	 * that will break us out of the poll early.
 	 */
 	do {
-		msleep(10);
+		wait_for_completion_timeout(&wm8903->wseq,
+					    msecs_to_jiffies(10));
 
 		reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4);
 	} while (reg[4] & WM8903_WSEQ_BUSY);
@@ -1433,6 +1437,22 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static irqreturn_t wm8903_irq(int irq, void *data)
+{
+	struct wm8903_priv *wm8903 = data;
+	struct snd_soc_codec *codec = &wm8903->codec;
+	int reg;
+
+	reg = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1);
+
+	if (reg & WM8903_WSEQ_BUSY_EINT) {
+		dev_dbg(codec->dev, "Write sequencer done\n");
+		complete(&wm8903->wseq);
+	}
+
+	return IRQ_HANDLED;
+}
+
 #define WM8903_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\
 			       SNDRV_PCM_RATE_11025 |	\
 			       SNDRV_PCM_RATE_16000 |	\
@@ -1527,9 +1547,11 @@ static struct snd_soc_codec *wm8903_codec;
 static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *id)
 {
+	struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8903_priv *wm8903;
 	struct snd_soc_codec *codec;
 	int ret, i;
+	int trigger, irq_pol;
 	u16 val;
 
 	wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
@@ -1553,6 +1575,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 	codec->reg_cache = &wm8903->reg_cache[0];
 	codec->private_data = wm8903;
 	codec->volatile_register = wm8903_volatile_register;
+	init_completion(&wm8903->wseq);
 
 	i2c_set_clientdata(i2c, codec);
 	codec->control_data = i2c;
@@ -1596,6 +1619,32 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 		wm8903->mic_delay = pdata->micdet_delay;
 	}
+	
+	if (i2c->irq) {
+		if (pdata && pdata->irq_active_low) {
+			trigger = IRQF_TRIGGER_LOW;
+			irq_pol = WM8903_IRQ_POL;
+		} else {
+			trigger = IRQF_TRIGGER_HIGH;
+			irq_pol = 0;
+		}
+
+		snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL,
+				    WM8903_IRQ_POL, irq_pol);
+		
+		ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq,
+					   trigger | IRQF_ONESHOT,
+					   "wm8903", wm8903);
+		if (ret != 0) {
+			dev_err(&i2c->dev, "Failed to request IRQ: %d\n",
+				ret);
+			goto err;
+		}
+
+		/* Enable write sequencer interrupts */
+		snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK,
+				    WM8903_IM_WSEQ_BUSY_EINT, 0);
+	}
 
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1637,7 +1686,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
-		goto err;
+		goto err_irq;
 	}
 
 	ret = snd_soc_register_dai(&wm8903_dai);
@@ -1650,6 +1699,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 err_codec:
 	snd_soc_unregister_codec(codec);
+err_irq:
+	if (i2c->irq)
+		free_irq(i2c->irq, wm8903);
 err:
 	wm8903_codec = NULL;
 	kfree(wm8903);
@@ -1659,12 +1711,16 @@ err:
 static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	struct wm8903_priv *priv = codec->private_data;
 
 	snd_soc_unregister_dai(&wm8903_dai);
 	snd_soc_unregister_codec(codec);
 
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
+	if (client->irq)
+		free_irq(client->irq, priv);
+
 	kfree(codec->private_data);
 
 	wm8903_codec = NULL;
-- 
cgit v1.2.3


From 7245387e36e16ae918467685c34510606fd74b7c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 15 Mar 2010 21:22:58 +0000
Subject: ASoC: Implement interrupt driven microphone detection for WM8903

Support use of the WM8903 IRQ for reporting of microphone presence
and short detection.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8903.h    |   2 +
 sound/soc/codecs/wm8903.c | 106 ++++++++++++++++++++++++++++++++++++++++++++--
 sound/soc/codecs/wm8903.h |   4 ++
 3 files changed, 109 insertions(+), 3 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index 49bb1a844bfc..b4a0db2307ef 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -241,6 +241,8 @@ struct wm8903_platform_data {
 	 */
 	u16 micdet_cfg;
 
+	int micdet_delay;      /* Delay after microphone detection (ms) */
+
 	u32 gpio_cfg[5];       /* Default register values for GPIO pin mux */
 };
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 7aa2adbe6fba..b5427b47d6fd 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -23,6 +23,7 @@
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
+#include <sound/jack.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/tlv.h>
@@ -223,6 +224,12 @@ struct wm8903_priv {
 
 	struct completion wseq;
 
+	struct snd_soc_jack *mic_jack;
+	int mic_det;
+	int mic_short;
+	int mic_last_report;
+	int mic_delay;
+
 	struct snd_pcm_substream *master_substream;
 	struct snd_pcm_substream *slave_substream;
 };
@@ -1437,19 +1444,112 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+/**
+ * wm8903_mic_detect - Enable microphone detection via the WM8903 IRQ
+ *
+ * @codec:  WM8903 codec
+ * @jack:   jack to report detection events on
+ * @det:    value to report for presence detection
+ * @shrt:   value to report for short detection
+ *
+ * Enable microphone detection via IRQ on the WM8903.  If GPIOs are
+ * being used to bring out signals to the processor then only platform
+ * data configuration is needed for WM8903 and processor GPIOs should
+ * be configured using snd_soc_jack_add_gpios() instead.
+ *
+ * The current threasholds for detection should be configured using
+ * micdet_cfg in the platform data.  Using this function will force on
+ * the microphone bias for the device.
+ */
+int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
+		      int det, int shrt)
+{
+	struct wm8903_priv *wm8903 = codec->private_data;
+	int irq_mask = 0;
+
+	dev_dbg(codec->dev, "Enabling microphone detection: %x %x\n",
+		det, shrt);
+
+	/* Store the configuration */
+	wm8903->mic_jack = jack;
+	wm8903->mic_det = det;
+	wm8903->mic_short = shrt;
+
+	/* Enable interrupts we've got a report configured for */
+	if (det)
+		irq_mask &= ~WM8903_MICDET_EINT;
+	if (shrt)
+		irq_mask &= ~WM8903_MICSHRT_EINT;
+
+	snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK,
+			    WM8903_MICDET_EINT | WM8903_MICSHRT_EINT,
+			    irq_mask);
+
+	/* Enable mic detection, this may not have been set through
+	 * platform data (eg, if the defaults are OK). */
+	snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
+			    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+	snd_soc_update_bits(codec, WM8903_MIC_BIAS_CONTROL_0,
+			    WM8903_MICDET_ENA, WM8903_MICDET_ENA);
+
+	/* Force the microphone bias on; this will trigger an initial
+	 * detection. */
+	snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm8903_mic_detect);
+
 static irqreturn_t wm8903_irq(int irq, void *data)
 {
 	struct wm8903_priv *wm8903 = data;
 	struct snd_soc_codec *codec = &wm8903->codec;
-	int reg;
+	int mic_report;
+	int int_pol;
+	int int_val = 0;
+	int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK);
 
-	reg = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1);
+	int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
 
-	if (reg & WM8903_WSEQ_BUSY_EINT) {
+	if (int_val & WM8903_WSEQ_BUSY_EINT) {
 		dev_dbg(codec->dev, "Write sequencer done\n");
 		complete(&wm8903->wseq);
 	}
 
+	/*
+	 * The rest is microphone jack detection.  We need to manually
+	 * invert the polarity of the interrupt after each event - to
+	 * simplify the code keep track of the last state we reported
+	 * and just invert the relevant bits in both the report and
+	 * the polarity register.
+	 */
+	mic_report = wm8903->mic_last_report;
+	int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1);
+
+	if (int_val & WM8903_MICSHRT_EINT) {
+		dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol);
+
+		mic_report ^= wm8903->mic_short;
+		int_pol ^= WM8903_MICSHRT_INV;
+	}
+
+	if (int_val & WM8903_MICDET_EINT) {
+		dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol);
+
+		mic_report ^= wm8903->mic_det;
+		int_pol ^= WM8903_MICDET_INV;
+
+		msleep(wm8903->mic_delay);
+	}
+
+	snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1,
+			    WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
+
+	snd_soc_jack_report(wm8903->mic_jack, mic_report,
+			    wm8903->mic_short | wm8903->mic_det);
+
+	wm8903->mic_last_report = mic_report;
+
 	return IRQ_HANDLED;
 }
 
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index 8f19a2413785..ce384a2ad820 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -18,6 +18,10 @@
 extern struct snd_soc_dai wm8903_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8903;
 
+extern int wm8903_mic_detect(struct snd_soc_codec *codec,
+			     struct snd_soc_jack *jack,
+			     int det, int shrt);
+
 #define WM8903_MCLK_DIV_2 1
 #define WM8903_CLK_SYS    2
 #define WM8903_BCLK       3
-- 
cgit v1.2.3


From cdce4e9ba7ff86a5f43b971a9af88b25a3f6c9a4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 16 Mar 2010 12:01:28 +0000
Subject: ASoC: Allow configuration of WM8904 GPIO pin functions

Provide platform data allowing the configuration of the GPIO pins
on the WM8904 to be selected, allowing alternate functions to be
enabled.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8904.h    | 74 +++++++++++++++++++++++++++++++++++++++++++++--
 sound/soc/codecs/wm8904.c | 12 ++++++++
 sound/soc/codecs/wm8904.h | 64 ----------------------------------------
 3 files changed, 84 insertions(+), 66 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h
index d66575a601be..ddeeebf7c426 100644
--- a/include/sound/wm8904.h
+++ b/include/sound/wm8904.h
@@ -15,8 +15,76 @@
 #ifndef __MFD_WM8994_PDATA_H__
 #define __MFD_WM8994_PDATA_H__
 
-#define WM8904_DRC_REGS 4
-#define WM8904_EQ_REGS  25
+/* Used to enable configuration of a GPIO to all zeros */
+#define WM8904_GPIO_NO_CONFIG 0x8000
+
+/*
+ * R121 (0x79) - GPIO Control 1
+ */
+#define WM8904_GPIO1_PU                         0x0020  /* GPIO1_PU */
+#define WM8904_GPIO1_PU_MASK                    0x0020  /* GPIO1_PU */
+#define WM8904_GPIO1_PU_SHIFT                        5  /* GPIO1_PU */
+#define WM8904_GPIO1_PU_WIDTH                        1  /* GPIO1_PU */
+#define WM8904_GPIO1_PD                         0x0010  /* GPIO1_PD */
+#define WM8904_GPIO1_PD_MASK                    0x0010  /* GPIO1_PD */
+#define WM8904_GPIO1_PD_SHIFT                        4  /* GPIO1_PD */
+#define WM8904_GPIO1_PD_WIDTH                        1  /* GPIO1_PD */
+#define WM8904_GPIO1_SEL_MASK                   0x000F  /* GPIO1_SEL - [3:0] */
+#define WM8904_GPIO1_SEL_SHIFT                       0  /* GPIO1_SEL - [3:0] */
+#define WM8904_GPIO1_SEL_WIDTH                       4  /* GPIO1_SEL - [3:0] */
+
+/*
+ * R122 (0x7A) - GPIO Control 2
+ */
+#define WM8904_GPIO2_PU                         0x0020  /* GPIO2_PU */
+#define WM8904_GPIO2_PU_MASK                    0x0020  /* GPIO2_PU */
+#define WM8904_GPIO2_PU_SHIFT                        5  /* GPIO2_PU */
+#define WM8904_GPIO2_PU_WIDTH                        1  /* GPIO2_PU */
+#define WM8904_GPIO2_PD                         0x0010  /* GPIO2_PD */
+#define WM8904_GPIO2_PD_MASK                    0x0010  /* GPIO2_PD */
+#define WM8904_GPIO2_PD_SHIFT                        4  /* GPIO2_PD */
+#define WM8904_GPIO2_PD_WIDTH                        1  /* GPIO2_PD */
+#define WM8904_GPIO2_SEL_MASK                   0x000F  /* GPIO2_SEL - [3:0] */
+#define WM8904_GPIO2_SEL_SHIFT                       0  /* GPIO2_SEL - [3:0] */
+#define WM8904_GPIO2_SEL_WIDTH                       4  /* GPIO2_SEL - [3:0] */
+
+/*
+ * R123 (0x7B) - GPIO Control 3
+ */
+#define WM8904_GPIO3_PU                         0x0020  /* GPIO3_PU */
+#define WM8904_GPIO3_PU_MASK                    0x0020  /* GPIO3_PU */
+#define WM8904_GPIO3_PU_SHIFT                        5  /* GPIO3_PU */
+#define WM8904_GPIO3_PU_WIDTH                        1  /* GPIO3_PU */
+#define WM8904_GPIO3_PD                         0x0010  /* GPIO3_PD */
+#define WM8904_GPIO3_PD_MASK                    0x0010  /* GPIO3_PD */
+#define WM8904_GPIO3_PD_SHIFT                        4  /* GPIO3_PD */
+#define WM8904_GPIO3_PD_WIDTH                        1  /* GPIO3_PD */
+#define WM8904_GPIO3_SEL_MASK                   0x000F  /* GPIO3_SEL - [3:0] */
+#define WM8904_GPIO3_SEL_SHIFT                       0  /* GPIO3_SEL - [3:0] */
+#define WM8904_GPIO3_SEL_WIDTH                       4  /* GPIO3_SEL - [3:0] */
+
+/*
+ * R124 (0x7C) - GPIO Control 4
+ */
+#define WM8904_GPI7_ENA                         0x0200  /* GPI7_ENA */
+#define WM8904_GPI7_ENA_MASK                    0x0200  /* GPI7_ENA */
+#define WM8904_GPI7_ENA_SHIFT                        9  /* GPI7_ENA */
+#define WM8904_GPI7_ENA_WIDTH                        1  /* GPI7_ENA */
+#define WM8904_GPI8_ENA                         0x0100  /* GPI8_ENA */
+#define WM8904_GPI8_ENA_MASK                    0x0100  /* GPI8_ENA */
+#define WM8904_GPI8_ENA_SHIFT                        8  /* GPI8_ENA */
+#define WM8904_GPI8_ENA_WIDTH                        1  /* GPI8_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA               0x0080  /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA_MASK          0x0080  /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT              7  /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH              1  /* GPIO_BCLK_MODE_ENA */
+#define WM8904_GPIO_BCLK_SEL_MASK               0x000F  /* GPIO_BCLK_SEL - [3:0] */
+#define WM8904_GPIO_BCLK_SEL_SHIFT                   0  /* GPIO_BCLK_SEL - [3:0] */
+#define WM8904_GPIO_BCLK_SEL_WIDTH                   4  /* GPIO_BCLK_SEL - [3:0] */
+
+#define WM8904_GPIO_REGS 4
+#define WM8904_DRC_REGS  4
+#define WM8904_EQ_REGS   25
 
 /**
  * DRC configurations are specified with a label and a set of register
@@ -52,6 +120,8 @@ struct wm8904_pdata {
 
 	int num_retune_mobile_cfgs;
 	struct wm8904_retune_mobile_cfg *retune_mobile_cfgs;
+
+	u32 gpio_cfg[WM8904_GPIO_REGS];
 };
 
 #endif
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 593e47d0e0eb..ce3f004217ae 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2425,6 +2425,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8904);
 static int wm8904_register(struct wm8904_priv *wm8904,
 			   enum snd_soc_control_type control)
 {
+	struct wm8904_pdata *pdata = wm8904->pdata;
 	int ret;
 	struct snd_soc_codec *codec = &wm8904->codec;
 	int i;
@@ -2530,6 +2531,17 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 		WM8904_LINEOUTRZC;
 	wm8904->reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE;
 
+	/* Apply configuration from the platform data. */
+	if (wm8904->pdata) {
+		for (i = 0; i < WM8904_GPIO_REGS; i++) {
+			if (!pdata->gpio_cfg[i])
+				continue;
+
+			wm8904->reg_cache[WM8904_GPIO_CONTROL_1 + i]
+				= pdata->gpio_cfg[i] & 0xffff;
+		}
+	}
+
 	/* Set Class W by default - this will be managed by the Class
 	 * G widget at runtime where bypass paths are available.
 	 */
diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h
index b68886df34e4..10603a75c1a5 100644
--- a/sound/soc/codecs/wm8904.h
+++ b/sound/soc/codecs/wm8904.h
@@ -1199,70 +1199,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8904;
 #define WM8904_FLL_CLK_REF_SRC_SHIFT                 0  /* FLL_CLK_REF_SRC - [1:0] */
 #define WM8904_FLL_CLK_REF_SRC_WIDTH                 2  /* FLL_CLK_REF_SRC - [1:0] */
 
-/*
- * R121 (0x79) - GPIO Control 1
- */
-#define WM8904_GPIO1_PU                         0x0020  /* GPIO1_PU */
-#define WM8904_GPIO1_PU_MASK                    0x0020  /* GPIO1_PU */
-#define WM8904_GPIO1_PU_SHIFT                        5  /* GPIO1_PU */
-#define WM8904_GPIO1_PU_WIDTH                        1  /* GPIO1_PU */
-#define WM8904_GPIO1_PD                         0x0010  /* GPIO1_PD */
-#define WM8904_GPIO1_PD_MASK                    0x0010  /* GPIO1_PD */
-#define WM8904_GPIO1_PD_SHIFT                        4  /* GPIO1_PD */
-#define WM8904_GPIO1_PD_WIDTH                        1  /* GPIO1_PD */
-#define WM8904_GPIO1_SEL_MASK                   0x000F  /* GPIO1_SEL - [3:0] */
-#define WM8904_GPIO1_SEL_SHIFT                       0  /* GPIO1_SEL - [3:0] */
-#define WM8904_GPIO1_SEL_WIDTH                       4  /* GPIO1_SEL - [3:0] */
-
-/*
- * R122 (0x7A) - GPIO Control 2
- */
-#define WM8904_GPIO2_PU                         0x0020  /* GPIO2_PU */
-#define WM8904_GPIO2_PU_MASK                    0x0020  /* GPIO2_PU */
-#define WM8904_GPIO2_PU_SHIFT                        5  /* GPIO2_PU */
-#define WM8904_GPIO2_PU_WIDTH                        1  /* GPIO2_PU */
-#define WM8904_GPIO2_PD                         0x0010  /* GPIO2_PD */
-#define WM8904_GPIO2_PD_MASK                    0x0010  /* GPIO2_PD */
-#define WM8904_GPIO2_PD_SHIFT                        4  /* GPIO2_PD */
-#define WM8904_GPIO2_PD_WIDTH                        1  /* GPIO2_PD */
-#define WM8904_GPIO2_SEL_MASK                   0x000F  /* GPIO2_SEL - [3:0] */
-#define WM8904_GPIO2_SEL_SHIFT                       0  /* GPIO2_SEL - [3:0] */
-#define WM8904_GPIO2_SEL_WIDTH                       4  /* GPIO2_SEL - [3:0] */
-
-/*
- * R123 (0x7B) - GPIO Control 3
- */
-#define WM8904_GPIO3_PU                         0x0020  /* GPIO3_PU */
-#define WM8904_GPIO3_PU_MASK                    0x0020  /* GPIO3_PU */
-#define WM8904_GPIO3_PU_SHIFT                        5  /* GPIO3_PU */
-#define WM8904_GPIO3_PU_WIDTH                        1  /* GPIO3_PU */
-#define WM8904_GPIO3_PD                         0x0010  /* GPIO3_PD */
-#define WM8904_GPIO3_PD_MASK                    0x0010  /* GPIO3_PD */
-#define WM8904_GPIO3_PD_SHIFT                        4  /* GPIO3_PD */
-#define WM8904_GPIO3_PD_WIDTH                        1  /* GPIO3_PD */
-#define WM8904_GPIO3_SEL_MASK                   0x000F  /* GPIO3_SEL - [3:0] */
-#define WM8904_GPIO3_SEL_SHIFT                       0  /* GPIO3_SEL - [3:0] */
-#define WM8904_GPIO3_SEL_WIDTH                       4  /* GPIO3_SEL - [3:0] */
-
-/*
- * R124 (0x7C) - GPIO Control 4
- */
-#define WM8904_GPI7_ENA                         0x0200  /* GPI7_ENA */
-#define WM8904_GPI7_ENA_MASK                    0x0200  /* GPI7_ENA */
-#define WM8904_GPI7_ENA_SHIFT                        9  /* GPI7_ENA */
-#define WM8904_GPI7_ENA_WIDTH                        1  /* GPI7_ENA */
-#define WM8904_GPI8_ENA                         0x0100  /* GPI8_ENA */
-#define WM8904_GPI8_ENA_MASK                    0x0100  /* GPI8_ENA */
-#define WM8904_GPI8_ENA_SHIFT                        8  /* GPI8_ENA */
-#define WM8904_GPI8_ENA_WIDTH                        1  /* GPI8_ENA */
-#define WM8904_GPIO_BCLK_MODE_ENA               0x0080  /* GPIO_BCLK_MODE_ENA */
-#define WM8904_GPIO_BCLK_MODE_ENA_MASK          0x0080  /* GPIO_BCLK_MODE_ENA */
-#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT              7  /* GPIO_BCLK_MODE_ENA */
-#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH              1  /* GPIO_BCLK_MODE_ENA */
-#define WM8904_GPIO_BCLK_SEL_MASK               0x000F  /* GPIO_BCLK_SEL - [3:0] */
-#define WM8904_GPIO_BCLK_SEL_SHIFT                   0  /* GPIO_BCLK_SEL - [3:0] */
-#define WM8904_GPIO_BCLK_SEL_WIDTH                   4  /* GPIO_BCLK_SEL - [3:0] */
-
 /*
  * R126 (0x7E) - Digital Pulls
  */
-- 
cgit v1.2.3


From fbc2dae854da5fc44e2ce2f97446c8bd283c7e17 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 16 Mar 2010 14:26:45 +0000
Subject: ASoC: Support GPIO based microphone detection for WM8904

The WM8904 allows microphone detection signals to be brought out as
alternate functions of the GPIO signals which can be detected using
interrupt inputs on the CPU. Allow this to be configured using
platform data.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm8904.h    | 36 ++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8904.c |  5 +++++
 sound/soc/codecs/wm8904.h | 33 ---------------------------------
 3 files changed, 41 insertions(+), 33 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h
index ddeeebf7c426..898be3a8db9a 100644
--- a/include/sound/wm8904.h
+++ b/include/sound/wm8904.h
@@ -18,6 +18,40 @@
 /* Used to enable configuration of a GPIO to all zeros */
 #define WM8904_GPIO_NO_CONFIG 0x8000
 
+/*
+ * R6 (0x06) - Mic Bias Control 0
+ */
+#define WM8904_MICDET_THR_MASK                  0x0070  /* MICDET_THR - [6:4] */
+#define WM8904_MICDET_THR_SHIFT                      4  /* MICDET_THR - [6:4] */
+#define WM8904_MICDET_THR_WIDTH                      3  /* MICDET_THR - [6:4] */
+#define WM8904_MICSHORT_THR_MASK                0x000C  /* MICSHORT_THR - [3:2] */
+#define WM8904_MICSHORT_THR_SHIFT                    2  /* MICSHORT_THR - [3:2] */
+#define WM8904_MICSHORT_THR_WIDTH                    2  /* MICSHORT_THR - [3:2] */
+#define WM8904_MICDET_ENA                       0x0002  /* MICDET_ENA */
+#define WM8904_MICDET_ENA_MASK                  0x0002  /* MICDET_ENA */
+#define WM8904_MICDET_ENA_SHIFT                      1  /* MICDET_ENA */
+#define WM8904_MICDET_ENA_WIDTH                      1  /* MICDET_ENA */
+#define WM8904_MICBIAS_ENA                      0x0001  /* MICBIAS_ENA */
+#define WM8904_MICBIAS_ENA_MASK                 0x0001  /* MICBIAS_ENA */
+#define WM8904_MICBIAS_ENA_SHIFT                     0  /* MICBIAS_ENA */
+#define WM8904_MICBIAS_ENA_WIDTH                     1  /* MICBIAS_ENA */
+
+/*
+ * R7 (0x07) - Mic Bias Control 1
+ */
+#define WM8904_MIC_DET_FILTER_ENA               0x8000  /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_DET_FILTER_ENA_MASK          0x8000  /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_DET_FILTER_ENA_SHIFT             15  /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_DET_FILTER_ENA_WIDTH              1  /* MIC_DET_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA             0x4000  /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA_MASK        0x4000  /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT           14  /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH            1  /* MIC_SHORT_FILTER_ENA */
+#define WM8904_MICBIAS_SEL_MASK                 0x0007  /* MICBIAS_SEL - [2:0] */
+#define WM8904_MICBIAS_SEL_SHIFT                     0  /* MICBIAS_SEL - [2:0] */
+#define WM8904_MICBIAS_SEL_WIDTH                     3  /* MICBIAS_SEL - [2:0] */
+
+
 /*
  * R121 (0x79) - GPIO Control 1
  */
@@ -82,6 +116,7 @@
 #define WM8904_GPIO_BCLK_SEL_SHIFT                   0  /* GPIO_BCLK_SEL - [3:0] */
 #define WM8904_GPIO_BCLK_SEL_WIDTH                   4  /* GPIO_BCLK_SEL - [3:0] */
 
+#define WM8904_MIC_REGS  2
 #define WM8904_GPIO_REGS 4
 #define WM8904_DRC_REGS  4
 #define WM8904_EQ_REGS   25
@@ -122,6 +157,7 @@ struct wm8904_pdata {
 	struct wm8904_retune_mobile_cfg *retune_mobile_cfgs;
 
 	u32 gpio_cfg[WM8904_GPIO_REGS];
+	u32 mic_cfg[WM8904_MIC_REGS];
 };
 
 #endif
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index ce3f004217ae..c5b50d73f2d5 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2540,6 +2540,11 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 			wm8904->reg_cache[WM8904_GPIO_CONTROL_1 + i]
 				= pdata->gpio_cfg[i] & 0xffff;
 		}
+
+		/* Zero is the default value for these anyway */
+		for (i = 0; i < WM8904_MIC_REGS; i++)
+			wm8904->reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i]
+				= pdata->mic_cfg[i];
 	}
 
 	/* Set Class W by default - this will be managed by the Class
diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h
index 10603a75c1a5..abe5059b3004 100644
--- a/sound/soc/codecs/wm8904.h
+++ b/sound/soc/codecs/wm8904.h
@@ -185,39 +185,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8904;
 #define WM8904_VMID_ENA_SHIFT                        0  /* VMID_ENA */
 #define WM8904_VMID_ENA_WIDTH                        1  /* VMID_ENA */
 
-/*
- * R6 (0x06) - Mic Bias Control 0
- */
-#define WM8904_MICDET_THR_MASK                  0x0070  /* MICDET_THR - [6:4] */
-#define WM8904_MICDET_THR_SHIFT                      4  /* MICDET_THR - [6:4] */
-#define WM8904_MICDET_THR_WIDTH                      3  /* MICDET_THR - [6:4] */
-#define WM8904_MICSHORT_THR_MASK                0x000C  /* MICSHORT_THR - [3:2] */
-#define WM8904_MICSHORT_THR_SHIFT                    2  /* MICSHORT_THR - [3:2] */
-#define WM8904_MICSHORT_THR_WIDTH                    2  /* MICSHORT_THR - [3:2] */
-#define WM8904_MICDET_ENA                       0x0002  /* MICDET_ENA */
-#define WM8904_MICDET_ENA_MASK                  0x0002  /* MICDET_ENA */
-#define WM8904_MICDET_ENA_SHIFT                      1  /* MICDET_ENA */
-#define WM8904_MICDET_ENA_WIDTH                      1  /* MICDET_ENA */
-#define WM8904_MICBIAS_ENA                      0x0001  /* MICBIAS_ENA */
-#define WM8904_MICBIAS_ENA_MASK                 0x0001  /* MICBIAS_ENA */
-#define WM8904_MICBIAS_ENA_SHIFT                     0  /* MICBIAS_ENA */
-#define WM8904_MICBIAS_ENA_WIDTH                     1  /* MICBIAS_ENA */
-
-/*
- * R7 (0x07) - Mic Bias Control 1
- */
-#define WM8904_MIC_DET_FILTER_ENA               0x8000  /* MIC_DET_FILTER_ENA */
-#define WM8904_MIC_DET_FILTER_ENA_MASK          0x8000  /* MIC_DET_FILTER_ENA */
-#define WM8904_MIC_DET_FILTER_ENA_SHIFT             15  /* MIC_DET_FILTER_ENA */
-#define WM8904_MIC_DET_FILTER_ENA_WIDTH              1  /* MIC_DET_FILTER_ENA */
-#define WM8904_MIC_SHORT_FILTER_ENA             0x4000  /* MIC_SHORT_FILTER_ENA */
-#define WM8904_MIC_SHORT_FILTER_ENA_MASK        0x4000  /* MIC_SHORT_FILTER_ENA */
-#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT           14  /* MIC_SHORT_FILTER_ENA */
-#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH            1  /* MIC_SHORT_FILTER_ENA */
-#define WM8904_MICBIAS_SEL_MASK                 0x0007  /* MICBIAS_SEL - [2:0] */
-#define WM8904_MICBIAS_SEL_SHIFT                     0  /* MICBIAS_SEL - [2:0] */
-#define WM8904_MICBIAS_SEL_WIDTH                     3  /* MICBIAS_SEL - [2:0] */
-
 /*
  * R8 (0x08) - Analogue DAC 0
  */
-- 
cgit v1.2.3


From 2a0761a35b70f170a9c14ecbcb6a959fb6e7bdf3 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 16 Mar 2010 15:54:12 +0000
Subject: ASoC: Implement WM835x microphone jack detection support

The WM8350 provides microphone presence and short circuit detection.
Integrate this with the ASoC jack reporting API.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/linux/mfd/wm8350/audio.h |  2 ++
 sound/soc/codecs/wm8350.c        | 58 ++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8350.h        |  3 +++
 3 files changed, 63 insertions(+)

(limited to 'sound/soc')

diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index d899dc0223ba..a95141eafce3 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -492,6 +492,8 @@
  */
 #define WM8350_JACK_L_LVL			0x0800
 #define WM8350_JACK_R_LVL                       0x0400
+#define WM8350_JACK_MICSCD_LVL			0x0200
+#define WM8350_JACK_MICSD_LVL			0x0100
 
 /*
  * WM8350 Platform setup
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index df2c6d9617fb..ff6c10155ee6 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -54,6 +54,7 @@ struct wm8350_output {
 struct wm8350_jack_data {
 	struct snd_soc_jack *jack;
 	int report;
+	int short_report;
 };
 
 struct wm8350_data {
@@ -62,6 +63,7 @@ struct wm8350_data {
 	struct wm8350_output out2;
 	struct wm8350_jack_data hpl;
 	struct wm8350_jack_data hpr;
+	struct wm8350_jack_data mic;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 	int fll_freq_out;
 	int fll_freq_in;
@@ -1430,6 +1432,55 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 }
 EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
 
+static irqreturn_t wm8350_mic_handler(int irq, void *data)
+{
+	struct wm8350_data *priv = data;
+	struct wm8350 *wm8350 = priv->codec.control_data;
+	u16 reg;
+	int report = 0;
+
+	reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
+	if (reg & WM8350_JACK_MICSCD_LVL)
+		report |= priv->mic.short_report;
+	if (reg & WM8350_JACK_MICSD_LVL)
+		report |= priv->mic.report;
+
+	snd_soc_jack_report(priv->mic.jack, report,
+			    priv->mic.report | priv->mic.short_report);
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * wm8350_mic_jack_detect - Enable microphone jack detection.
+ *
+ * @codec:         WM8350 codec
+ * @jack:          jack to report detection events on
+ * @detect_report: value to report when presence detected
+ * @short_report:  value to report when microphone short detected
+ *
+ * Enables the microphone jack detection of the WM8350.
+ */
+int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
+			   struct snd_soc_jack *jack,
+			   int detect_report, int short_report)
+{
+	struct wm8350_data *priv = codec->private_data;
+	struct wm8350 *wm8350 = codec->control_data;
+
+	priv->mic.jack = jack;
+	priv->mic.report = detect_report;
+	priv->mic.short_report = short_report;
+
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_1, WM8350_MIC_DET_ENA);
+
+	snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
+
 static struct snd_soc_codec *wm8350_codec;
 
 static int wm8350_probe(struct platform_device *pdev)
@@ -1493,6 +1544,10 @@ static int wm8350_probe(struct platform_device *pdev)
 	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
 			    wm8350_hp_jack_handler, 0, "Right jack detect",
 			    priv);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD,
+			    wm8350_mic_handler, 0, "Microphone short", priv);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD,
+			    wm8350_mic_handler, 0, "Microphone detect", priv);
 
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -1521,11 +1576,14 @@ static int wm8350_remove(struct platform_device *pdev)
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
 
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICD, priv);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv);
 	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv);
 	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv);
 
 	priv->hpl.jack = NULL;
 	priv->hpr.jack = NULL;
+	priv->mic.jack = NULL;
 
 	/* cancel any work waiting to be queued. */
 	ret = cancel_delayed_work(&codec->delayed_work);
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index d088eb4b88bb..9ed0467c71db 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -25,5 +25,8 @@ enum wm8350_jack {
 
 int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 			  struct snd_soc_jack *jack, int report);
+int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
+			   struct snd_soc_jack *jack,
+			   int detect_report, int short_report);
 
 #endif
-- 
cgit v1.2.3


From a6c65736bc2e63392334bd800f1b2754ab55db1e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Mar 2010 17:45:21 +0000
Subject: ASoC: Remove current PGA control handling

A code audit reveals that there are currently no users of the widget
controls on PGAs. This is likely to continue to be the case since
while there are useful things that can be done with integrating the
PGA gain and mute controls with the power sequencing userspace
generally wants stereo controls for output stages which this doesn't
map onto well.

In preparation for implementing something more useful strip out the
existing code, leaving the parameters there for use by the new code.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/soc-dapm.c | 93 ++--------------------------------------------------
 1 file changed, 3 insertions(+), 90 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index bbb2135a0b21..611284194000 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -321,54 +321,6 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
 	return change;
 }
 
-/* ramps the volume up or down to minimise pops before or after a
- * DAPM power event */
-static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power)
-{
-	const struct snd_kcontrol_new *k = widget->kcontrols;
-
-	if (widget->muted && !power)
-		return 0;
-	if (!widget->muted && power)
-		return 0;
-
-	if (widget->num_kcontrols && k) {
-		struct soc_mixer_control *mc =
-			(struct soc_mixer_control *)k->private_value;
-		unsigned int reg = mc->reg;
-		unsigned int shift = mc->shift;
-		int max = mc->max;
-		unsigned int mask = (1 << fls(max)) - 1;
-		unsigned int invert = mc->invert;
-
-		if (power) {
-			int i;
-			/* power up has happended, increase volume to last level */
-			if (invert) {
-				for (i = max; i > widget->saved_value; i--)
-					snd_soc_update_bits(widget->codec, reg, mask, i);
-			} else {
-				for (i = 0; i < widget->saved_value; i++)
-					snd_soc_update_bits(widget->codec, reg, mask, i);
-			}
-			widget->muted = 0;
-		} else {
-			/* power down is about to occur, decrease volume to mute */
-			int val = snd_soc_read(widget->codec, reg);
-			int i = widget->saved_value = (val >> shift) & mask;
-			if (invert) {
-				for (; i < mask; i++)
-					snd_soc_update_bits(widget->codec, reg, mask, i);
-			} else {
-				for (; i > 0; i--)
-					snd_soc_update_bits(widget->codec, reg, mask, i);
-			}
-			widget->muted = 1;
-		}
-	}
-	return 0;
-}
-
 /* create new dapm mixer control */
 static int dapm_new_mixer(struct snd_soc_codec *codec,
 	struct snd_soc_dapm_widget *w)
@@ -463,20 +415,10 @@ err:
 static int dapm_new_pga(struct snd_soc_codec *codec,
 	struct snd_soc_dapm_widget *w)
 {
-	struct snd_kcontrol *kcontrol;
-	int ret = 0;
-
-	if (!w->num_kcontrols)
-		return -EINVAL;
+	if (w->num_kcontrols)
+		pr_err("asoc: PGA controls not supported: '%s'\n", w->name);
 
-	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
-	ret = snd_ctl_add(codec->card, kcontrol);
-	if (ret < 0) {
-		printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
-		return ret;
-	}
-
-	return ret;
+	return 0;
 }
 
 /* reset 'walked' bit for each dapm path */
@@ -632,16 +574,8 @@ static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w)
 			return ret;
 	}
 
-	/* Lower PGA volume to reduce pops */
-	if (w->id == snd_soc_dapm_pga && !w->power)
-		dapm_set_pga(w, w->power);
-
 	dapm_update_bits(w);
 
-	/* Raise PGA volume to reduce pops */
-	if (w->id == snd_soc_dapm_pga && w->power)
-		dapm_set_pga(w, w->power);
-
 	/* power up post event */
 	if (w->power && w->event &&
 	    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
@@ -808,10 +742,6 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
 				pr_err("%s: pre event failed: %d\n",
 				       w->name, ret);
 		}
-
-		/* Lower PGA volume to reduce pops */
-		if (w->id == snd_soc_dapm_pga && !w->power)
-			dapm_set_pga(w, w->power);
 	}
 
 	if (reg >= 0) {
@@ -823,10 +753,6 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec,
 	}
 
 	list_for_each_entry(w, pending, power_list) {
-		/* Raise PGA volume to reduce pops */
-		if (w->id == snd_soc_dapm_pga && w->power)
-			dapm_set_pga(w, w->power);
-
 		/* power up post event */
 		if (w->power && w->event &&
 		    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
@@ -1596,12 +1522,6 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
 	unsigned int invert = mc->invert;
 	unsigned int mask = (1 << fls(max)) - 1;
 
-	/* return the saved value if we are powered down */
-	if (widget->id == snd_soc_dapm_pga && !widget->power) {
-		ucontrol->value.integer.value[0] = widget->saved_value;
-		return 0;
-	}
-
 	ucontrol->value.integer.value[0] =
 		(snd_soc_read(widget->codec, reg) >> shift) & mask;
 	if (shift != rshift)
@@ -1661,13 +1581,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	mutex_lock(&widget->codec->mutex);
 	widget->value = val;
 
-	/* save volume value if the widget is powered down */
-	if (widget->id == snd_soc_dapm_pga && !widget->power) {
-		widget->saved_value = val;
-		mutex_unlock(&widget->codec->mutex);
-		return 1;
-	}
-
 	if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) {
 		if (val)
 			/* new connection */
-- 
cgit v1.2.3


From dd76769dd53bdb804a4b02a3eb256f4740f78720 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 17 Mar 2010 11:51:33 +0000
Subject: ASoC: Refresh WM8750 bias management

The WM8750 is using some delayed work to manage the ramping of the bias
at startup and resume out of line from the normal flow.  This predates
the support within ASoC core for moving the resume out of line from the
main system resume which provides equivalent functionality with better
interaction with applications.  Change to doing the ramp in line to make
use of the core functionality.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8750.c | 50 +++++++++--------------------------------------
 1 file changed, 9 insertions(+), 41 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index c0c487501248..ee084083a49d 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -611,10 +611,16 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		/* set vmid to 5k for quick power up */
-		snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Set VMID to 5k */
+			snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
+
+			/* ...and ramp */
+			msleep(1000);
+		}
+
 		/* mute dac and set vmid to 500k, enable VREF */
 		snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
 		break;
@@ -658,13 +664,6 @@ struct snd_soc_dai wm8750_dai = {
 };
 EXPORT_SYMBOL_GPL(wm8750_dai);
 
-static void wm8750_work(struct work_struct *work)
-{
-	struct snd_soc_codec *codec =
-		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8750_set_bias_level(codec, codec->bias_level);
-}
-
 static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -693,14 +692,6 @@ static int wm8750_resume(struct platform_device *pdev)
 
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	/* charge wm8750 caps */
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
-		wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-		codec->bias_level = SND_SOC_BIAS_ON;
-		schedule_delayed_work(&codec->delayed_work,
-					msecs_to_jiffies(1000));
-	}
-
 	return 0;
 }
 
@@ -744,9 +735,7 @@ static int wm8750_init(struct snd_soc_device *socdev,
 	}
 
 	/* charge output caps */
-	wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-	codec->bias_level = SND_SOC_BIAS_STANDBY;
-	schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* set the update bits */
 	reg = snd_soc_read(codec, WM8750_LDAC);
@@ -926,7 +915,6 @@ static int wm8750_probe(struct platform_device *pdev)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 	wm8750_socdev = socdev;
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
 
 	ret = -ENODEV;
 
@@ -950,25 +938,6 @@ static int wm8750_probe(struct platform_device *pdev)
 	return ret;
 }
 
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
-{
-	int ret;
-
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(dwork);
-
-	/* if there was any work waiting then we run it now and
-	 * wait for it's completion */
-	if (ret) {
-		schedule_delayed_work(dwork, 0);
-		flush_scheduled_work();
-	}
-	return ret;
-}
-
 /* power down chip */
 static int wm8750_remove(struct platform_device *pdev)
 {
@@ -977,7 +946,6 @@ static int wm8750_remove(struct platform_device *pdev)
 
 	if (codec->control_data)
 		wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-- 
cgit v1.2.3


From 85dfcdffc227717fd218b9a86b507d9b7a3c321d Mon Sep 17 00:00:00 2001
From: Cliff Cai <cliff.cai@analog.com>
Date: Thu, 18 Mar 2010 16:17:00 +0800
Subject: ASoC: soc-cache: add i2c read entry for 8_8 mode

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-cache.c | 35 +++++++++++++++++++++++++++++++++++
 1 file changed, 35 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index bf593a834f5a..91db7afb1210 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -225,6 +225,40 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
 	}
 }
 
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
+					  unsigned int r)
+{
+	struct i2c_msg xfer[2];
+	u8 reg = r;
+	u8 data;
+	int ret;
+	struct i2c_client *client = codec->control_data;
+
+	/* Write register */
+	xfer[0].addr = client->addr;
+	xfer[0].flags = 0;
+	xfer[0].len = 1;
+	xfer[0].buf = &reg;
+
+	/* Read data */
+	xfer[1].addr = client->addr;
+	xfer[1].flags = I2C_M_RD;
+	xfer[1].len = 1;
+	xfer[1].buf = &data;
+
+	ret = i2c_transfer(client->adapter, xfer, 2);
+	if (ret != 2) {
+		dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+		return 0;
+	}
+
+	return data;
+}
+#else
+#define snd_soc_8_8_read_i2c NULL
+#endif
+
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
 					  unsigned int r)
@@ -466,6 +500,7 @@ static struct {
 	{
 		.addr_bits = 8, .data_bits = 8,
 		.write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
+		.i2c_read = snd_soc_8_8_read_i2c,
 	},
 	{
 		.addr_bits = 8, .data_bits = 16,
-- 
cgit v1.2.3


From f4bee1bb0083869b1b7d73c6eda7703a7aae0506 Mon Sep 17 00:00:00 2001
From: Barry Song <21cnbao@gmail.com>
Date: Thu, 18 Mar 2010 16:17:01 +0800
Subject: ASoC: soc-cache: let reg be AND'ed by 0xff instead of data buffer for
 8_8 mode

The registers for AD193X are defined as 0x800-0x810 for spi which uses
16_8 mode, for i2c to support AD1937, we will use 8_8 mode, only the low
byte of 0x800-0x810 is valid.  The patch will not destory other codecs,
but make soc cache interface more useful.

Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-cache.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 91db7afb1210..9dfe9a58a314 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -159,7 +159,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	BUG_ON(codec->volatile_register);
 
-	data[0] = reg & 0xff;
+	reg &= 0xff;
+	data[0] = reg;
 	data[1] = value & 0xff;
 
 	if (reg < codec->reg_cache_size)
@@ -180,6 +181,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
 {
 	u8 *cache = codec->reg_cache;
+	reg &= 0xff;
 	if (reg >= codec->reg_cache_size)
 		return -1;
 	return cache[reg];
-- 
cgit v1.2.3


From 9fc71e8f586efc99472337a8b089a4915877b4b8 Mon Sep 17 00:00:00 2001
From: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Date: Thu, 18 Mar 2010 19:45:41 -0600
Subject: ASoC: McPDM: Use tabs for indentation

Indentation in initial support for McPDM driver was converted to spaces.
Use tabs to comply with open source coding-style.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/mcpdm.c | 548 ++++++++++++++++++++++++-------------------------
 1 file changed, 274 insertions(+), 274 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c
index ad8df6cfae88..479d3b6306d7 100644
--- a/sound/soc/omap/mcpdm.c
+++ b/sound/soc/omap/mcpdm.c
@@ -1,5 +1,5 @@
 /*
- * mcpdm.c  --  McPDM interface driver
+ * mcpdm.c  --	McPDM interface driver
  *
  * Author: Jorge Eduardo Candelaria <x0107209@ti.com>
  * Copyright (C) 2009 - Texas Instruments, Inc.
@@ -38,46 +38,46 @@ static struct omap_mcpdm *mcpdm;
 
 static inline void omap_mcpdm_write(u16 reg, u32 val)
 {
-       __raw_writel(val, mcpdm->io_base + reg);
+	__raw_writel(val, mcpdm->io_base + reg);
 }
 
 static inline int omap_mcpdm_read(u16 reg)
 {
-       return __raw_readl(mcpdm->io_base + reg);
+	return __raw_readl(mcpdm->io_base + reg);
 }
 
 static void omap_mcpdm_reg_dump(void)
 {
-       dev_dbg(mcpdm->dev, "***********************\n");
-       dev_dbg(mcpdm->dev, "IRQSTATUS_RAW:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_IRQSTATUS_RAW));
-       dev_dbg(mcpdm->dev, "IRQSTATUS:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_IRQSTATUS));
-       dev_dbg(mcpdm->dev, "IRQENABLE_SET:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_IRQENABLE_SET));
-       dev_dbg(mcpdm->dev, "IRQENABLE_CLR:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_IRQENABLE_CLR));
-       dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
-                       omap_mcpdm_read(MCPDM_IRQWAKE_EN));
-       dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
-                       omap_mcpdm_read(MCPDM_DMAENABLE_SET));
-       dev_dbg(mcpdm->dev, "DMAENABLE_CLR:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_DMAENABLE_CLR));
-       dev_dbg(mcpdm->dev, "DMAWAKEEN:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_DMAWAKEEN));
-       dev_dbg(mcpdm->dev, "CTRL:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_CTRL));
-       dev_dbg(mcpdm->dev, "DN_DATA:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_DN_DATA));
-       dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
-                       omap_mcpdm_read(MCPDM_UP_DATA));
-       dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
-                       omap_mcpdm_read(MCPDM_FIFO_CTRL_DN));
-       dev_dbg(mcpdm->dev, "FIFO_CTRL_UP:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_FIFO_CTRL_UP));
-       dev_dbg(mcpdm->dev, "DN_OFFSET:  0x%04x\n",
-                       omap_mcpdm_read(MCPDM_DN_OFFSET));
-       dev_dbg(mcpdm->dev, "***********************\n");
+	dev_dbg(mcpdm->dev, "***********************\n");
+	dev_dbg(mcpdm->dev, "IRQSTATUS_RAW:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQSTATUS_RAW));
+	dev_dbg(mcpdm->dev, "IRQSTATUS:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQSTATUS));
+	dev_dbg(mcpdm->dev, "IRQENABLE_SET:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQENABLE_SET));
+	dev_dbg(mcpdm->dev, "IRQENABLE_CLR:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQENABLE_CLR));
+	dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_IRQWAKE_EN));
+	dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_DMAENABLE_SET));
+	dev_dbg(mcpdm->dev, "DMAENABLE_CLR:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_DMAENABLE_CLR));
+	dev_dbg(mcpdm->dev, "DMAWAKEEN:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_DMAWAKEEN));
+	dev_dbg(mcpdm->dev, "CTRL:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_CTRL));
+	dev_dbg(mcpdm->dev, "DN_DATA:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_DN_DATA));
+	dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_UP_DATA));
+	dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
+			omap_mcpdm_read(MCPDM_FIFO_CTRL_DN));
+	dev_dbg(mcpdm->dev, "FIFO_CTRL_UP:  0x%04x\n",
+			omap_mcpdm_read(MCPDM_FIFO_CTRL_UP));
+	dev_dbg(mcpdm->dev, "DN_OFFSET:	0x%04x\n",
+			omap_mcpdm_read(MCPDM_DN_OFFSET));
+	dev_dbg(mcpdm->dev, "***********************\n");
 }
 
 /*
@@ -86,26 +86,26 @@ static void omap_mcpdm_reg_dump(void)
  */
 static void omap_mcpdm_reset_capture(int reset)
 {
-       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
 
-       if (reset)
-               ctrl |= SW_UP_RST;
-       else
-               ctrl &= ~SW_UP_RST;
+	if (reset)
+		ctrl |= SW_UP_RST;
+	else
+		ctrl &= ~SW_UP_RST;
 
-       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
 }
 
 static void omap_mcpdm_reset_playback(int reset)
 {
-       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
 
-       if (reset)
-               ctrl |= SW_DN_RST;
-       else
-               ctrl &= ~SW_DN_RST;
+	if (reset)
+		ctrl |= SW_DN_RST;
+	else
+		ctrl &= ~SW_DN_RST;
 
-       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
 }
 
 /*
@@ -114,14 +114,14 @@ static void omap_mcpdm_reset_playback(int reset)
  */
 void omap_mcpdm_start(int stream)
 {
-       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
 
-       if (stream)
-               ctrl |= mcpdm->up_channels;
-       else
-               ctrl |= mcpdm->dn_channels;
+	if (stream)
+		ctrl |= mcpdm->up_channels;
+	else
+		ctrl |= mcpdm->dn_channels;
 
-       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
 }
 
 /*
@@ -130,14 +130,14 @@ void omap_mcpdm_start(int stream)
  */
 void omap_mcpdm_stop(int stream)
 {
-       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	int ctrl = omap_mcpdm_read(MCPDM_CTRL);
 
-       if (stream)
-               ctrl &= ~mcpdm->up_channels;
-       else
-               ctrl &= ~mcpdm->dn_channels;
+	if (stream)
+		ctrl &= ~mcpdm->up_channels;
+	else
+		ctrl &= ~mcpdm->dn_channels;
 
-       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
 }
 
 /*
@@ -146,38 +146,38 @@ void omap_mcpdm_stop(int stream)
  */
 int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink)
 {
-       int irq_mask = 0;
-       int ctrl;
+	int irq_mask = 0;
+	int ctrl;
 
-       if (!uplink)
-               return -EINVAL;
+	if (!uplink)
+		return -EINVAL;
 
-       mcpdm->uplink = uplink;
+	mcpdm->uplink = uplink;
 
-       /* Enable irq request generation */
-       irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
-       omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+	/* Enable irq request generation */
+	irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
 
-       /* Configure uplink threshold */
-       if (uplink->threshold > UP_THRES_MAX)
-               uplink->threshold = UP_THRES_MAX;
+	/* Configure uplink threshold */
+	if (uplink->threshold > UP_THRES_MAX)
+		uplink->threshold = UP_THRES_MAX;
 
-       omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold);
+	omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold);
 
-       /* Configure DMA controller */
-       omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE);
+	/* Configure DMA controller */
+	omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE);
 
-       /* Set pdm out format */
-       ctrl = omap_mcpdm_read(MCPDM_CTRL);
-       ctrl &= ~PDMOUTFORMAT;
-       ctrl |= uplink->format & PDMOUTFORMAT;
+	/* Set pdm out format */
+	ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	ctrl &= ~PDMOUTFORMAT;
+	ctrl |= uplink->format & PDMOUTFORMAT;
 
-       /* Uplink channels */
-       mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK);
+	/* Uplink channels */
+	mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK);
 
-       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
 
-       return 0;
+	return 0;
 }
 
 /*
@@ -186,38 +186,38 @@ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink)
  */
 int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink)
 {
-       int irq_mask = 0;
-       int ctrl;
+	int irq_mask = 0;
+	int ctrl;
 
-       if (!downlink)
-               return -EINVAL;
+	if (!downlink)
+		return -EINVAL;
 
-       mcpdm->downlink = downlink;
+	mcpdm->downlink = downlink;
 
-       /* Enable irq request generation */
-       irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
-       omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+	/* Enable irq request generation */
+	irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
 
-       /* Configure uplink threshold */
-       if (downlink->threshold > DN_THRES_MAX)
-               downlink->threshold = DN_THRES_MAX;
+	/* Configure uplink threshold */
+	if (downlink->threshold > DN_THRES_MAX)
+		downlink->threshold = DN_THRES_MAX;
 
-       omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold);
+	omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold);
 
-       /* Enable DMA request generation */
-       omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE);
+	/* Enable DMA request generation */
+	omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE);
 
-       /* Set pdm out format */
-       ctrl = omap_mcpdm_read(MCPDM_CTRL);
-       ctrl &= ~PDMOUTFORMAT;
-       ctrl |= downlink->format & PDMOUTFORMAT;
+	/* Set pdm out format */
+	ctrl = omap_mcpdm_read(MCPDM_CTRL);
+	ctrl &= ~PDMOUTFORMAT;
+	ctrl |= downlink->format & PDMOUTFORMAT;
 
-       /* Downlink channels */
-       mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK);
+	/* Downlink channels */
+	mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK);
 
-       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+	omap_mcpdm_write(MCPDM_CTRL, ctrl);
 
-       return 0;
+	return 0;
 }
 
 /*
@@ -226,24 +226,24 @@ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink)
  */
 int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink)
 {
-       int irq_mask = 0;
+	int irq_mask = 0;
 
-       if (!uplink)
-               return -EINVAL;
+	if (!uplink)
+		return -EINVAL;
 
-       /* Disable irq request generation */
-       irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
-       omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+	/* Disable irq request generation */
+	irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
 
-       /* Disable DMA request generation */
-       omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE);
+	/* Disable DMA request generation */
+	omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE);
 
-       /* Clear Downlink channels */
-       mcpdm->up_channels = 0;
+	/* Clear Downlink channels */
+	mcpdm->up_channels = 0;
 
-       mcpdm->uplink = NULL;
+	mcpdm->uplink = NULL;
 
-       return 0;
+	return 0;
 }
 
 /*
@@ -252,124 +252,124 @@ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink)
  */
 int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink)
 {
-       int irq_mask = 0;
+	int irq_mask = 0;
 
-       if (!downlink)
-               return -EINVAL;
+	if (!downlink)
+		return -EINVAL;
 
-       /* Disable irq request generation */
-       irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
-       omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+	/* Disable irq request generation */
+	irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+	omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
 
-       /* Disable DMA request generation */
-       omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE);
+	/* Disable DMA request generation */
+	omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE);
 
-       /* clear Downlink channels */
-       mcpdm->dn_channels = 0;
+	/* clear Downlink channels */
+	mcpdm->dn_channels = 0;
 
-       mcpdm->downlink = NULL;
+	mcpdm->downlink = NULL;
 
-       return 0;
+	return 0;
 }
 
 static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id)
 {
-       struct omap_mcpdm *mcpdm_irq = dev_id;
-       int irq_status;
-
-       irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS);
-
-       /* Acknowledge irq event */
-       omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status);
-
-       if (irq & MCPDM_DN_IRQ_FULL) {
-               dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
-               omap_mcpdm_reset_playback(1);
-               omap_mcpdm_playback_open(mcpdm_irq->downlink);
-               omap_mcpdm_reset_playback(0);
-       }
-
-       if (irq & MCPDM_DN_IRQ_EMPTY) {
-               dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
-               omap_mcpdm_reset_playback(1);
-               omap_mcpdm_playback_open(mcpdm_irq->downlink);
-               omap_mcpdm_reset_playback(0);
-       }
-
-       if (irq & MCPDM_DN_IRQ) {
-               dev_dbg(mcpdm_irq->dev, "DN write request\n");
-       }
-
-       if (irq & MCPDM_UP_IRQ_FULL) {
-               dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
-               omap_mcpdm_reset_capture(1);
-               omap_mcpdm_capture_open(mcpdm_irq->uplink);
-               omap_mcpdm_reset_capture(0);
-       }
-
-       if (irq & MCPDM_UP_IRQ_EMPTY) {
-               dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
-               omap_mcpdm_reset_capture(1);
-               omap_mcpdm_capture_open(mcpdm_irq->uplink);
-               omap_mcpdm_reset_capture(0);
-       }
-
-       if (irq & MCPDM_UP_IRQ) {
-               dev_dbg(mcpdm_irq->dev, "UP write request\n");
-       }
-
-       return IRQ_HANDLED;
+	struct omap_mcpdm *mcpdm_irq = dev_id;
+	int irq_status;
+
+	irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS);
+
+	/* Acknowledge irq event */
+	omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status);
+
+	if (irq & MCPDM_DN_IRQ_FULL) {
+		dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_playback(1);
+		omap_mcpdm_playback_open(mcpdm_irq->downlink);
+		omap_mcpdm_reset_playback(0);
+	}
+
+	if (irq & MCPDM_DN_IRQ_EMPTY) {
+		dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_playback(1);
+		omap_mcpdm_playback_open(mcpdm_irq->downlink);
+		omap_mcpdm_reset_playback(0);
+	}
+
+	if (irq & MCPDM_DN_IRQ) {
+		dev_dbg(mcpdm_irq->dev, "DN write request\n");
+	}
+
+	if (irq & MCPDM_UP_IRQ_FULL) {
+		dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_capture(1);
+		omap_mcpdm_capture_open(mcpdm_irq->uplink);
+		omap_mcpdm_reset_capture(0);
+	}
+
+	if (irq & MCPDM_UP_IRQ_EMPTY) {
+		dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+		omap_mcpdm_reset_capture(1);
+		omap_mcpdm_capture_open(mcpdm_irq->uplink);
+		omap_mcpdm_reset_capture(0);
+	}
+
+	if (irq & MCPDM_UP_IRQ) {
+		dev_dbg(mcpdm_irq->dev, "UP write request\n");
+	}
+
+	return IRQ_HANDLED;
 }
 
 int omap_mcpdm_request(void)
 {
-       int ret;
+	int ret;
 
-       clk_enable(mcpdm->clk);
+	clk_enable(mcpdm->clk);
 
-       spin_lock(&mcpdm->lock);
+	spin_lock(&mcpdm->lock);
 
-       if (!mcpdm->free) {
-               dev_err(mcpdm->dev, "McPDM interface is in use\n");
-               spin_unlock(&mcpdm->lock);
-               ret = -EBUSY;
-               goto err;
-       }
-       mcpdm->free = 0;
+	if (!mcpdm->free) {
+		dev_err(mcpdm->dev, "McPDM interface is in use\n");
+		spin_unlock(&mcpdm->lock);
+		ret = -EBUSY;
+		goto err;
+	}
+	mcpdm->free = 0;
 
-       spin_unlock(&mcpdm->lock);
+	spin_unlock(&mcpdm->lock);
 
-       /* Disable lines while request is ongoing */
-       omap_mcpdm_write(MCPDM_CTRL, 0x00);
+	/* Disable lines while request is ongoing */
+	omap_mcpdm_write(MCPDM_CTRL, 0x00);
 
-       ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
-                               0, "McPDM", (void *)mcpdm);
-       if (ret) {
-               dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n");
-               goto err;
-       }
+	ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+				0, "McPDM", (void *)mcpdm);
+	if (ret) {
+		dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n");
+		goto err;
+	}
 
-       return 0;
+	return 0;
 
 err:
-       clk_disable(mcpdm->clk);
-       return ret;
+	clk_disable(mcpdm->clk);
+	return ret;
 }
 
 void omap_mcpdm_free(void)
 {
-       spin_lock(&mcpdm->lock);
-       if (mcpdm->free) {
-               dev_err(mcpdm->dev, "McPDM interface is already free\n");
-               spin_unlock(&mcpdm->lock);
-               return;
-       }
-       mcpdm->free = 1;
-       spin_unlock(&mcpdm->lock);
-
-       clk_disable(mcpdm->clk);
-
-       free_irq(mcpdm->irq, (void *)mcpdm);
+	spin_lock(&mcpdm->lock);
+	if (mcpdm->free) {
+		dev_err(mcpdm->dev, "McPDM interface is already free\n");
+		spin_unlock(&mcpdm->lock);
+		return;
+	}
+	mcpdm->free = 1;
+	spin_unlock(&mcpdm->lock);
+
+	clk_disable(mcpdm->clk);
+
+	free_irq(mcpdm->irq, (void *)mcpdm);
 }
 
 /* Enable/disable DC offset cancelation for the analog
@@ -377,108 +377,108 @@ void omap_mcpdm_free(void)
  */
 int omap_mcpdm_set_offset(int offset1, int offset2)
 {
-       int offset;
+	int offset;
 
-       if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX))
-               return -EINVAL;
+	if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX))
+		return -EINVAL;
 
-       offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2);
+	offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2);
 
-       /* offset cancellation for channel 1 */
-       if (offset1)
-               offset |= DN_OFST_RX1_EN;
-       else
-               offset &= ~DN_OFST_RX1_EN;
+	/* offset cancellation for channel 1 */
+	if (offset1)
+		offset |= DN_OFST_RX1_EN;
+	else
+		offset &= ~DN_OFST_RX1_EN;
 
-       /* offset cancellation for channel 2 */
-       if (offset2)
-               offset |= DN_OFST_RX2_EN;
-       else
-               offset &= ~DN_OFST_RX2_EN;
+	/* offset cancellation for channel 2 */
+	if (offset2)
+		offset |= DN_OFST_RX2_EN;
+	else
+		offset &= ~DN_OFST_RX2_EN;
 
-       omap_mcpdm_write(MCPDM_DN_OFFSET, offset);
+	omap_mcpdm_write(MCPDM_DN_OFFSET, offset);
 
-       return 0;
+	return 0;
 }
 
 static int __devinit omap_mcpdm_probe(struct platform_device *pdev)
 {
-       struct resource *res;
-       int ret = 0;
-
-       mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
-       if (!mcpdm) {
-               ret = -ENOMEM;
-               goto exit;
-       }
-
-       res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-       if (res == NULL) {
-               dev_err(&pdev->dev, "no resource\n");
-               goto err_resource;
-       }
-
-       spin_lock_init(&mcpdm->lock);
-       mcpdm->free = 1;
-       mcpdm->io_base = ioremap(res->start, resource_size(res));
-       if (!mcpdm->io_base) {
-               ret = -ENOMEM;
-               goto err_resource;
-       }
-
-       mcpdm->irq = platform_get_irq(pdev, 0);
-
-       mcpdm->clk = clk_get(&pdev->dev, "pdm_ck");
-       if (IS_ERR(mcpdm->clk)) {
-               ret = PTR_ERR(mcpdm->clk);
-               dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret);
-               goto err_clk;
-       }
-
-       mcpdm->dev = &pdev->dev;
-       platform_set_drvdata(pdev, mcpdm);
-
-       return 0;
+	struct resource *res;
+	int ret = 0;
+
+	mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
+	if (!mcpdm) {
+		ret = -ENOMEM;
+		goto exit;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "no resource\n");
+		goto err_resource;
+	}
+
+	spin_lock_init(&mcpdm->lock);
+	mcpdm->free = 1;
+	mcpdm->io_base = ioremap(res->start, resource_size(res));
+	if (!mcpdm->io_base) {
+		ret = -ENOMEM;
+		goto err_resource;
+	}
+
+	mcpdm->irq = platform_get_irq(pdev, 0);
+
+	mcpdm->clk = clk_get(&pdev->dev, "pdm_ck");
+	if (IS_ERR(mcpdm->clk)) {
+		ret = PTR_ERR(mcpdm->clk);
+		dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret);
+		goto err_clk;
+	}
+
+	mcpdm->dev = &pdev->dev;
+	platform_set_drvdata(pdev, mcpdm);
+
+	return 0;
 
 err_clk:
-       iounmap(mcpdm->io_base);
+	iounmap(mcpdm->io_base);
 err_resource:
-       kfree(mcpdm);
+	kfree(mcpdm);
 exit:
-       return ret;
+	return ret;
 }
 
 static int __devexit omap_mcpdm_remove(struct platform_device *pdev)
 {
-       struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev);
+	struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev);
 
-       platform_set_drvdata(pdev, NULL);
+	platform_set_drvdata(pdev, NULL);
 
-       clk_put(mcpdm_ptr->clk);
+	clk_put(mcpdm_ptr->clk);
 
-       iounmap(mcpdm_ptr->io_base);
+	iounmap(mcpdm_ptr->io_base);
 
-       mcpdm_ptr->clk = NULL;
-       mcpdm_ptr->free = 0;
-       mcpdm_ptr->dev = NULL;
+	mcpdm_ptr->clk = NULL;
+	mcpdm_ptr->free = 0;
+	mcpdm_ptr->dev = NULL;
 
-       kfree(mcpdm_ptr);
+	kfree(mcpdm_ptr);
 
-       return 0;
+	return 0;
 }
 
 static struct platform_driver omap_mcpdm_driver = {
-       .probe = omap_mcpdm_probe,
-       .remove = __devexit_p(omap_mcpdm_remove),
-       .driver = {
-               .name = "omap-mcpdm",
-       },
+	.probe = omap_mcpdm_probe,
+	.remove = __devexit_p(omap_mcpdm_remove),
+	.driver = {
+		.name = "omap-mcpdm",
+	},
 };
 
 static struct platform_device *omap_mcpdm_device;
 
 static int __init omap_mcpdm_init(void)
 {
-       return platform_driver_register(&omap_mcpdm_driver);
+	return platform_driver_register(&omap_mcpdm_driver);
 }
 arch_initcall(omap_mcpdm_init);
-- 
cgit v1.2.3


From 299a151f53ed9272af616076305d3ee762c2eba0 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 19 Mar 2010 12:27:31 +0200
Subject: ASoC: omap-mcbsp: Add support for Left Justified format

Basic support for Left Justified coding for OMAP McBSP.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap-mcbsp.c | 12 +++++++++++-
 1 file changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 2952fb046ed6..0199e95a5d63 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -331,7 +331,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 
 	format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
 	wpf = channels = params_channels(params);
-	if (channels == 2 && format == SND_SOC_DAIFMT_I2S) {
+	if (channels == 2 && (format == SND_SOC_DAIFMT_I2S ||
+			      format == SND_SOC_DAIFMT_LEFT_J)) {
 		/* Use dual-phase frames */
 		regs->rcr2	|= RPHASE;
 		regs->xcr2	|= XPHASE;
@@ -376,6 +377,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	/* Set FS period and length in terms of bit clock periods */
 	switch (format) {
 	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
 		regs->srgr2	|= FPER(framesize - 1);
 		regs->srgr1	|= FWID((framesize >> 1) - 1);
 		break;
@@ -427,6 +429,14 @@ 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_LEFT_J:
+		/* 0-bit data delay */
+		regs->rcr2	|= RDATDLY(0);
+		regs->xcr2	|= XDATDLY(0);
+		regs->spcr1	|= RJUST(2);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* 1-bit data delay */
 		regs->rcr2      |= RDATDLY(1);
-- 
cgit v1.2.3


From 27648b2f1c464f8827cc01ba4eb21580b5402947 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 16 Mar 2010 16:09:47 +0000
Subject: ASoC: Correct typoed Mic2 connections on 1133-EV1 board

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/imx/wm1133-ev1.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
index b75fcde85e88..0af987e94223 100644
--- a/sound/soc/imx/wm1133-ev1.c
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -181,8 +181,8 @@ static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
 
 	/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
 	{ "IN1RN", NULL, "Mic Bias" },
-	{ "IN1RP", NULL, "Mic1 Jack" },
-	{ "Mic Bias", NULL, "Mic1 Jack" },
+	{ "IN1RP", NULL, "Mic2 Jack" },
+	{ "Mic Bias", NULL, "Mic2 Jack" },
 
 	/* Line in Jack --> AUX (L+R) */
 	{ "IN3R", NULL, "Line In Jack" },
-- 
cgit v1.2.3


From 8ecbabd977dc64d2d8c9980420a3d72db06fe551 Mon Sep 17 00:00:00 2001
From: Misael Lopez Cruz <x0052729@ti.com>
Date: Fri, 19 Mar 2010 11:25:51 +0000
Subject: ASoC: TWL6040: Add twl6040 codec driver

Initial version of TWL6040 codec driver.

The TWL6040 codec uses a proprietary PDM-based digital audio interface.
Audio paths supported are:

- Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right
- Output: Headset Left/Right, Handsfree Left/Right

TWL6040 codec supports power-up/down manual and automatic sequence.
Manual sequence is done through a specific register writes sequence.
Automatic sequence is done when the codec is powered-up through the
external AUDPWRON line. The completion of the sequence is signaled
through the audio interrupt.

TWL6040 codec sysclk can be provided by: low-power or high
performance PLL:

- The low-power PLL takes a low-frequency input at 32,768 Hz and
generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz
respectively)

- The high-performance PLL generates an exact 19.2 MHz clock signal
from high-frequency input at 12/19.2/26/38.4 MHz.

Low-power playback mode is a special scenario where only headset path
(headset DAC and driver) is active.

For the particular case of headset path, PLL being used defines the
headset power mode: low-power, high-performance.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig   |    4 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/twl6040.c | 1227 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/twl6040.h |  141 +++++
 4 files changed, 1374 insertions(+)
 create mode 100755 sound/soc/codecs/twl6040.c
 create mode 100644 sound/soc/codecs/twl6040.h

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 16c47edb8266..398cbb0e79fd 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -35,6 +35,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
+	select SND_SOC_TWL6040 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WM2000 if I2C
@@ -168,6 +169,9 @@ config SND_SOC_TWL4030
 	select TWL4030_CODEC
 	tristate
 
+config SND_SOC_TWL6040
+	tristate
+
 config SND_SOC_UDA134X
        tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 69817778b2bd..98bd10c8fd3a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -22,6 +22,7 @@ snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
 snd-soc-twl4030-objs := twl4030.o
+snd-soc-twl6040-objs := twl6040.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wm8350-objs := wm8350.o
@@ -85,6 +86,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
new file mode 100755
index 000000000000..0e4dce7de052
--- /dev/null
+++ b/sound/soc/codecs/twl6040.c
@@ -0,0 +1,1227 @@
+/*
+ * ALSA SoC TWL6040 codec driver
+ *
+ * Author:	 Misael Lopez Cruz <x0052729@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/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/platform_device.h>
+#include <linux/i2c/twl.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/initval.h>
+#include <sound/tlv.h>
+
+#include "twl6040.h"
+
+#define TWL6040_RATES	 (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+#define TWL6040_FORMATS	 (SNDRV_PCM_FMTBIT_S32_LE)
+
+/* codec private data */
+struct twl6040_data {
+	struct snd_soc_codec codec;
+	int audpwron;
+	int naudint;
+	int codec_powered;
+	int pll;
+	int non_lp;
+	unsigned int sysclk;
+	struct snd_pcm_hw_constraint_list *sysclk_constraints;
+	struct completion ready;
+};
+
+/*
+ * twl6040 register cache & default register settings
+ */
+static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
+	0x00, /* not used		0x00	*/
+	0x4B, /* TWL6040_ASICID (ro)	0x01	*/
+	0x00, /* TWL6040_ASICREV (ro)	0x02	*/
+	0x00, /* TWL6040_INTID		0x03	*/
+	0x00, /* TWL6040_INTMR		0x04	*/
+	0x00, /* TWL6040_NCPCTRL	0x05	*/
+	0x00, /* TWL6040_LDOCTL		0x06	*/
+	0x60, /* TWL6040_HPPLLCTL	0x07	*/
+	0x00, /* TWL6040_LPPLLCTL	0x08	*/
+	0x4A, /* TWL6040_LPPLLDIV	0x09	*/
+	0x00, /* TWL6040_AMICBCTL	0x0A	*/
+	0x00, /* TWL6040_DMICBCTL	0x0B	*/
+	0x18, /* TWL6040_MICLCTL	0x0C	- No input selected on Left Mic */
+	0x18, /* TWL6040_MICRCTL	0x0D	- No input selected on Right Mic */
+	0x00, /* TWL6040_MICGAIN	0x0E	*/
+	0x1B, /* TWL6040_LINEGAIN	0x0F	*/
+	0x00, /* TWL6040_HSLCTL		0x10	*/
+	0x00, /* TWL6040_HSRCTL		0x11	*/
+	0x00, /* TWL6040_HSGAIN		0x12	*/
+	0x00, /* TWL6040_EARCTL		0x13	*/
+	0x00, /* TWL6040_HFLCTL		0x14	*/
+	0x00, /* TWL6040_HFLGAIN	0x15	*/
+	0x00, /* TWL6040_HFRCTL		0x16	*/
+	0x00, /* TWL6040_HFRGAIN	0x17	*/
+	0x00, /* TWL6040_VIBCTLL	0x18	*/
+	0x00, /* TWL6040_VIBDATL	0x19	*/
+	0x00, /* TWL6040_VIBCTLR	0x1A	*/
+	0x00, /* TWL6040_VIBDATR	0x1B	*/
+	0x00, /* TWL6040_HKCTL1		0x1C	*/
+	0x00, /* TWL6040_HKCTL2		0x1D	*/
+	0x00, /* TWL6040_GPOCTL		0x1E	*/
+	0x00, /* TWL6040_ALB		0x1F	*/
+	0x00, /* TWL6040_DLB		0x20	*/
+	0x00, /* not used		0x21	*/
+	0x00, /* not used		0x22	*/
+	0x00, /* not used		0x23	*/
+	0x00, /* not used		0x24	*/
+	0x00, /* not used		0x25	*/
+	0x00, /* not used		0x26	*/
+	0x00, /* not used		0x27	*/
+	0x00, /* TWL6040_TRIM1		0x28	*/
+	0x00, /* TWL6040_TRIM2		0x29	*/
+	0x00, /* TWL6040_TRIM3		0x2A	*/
+	0x00, /* TWL6040_HSOTRIM	0x2B	*/
+	0x00, /* TWL6040_HFOTRIM	0x2C	*/
+	0x09, /* TWL6040_ACCCTL		0x2D	*/
+	0x00, /* TWL6040_STATUS (ro)	0x2E	*/
+};
+
+/*
+ * twl6040 vio/gnd registers:
+ * registers under vio/gnd supply can be accessed
+ * before the power-up sequence, after NRESPWRON goes high
+ */
+static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = {
+	TWL6040_REG_ASICID,
+	TWL6040_REG_ASICREV,
+	TWL6040_REG_INTID,
+	TWL6040_REG_INTMR,
+	TWL6040_REG_NCPCTL,
+	TWL6040_REG_LDOCTL,
+	TWL6040_REG_AMICBCTL,
+	TWL6040_REG_DMICBCTL,
+	TWL6040_REG_HKCTL1,
+	TWL6040_REG_HKCTL2,
+	TWL6040_REG_GPOCTL,
+	TWL6040_REG_TRIM1,
+	TWL6040_REG_TRIM2,
+	TWL6040_REG_TRIM3,
+	TWL6040_REG_HSOTRIM,
+	TWL6040_REG_HFOTRIM,
+	TWL6040_REG_ACCCTL,
+	TWL6040_REG_STATUS,
+};
+
+/*
+ * twl6040 vdd/vss registers:
+ * registers under vdd/vss supplies can only be accessed
+ * after the power-up sequence
+ */
+static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = {
+	TWL6040_REG_HPPLLCTL,
+	TWL6040_REG_LPPLLCTL,
+	TWL6040_REG_LPPLLDIV,
+	TWL6040_REG_MICLCTL,
+	TWL6040_REG_MICRCTL,
+	TWL6040_REG_MICGAIN,
+	TWL6040_REG_LINEGAIN,
+	TWL6040_REG_HSLCTL,
+	TWL6040_REG_HSRCTL,
+	TWL6040_REG_HSGAIN,
+	TWL6040_REG_EARCTL,
+	TWL6040_REG_HFLCTL,
+	TWL6040_REG_HFLGAIN,
+	TWL6040_REG_HFRCTL,
+	TWL6040_REG_HFRGAIN,
+	TWL6040_REG_VIBCTLL,
+	TWL6040_REG_VIBDATL,
+	TWL6040_REG_VIBCTLR,
+	TWL6040_REG_VIBDATR,
+	TWL6040_REG_ALB,
+	TWL6040_REG_DLB,
+};
+
+/*
+ * read twl6040 register cache
+ */
+static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec,
+						unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= TWL6040_CACHEREGNUM)
+		return -EIO;
+
+	return cache[reg];
+}
+
+/*
+ * write twl6040 register cache
+ */
+static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec,
+						u8 reg, u8 value)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= TWL6040_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * read from twl6040 hardware register
+ */
+static int twl6040_read_reg_volatile(struct snd_soc_codec *codec,
+			unsigned int reg)
+{
+	u8 value;
+
+	if (reg >= TWL6040_CACHEREGNUM)
+		return -EIO;
+
+	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg);
+	twl6040_write_reg_cache(codec, reg, value);
+
+	return value;
+}
+
+/*
+ * write to the twl6040 register space
+ */
+static int twl6040_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value)
+{
+	if (reg >= TWL6040_CACHEREGNUM)
+		return -EIO;
+
+	twl6040_write_reg_cache(codec, reg, value);
+	return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+}
+
+static void twl6040_init_vio_regs(struct snd_soc_codec *codec)
+{
+	u8 *cache = codec->reg_cache;
+	int reg, i;
+
+	/* allow registers to be accessed by i2c */
+	twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]);
+
+	for (i = 0; i < TWL6040_VIOREGNUM; i++) {
+		reg = twl6040_vio_reg[i];
+		/* skip read-only registers (ASICID, ASICREV, STATUS) */
+		switch (reg) {
+		case TWL6040_REG_ASICID:
+		case TWL6040_REG_ASICREV:
+		case TWL6040_REG_STATUS:
+			continue;
+		default:
+			break;
+		}
+		twl6040_write(codec, reg, cache[reg]);
+	}
+}
+
+static void twl6040_init_vdd_regs(struct snd_soc_codec *codec)
+{
+	u8 *cache = codec->reg_cache;
+	int reg, i;
+
+	for (i = 0; i < TWL6040_VDDREGNUM; i++) {
+		reg = twl6040_vdd_reg[i];
+		twl6040_write(codec, reg, cache[reg]);
+	}
+}
+
+/* twl6040 codec manual power-up sequence */
+static void twl6040_power_up(struct snd_soc_codec *codec)
+{
+	u8 ncpctl, ldoctl, lppllctl, accctl;
+
+	ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL);
+	ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL);
+	lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
+	accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL);
+
+	/* enable reference system */
+	ldoctl |= TWL6040_REFENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	msleep(10);
+	/* enable internal oscillator */
+	ldoctl |= TWL6040_OSCENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	udelay(10);
+	/* enable high-side ldo */
+	ldoctl |= TWL6040_HSLDOENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	udelay(244);
+	/* enable negative charge pump */
+	ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN;
+	twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl);
+	udelay(488);
+	/* enable low-side ldo */
+	ldoctl |= TWL6040_LSLDOENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	udelay(244);
+	/* enable low-power pll */
+	lppllctl |= TWL6040_LPLLENA;
+	twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+	/* reset state machine */
+	accctl |= TWL6040_RESETSPLIT;
+	twl6040_write(codec, TWL6040_REG_ACCCTL, accctl);
+	mdelay(5);
+	accctl &= ~TWL6040_RESETSPLIT;
+	twl6040_write(codec, TWL6040_REG_ACCCTL, accctl);
+	/* disable internal oscillator */
+	ldoctl &= ~TWL6040_OSCENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+}
+
+/* twl6040 codec manual power-down sequence */
+static void twl6040_power_down(struct snd_soc_codec *codec)
+{
+	u8 ncpctl, ldoctl, lppllctl, accctl;
+
+	ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL);
+	ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL);
+	lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
+	accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL);
+
+	/* enable internal oscillator */
+	ldoctl |= TWL6040_OSCENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	udelay(10);
+	/* disable low-power pll */
+	lppllctl &= ~TWL6040_LPLLENA;
+	twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+	/* disable low-side ldo */
+	ldoctl &= ~TWL6040_LSLDOENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	udelay(244);
+	/* disable negative charge pump */
+	ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN);
+	twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl);
+	udelay(488);
+	/* disable high-side ldo */
+	ldoctl &= ~TWL6040_HSLDOENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	udelay(244);
+	/* disable internal oscillator */
+	ldoctl &= ~TWL6040_OSCENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	/* disable reference system */
+	ldoctl &= ~TWL6040_REFENA;
+	twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl);
+	msleep(10);
+}
+
+/* set headset dac and driver power mode */
+static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
+{
+	int hslctl, hsrctl;
+	int mask = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL;
+
+	hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL);
+	hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL);
+
+	if (high_perf) {
+		hslctl &= ~mask;
+		hsrctl &= ~mask;
+	} else {
+		hslctl |= mask;
+		hsrctl |= mask;
+	}
+
+	twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl);
+	twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl);
+
+	return 0;
+}
+
+static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct twl6040_data *priv = codec->private_data;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		priv->non_lp++;
+	else
+		priv->non_lp--;
+
+	return 0;
+}
+
+/* audio interrupt handler */
+static irqreturn_t twl6040_naudint_handler(int irq, void *data)
+{
+	struct snd_soc_codec *codec = data;
+	struct twl6040_data *priv = codec->private_data;
+	u8 intid;
+
+	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID);
+
+	switch (intid) {
+	case TWL6040_THINT:
+		dev_alert(codec->dev, "die temp over-limit detection\n");
+		break;
+	case TWL6040_PLUGINT:
+	case TWL6040_UNPLUGINT:
+	case TWL6040_HOOKINT:
+		break;
+	case TWL6040_HFINT:
+		dev_alert(codec->dev, "hf drivers over current detection\n");
+		break;
+	case TWL6040_VIBINT:
+		dev_alert(codec->dev, "vib drivers over current detection\n");
+		break;
+	case TWL6040_READYINT:
+		complete(&priv->ready);
+		break;
+	default:
+		dev_err(codec->dev, "unknown audio interrupt %d\n", intid);
+		break;
+	}
+
+	return IRQ_HANDLED;
+}
+
+/*
+ * MICATT volume control:
+ * from -6 to 0 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0);
+
+/*
+ * MICGAIN volume control:
+ * from 6 to 30 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0);
+
+/*
+ * HSGAIN volume control:
+ * from -30 to 0 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0);
+
+/*
+ * HFGAIN volume control:
+ * from -52 to 6 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0);
+
+/* Left analog microphone selection */
+static const char *twl6040_amicl_texts[] =
+	{"Headset Mic", "Main Mic", "Aux/FM Left", "Off"};
+
+/* Right analog microphone selection */
+static const char *twl6040_amicr_texts[] =
+	{"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"};
+
+static const struct soc_enum twl6040_enum[] = {
+	SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts),
+	SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts),
+};
+
+static const struct snd_kcontrol_new amicl_control =
+	SOC_DAPM_ENUM("Route", twl6040_enum[0]);
+
+static const struct snd_kcontrol_new amicr_control =
+	SOC_DAPM_ENUM("Route", twl6040_enum[1]);
+
+/* Headset DAC playback switches */
+static const struct snd_kcontrol_new hsdacl_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0);
+
+static const struct snd_kcontrol_new hsdacr_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0);
+
+/* Handsfree DAC playback switches */
+static const struct snd_kcontrol_new hfdacl_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0);
+
+static const struct snd_kcontrol_new hfdacr_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0);
+
+/* Headset driver switches */
+static const struct snd_kcontrol_new hsl_driver_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0);
+
+static const struct snd_kcontrol_new hsr_driver_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0);
+
+/* Handsfree driver switches */
+static const struct snd_kcontrol_new hfl_driver_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0);
+
+static const struct snd_kcontrol_new hfr_driver_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0);
+
+static const struct snd_kcontrol_new twl6040_snd_controls[] = {
+	/* Capture gains */
+	SOC_DOUBLE_TLV("Capture Preamplifier Volume",
+		TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv),
+	SOC_DOUBLE_TLV("Capture Volume",
+		TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv),
+
+	/* Playback gains */
+	SOC_DOUBLE_TLV("Headset Playback Volume",
+		TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
+	SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+		TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv),
+
+};
+
+static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MAINMIC"),
+	SND_SOC_DAPM_INPUT("HSMIC"),
+	SND_SOC_DAPM_INPUT("SUBMIC"),
+	SND_SOC_DAPM_INPUT("AFML"),
+	SND_SOC_DAPM_INPUT("AFMR"),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HSOL"),
+	SND_SOC_DAPM_OUTPUT("HSOR"),
+	SND_SOC_DAPM_OUTPUT("HFL"),
+	SND_SOC_DAPM_OUTPUT("HFR"),
+
+	/* Analog input muxes for the capture amplifiers */
+	SND_SOC_DAPM_MUX("Analog Left Capture Route",
+			SND_SOC_NOPM, 0, 0, &amicl_control),
+	SND_SOC_DAPM_MUX("Analog Right Capture Route",
+			SND_SOC_NOPM, 0, 0, &amicr_control),
+
+	/* Analog capture PGAs */
+	SND_SOC_DAPM_PGA("MicAmpL",
+			TWL6040_REG_MICLCTL, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MicAmpR",
+			TWL6040_REG_MICRCTL, 0, 0, NULL, 0),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture",
+			TWL6040_REG_MICLCTL, 2, 0),
+	SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture",
+			TWL6040_REG_MICRCTL, 2, 0),
+
+	/* Microphone bias */
+	SND_SOC_DAPM_MICBIAS("Headset Mic Bias",
+			TWL6040_REG_AMICBCTL, 0, 0),
+	SND_SOC_DAPM_MICBIAS("Main Mic Bias",
+			TWL6040_REG_AMICBCTL, 4, 0),
+	SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias",
+			TWL6040_REG_DMICBCTL, 0, 0),
+	SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias",
+			TWL6040_REG_DMICBCTL, 4, 0),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback",
+			TWL6040_REG_HSLCTL, 0, 0),
+	SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback",
+			TWL6040_REG_HSRCTL, 0, 0),
+	SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback",
+			TWL6040_REG_HFLCTL, 0, 0,
+			twl6040_power_mode_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback",
+			TWL6040_REG_HFRCTL, 0, 0,
+			twl6040_power_mode_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Analog playback switches */
+	SND_SOC_DAPM_SWITCH("HSDAC Left Playback",
+			SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls),
+	SND_SOC_DAPM_SWITCH("HSDAC Right Playback",
+			SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls),
+	SND_SOC_DAPM_SWITCH("HFDAC Left Playback",
+			SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls),
+	SND_SOC_DAPM_SWITCH("HFDAC Right Playback",
+			SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls),
+
+	SND_SOC_DAPM_SWITCH("Headset Left Driver",
+			SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls),
+	SND_SOC_DAPM_SWITCH("Headset Right Driver",
+			SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls),
+	SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver",
+			SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls,
+			twl6040_power_mode_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver",
+			SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls,
+			twl6040_power_mode_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Analog playback PGAs */
+	SND_SOC_DAPM_PGA("HFDAC Left PGA",
+			TWL6040_REG_HFLCTL, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HFDAC Right PGA",
+			TWL6040_REG_HFRCTL, 1, 0, NULL, 0),
+
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* Capture path */
+	{"Analog Left Capture Route", "Headset Mic", "HSMIC"},
+	{"Analog Left Capture Route", "Main Mic", "MAINMIC"},
+	{"Analog Left Capture Route", "Aux/FM Left", "AFML"},
+
+	{"Analog Right Capture Route", "Headset Mic", "HSMIC"},
+	{"Analog Right Capture Route", "Sub Mic", "SUBMIC"},
+	{"Analog Right Capture Route", "Aux/FM Right", "AFMR"},
+
+	{"MicAmpL", NULL, "Analog Left Capture Route"},
+	{"MicAmpR", NULL, "Analog Right Capture Route"},
+
+	{"ADC Left", NULL, "MicAmpL"},
+	{"ADC Right", NULL, "MicAmpR"},
+
+	/* Headset playback path */
+	{"HSDAC Left Playback", "Switch", "HSDAC Left"},
+	{"HSDAC Right Playback", "Switch", "HSDAC Right"},
+
+	{"Headset Left Driver", "Switch", "HSDAC Left Playback"},
+	{"Headset Right Driver", "Switch", "HSDAC Right Playback"},
+
+	{"HSOL", NULL, "Headset Left Driver"},
+	{"HSOR", NULL, "Headset Right Driver"},
+
+	/* Handsfree playback path */
+	{"HFDAC Left Playback", "Switch", "HFDAC Left"},
+	{"HFDAC Right Playback", "Switch", "HFDAC Right"},
+
+	{"HFDAC Left PGA", NULL, "HFDAC Left Playback"},
+	{"HFDAC Right PGA", NULL, "HFDAC Right Playback"},
+
+	{"Handsfree Left Driver", "Switch", "HFDAC Left PGA"},
+	{"Handsfree Right Driver", "Switch", "HFDAC Right PGA"},
+
+	{"HFL", NULL, "Handsfree Left Driver"},
+	{"HFR", NULL, "Handsfree Right Driver"},
+};
+
+static int twl6040_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets,
+				 ARRAY_SIZE(twl6040_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+static int twl6040_power_up_completion(struct snd_soc_codec *codec,
+					int naudint)
+{
+	struct twl6040_data *priv = codec->private_data;
+	int time_left;
+	u8 intid;
+
+	time_left = wait_for_completion_timeout(&priv->ready,
+				msecs_to_jiffies(48));
+
+	if (!time_left) {
+		twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid,
+							TWL6040_REG_INTID);
+		if (!(intid & TWL6040_READYINT)) {
+			dev_err(codec->dev, "timeout waiting for READYINT\n");
+			return -ETIMEDOUT;
+		}
+	}
+
+	priv->codec_powered = 1;
+
+	return 0;
+}
+
+static int twl6040_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
+{
+	struct twl6040_data *priv = codec->private_data;
+	int audpwron = priv->audpwron;
+	int naudint = priv->naudint;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (priv->codec_powered)
+			break;
+
+		if (gpio_is_valid(audpwron)) {
+			/* use AUDPWRON line */
+			gpio_set_value(audpwron, 1);
+
+			/* wait for power-up completion */
+			ret = twl6040_power_up_completion(codec, naudint);
+			if (ret)
+				return ret;
+
+			/* sync registers updated during power-up sequence */
+			twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL);
+			twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL);
+			twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL);
+		} else {
+			/* use manual power-up sequence */
+			twl6040_power_up(codec);
+			priv->codec_powered = 1;
+		}
+
+		/* initialize vdd/vss registers with reg_cache */
+		twl6040_init_vdd_regs(codec);
+		break;
+	case SND_SOC_BIAS_OFF:
+		if (!priv->codec_powered)
+			break;
+
+		if (gpio_is_valid(audpwron)) {
+			/* use AUDPWRON line */
+			gpio_set_value(audpwron, 0);
+
+			/* power-down sequence latency */
+			udelay(500);
+
+			/* sync registers updated during power-down sequence */
+			twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL);
+			twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL);
+			twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL,
+						0x00);
+		} else {
+			/* use manual power-down sequence */
+			twl6040_power_down(codec);
+		}
+
+		priv->codec_powered = 0;
+		break;
+	}
+
+	codec->bias_level = level;
+
+	return 0;
+}
+
+/* set of rates for each pll: low-power and high-performance */
+
+static unsigned int lp_rates[] = {
+	88200,
+	96000,
+};
+
+static struct snd_pcm_hw_constraint_list lp_constraints = {
+	.count	= ARRAY_SIZE(lp_rates),
+	.list	= lp_rates,
+};
+
+static unsigned int hp_rates[] = {
+	96000,
+};
+
+static struct snd_pcm_hw_constraint_list hp_constraints = {
+	.count	= ARRAY_SIZE(hp_rates),
+	.list	= hp_rates,
+};
+
+static int twl6040_startup(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl6040_data *priv = codec->private_data;
+
+	if (!priv->sysclk) {
+		dev_err(codec->dev,
+			"no mclk configured, call set_sysclk() on init\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * capture is not supported at 17.64 MHz,
+	 * it's reserved for headset low-power playback scenario
+	 */
+	if ((priv->sysclk == 17640000) && substream->stream) {
+		dev_err(codec->dev,
+			"capture mode is not supported at %dHz\n",
+			priv->sysclk);
+		return -EINVAL;
+	}
+
+	snd_pcm_hw_constraint_list(substream->runtime, 0,
+				SNDRV_PCM_HW_PARAM_RATE,
+				priv->sysclk_constraints);
+
+	return 0;
+}
+
+static int twl6040_hw_params(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *params,
+			struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl6040_data *priv = codec->private_data;
+	u8 lppllctl;
+	int rate;
+
+	/* nothing to do for high-perf pll, it supports only 48 kHz */
+	if (priv->pll == TWL6040_HPPLL_ID)
+		return 0;
+
+	lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
+
+	rate = params_rate(params);
+	switch (rate) {
+	case 88200:
+		lppllctl |= TWL6040_LPLLFIN;
+		priv->sysclk = 17640000;
+		break;
+	case 96000:
+		lppllctl &= ~TWL6040_LPLLFIN;
+		priv->sysclk = 19200000;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported rate %d\n", rate);
+		return -EINVAL;
+	}
+
+	twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+
+	return 0;
+}
+
+static int twl6040_trigger(struct snd_pcm_substream *substream,
+			int cmd, struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl6040_data *priv = codec->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		/*
+		 * low-power playback mode is restricted
+		 * for headset path only
+		 */
+		if ((priv->sysclk == 17640000) && priv->non_lp) {
+			dev_err(codec->dev,
+				"some enabled paths aren't supported at %dHz\n",
+				priv->sysclk);
+			return -EPERM;
+		}
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct twl6040_data *priv = codec->private_data;
+	u8 hppllctl, lppllctl;
+
+	hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL);
+	lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL);
+
+	switch (clk_id) {
+	case TWL6040_SYSCLK_SEL_LPPLL:
+		switch (freq) {
+		case 32768:
+			/* headset dac and driver must be in low-power mode */
+			headset_power_mode(codec, 0);
+
+			/* clk32k input requires low-power pll */
+			lppllctl |= TWL6040_LPLLENA;
+			twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+			mdelay(5);
+			lppllctl &= ~TWL6040_HPLLSEL;
+			twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+			hppllctl &= ~TWL6040_HPLLENA;
+			twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl);
+			break;
+		default:
+			dev_err(codec->dev, "unknown mclk freq %d\n", freq);
+			return -EINVAL;
+		}
+
+		/* lppll divider */
+		switch (priv->sysclk) {
+		case 17640000:
+			lppllctl |= TWL6040_LPLLFIN;
+			break;
+		case 19200000:
+			lppllctl &= ~TWL6040_LPLLFIN;
+			break;
+		default:
+			/* sysclk not yet configured */
+			lppllctl &= ~TWL6040_LPLLFIN;
+			priv->sysclk = 19200000;
+			break;
+		}
+
+		twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+
+		priv->pll = TWL6040_LPPLL_ID;
+		priv->sysclk_constraints = &lp_constraints;
+		break;
+	case TWL6040_SYSCLK_SEL_HPPLL:
+		hppllctl &= ~TWL6040_MCLK_MSK;
+
+		switch (freq) {
+		case 12000000:
+			/* mclk input, pll enabled */
+			hppllctl |= TWL6040_MCLK_12000KHZ |
+				    TWL6040_HPLLSQRBP |
+				    TWL6040_HPLLENA;
+			break;
+		case 19200000:
+			/* mclk input, pll disabled */
+			hppllctl |= TWL6040_MCLK_19200KHZ |
+				    TWL6040_HPLLSQRBP |
+				    TWL6040_HPLLBP;
+			break;
+		case 26000000:
+			/* mclk input, pll enabled */
+			hppllctl |= TWL6040_MCLK_26000KHZ |
+				    TWL6040_HPLLSQRBP |
+				    TWL6040_HPLLENA;
+			break;
+		case 38400000:
+			/* clk slicer, pll disabled */
+			hppllctl |= TWL6040_MCLK_38400KHZ |
+				    TWL6040_HPLLSQRENA |
+				    TWL6040_HPLLBP;
+			break;
+		default:
+			dev_err(codec->dev, "unknown mclk freq %d\n", freq);
+			return -EINVAL;
+		}
+
+		/* headset dac and driver must be in high-performance mode */
+		headset_power_mode(codec, 1);
+
+		twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl);
+		udelay(500);
+		lppllctl |= TWL6040_HPLLSEL;
+		twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+		lppllctl &= ~TWL6040_LPLLENA;
+		twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl);
+
+		/* high-performance pll can provide only 19.2 MHz */
+		priv->pll = TWL6040_HPPLL_ID;
+		priv->sysclk = 19200000;
+		priv->sysclk_constraints = &hp_constraints;
+		break;
+	default:
+		dev_err(codec->dev, "unknown clk_id %d\n", clk_id);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops twl6040_dai_ops = {
+	.startup	= twl6040_startup,
+	.hw_params	= twl6040_hw_params,
+	.trigger	= twl6040_trigger,
+	.set_sysclk	= twl6040_set_dai_sysclk,
+};
+
+struct snd_soc_dai twl6040_dai = {
+	.name = "twl6040",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 4,
+		.rates = TWL6040_RATES,
+		.formats = TWL6040_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = TWL6040_RATES,
+		.formats = TWL6040_FORMATS,
+	},
+	.ops = &twl6040_dai_ops,
+};
+EXPORT_SYMBOL_GPL(twl6040_dai);
+
+#ifdef CONFIG_PM
+static int twl6040_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int twl6040_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	twl6040_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+#else
+#define twl6040_suspend NULL
+#define twl6040_resume NULL
+#endif
+
+static struct snd_soc_codec *twl6040_codec;
+
+static int twl6040_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	BUG_ON(!twl6040_codec);
+
+	codec = twl6040_codec;
+	socdev->card->codec = codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		return ret;
+	}
+
+	snd_soc_add_controls(codec, twl6040_snd_controls,
+				ARRAY_SIZE(twl6040_snd_controls));
+	twl6040_add_widgets(codec);
+
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+	return ret;
+}
+
+static int twl6040_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_twl6040 = {
+	.probe = twl6040_probe,
+	.remove = twl6040_remove,
+	.suspend = twl6040_suspend,
+	.resume = twl6040_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040);
+
+static int __devinit twl6040_codec_probe(struct platform_device *pdev)
+{
+	struct twl4030_codec_data *twl_codec = pdev->dev.platform_data;
+	struct snd_soc_codec *codec;
+	struct twl6040_data *priv;
+	int audpwron, naudint;
+	int ret = 0;
+
+	priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	if (twl_codec) {
+		audpwron = twl_codec->audpwron_gpio;
+		naudint = twl_codec->naudint_irq;
+	} else {
+		audpwron = -EINVAL;
+		naudint = 0;
+	}
+
+	priv->audpwron = audpwron;
+	priv->naudint = naudint;
+
+	codec = &priv->codec;
+	codec->dev = &pdev->dev;
+	twl6040_dai.dev = &pdev->dev;
+
+	codec->name = "twl6040";
+	codec->owner = THIS_MODULE;
+	codec->read = twl6040_read_reg_cache;
+	codec->write = twl6040_write;
+	codec->set_bias_level = twl6040_set_bias_level;
+	codec->private_data = priv;
+	codec->dai = &twl6040_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(twl6040_reg);
+	codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_reg),
+					GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	init_completion(&priv->ready);
+
+	if (gpio_is_valid(audpwron)) {
+		ret = gpio_request(audpwron, "audpwron");
+		if (ret)
+			goto gpio1_err;
+
+		ret = gpio_direction_output(audpwron, 0);
+		if (ret)
+			goto gpio2_err;
+
+		priv->codec_powered = 0;
+	}
+
+	if (naudint) {
+		/* audio interrupt */
+		ret = request_threaded_irq(naudint, NULL,
+				twl6040_naudint_handler,
+				IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+				"twl6040_codec", codec);
+		if (ret)
+			goto gpio2_err;
+	} else {
+		if (gpio_is_valid(audpwron)) {
+			/* enable only codec ready interrupt */
+			twl6040_write_reg_cache(codec, TWL6040_REG_INTMR,
+					~TWL6040_READYMSK & TWL6040_ALLINT_MSK);
+		} else {
+			/* no interrupts at all */
+			twl6040_write_reg_cache(codec, TWL6040_REG_INTMR,
+						TWL6040_ALLINT_MSK);
+		}
+	}
+
+	/* init vio registers */
+	twl6040_init_vio_regs(codec);
+
+	/* power on device */
+	ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	if (ret)
+		goto irq_err;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret)
+		goto reg_err;
+
+	twl6040_codec = codec;
+
+	ret = snd_soc_register_dai(&twl6040_dai);
+	if (ret)
+		goto dai_err;
+
+	return 0;
+
+dai_err:
+	snd_soc_unregister_codec(codec);
+	twl6040_codec = NULL;
+reg_err:
+	twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
+irq_err:
+	if (naudint)
+		free_irq(naudint, codec);
+gpio2_err:
+	if (gpio_is_valid(audpwron))
+		gpio_free(audpwron);
+gpio1_err:
+	kfree(codec->reg_cache);
+cache_err:
+	kfree(priv);
+	return ret;
+}
+
+static int __devexit twl6040_codec_remove(struct platform_device *pdev)
+{
+	struct twl6040_data *priv = twl6040_codec->private_data;
+	int audpwron = priv->audpwron;
+	int naudint = priv->naudint;
+
+	if (gpio_is_valid(audpwron))
+		gpio_free(audpwron);
+
+	if (naudint)
+		free_irq(naudint, twl6040_codec);
+
+	snd_soc_unregister_dai(&twl6040_dai);
+	snd_soc_unregister_codec(twl6040_codec);
+
+	kfree(twl6040_codec);
+	twl6040_codec = NULL;
+
+	return 0;
+}
+
+static struct platform_driver twl6040_codec_driver = {
+	.driver = {
+		.name = "twl6040_codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = twl6040_codec_probe,
+	.remove = __devexit_p(twl6040_codec_remove),
+};
+
+static int __init twl6040_codec_init(void)
+{
+	return platform_driver_register(&twl6040_codec_driver);
+}
+module_init(twl6040_codec_init);
+
+static void __exit twl6040_codec_exit(void)
+{
+	platform_driver_unregister(&twl6040_codec_driver);
+}
+module_exit(twl6040_codec_exit);
+
+MODULE_DESCRIPTION("ASoC TWL6040 codec driver");
+MODULE_AUTHOR("Misael Lopez Cruz");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h
new file mode 100644
index 000000000000..c472070a1da2
--- /dev/null
+++ b/sound/soc/codecs/twl6040.h
@@ -0,0 +1,141 @@
+/*
+ * ALSA SoC TWL6040 codec driver
+ *
+ * Author:	Misael Lopez Cruz <x0052729@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
+ *
+ */
+
+#ifndef __TWL6040_H__
+#define __TWL6040_H__
+
+#define TWL6040_REG_ASICID		0x01
+#define TWL6040_REG_ASICREV		0x02
+#define TWL6040_REG_INTID		0x03
+#define TWL6040_REG_INTMR		0x04
+#define TWL6040_REG_NCPCTL		0x05
+#define TWL6040_REG_LDOCTL		0x06
+#define TWL6040_REG_HPPLLCTL		0x07
+#define TWL6040_REG_LPPLLCTL		0x08
+#define TWL6040_REG_LPPLLDIV		0x09
+#define TWL6040_REG_AMICBCTL		0x0A
+#define TWL6040_REG_DMICBCTL		0x0B
+#define TWL6040_REG_MICLCTL		0x0C
+#define TWL6040_REG_MICRCTL		0x0D
+#define TWL6040_REG_MICGAIN		0x0E
+#define TWL6040_REG_LINEGAIN		0x0F
+#define TWL6040_REG_HSLCTL		0x10
+#define TWL6040_REG_HSRCTL		0x11
+#define TWL6040_REG_HSGAIN		0x12
+#define TWL6040_REG_EARCTL		0x13
+#define TWL6040_REG_HFLCTL		0x14
+#define TWL6040_REG_HFLGAIN		0x15
+#define TWL6040_REG_HFRCTL		0x16
+#define TWL6040_REG_HFRGAIN		0x17
+#define TWL6040_REG_VIBCTLL		0x18
+#define TWL6040_REG_VIBDATL		0x19
+#define TWL6040_REG_VIBCTLR		0x1A
+#define TWL6040_REG_VIBDATR		0x1B
+#define TWL6040_REG_HKCTL1		0x1C
+#define TWL6040_REG_HKCTL2		0x1D
+#define TWL6040_REG_GPOCTL		0x1E
+#define TWL6040_REG_ALB			0x1F
+#define TWL6040_REG_DLB			0x20
+#define TWL6040_REG_TRIM1		0x28
+#define TWL6040_REG_TRIM2		0x29
+#define TWL6040_REG_TRIM3		0x2A
+#define TWL6040_REG_HSOTRIM		0x2B
+#define TWL6040_REG_HFOTRIM		0x2C
+#define TWL6040_REG_ACCCTL		0x2D
+#define TWL6040_REG_STATUS		0x2E
+
+#define TWL6040_CACHEREGNUM		(TWL6040_REG_STATUS + 1)
+
+#define TWL6040_VIOREGNUM		18
+#define TWL6040_VDDREGNUM		21
+
+/* INTID (0x03) fields */
+
+#define TWL6040_THINT			0x01
+#define TWL6040_PLUGINT			0x02
+#define TWL6040_UNPLUGINT		0x04
+#define TWL6040_HOOKINT			0x08
+#define TWL6040_HFINT			0x10
+#define TWL6040_VIBINT			0x20
+#define TWL6040_READYINT		0x40
+
+/* INTMR (0x04) fields */
+
+#define TWL6040_READYMSK		0x40
+#define TWL6040_ALLINT_MSK		0x7B
+
+/* NCPCTL (0x05) fields */
+
+#define TWL6040_NCPENA			0x01
+#define TWL6040_NCPOPEN			0x40
+
+/* LDOCTL (0x06) fields */
+
+#define TWL6040_LSLDOENA		0x01
+#define TWL6040_HSLDOENA		0x04
+#define TWL6040_REFENA			0x40
+#define TWL6040_OSCENA			0x80
+
+/* HPPLLCTL (0x07) fields */
+
+#define TWL6040_HPLLENA			0x01
+#define TWL6040_HPLLRST			0x02
+#define TWL6040_HPLLBP			0x04
+#define TWL6040_HPLLSQRENA		0x08
+#define TWL6040_HPLLSQRBP		0x10
+#define TWL6040_MCLK_12000KHZ		(0 << 5)
+#define TWL6040_MCLK_19200KHZ		(1 << 5)
+#define TWL6040_MCLK_26000KHZ		(2 << 5)
+#define TWL6040_MCLK_38400KHZ		(3 << 5)
+#define TWL6040_MCLK_MSK		0x60
+
+/* LPPLLCTL (0x08) fields */
+
+#define TWL6040_LPLLENA			0x01
+#define TWL6040_LPLLRST			0x02
+#define TWL6040_LPLLSEL			0x04
+#define TWL6040_LPLLFIN			0x08
+#define TWL6040_HPLLSEL			0x10
+
+/* HSLCTL (0x10) fields */
+
+#define TWL6040_HSDACMODEL		0x02
+#define TWL6040_HSDRVMODEL		0x08
+
+/* HSRCTL (0x11) fields */
+
+#define TWL6040_HSDACMODER		0x02
+#define TWL6040_HSDRVMODER		0x08
+
+/* ACCCTL (0x2D) fields */
+
+#define TWL6040_RESETSPLIT		0x04
+
+#define TWL6040_SYSCLK_SEL_LPPLL	1
+#define TWL6040_SYSCLK_SEL_HPPLL	2
+
+#define TWL6040_HPPLL_ID		1
+#define TWL6040_LPPLL_ID		2
+
+extern struct snd_soc_dai twl6040_dai;
+extern struct snd_soc_codec_device soc_codec_dev_twl6040;
+
+#endif /* End of __TWL6040_H__ */
-- 
cgit v1.2.3


From a1533d94c6b930ef4263941adfd1e1132ca95627 Mon Sep 17 00:00:00 2001
From: Barry Song <21cnbao@gmail.com>
Date: Fri, 19 Mar 2010 14:48:33 +0800
Subject: ASoC: rename ad1938 to ad193x and add support for ad1936/7/8/9

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Yi Li <yi.li@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   4 +-
 sound/soc/codecs/Makefile |   4 +-
 sound/soc/codecs/ad1938.c | 521 ------------------------------------------
 sound/soc/codecs/ad1938.h | 100 ---------
 sound/soc/codecs/ad193x.c | 560 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/ad193x.h |  83 +++++++
 6 files changed, 647 insertions(+), 625 deletions(-)
 delete mode 100644 sound/soc/codecs/ad1938.c
 delete mode 100644 sound/soc/codecs/ad1938.h
 create mode 100644 sound/soc/codecs/ad193x.c
 create mode 100644 sound/soc/codecs/ad193x.h

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 398cbb0e79fd..bc0ab47e156b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -13,7 +13,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1836 if SPI_MASTER
-	select SND_SOC_AD1938 if SPI_MASTER
+	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_ADS117X
 	select SND_SOC_AD73311 if I2C
@@ -92,7 +92,7 @@ config SND_SOC_AC97_CODEC
 config SND_SOC_AD1836
 	tristate
 
-config SND_SOC_AD1938
+config SND_SOC_AD193X
 	tristate
 
 config SND_SOC_AD1980
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 98bd10c8fd3a..337904167358 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,6 +1,6 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1836-objs := ad1836.o
-snd-soc-ad1938-objs := ad1938.o
+snd-soc-ad193x-objs := ad193x.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
 snd-soc-ads117x-objs := ads117x.o
@@ -64,7 +64,7 @@ snd-soc-wm2000-objs := wm2000.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
-obj-$(CONFIG_SND_SOC_AD1938)	+= snd-soc-ad1938.o
+obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
deleted file mode 100644
index c233810d463d..000000000000
--- a/sound/soc/codecs/ad1938.c
+++ /dev/null
@@ -1,521 +0,0 @@
-/*
- * File:         sound/soc/codecs/ad1938.c
- * Author:       Barry Song <Barry.Song@analog.com>
- *
- * Created:      June 04 2009
- * Description:  Driver for AD1938 sound chip
- *
- * Modified:
- *               Copyright 2009 Analog Devices Inc.
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, see the file COPYING, or write
- * to the Free Software Foundation, Inc.,
- * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/kernel.h>
-#include <linux/device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-#include <sound/tlv.h>
-#include <sound/soc-dapm.h>
-#include <linux/spi/spi.h>
-#include "ad1938.h"
-
-/* codec private data */
-struct ad1938_priv {
-	struct snd_soc_codec codec;
-	u8 reg_cache[AD1938_NUM_REGS];
-};
-
-/* ad1938 register cache & default register settings */
-static const u8 ad1938_reg[AD1938_NUM_REGS] = {
-	0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
-static struct snd_soc_codec *ad1938_codec;
-struct snd_soc_codec_device soc_codec_dev_ad1938;
-static int ad1938_register(struct ad1938_priv *ad1938);
-static void ad1938_unregister(struct ad1938_priv *ad1938);
-
-/*
- * AD1938 volume/mute/de-emphasis etc. controls
- */
-static const char *ad1938_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"};
-
-static const struct soc_enum ad1938_deemp_enum =
-	SOC_ENUM_SINGLE(AD1938_DAC_CTRL2, 1, 4, ad1938_deemp);
-
-static const struct snd_kcontrol_new ad1938_snd_controls[] = {
-	/* DAC volume control */
-	SOC_DOUBLE_R("DAC1  Volume", AD1938_DAC_L1_VOL,
-			AD1938_DAC_R1_VOL, 0, 0xFF, 1),
-	SOC_DOUBLE_R("DAC2  Volume", AD1938_DAC_L2_VOL,
-			AD1938_DAC_R2_VOL, 0, 0xFF, 1),
-	SOC_DOUBLE_R("DAC3  Volume", AD1938_DAC_L3_VOL,
-			AD1938_DAC_R3_VOL, 0, 0xFF, 1),
-	SOC_DOUBLE_R("DAC4  Volume", AD1938_DAC_L4_VOL,
-			AD1938_DAC_R4_VOL, 0, 0xFF, 1),
-
-	/* ADC switch control */
-	SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE,
-		AD1938_ADCR1_MUTE, 1, 1),
-	SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE,
-		AD1938_ADCR2_MUTE, 1, 1),
-
-	/* DAC switch control */
-	SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE,
-		AD1938_DACR1_MUTE, 1, 1),
-	SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE,
-		AD1938_DACR2_MUTE, 1, 1),
-	SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE,
-		AD1938_DACR3_MUTE, 1, 1),
-	SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE,
-		AD1938_DACR4_MUTE, 1, 1),
-
-	/* ADC high-pass filter */
-	SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0,
-			AD1938_ADC_HIGHPASS_FILTER, 1, 0),
-
-	/* DAC de-emphasis */
-	SOC_ENUM("Playback Deemphasis", ad1938_deemp_enum),
-};
-
-static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = {
-	SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1),
-	SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
-	SND_SOC_DAPM_SUPPLY("PLL_PWR", AD1938_PLL_CLK_CTRL0, 0, 1, NULL, 0),
-	SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0),
-	SND_SOC_DAPM_OUTPUT("DAC1OUT"),
-	SND_SOC_DAPM_OUTPUT("DAC2OUT"),
-	SND_SOC_DAPM_OUTPUT("DAC3OUT"),
-	SND_SOC_DAPM_OUTPUT("DAC4OUT"),
-	SND_SOC_DAPM_INPUT("ADC1IN"),
-	SND_SOC_DAPM_INPUT("ADC2IN"),
-};
-
-static const struct snd_soc_dapm_route audio_paths[] = {
-	{ "DAC", NULL, "PLL_PWR" },
-	{ "ADC", NULL, "PLL_PWR" },
-	{ "DAC", NULL, "ADC_PWR" },
-	{ "ADC", NULL, "ADC_PWR" },
-	{ "DAC1OUT", "DAC1 Switch", "DAC" },
-	{ "DAC2OUT", "DAC2 Switch", "DAC" },
-	{ "DAC3OUT", "DAC3 Switch", "DAC" },
-	{ "DAC4OUT", "DAC4 Switch", "DAC" },
-	{ "ADC", "ADC1 Switch", "ADC1IN" },
-	{ "ADC", "ADC2 Switch", "ADC2IN" },
-};
-
-/*
- * DAI ops entries
- */
-
-static int ad1938_mute(struct snd_soc_dai *dai, int mute)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	int reg;
-
-	reg = snd_soc_read(codec, AD1938_DAC_CTRL2);
-	reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg &
-		(~AD1938_DAC_MASTER_MUTE);
-	snd_soc_write(codec, AD1938_DAC_CTRL2, reg);
-
-	return 0;
-}
-
-static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
-			       unsigned int rx_mask, int slots, int width)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	int dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1);
-	int adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2);
-
-	dac_reg &= ~AD1938_DAC_CHAN_MASK;
-	adc_reg &= ~AD1938_ADC_CHAN_MASK;
-
-	switch (slots) {
-	case 2:
-		dac_reg |= AD1938_DAC_2_CHANNELS << AD1938_DAC_CHAN_SHFT;
-		adc_reg |= AD1938_ADC_2_CHANNELS << AD1938_ADC_CHAN_SHFT;
-		break;
-	case 4:
-		dac_reg |= AD1938_DAC_4_CHANNELS << AD1938_DAC_CHAN_SHFT;
-		adc_reg |= AD1938_ADC_4_CHANNELS << AD1938_ADC_CHAN_SHFT;
-		break;
-	case 8:
-		dac_reg |= AD1938_DAC_8_CHANNELS << AD1938_DAC_CHAN_SHFT;
-		adc_reg |= AD1938_ADC_8_CHANNELS << AD1938_ADC_CHAN_SHFT;
-		break;
-	case 16:
-		dac_reg |= AD1938_DAC_16_CHANNELS << AD1938_DAC_CHAN_SHFT;
-		adc_reg |= AD1938_ADC_16_CHANNELS << AD1938_ADC_CHAN_SHFT;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg);
-	snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg);
-
-	return 0;
-}
-
-static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai,
-		unsigned int fmt)
-{
-	struct snd_soc_codec *codec = codec_dai->codec;
-	int adc_reg, dac_reg;
-
-	adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2);
-	dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1);
-
-	/* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S
-	 * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A)
-	 */
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_I2S:
-		adc_reg &= ~AD1938_ADC_SERFMT_MASK;
-		adc_reg |= AD1938_ADC_SERFMT_TDM;
-		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		adc_reg &= ~AD1938_ADC_SERFMT_MASK;
-		adc_reg |= AD1938_ADC_SERFMT_AUX;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
-		adc_reg &= ~AD1938_ADC_LEFT_HIGH;
-		adc_reg &= ~AD1938_ADC_BCLK_INV;
-		dac_reg &= ~AD1938_DAC_LEFT_HIGH;
-		dac_reg &= ~AD1938_DAC_BCLK_INV;
-		break;
-	case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */
-		adc_reg |= AD1938_ADC_LEFT_HIGH;
-		adc_reg &= ~AD1938_ADC_BCLK_INV;
-		dac_reg |= AD1938_DAC_LEFT_HIGH;
-		dac_reg &= ~AD1938_DAC_BCLK_INV;
-		break;
-	case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */
-		adc_reg &= ~AD1938_ADC_LEFT_HIGH;
-		adc_reg |= AD1938_ADC_BCLK_INV;
-		dac_reg &= ~AD1938_DAC_LEFT_HIGH;
-		dac_reg |= AD1938_DAC_BCLK_INV;
-		break;
-
-	case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */
-		adc_reg |= AD1938_ADC_LEFT_HIGH;
-		adc_reg |= AD1938_ADC_BCLK_INV;
-		dac_reg |= AD1938_DAC_LEFT_HIGH;
-		dac_reg |= AD1938_DAC_BCLK_INV;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
-		adc_reg |= AD1938_ADC_LCR_MASTER;
-		adc_reg |= AD1938_ADC_BCLK_MASTER;
-		dac_reg |= AD1938_DAC_LCR_MASTER;
-		dac_reg |= AD1938_DAC_BCLK_MASTER;
-		break;
-	case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */
-		adc_reg |= AD1938_ADC_LCR_MASTER;
-		adc_reg &= ~AD1938_ADC_BCLK_MASTER;
-		dac_reg |= AD1938_DAC_LCR_MASTER;
-		dac_reg &= ~AD1938_DAC_BCLK_MASTER;
-		break;
-	case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
-		adc_reg &= ~AD1938_ADC_LCR_MASTER;
-		adc_reg |= AD1938_ADC_BCLK_MASTER;
-		dac_reg &= ~AD1938_DAC_LCR_MASTER;
-		dac_reg |= AD1938_DAC_BCLK_MASTER;
-		break;
-	case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */
-		adc_reg &= ~AD1938_ADC_LCR_MASTER;
-		adc_reg &= ~AD1938_ADC_BCLK_MASTER;
-		dac_reg &= ~AD1938_DAC_LCR_MASTER;
-		dac_reg &= ~AD1938_DAC_BCLK_MASTER;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg);
-	snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg);
-
-	return 0;
-}
-
-static int ad1938_hw_params(struct snd_pcm_substream *substream,
-		struct snd_pcm_hw_params *params,
-		struct snd_soc_dai *dai)
-{
-	int word_len = 0, reg = 0;
-
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	/* bit size */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		word_len = 3;
-		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-		word_len = 1;
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-	case SNDRV_PCM_FORMAT_S32_LE:
-		word_len = 0;
-		break;
-	}
-
-	reg = snd_soc_read(codec, AD1938_DAC_CTRL2);
-	reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len;
-	snd_soc_write(codec, AD1938_DAC_CTRL2, reg);
-
-	reg = snd_soc_read(codec, AD1938_ADC_CTRL1);
-	reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len;
-	snd_soc_write(codec, AD1938_ADC_CTRL1, reg);
-
-	return 0;
-}
-
-static int __devinit ad1938_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_codec *codec;
-	struct ad1938_priv *ad1938;
-
-	ad1938 = kzalloc(sizeof(struct ad1938_priv), GFP_KERNEL);
-	if (ad1938 == NULL)
-		return -ENOMEM;
-
-	codec = &ad1938->codec;
-	codec->control_data = spi;
-	codec->dev = &spi->dev;
-
-	dev_set_drvdata(&spi->dev, ad1938);
-
-	return ad1938_register(ad1938);
-}
-
-static int __devexit ad1938_spi_remove(struct spi_device *spi)
-{
-	struct ad1938_priv *ad1938 = dev_get_drvdata(&spi->dev);
-
-	ad1938_unregister(ad1938);
-	return 0;
-}
-
-static struct spi_driver ad1938_spi_driver = {
-	.driver = {
-		.name	= "ad1938",
-		.owner	= THIS_MODULE,
-	},
-	.probe		= ad1938_spi_probe,
-	.remove		= __devexit_p(ad1938_spi_remove),
-};
-
-static struct snd_soc_dai_ops ad1938_dai_ops = {
-	.hw_params = ad1938_hw_params,
-	.digital_mute = ad1938_mute,
-	.set_tdm_slot = ad1938_set_tdm_slot,
-	.set_fmt = ad1938_set_dai_fmt,
-};
-
-/* codec DAI instance */
-struct snd_soc_dai ad1938_dai = {
-	.name = "AD1938",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 2,
-		.channels_max = 8,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 2,
-		.channels_max = 4,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.ops = &ad1938_dai_ops,
-};
-EXPORT_SYMBOL_GPL(ad1938_dai);
-
-static int ad1938_register(struct ad1938_priv *ad1938)
-{
-	int ret;
-	struct snd_soc_codec *codec = &ad1938->codec;
-
-	if (ad1938_codec) {
-		dev_err(codec->dev, "Another ad1938 is registered\n");
-		return -EINVAL;
-	}
-
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	codec->private_data = ad1938;
-	codec->reg_cache = ad1938->reg_cache;
-	codec->reg_cache_size = AD1938_NUM_REGS;
-	codec->name = "AD1938";
-	codec->owner = THIS_MODULE;
-	codec->dai = &ad1938_dai;
-	codec->num_dai = 1;
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	ad1938_dai.dev = codec->dev;
-	ad1938_codec = codec;
-
-	memcpy(codec->reg_cache, ad1938_reg, AD1938_NUM_REGS);
-
-	ret = snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_SPI);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to set cache I/O: %d\n",
-				ret);
-		kfree(ad1938);
-		return ret;
-	}
-
-	/* default setting for ad1938 */
-
-	/* unmute dac channels */
-	snd_soc_write(codec, AD1938_DAC_CHNL_MUTE, 0x0);
-	/* de-emphasis: 48kHz, powedown dac */
-	snd_soc_write(codec, AD1938_DAC_CTRL2, 0x1A);
-	/* powerdown dac, dac in tdm mode */
-	snd_soc_write(codec, AD1938_DAC_CTRL0, 0x41);
-	/* high-pass filter enable */
-	snd_soc_write(codec, AD1938_ADC_CTRL0, 0x3);
-	/* sata delay=1, adc aux mode */
-	snd_soc_write(codec, AD1938_ADC_CTRL1, 0x43);
-	/* pll input: mclki/xi */
-	snd_soc_write(codec, AD1938_PLL_CLK_CTRL0, 0x9D);
-	snd_soc_write(codec, AD1938_PLL_CLK_CTRL1, 0x04);
-
-	ret = snd_soc_register_codec(codec);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		kfree(ad1938);
-		return ret;
-	}
-
-	ret = snd_soc_register_dai(&ad1938_dai);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-		snd_soc_unregister_codec(codec);
-		kfree(ad1938);
-		return ret;
-	}
-
-	return 0;
-}
-
-static void ad1938_unregister(struct ad1938_priv *ad1938)
-{
-	snd_soc_unregister_dai(&ad1938_dai);
-	snd_soc_unregister_codec(&ad1938->codec);
-	kfree(ad1938);
-	ad1938_codec = NULL;
-}
-
-static int ad1938_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec;
-	int ret = 0;
-
-	if (ad1938_codec == NULL) {
-		dev_err(&pdev->dev, "Codec device not registered\n");
-		return -ENODEV;
-	}
-
-	socdev->card->codec = ad1938_codec;
-	codec = ad1938_codec;
-
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
-		goto pcm_err;
-	}
-
-	snd_soc_add_controls(codec, ad1938_snd_controls,
-			     ARRAY_SIZE(ad1938_snd_controls));
-	snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets,
-				  ARRAY_SIZE(ad1938_dapm_widgets));
-	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
-
-
-pcm_err:
-	return ret;
-}
-
-/* power down chip */
-static int ad1938_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-
-	return 0;
-}
-
-struct snd_soc_codec_device soc_codec_dev_ad1938 = {
-	.probe = 	ad1938_probe,
-	.remove = 	ad1938_remove,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938);
-
-static int __init ad1938_init(void)
-{
-	int ret;
-
-	ret = spi_register_driver(&ad1938_spi_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register ad1938 SPI driver: %d\n",
-				ret);
-	}
-
-	return ret;
-}
-module_init(ad1938_init);
-
-static void __exit ad1938_exit(void)
-{
-	spi_unregister_driver(&ad1938_spi_driver);
-}
-module_exit(ad1938_exit);
-
-MODULE_DESCRIPTION("ASoC ad1938 driver");
-MODULE_AUTHOR("Barry Song ");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1938.h b/sound/soc/codecs/ad1938.h
deleted file mode 100644
index fe3c48cd2d5b..000000000000
--- a/sound/soc/codecs/ad1938.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * File:         sound/soc/codecs/ad1836.h
- * Based on:
- * Author:       Barry Song <Barry.Song@analog.com>
- *
- * Created:      May 25, 2009
- * Description:  definitions for AD1938 registers
- *
- * Modified:
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, see the file COPYING, or write
- * to the Free Software Foundation, Inc.,
- * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- */
-
-#ifndef __AD1938_H__
-#define __AD1938_H__
-
-#define AD1938_PLL_CLK_CTRL0    0
-#define AD1938_PLL_POWERDOWN           0x01
-#define AD1938_PLL_CLK_CTRL1    1
-#define AD1938_DAC_CTRL0        2
-#define AD1938_DAC_POWERDOWN           0x01
-#define AD1938_DAC_SERFMT_MASK		0xC0
-#define AD1938_DAC_SERFMT_STEREO	(0 << 6)
-#define AD1938_DAC_SERFMT_TDM		(1 << 6)
-#define AD1938_DAC_CTRL1        3
-#define AD1938_DAC_2_CHANNELS   0
-#define AD1938_DAC_4_CHANNELS   1
-#define AD1938_DAC_8_CHANNELS   2
-#define AD1938_DAC_16_CHANNELS  3
-#define AD1938_DAC_CHAN_SHFT    1
-#define AD1938_DAC_CHAN_MASK    (3 << AD1938_DAC_CHAN_SHFT)
-#define AD1938_DAC_LCR_MASTER   (1 << 4)
-#define AD1938_DAC_BCLK_MASTER  (1 << 5)
-#define AD1938_DAC_LEFT_HIGH    (1 << 3)
-#define AD1938_DAC_BCLK_INV     (1 << 7)
-#define AD1938_DAC_CTRL2        4
-#define AD1938_DAC_WORD_LEN_MASK	0xC
-#define AD1938_DAC_MASTER_MUTE  1
-#define AD1938_DAC_CHNL_MUTE    5
-#define AD1938_DACL1_MUTE       0
-#define AD1938_DACR1_MUTE       1
-#define AD1938_DACL2_MUTE       2
-#define AD1938_DACR2_MUTE       3
-#define AD1938_DACL3_MUTE       4
-#define AD1938_DACR3_MUTE       5
-#define AD1938_DACL4_MUTE       6
-#define AD1938_DACR4_MUTE       7
-#define AD1938_DAC_L1_VOL       6
-#define AD1938_DAC_R1_VOL       7
-#define AD1938_DAC_L2_VOL       8
-#define AD1938_DAC_R2_VOL       9
-#define AD1938_DAC_L3_VOL       10
-#define AD1938_DAC_R3_VOL       11
-#define AD1938_DAC_L4_VOL       12
-#define AD1938_DAC_R4_VOL       13
-#define AD1938_ADC_CTRL0        14
-#define AD1938_ADC_POWERDOWN           0x01
-#define AD1938_ADC_HIGHPASS_FILTER	1
-#define AD1938_ADCL1_MUTE 		2
-#define AD1938_ADCR1_MUTE 		3
-#define AD1938_ADCL2_MUTE 		4
-#define AD1938_ADCR2_MUTE 		5
-#define AD1938_ADC_CTRL1        15
-#define AD1938_ADC_SERFMT_MASK		0x60
-#define AD1938_ADC_SERFMT_STEREO	(0 << 5)
-#define AD1938_ADC_SERFMT_TDM		(1 << 2)
-#define AD1938_ADC_SERFMT_AUX		(2 << 5)
-#define AD1938_ADC_WORD_LEN_MASK	0x3
-#define AD1938_ADC_CTRL2        16
-#define AD1938_ADC_2_CHANNELS   0
-#define AD1938_ADC_4_CHANNELS   1
-#define AD1938_ADC_8_CHANNELS   2
-#define AD1938_ADC_16_CHANNELS  3
-#define AD1938_ADC_CHAN_SHFT    4
-#define AD1938_ADC_CHAN_MASK    (3 << AD1938_ADC_CHAN_SHFT)
-#define AD1938_ADC_LCR_MASTER   (1 << 3)
-#define AD1938_ADC_BCLK_MASTER  (1 << 6)
-#define AD1938_ADC_LEFT_HIGH    (1 << 2)
-#define AD1938_ADC_BCLK_INV     (1 << 1)
-
-#define AD1938_NUM_REGS          17
-
-extern struct snd_soc_dai ad1938_dai;
-extern struct snd_soc_codec_device soc_codec_dev_ad1938;
-#endif
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
new file mode 100644
index 000000000000..d7426990b66e
--- /dev/null
+++ b/sound/soc/codecs/ad193x.c
@@ -0,0 +1,560 @@
+/*
+ * AD193X Audio Codec driver supporting AD1936/7/8/9
+ *
+ * Copyright 2010 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-dapm.h>
+#include "ad193x.h"
+
+/* codec private data */
+struct ad193x_priv {
+	struct snd_soc_codec codec;
+	u8 reg_cache[AD193X_NUM_REGS];
+};
+
+/* ad193x register cache & default register settings */
+static const u8 ad193x_reg[AD193X_NUM_REGS] = {
+	0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
+};
+
+static struct snd_soc_codec *ad193x_codec;
+struct snd_soc_codec_device soc_codec_dev_ad193x;
+static int ad193x_register(struct ad193x_priv *ad193x, int bus_type);
+static void ad193x_unregister(struct ad193x_priv *ad193x);
+
+/*
+ * AD193X volume/mute/de-emphasis etc. controls
+ */
+static const char *ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"};
+
+static const struct soc_enum ad193x_deemp_enum =
+	SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp);
+
+static const struct snd_kcontrol_new ad193x_snd_controls[] = {
+	/* DAC volume control */
+	SOC_DOUBLE_R("DAC1  Volume", AD193X_DAC_L1_VOL,
+			AD193X_DAC_R1_VOL, 0, 0xFF, 1),
+	SOC_DOUBLE_R("DAC2  Volume", AD193X_DAC_L2_VOL,
+			AD193X_DAC_R2_VOL, 0, 0xFF, 1),
+	SOC_DOUBLE_R("DAC3  Volume", AD193X_DAC_L3_VOL,
+			AD193X_DAC_R3_VOL, 0, 0xFF, 1),
+	SOC_DOUBLE_R("DAC4  Volume", AD193X_DAC_L4_VOL,
+			AD193X_DAC_R4_VOL, 0, 0xFF, 1),
+
+	/* ADC switch control */
+	SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
+		AD193X_ADCR1_MUTE, 1, 1),
+	SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
+		AD193X_ADCR2_MUTE, 1, 1),
+
+	/* DAC switch control */
+	SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE,
+		AD193X_DACR1_MUTE, 1, 1),
+	SOC_DOUBLE("DAC2 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL2_MUTE,
+		AD193X_DACR2_MUTE, 1, 1),
+	SOC_DOUBLE("DAC3 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL3_MUTE,
+		AD193X_DACR3_MUTE, 1, 1),
+	SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE,
+		AD193X_DACR4_MUTE, 1, 1),
+
+	/* ADC high-pass filter */
+	SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0,
+			AD193X_ADC_HIGHPASS_FILTER, 1, 0),
+
+	/* DAC de-emphasis */
+	SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
+};
+
+static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1),
+	SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("DAC1OUT"),
+	SND_SOC_DAPM_OUTPUT("DAC2OUT"),
+	SND_SOC_DAPM_OUTPUT("DAC3OUT"),
+	SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+	SND_SOC_DAPM_INPUT("ADC1IN"),
+	SND_SOC_DAPM_INPUT("ADC2IN"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+	{ "DAC", NULL, "PLL_PWR" },
+	{ "ADC", NULL, "PLL_PWR" },
+	{ "DAC", NULL, "ADC_PWR" },
+	{ "ADC", NULL, "ADC_PWR" },
+	{ "DAC1OUT", "DAC1 Switch", "DAC" },
+	{ "DAC2OUT", "DAC2 Switch", "DAC" },
+	{ "DAC3OUT", "DAC3 Switch", "DAC" },
+	{ "DAC4OUT", "DAC4 Switch", "DAC" },
+	{ "ADC", "ADC1 Switch", "ADC1IN" },
+	{ "ADC", "ADC2 Switch", "ADC2IN" },
+};
+
+/*
+ * DAI ops entries
+ */
+
+static int ad193x_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int reg;
+
+	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
+	reg = (mute > 0) ? reg | AD193X_DAC_MASTER_MUTE : reg &
+		(~AD193X_DAC_MASTER_MUTE);
+	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
+
+	return 0;
+}
+
+static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+			       unsigned int rx_mask, int slots, int width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1);
+	int adc_reg = snd_soc_read(codec, AD193X_ADC_CTRL2);
+
+	dac_reg &= ~AD193X_DAC_CHAN_MASK;
+	adc_reg &= ~AD193X_ADC_CHAN_MASK;
+
+	switch (slots) {
+	case 2:
+		dac_reg |= AD193X_DAC_2_CHANNELS << AD193X_DAC_CHAN_SHFT;
+		adc_reg |= AD193X_ADC_2_CHANNELS << AD193X_ADC_CHAN_SHFT;
+		break;
+	case 4:
+		dac_reg |= AD193X_DAC_4_CHANNELS << AD193X_DAC_CHAN_SHFT;
+		adc_reg |= AD193X_ADC_4_CHANNELS << AD193X_ADC_CHAN_SHFT;
+		break;
+	case 8:
+		dac_reg |= AD193X_DAC_8_CHANNELS << AD193X_DAC_CHAN_SHFT;
+		adc_reg |= AD193X_ADC_8_CHANNELS << AD193X_ADC_CHAN_SHFT;
+		break;
+	case 16:
+		dac_reg |= AD193X_DAC_16_CHANNELS << AD193X_DAC_CHAN_SHFT;
+		adc_reg |= AD193X_ADC_16_CHANNELS << AD193X_ADC_CHAN_SHFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg);
+	snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg);
+
+	return 0;
+}
+
+static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int adc_reg, dac_reg;
+
+	adc_reg = snd_soc_read(codec, AD193X_ADC_CTRL2);
+	dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1);
+
+	/* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S
+	 * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A)
+	 */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		adc_reg &= ~AD193X_ADC_SERFMT_MASK;
+		adc_reg |= AD193X_ADC_SERFMT_TDM;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		adc_reg &= ~AD193X_ADC_SERFMT_MASK;
+		adc_reg |= AD193X_ADC_SERFMT_AUX;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
+		adc_reg &= ~AD193X_ADC_LEFT_HIGH;
+		adc_reg &= ~AD193X_ADC_BCLK_INV;
+		dac_reg &= ~AD193X_DAC_LEFT_HIGH;
+		dac_reg &= ~AD193X_DAC_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */
+		adc_reg |= AD193X_ADC_LEFT_HIGH;
+		adc_reg &= ~AD193X_ADC_BCLK_INV;
+		dac_reg |= AD193X_DAC_LEFT_HIGH;
+		dac_reg &= ~AD193X_DAC_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */
+		adc_reg &= ~AD193X_ADC_LEFT_HIGH;
+		adc_reg |= AD193X_ADC_BCLK_INV;
+		dac_reg &= ~AD193X_DAC_LEFT_HIGH;
+		dac_reg |= AD193X_DAC_BCLK_INV;
+		break;
+
+	case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */
+		adc_reg |= AD193X_ADC_LEFT_HIGH;
+		adc_reg |= AD193X_ADC_BCLK_INV;
+		dac_reg |= AD193X_DAC_LEFT_HIGH;
+		dac_reg |= AD193X_DAC_BCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
+		adc_reg |= AD193X_ADC_LCR_MASTER;
+		adc_reg |= AD193X_ADC_BCLK_MASTER;
+		dac_reg |= AD193X_DAC_LCR_MASTER;
+		dac_reg |= AD193X_DAC_BCLK_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */
+		adc_reg |= AD193X_ADC_LCR_MASTER;
+		adc_reg &= ~AD193X_ADC_BCLK_MASTER;
+		dac_reg |= AD193X_DAC_LCR_MASTER;
+		dac_reg &= ~AD193X_DAC_BCLK_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
+		adc_reg &= ~AD193X_ADC_LCR_MASTER;
+		adc_reg |= AD193X_ADC_BCLK_MASTER;
+		dac_reg &= ~AD193X_DAC_LCR_MASTER;
+		dac_reg |= AD193X_DAC_BCLK_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */
+		adc_reg &= ~AD193X_ADC_LCR_MASTER;
+		adc_reg &= ~AD193X_ADC_BCLK_MASTER;
+		dac_reg &= ~AD193X_DAC_LCR_MASTER;
+		dac_reg &= ~AD193X_DAC_BCLK_MASTER;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg);
+	snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg);
+
+	return 0;
+}
+
+static int ad193x_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *dai)
+{
+	int word_len = 0, reg = 0;
+
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word_len = 3;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		word_len = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+	case SNDRV_PCM_FORMAT_S32_LE:
+		word_len = 0;
+		break;
+	}
+
+	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
+	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
+
+	reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
+	reg = (reg & (~AD193X_ADC_WORD_LEN_MASK)) | word_len;
+	snd_soc_write(codec, AD193X_ADC_CTRL1, reg);
+
+	return 0;
+}
+
+int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
+{
+	struct snd_soc_codec *codec;
+	struct ad193x_priv *ad193x;
+
+	ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL);
+	if (ad193x == NULL)
+		return -ENOMEM;
+
+	codec = &ad193x->codec;
+	codec->control_data = ctrl_data;
+	codec->dev = dev;
+
+	dev_set_drvdata(dev, ad193x);
+
+	return ad193x_register(ad193x, bus_type);
+}
+EXPORT_SYMBOL_GPL(ad193x_bus_probe);
+
+int ad193x_bus_remove(struct device *dev)
+{
+	struct ad193x_priv *ad193x = dev_get_drvdata(dev);
+
+	ad193x_unregister(ad193x);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ad193x_bus_remove);
+
+static struct snd_soc_dai_ops ad193x_dai_ops = {
+	.hw_params = ad193x_hw_params,
+	.digital_mute = ad193x_mute,
+	.set_tdm_slot = ad193x_set_tdm_slot,
+	.set_fmt = ad193x_set_dai_fmt,
+};
+
+/* codec DAI instance */
+struct snd_soc_dai ad193x_dai = {
+	.name = "AD193X",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 4,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.ops = &ad193x_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ad193x_dai);
+
+static int ad193x_register(struct ad193x_priv *ad193x, int bus_type)
+{
+	int ret;
+	struct snd_soc_codec *codec = &ad193x->codec;
+
+	if (ad193x_codec) {
+		dev_err(codec->dev, "Another ad193x is registered\n");
+		return -EINVAL;
+	}
+
+	mutex_init(&codec->mutex);
+	codec->private_data = ad193x;
+	codec->reg_cache = ad193x->reg_cache;
+	codec->reg_cache_size = AD193X_NUM_REGS;
+	codec->name = "AD193X";
+	codec->owner = THIS_MODULE;
+	codec->dai = &ad193x_dai;
+	codec->num_dai = 1;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ad193x_dai.dev = codec->dev;
+	ad193x_codec = codec;
+
+	memcpy(codec->reg_cache, ad193x_reg, AD193X_NUM_REGS);
+
+	if (bus_type == SND_SOC_I2C)
+		ret = snd_soc_codec_set_cache_io(codec, 8, 8, bus_type);
+	else
+		ret = snd_soc_codec_set_cache_io(codec, 16, 8, bus_type);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to set cache I/O: %d\n",
+				ret);
+		kfree(ad193x);
+		return ret;
+	}
+
+	/* default setting for ad193x */
+
+	/* unmute dac channels */
+	snd_soc_write(codec, AD193X_DAC_CHNL_MUTE, 0x0);
+	/* de-emphasis: 48kHz, powedown dac */
+	snd_soc_write(codec, AD193X_DAC_CTRL2, 0x1A);
+	/* powerdown dac, dac in tdm mode */
+	snd_soc_write(codec, AD193X_DAC_CTRL0, 0x41);
+	/* high-pass filter enable */
+	snd_soc_write(codec, AD193X_ADC_CTRL0, 0x3);
+	/* sata delay=1, adc aux mode */
+	snd_soc_write(codec, AD193X_ADC_CTRL1, 0x43);
+	/* pll input: mclki/xi */
+	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
+	snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04);
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		kfree(ad193x);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&ad193x_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		kfree(ad193x);
+		return ret;
+	}
+
+	return 0;
+}
+
+static void ad193x_unregister(struct ad193x_priv *ad193x)
+{
+	snd_soc_unregister_dai(&ad193x_dai);
+	snd_soc_unregister_codec(&ad193x->codec);
+	kfree(ad193x);
+	ad193x_codec = NULL;
+}
+
+static int ad193x_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (ad193x_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = ad193x_codec;
+	codec = ad193x_codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, ad193x_snd_controls,
+			     ARRAY_SIZE(ad193x_snd_controls));
+	snd_soc_dapm_new_controls(codec, ad193x_dapm_widgets,
+				  ARRAY_SIZE(ad193x_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+
+pcm_err:
+	return ret;
+}
+
+/* power down chip */
+static int ad193x_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad193x = {
+	.probe = 	ad193x_probe,
+	.remove = 	ad193x_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad193x);
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit ad193x_spi_probe(struct spi_device *spi)
+{
+	return ad193x_bus_probe(&spi->dev, spi, SND_SOC_SPI);
+}
+
+static int __devexit ad193x_spi_remove(struct spi_device *spi)
+{
+	return ad193x_bus_remove(&spi->dev);
+}
+
+static struct spi_driver ad193x_spi_driver = {
+	.driver = {
+		.name	= "ad193x",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= ad193x_spi_probe,
+	.remove		= __devexit_p(ad193x_spi_remove),
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id ad193x_id[] = {
+	{ "ad1936", 0 },
+	{ "ad1937", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ad193x_id);
+
+static int __devinit ad193x_i2c_probe(struct i2c_client *client,
+		const struct i2c_device_id *id)
+{
+	return ad193x_bus_probe(&client->dev, client, SND_SOC_I2C);
+}
+
+static int __devexit ad193x_i2c_remove(struct i2c_client *client)
+{
+	return ad193x_bus_remove(&client->dev);
+}
+
+static struct i2c_driver ad193x_i2c_driver = {
+	.driver = {
+		.name = "ad193x",
+	},
+	.probe    = ad193x_i2c_probe,
+	.remove   = __devexit_p(ad193x_i2c_remove),
+	.id_table = ad193x_id,
+};
+#endif
+
+static int __init ad193x_modinit(void)
+{
+	int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret =  i2c_add_driver(&ad193x_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n",
+				ret);
+	}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&ad193x_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register AD193X SPI driver: %d\n",
+				ret);
+	}
+#endif
+	return ret;
+}
+module_init(ad193x_modinit);
+
+static void __exit ad193x_modexit(void)
+{
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&ad193x_spi_driver);
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&ad193x_i2c_driver);
+#endif
+}
+module_exit(ad193x_modexit);
+
+MODULE_DESCRIPTION("ASoC ad193x driver");
+MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
new file mode 100644
index 000000000000..b240d1d67c22
--- /dev/null
+++ b/sound/soc/codecs/ad193x.h
@@ -0,0 +1,83 @@
+/*
+ * AD193X Audio Codec driver
+ *
+ * Copyright 2010 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef __AD193X_H__
+#define __AD193X_H__
+
+#define AD193X_PLL_CLK_CTRL0    0x800
+#define AD193X_PLL_POWERDOWN           0x01
+#define AD193X_PLL_CLK_CTRL1    0x801
+#define AD193X_DAC_CTRL0        0x802
+#define AD193X_DAC_POWERDOWN           0x01
+#define AD193X_DAC_SERFMT_MASK		0xC0
+#define AD193X_DAC_SERFMT_STEREO	(0 << 6)
+#define AD193X_DAC_SERFMT_TDM		(1 << 6)
+#define AD193X_DAC_CTRL1        0x803
+#define AD193X_DAC_2_CHANNELS   0
+#define AD193X_DAC_4_CHANNELS   1
+#define AD193X_DAC_8_CHANNELS   2
+#define AD193X_DAC_16_CHANNELS  3
+#define AD193X_DAC_CHAN_SHFT    1
+#define AD193X_DAC_CHAN_MASK    (3 << AD193X_DAC_CHAN_SHFT)
+#define AD193X_DAC_LCR_MASTER   (1 << 4)
+#define AD193X_DAC_BCLK_MASTER  (1 << 5)
+#define AD193X_DAC_LEFT_HIGH    (1 << 3)
+#define AD193X_DAC_BCLK_INV     (1 << 7)
+#define AD193X_DAC_CTRL2        0x804
+#define AD193X_DAC_WORD_LEN_MASK	0xC
+#define AD193X_DAC_MASTER_MUTE  1
+#define AD193X_DAC_CHNL_MUTE    0x805
+#define AD193X_DACL1_MUTE       0
+#define AD193X_DACR1_MUTE       1
+#define AD193X_DACL2_MUTE       2
+#define AD193X_DACR2_MUTE       3
+#define AD193X_DACL3_MUTE       4
+#define AD193X_DACR3_MUTE       5
+#define AD193X_DACL4_MUTE       6
+#define AD193X_DACR4_MUTE       7
+#define AD193X_DAC_L1_VOL       0x806
+#define AD193X_DAC_R1_VOL       0x807
+#define AD193X_DAC_L2_VOL       0x808
+#define AD193X_DAC_R2_VOL       0x809
+#define AD193X_DAC_L3_VOL       0x80a
+#define AD193X_DAC_R3_VOL       0x80b
+#define AD193X_DAC_L4_VOL       0x80c
+#define AD193X_DAC_R4_VOL       0x80d
+#define AD193X_ADC_CTRL0        0x80e
+#define AD193X_ADC_POWERDOWN           0x01
+#define AD193X_ADC_HIGHPASS_FILTER	1
+#define AD193X_ADCL1_MUTE 		2
+#define AD193X_ADCR1_MUTE 		3
+#define AD193X_ADCL2_MUTE 		4
+#define AD193X_ADCR2_MUTE 		5
+#define AD193X_ADC_CTRL1        0x80f
+#define AD193X_ADC_SERFMT_MASK		0x60
+#define AD193X_ADC_SERFMT_STEREO	(0 << 5)
+#define AD193X_ADC_SERFMT_TDM		(1 << 2)
+#define AD193X_ADC_SERFMT_AUX		(2 << 5)
+#define AD193X_ADC_WORD_LEN_MASK	0x3
+#define AD193X_ADC_CTRL2        0x810
+#define AD193X_ADC_2_CHANNELS   0
+#define AD193X_ADC_4_CHANNELS   1
+#define AD193X_ADC_8_CHANNELS   2
+#define AD193X_ADC_16_CHANNELS  3
+#define AD193X_ADC_CHAN_SHFT    4
+#define AD193X_ADC_CHAN_MASK    (3 << AD193X_ADC_CHAN_SHFT)
+#define AD193X_ADC_LCR_MASTER   (1 << 3)
+#define AD193X_ADC_BCLK_MASTER  (1 << 6)
+#define AD193X_ADC_LEFT_HIGH    (1 << 2)
+#define AD193X_ADC_BCLK_INV     (1 << 1)
+
+#define AD193X_NUM_REGS          17
+
+extern struct snd_soc_dai ad193x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad193x;
+extern int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type);
+extern int ad193x_bus_remove(struct device *dev);
+
+#endif
-- 
cgit v1.2.3


From cffce322be94ecdd3bbb74dfd3f3834c51b4232a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 19 Mar 2010 12:22:03 +0000
Subject: ASoC: Unexport AD193x bus probe/remove functions

The export is not needed since the per-bus code lives in the same
module.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/ad193x.c | 6 ++----
 sound/soc/codecs/ad193x.h | 2 --
 2 files changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index d7426990b66e..d034464e26ff 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -285,7 +285,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
+static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
 {
 	struct snd_soc_codec *codec;
 	struct ad193x_priv *ad193x;
@@ -302,16 +302,14 @@ int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
 
 	return ad193x_register(ad193x, bus_type);
 }
-EXPORT_SYMBOL_GPL(ad193x_bus_probe);
 
-int ad193x_bus_remove(struct device *dev)
+static int ad193x_bus_remove(struct device *dev)
 {
 	struct ad193x_priv *ad193x = dev_get_drvdata(dev);
 
 	ad193x_unregister(ad193x);
 	return 0;
 }
-EXPORT_SYMBOL_GPL(ad193x_bus_remove);
 
 static struct snd_soc_dai_ops ad193x_dai_ops = {
 	.hw_params = ad193x_hw_params,
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index b240d1d67c22..a03c880d52f9 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -77,7 +77,5 @@
 
 extern struct snd_soc_dai ad193x_dai;
 extern struct snd_soc_codec_device soc_codec_dev_ad193x;
-extern int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type);
-extern int ad193x_bus_remove(struct device *dev);
 
 #endif
-- 
cgit v1.2.3


From 698c3756660f7afd6fa5051f667a259a953eea77 Mon Sep 17 00:00:00 2001
From: Barry Song <21cnbao@gmail.com>
Date: Fri, 19 Mar 2010 11:18:40 +0800
Subject: ASoC: change bf5xx-ad1938 machine driver to bf5xx-ad193x machine
 driver

Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/Kconfig        |   9 ++-
 sound/soc/blackfin/Makefile       |   4 +-
 sound/soc/blackfin/bf5xx-ad1938.c | 149 --------------------------------------
 sound/soc/blackfin/bf5xx-ad193x.c | 149 ++++++++++++++++++++++++++++++++++++++
 4 files changed, 156 insertions(+), 155 deletions(-)
 delete mode 100644 sound/soc/blackfin/bf5xx-ad1938.c
 create mode 100644 sound/soc/blackfin/bf5xx-ad193x.c

(limited to 'sound/soc')

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 97f1a251e446..8ef25025f3dc 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -49,13 +49,14 @@ config SND_BF5XX_SOC_AD1836
 	help
 	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
 
-config SND_BF5XX_SOC_AD1938
-	tristate "SoC AD1938 Audio support for Blackfin"
+config SND_BF5XX_SOC_AD193X
+	tristate "SoC AD193X Audio support for Blackfin"
 	depends on SND_BF5XX_TDM
 	select SND_BF5XX_SOC_TDM
-	select SND_SOC_AD1938
+	select SND_SOC_AD193X
 	help
-	  Say Y if you want to add support for AD1938 codec on Blackfin.
+	  Say Y if you want to add support for AD193X codec on Blackfin.
+	  This driver supports AD1936, AD1937, AD1938 and AD1939.
 
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 87e30423912f..49af3f32aec8 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -20,10 +20,10 @@ snd-ad1836-objs := bf5xx-ad1836.o
 snd-ad1980-objs := bf5xx-ad1980.o
 snd-ssm2602-objs := bf5xx-ssm2602.o
 snd-ad73311-objs := bf5xx-ad73311.o
-snd-ad1938-objs := bf5xx-ad1938.o
+snd-ad193x-objs := bf5xx-ad193x.o
 
 obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
 obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
-obj-$(CONFIG_SND_BF5XX_SOC_AD1938) += snd-ad1938.o
+obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c
deleted file mode 100644
index 2ef1e5013b8c..000000000000
--- a/sound/soc/blackfin/bf5xx-ad1938.c
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * File:         sound/soc/blackfin/bf5xx-ad1938.c
- * Author:       Barry Song <Barry.Song@analog.com>
- *
- * Created:      Thur June 4 2009
- * Description:  Board driver for ad1938 sound chip
- *
- * Bugs:         Enter bugs at http://blackfin.uclinux.org/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, see the file COPYING, or write
- * to the Free Software Foundation, Inc.,
- * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <sound/pcm_params.h>
-
-#include <asm/blackfin.h>
-#include <asm/cacheflush.h>
-#include <asm/irq.h>
-#include <asm/dma.h>
-#include <asm/portmux.h>
-
-#include "../codecs/ad1938.h"
-#include "bf5xx-sport.h"
-
-#include "bf5xx-tdm-pcm.h"
-#include "bf5xx-tdm.h"
-
-static struct snd_soc_card bf5xx_ad1938;
-
-static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-
-	cpu_dai->private_data = sport_handle;
-	return 0;
-}
-
-static int bf5xx_ad1938_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->dai->cpu_dai;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7};
-	int ret = 0;
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-		SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
-		SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set codec DAI slots, 8 channels, all channels are enabled */
-	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI channel mapping */
-	ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
-		channel_map, ARRAY_SIZE(channel_map), channel_map);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static struct snd_soc_ops bf5xx_ad1938_ops = {
-	.startup = bf5xx_ad1938_startup,
-	.hw_params = bf5xx_ad1938_hw_params,
-};
-
-static struct snd_soc_dai_link bf5xx_ad1938_dai = {
-	.name = "ad1938",
-	.stream_name = "AD1938",
-	.cpu_dai = &bf5xx_tdm_dai,
-	.codec_dai = &ad1938_dai,
-	.ops = &bf5xx_ad1938_ops,
-};
-
-static struct snd_soc_card bf5xx_ad1938 = {
-	.name = "bf5xx_ad1938",
-	.platform = &bf5xx_tdm_soc_platform,
-	.dai_link = &bf5xx_ad1938_dai,
-	.num_links = 1,
-};
-
-static struct snd_soc_device bf5xx_ad1938_snd_devdata = {
-	.card = &bf5xx_ad1938,
-	.codec_dev = &soc_codec_dev_ad1938,
-};
-
-static struct platform_device *bfxx_ad1938_snd_device;
-
-static int __init bf5xx_ad1938_init(void)
-{
-	int ret;
-
-	bfxx_ad1938_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!bfxx_ad1938_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(bfxx_ad1938_snd_device, &bf5xx_ad1938_snd_devdata);
-	bf5xx_ad1938_snd_devdata.dev = &bfxx_ad1938_snd_device->dev;
-	ret = platform_device_add(bfxx_ad1938_snd_device);
-
-	if (ret)
-		platform_device_put(bfxx_ad1938_snd_device);
-
-	return ret;
-}
-
-static void __exit bf5xx_ad1938_exit(void)
-{
-	platform_device_unregister(bfxx_ad1938_snd_device);
-}
-
-module_init(bf5xx_ad1938_init);
-module_exit(bf5xx_ad1938_exit);
-
-/* Module information */
-MODULE_AUTHOR("Barry Song");
-MODULE_DESCRIPTION("ALSA SoC AD1938 board driver");
-MODULE_LICENSE("GPL");
-
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
new file mode 100644
index 000000000000..b8c9060cfd8e
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -0,0 +1,149 @@
+/*
+ * File:         sound/soc/blackfin/bf5xx-ad193x.c
+ * Author:       Barry Song <Barry.Song@analog.com>
+ *
+ * Created:      Thur June 4 2009
+ * Description:  Board driver for ad193x sound chip
+ *
+ * Bugs:         Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+#include <asm/irq.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad193x.h"
+#include "bf5xx-sport.h"
+
+#include "bf5xx-tdm-pcm.h"
+#include "bf5xx-tdm.h"
+
+static struct snd_soc_card bf5xx_ad193x;
+
+static int bf5xx_ad193x_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	cpu_dai->private_data = sport_handle;
+	return 0;
+}
+
+static int bf5xx_ad193x_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->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7};
+	int ret = 0;
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+		SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
+		SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set codec DAI slots, 8 channels, all channels are enabled */
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI channel mapping */
+	ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+		channel_map, ARRAY_SIZE(channel_map), channel_map);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops bf5xx_ad193x_ops = {
+	.startup = bf5xx_ad193x_startup,
+	.hw_params = bf5xx_ad193x_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ad193x_dai = {
+	.name = "ad193x",
+	.stream_name = "AD193X",
+	.cpu_dai = &bf5xx_tdm_dai,
+	.codec_dai = &ad193x_dai,
+	.ops = &bf5xx_ad193x_ops,
+};
+
+static struct snd_soc_card bf5xx_ad193x = {
+	.name = "bf5xx_ad193x",
+	.platform = &bf5xx_tdm_soc_platform,
+	.dai_link = &bf5xx_ad193x_dai,
+	.num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ad193x_snd_devdata = {
+	.card = &bf5xx_ad193x,
+	.codec_dev = &soc_codec_dev_ad193x,
+};
+
+static struct platform_device *bfxx_ad193x_snd_device;
+
+static int __init bf5xx_ad193x_init(void)
+{
+	int ret;
+
+	bfxx_ad193x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!bfxx_ad193x_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(bfxx_ad193x_snd_device, &bf5xx_ad193x_snd_devdata);
+	bf5xx_ad193x_snd_devdata.dev = &bfxx_ad193x_snd_device->dev;
+	ret = platform_device_add(bfxx_ad193x_snd_device);
+
+	if (ret)
+		platform_device_put(bfxx_ad193x_snd_device);
+
+	return ret;
+}
+
+static void __exit bf5xx_ad193x_exit(void)
+{
+	platform_device_unregister(bfxx_ad193x_snd_device);
+}
+
+module_init(bf5xx_ad193x_init);
+module_exit(bf5xx_ad193x_exit);
+
+/* Module information */
+MODULE_AUTHOR("Barry Song");
+MODULE_DESCRIPTION("ALSA SoC AD193X board driver");
+MODULE_LICENSE("GPL");
+
-- 
cgit v1.2.3


From 093208f5d03980d7216b706e3c54432d0f299e26 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 16 Mar 2010 16:11:14 +0000
Subject: ASoC: Hook up microphone jack detection on 1133-EV1 board

Note that since all the microphones share a bias there is a single
jack exported for all three, even though there are two physical
connectors plus the soldered down silicon mic.  Note also that the SiMic
is always present by default.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/imx/wm1133-ev1.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
index 0af987e94223..cec8bae7ac18 100644
--- a/sound/soc/imx/wm1133-ev1.c
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -203,6 +203,13 @@ static struct snd_soc_jack_pin hp_jack_pins[] = {
 	{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
 };
 
+static struct snd_soc_jack mic_jack;
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
+	{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
+};
+
 static int wm1133_ev1_init(struct snd_soc_codec *codec)
 {
 	struct snd_soc_card *card = codec->socdev->card;
@@ -219,6 +226,14 @@ static int wm1133_ev1_init(struct snd_soc_codec *codec)
 			      hp_jack_pins);
 	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
 
+	/* Microphone jack detection */
+	snd_soc_jack_new(card, "Microphone",
+			 SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
+			       SND_JACK_BTN_0);
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From fd23b7dee5e4d369f620979cb120f53629389355 Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Fri, 19 Mar 2010 14:52:55 +0000
Subject: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream

This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.

All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.

Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h           | 18 +++++++++++++++++-
 include/sound/soc.h               |  1 +
 sound/soc/atmel/atmel-pcm.c       |  2 +-
 sound/soc/atmel/atmel_ssc_dai.c   |  6 +++---
 sound/soc/davinci/davinci-i2s.c   |  3 ++-
 sound/soc/davinci/davinci-mcasp.c |  3 ++-
 sound/soc/davinci/davinci-pcm.c   |  4 +++-
 sound/soc/davinci/davinci-vcif.c  |  3 ++-
 sound/soc/imx/imx-pcm-dma-mx2.c   |  8 ++++++--
 sound/soc/imx/imx-ssi.c           |  7 +++++--
 sound/soc/omap/omap-mcbsp.c       |  4 +++-
 sound/soc/omap/omap-mcpdm.c       |  3 ++-
 sound/soc/omap/omap-pcm.c         |  4 +++-
 sound/soc/pxa/pxa-ssp.c           | 23 +++++++++++-----------
 sound/soc/pxa/pxa2xx-ac97.c       | 17 ++++++++++++-----
 sound/soc/pxa/pxa2xx-i2s.c        |  7 +++++--
 sound/soc/pxa/pxa2xx-pcm.c        |  4 +++-
 sound/soc/s3c24xx/s3c-ac97.c      | 21 +++++++++++---------
 sound/soc/s3c24xx/s3c-dma.c       |  4 +++-
 sound/soc/s3c24xx/s3c-i2s-v2.c    | 13 ++++++++-----
 sound/soc/s3c24xx/s3c-pcm.c       |  7 +++++--
 sound/soc/s3c24xx/s3c2412-i2s.c   |  7 +++++--
 sound/soc/s3c24xx/s3c24xx-i2s.c   | 19 ++++++++++---------
 sound/soc/s6000/s6000-i2s.c       |  3 ++-
 sound/soc/s6000/s6000-pcm.c       | 40 ++++++++++++++++++++++++++++-----------
 25 files changed, 156 insertions(+), 75 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 6cf76a41501e..377693a14385 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -224,7 +224,6 @@ struct snd_soc_dai {
 	struct snd_soc_codec *codec;
 	unsigned int active;
 	unsigned char pop_wait:1;
-	void *dma_data;
 
 	/* DAI private data */
 	void *private_data;
@@ -235,4 +234,21 @@ struct snd_soc_dai {
 	struct list_head list;
 };
 
+static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
+					     const struct snd_pcm_substream *ss)
+{
+	return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		dai->playback.dma_data : dai->capture.dma_data;
+}
+
+static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
+					    const struct snd_pcm_substream *ss,
+					    void *data)
+{
+	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dai->playback.dma_data = data;
+	else
+		dai->capture.dma_data = data;
+}
+
 #endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index dbfec16015de..b8bac6ae6244 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -376,6 +376,7 @@ struct snd_soc_pcm_stream {
 	unsigned int channels_min;	/* min channels */
 	unsigned int channels_max;	/* max channels */
 	unsigned int active;		/* num of active users of the stream */
+	void *dma_data;			/* used by platform code */
 };
 
 /* SoC audio ops */
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index fdb255372127..f6b3cc04b34b 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	runtime->dma_bytes = params_buffer_bytes(params);
 
-	prtd->params = rtd->dai->cpu_dai->dma_data;
+	prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	prtd->params->dma_intr_handler = atmel_pcm_dma_irq;
 
 	prtd->dma_buffer = runtime->dma_addr;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e588e63f18d2..0b59806905d1 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 	ssc_p->dma_params[dir] = dma_params;
 
 	/*
-	 * The cpu_dai->dma_data field is only used to communicate the
-	 * appropriate DMA parameters to the pcm driver hw_params()
+	 * The snd_soc_pcm_stream->dma_data field is only used to communicate
+	 * the appropriate DMA parameters to the pcm driver hw_params()
 	 * function.  It should not be used for other purposes
 	 * as it is common to all substreams.
 	 */
-	rtd->dai->cpu_dai->dma_data = dma_params;
+	snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params);
 
 	channels = params_channels(params);
 
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 6362ca05506e..4aad7ecc90a2 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -585,7 +585,8 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 
 	davinci_i2s_dai.private_data = dev;
-	davinci_i2s_dai.dma_data = dev->dma_params;
+	davinci_i2s_dai.capture.dma_data = dev->dma_params;
+	davinci_i2s_dai.playback.dma_data = dev->dma_params;
 	ret = snd_soc_register_dai(&davinci_i2s_dai);
 	if (ret != 0)
 		goto err_free_mem;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index ab6518d86f18..c056bfbe0340 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -917,7 +917,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	dma_data->channel = res->start;
 	davinci_mcasp_dai[pdata->op_mode].private_data = dev;
-	davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params;
+	davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params;
+	davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params;
 	davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
 	ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 80c7fdf2f521..2dc406f42fe7 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_hardware *ppcm;
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data;
+	struct davinci_pcm_dma_params *pa;
 	struct davinci_pcm_dma_params *params;
+
+	pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	if (!pa)
 		return -ENODEV;
 	params = &pa[substream->stream];
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 03f3feb0fe2f..54b91e1768c4 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -222,7 +222,8 @@ static int davinci_vcif_probe(struct platform_device *pdev)
 					davinci_vc->davinci_vcif.dma_rx_addr;
 
 	davinci_vcif_dai.dev = &pdev->dev;
-	davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params;
+	davinci_vcif_dai.capture.dma_data = davinci_vcif_dev->dma_params;
+	davinci_vcif_dai.playback.dma_data = davinci_vcif_dev->dma_params;
 	davinci_vcif_dai.private_data = davinci_vcif_dev;
 
 	ret = snd_soc_register_dai(&davinci_vcif_dai);
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 19452e44afdc..c78c000e2afe 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -83,11 +83,13 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err)
 static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+	struct imx_pcm_dma_params *dma_params;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int ret;
 
+	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
 	if (iprtd->dma < 0) {
 		pr_err("Failed to claim the audio DMA\n");
@@ -192,10 +194,12 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+	struct imx_pcm_dma_params *dma_params;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int err;
 
+	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	iprtd->substream = substream;
 	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
 	iprtd->period_cnt = 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 56f46a75d297..28e55c7b14b4 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -234,17 +234,20 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *cpu_dai)
 {
 	struct imx_ssi *ssi = cpu_dai->private_data;
+	struct imx_pcm_dma_params *dma_data;
 	u32 reg, sccr;
 
 	/* Tx/Rx config */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		reg = SSI_STCCR;
-		cpu_dai->dma_data = &ssi->dma_params_tx;
+		dma_data = &ssi->dma_params_tx;
 	} else {
 		reg = SSI_SRCCR;
-		cpu_dai->dma_data = &ssi->dma_params_rx;
+		dma_data = &ssi->dma_params_rx;
 	}
 
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
 	sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;
 
 	/* DAI data (word) size */
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 0199e95a5d63..2d33a89f147a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -322,7 +322,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
 	omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
 							OMAP_DMA_DATA_TYPE_S16;
-	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream,
+		&omap_mcbsp_dai_dma_params[id][substream->stream]);
 
 	if (mcbsp_data->configured) {
 		/* McBSP already configured by another stream */
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 25f19e4728bf..b7f4f7e015f3 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -150,7 +150,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 	int stream = substream->stream;
 	int channels, err, link_mask = 0;
 
-	cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream];
+	snd_soc_dai_set_dma_data(cpu_dai, substream,
+				 &omap_mcpdm_dai_dma_params[stream]);
 
 	channels = params_channels(params);
 	switch (channels) {
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 825db385f01f..39538c0f81f0 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -100,9 +100,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct omap_runtime_data *prtd = runtime->private_data;
-	struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+	struct omap_pcm_dma_data *dma_data;
 	int err = 0;
 
+	dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
 	if (!dma_data)
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index e69397f40f72..5d65a00e4bc0 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -103,10 +103,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 		ssp_disable(&priv->dev);
 	}
 
-	if (cpu_dai->dma_data) {
-		kfree(cpu_dai->dma_data);
-		cpu_dai->dma_data = NULL;
-	}
+	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
+
 	return ret;
 }
 
@@ -122,10 +121,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 		clk_disable(priv->dev.ssp->clk);
 	}
 
-	if (cpu_dai->dma_data) {
-		kfree(cpu_dai->dma_data);
-		cpu_dai->dma_data = NULL;
-	}
+	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
+	snd_soc_dai_set_dma_data(cpu_dai, substream, NULL);
 }
 
 #ifdef CONFIG_PM
@@ -538,19 +535,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	u32 sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
 	int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
+	struct pxa2xx_pcm_dma_params *dma_data;
+
+	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 
 	/* generate correct DMA params */
-	if (cpu_dai->dma_data)
-		kfree(cpu_dai->dma_data);
+	kfree(dma_data);
 
 	/* Network mode with one active slot (ttsa == 1) can be used
 	 * to force 16-bit frame width on the wire (for S16_LE), even
 	 * with two channels. Use 16-bit DMA transfers for this case.
 	 */
-	cpu_dai->dma_data = ssp_get_dma_params(ssp,
+	dma_data = ssp_get_dma_params(ssp,
 			((chn == 2) && (ttsa != 1)) || (width == 32),
 			substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
+	snd_soc_dai_set_dma_data(dai, substream, dma_data);
+
 	/* we can only change the settings if the port is not in use */
 	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
 		return 0;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index e9ae7b3a7e00..d314115e3dd7 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct pxa2xx_pcm_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+		dma_data = &pxa2xx_ac97_pcm_stereo_out;
 	else
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+		dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	return 0;
 }
@@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct pxa2xx_pcm_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+		dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
 	else
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+		dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	return 0;
 }
@@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
 	else
-		cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+		snd_soc_dai_set_dma_data(cpu_dai, substream,
+					 &pxa2xx_ac97_pcm_mic_mono_in);
 
 	return 0;
 }
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 6b8f655d1ad8..c1a5275721e4 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct pxa2xx_pcm_dma_params *dma_data;
 
 	BUG_ON(IS_ERR(clk_i2s));
 	clk_enable(clk_i2s);
@@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	pxa_i2s_wait();
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+		dma_data = &pxa2xx_i2s_pcm_stereo_out;
 	else
-		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+		dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	/* is port used by another stream */
 	if (!(SACR0 & SACR0_ENB)) {
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index d38e39575f51..adc7e6f15f93 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct pxa2xx_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+	struct pxa2xx_pcm_dma_params *dma;
 	int ret;
 
+	dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
 	if (!dma)
diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c
index ee8ed9d7e703..ecf4fd04ae96 100644
--- a/sound/soc/s3c24xx/s3c-ac97.c
+++ b/sound/soc/s3c24xx/s3c-ac97.c
@@ -224,11 +224,14 @@ static int s3c_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct s3c_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = &s3c_ac97_pcm_out;
+		dma_data = &s3c_ac97_pcm_out;
 	else
-		cpu_dai->dma_data = &s3c_ac97_pcm_in;
+		dma_data = &s3c_ac97_pcm_in;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	return 0;
 }
@@ -238,8 +241,8 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	u32 ac_glbctrl;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -265,7 +268,7 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
 
-	s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+	s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 
 	return 0;
 }
@@ -280,7 +283,7 @@ static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
 	else
-		cpu_dai->dma_data = &s3c_ac97_mic_in;
+		snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in);
 
 	return 0;
 }
@@ -290,8 +293,8 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 {
 	u32 ac_glbctrl;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL);
 	ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK;
@@ -311,7 +314,7 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream,
 
 	writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL);
 
-	s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+	s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 
 	return 0;
 }
diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c
index 7725e26d6c91..1b61c23ff300 100644
--- a/sound/soc/s3c24xx/s3c-dma.c
+++ b/sound/soc/s3c24xx/s3c-dma.c
@@ -145,10 +145,12 @@ static int s3c_dma_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data;
 	unsigned long totbytes = params_buffer_bytes(params);
+	struct s3c_dma_params *dma =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	int ret = 0;
 
+
 	pr_debug("Entered %s\n", __func__);
 
 	/* return if this is a bufferless transfer e.g.
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index c3fcb63cbf25..865f93143bf1 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -333,14 +333,17 @@ static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_link *dai = rtd->dai;
 	struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai);
+	struct s3c_dma_params *dma_data;
 	u32 iismod;
 
 	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dai->cpu_dai->dma_data = i2s->dma_playback;
+		dma_data = i2s->dma_playback;
 	else
-		dai->cpu_dai->dma_data = i2s->dma_capture;
+		dma_data = i2s->dma_capture;
+
+	snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data);
 
 	/* Working copies of register */
 	iismod = readl(i2s->regs + S3C2412_IISMOD);
@@ -372,8 +375,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
 	int ret = 0;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -409,7 +412,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		 * of the auto reload mechanism of S3C24XX.
 		 * This call won't bother S3C64XX.
 		 */
-		s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+		s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 
 		break;
 
diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c
index a98f40c3cd29..326f0a9e7e30 100644
--- a/sound/soc/s3c24xx/s3c-pcm.c
+++ b/sound/soc/s3c24xx/s3c-pcm.c
@@ -178,6 +178,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai_link *dai = rtd->dai;
 	struct s3c_pcm_info *pcm = to_info(dai->cpu_dai);
+	struct s3c_dma_params *dma_data;
 	void __iomem *regs = pcm->regs;
 	struct clk *clk;
 	int sclk_div, sync_div;
@@ -187,9 +188,11 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
 	dev_dbg(pcm->dev, "Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dai->cpu_dai->dma_data = pcm->dma_playback;
+		dma_data = pcm->dma_playback;
 	else
-		dai->cpu_dai->dma_data = pcm->dma_capture;
+		dma_data = pcm->dma_capture;
+
+	snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data);
 
 	/* Strictly check for sample size */
 	switch (params_format(params)) {
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index a5b21f63261f..f3148f98b419 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -151,14 +151,17 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *cpu_dai)
 {
 	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	struct s3c_dma_params *dma_data;
 	u32 iismod;
 
 	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		cpu_dai->dma_data = i2s->dma_playback;
+		dma_data = i2s->dma_playback;
 	else
-		cpu_dai->dma_data = i2s->dma_capture;
+		dma_data = i2s->dma_capture;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	iismod = readl(i2s->regs + S3C2412_IISMOD);
 	pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 0bc5950b9f02..c3ac890a3986 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -242,14 +242,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s3c_dma_params *dma_data;
 	u32 iismod;
 
 	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
+		dma_data = &s3c24xx_i2s_pcm_stereo_out;
 	else
-		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in;
+		dma_data = &s3c24xx_i2s_pcm_stereo_in;
+
+	snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data);
 
 	/* Working copies of register */
 	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -258,13 +261,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		iismod &= ~S3C2410_IISMOD_16BIT;
-		((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->dma_size = 1;
+		dma_data->dma_size = 1;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		iismod |= S3C2410_IISMOD_16BIT;
-		((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->dma_size = 2;
+		dma_data->dma_size = 2;
 		break;
 	default:
 		return -EINVAL;
@@ -280,8 +281,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c_dma_params *)
-		  rtd->dai->cpu_dai->dma_data)->channel;
+	struct s3c_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -300,7 +301,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 		else
 			s3c24xx_snd_txctrl(1);
 
-		s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
+		s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index c5cda187ecab..fa23854c5f3a 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -518,7 +518,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
 
 	s6000_i2s_dai.dev = &pdev->dev;
 	s6000_i2s_dai.private_data = dev;
-	s6000_i2s_dai.dma_data = &dev->dma_params;
+	s6000_i2s_dai.capture.dma_data = &dev->dma_params;
+	s6000_i2s_dai.playback.dma_data = &dev->dma_params;
 
 	dev->sifbase = sifmem->start;
 	dev->scbbase = mmio;
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 1d61109e09fa..9c7f7f00cebb 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -58,13 +58,15 @@ static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s6000_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	int channel;
 	unsigned int period_size;
 	unsigned int dma_offset;
 	dma_addr_t dma_pos;
 	dma_addr_t src, dst;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	period_size = snd_pcm_lib_period_bytes(substream);
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
@@ -101,7 +103,8 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data)
 {
 	struct snd_pcm *pcm = data;
 	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
-	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *params =
+		snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 	struct s6000_runtime_data *prtd;
 	unsigned int has_xrun;
 	int i, ret = IRQ_NONE;
@@ -172,11 +175,13 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream)
 {
 	struct s6000_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	unsigned long flags;
 	int srcinc;
 	u32 dma;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	spin_lock_irqsave(&prtd->lock, flags);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -212,10 +217,12 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream)
 {
 	struct s6000_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	unsigned long flags;
 	u32 channel;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		channel = par->dma_out;
 	else
@@ -236,9 +243,11 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream)
 static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	int ret;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	ret = par->trigger(substream, cmd, 0);
 	if (ret < 0)
 		return ret;
@@ -275,13 +284,15 @@ static int s6000_pcm_prepare(struct snd_pcm_substream *substream)
 static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s6000_runtime_data *prtd = runtime->private_data;
 	unsigned long flags;
 	unsigned int offset;
 	dma_addr_t count;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	spin_lock_irqsave(&prtd->lock, flags);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -305,11 +316,12 @@ static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
 static int s6000_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s6000_runtime_data *prtd;
 	int ret;
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 	snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware);
 
 	ret = snd_pcm_hw_constraint_step(runtime, 0,
@@ -364,7 +376,7 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par;
 	int ret;
 	ret = snd_pcm_lib_malloc_pages(substream,
 				       params_buffer_bytes(hw_params));
@@ -373,6 +385,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	if (par->same_rate) {
 		spin_lock(&par->lock);
 		if (par->rate == -1 ||
@@ -392,7 +406,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
 static int s6000_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
-	struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *par =
+		snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 
 	spin_lock(&par->lock);
 	par->in_use &= ~(1 << substream->stream);
@@ -417,7 +432,8 @@ static struct snd_pcm_ops s6000_pcm_ops = {
 static void s6000_pcm_free(struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
-	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *params =
+		snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
 
 	free_irq(params->irq, pcm);
 	snd_pcm_lib_preallocate_free_for_all(pcm);
@@ -429,9 +445,11 @@ static int s6000_pcm_new(struct snd_card *card,
 			 struct snd_soc_dai *dai, struct snd_pcm *pcm)
 {
 	struct snd_soc_pcm_runtime *runtime = pcm->private_data;
-	struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+	struct s6000_pcm_dma_params *params;
 	int res;
 
+	params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream);
+
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &s6000_pcm_dmamask;
 	if (!card->dev->coherent_dma_mask)
-- 
cgit v1.2.3


From b2dfa62c525fbe4d89790b6fa90af7c3f1c052ae Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Date: Thu, 18 Mar 2010 08:23:33 +0100
Subject: ASoC: remove a card from the list, if instantiation failed

If instantiation of a card failed, we still have to remove it from the
card list on unregistration. This fixes an Oops on Migo-R, triggering,
when after a failed firmware load attempt the driver modules are removed
and re-inserted again.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 27 +++++++++++++--------------
 1 file changed, 13 insertions(+), 14 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index eb73aab78b94..1ff553bbc063 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1283,26 +1283,25 @@ static int soc_remove(struct platform_device *pdev)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
-	if (!card->instantiated)
-		return 0;
+	if (card->instantiated) {
+		run_delayed_work(&card->delayed_work);
 
-	run_delayed_work(&card->delayed_work);
+		if (platform->remove)
+			platform->remove(pdev);
 
-	if (platform->remove)
-		platform->remove(pdev);
+		if (codec_dev->remove)
+			codec_dev->remove(pdev);
 
-	if (codec_dev->remove)
-		codec_dev->remove(pdev);
+		for (i = 0; i < card->num_links; i++) {
+			struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+			if (cpu_dai->remove)
+				cpu_dai->remove(pdev, cpu_dai);
+		}
 
-	for (i = 0; i < card->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
-		if (cpu_dai->remove)
-			cpu_dai->remove(pdev, cpu_dai);
+		if (card->remove)
+			card->remove(pdev);
 	}
 
-	if (card->remove)
-		card->remove(pdev);
-
 	snd_soc_unregister_card(card);
 
 	return 0;
-- 
cgit v1.2.3


From c96907f21f26a1f8c1a1a9096a22500e4d158c4f Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Mon, 22 Mar 2010 17:46:37 +0200
Subject: ASoC: TWL4030: PM fix for output amplifiers

Gain controls on outputs affect the power consumption
when the gain is set to non 0 value.

Outputs with amps have one register to configure the
routing and the gain:
PREDL_CTL (0x25):
bit 0: Voice enable
bit 1: Audio L1 enable
bit 2: Audio L2 enable
bit 3: Audio R2 enable
bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB)

bit 0 - 3: is handled in DAPM domain (DAPM_MIXER)
bit 4 - 5: has simple volume control

If there is no audio activity (BIAS_STANDBY), and
user changes the volume, than the output amplifier will
be enabled.
If the user changes the routing (but the codec remains in
BIAS_STANDBY), than the cached gain value also be written
to the register, which enables the amplifier.

The existing workaround for this is to have virtual
PGAs associated with the outputs, and whit DAPM PMD
the gain on the output will be forced to 0 (off) by
bypassing the regcache.
This failed to disable the amplifiers in several
scenario (as mentioned above).

Also if the codec is in BIAS_ON state, and user modifies
a volume control, which path is actually not enabled, than
that amplifier will be enabled as well, but it will
be not turned off, since there is no DAPM path, which
would make mute it.

To prevent amps being enabled, when they are not
needed, introduce the following workaround:
Track the state of each of this type of output.
In twl4030_write only allow actual write, when the
given output is enabled, otherwise only update
the reg_cache.
The PGA event handlers on power up will write the cached
value to the chip (restoring gain, routing selection).
On power down 0 is written to the register (disabling
the amp, and also just in case clearing the routing).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 72 +++++++++++++++++++++++++++++++++++-----------
 1 file changed, 56 insertions(+), 16 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 6f5d4af20052..bf59b8a4d1d7 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -135,9 +135,11 @@ struct twl4030_priv {
 
 	unsigned int sysclk;
 
-	/* Headset output state handling */
-	unsigned int hsl_enabled;
-	unsigned int hsr_enabled;
+	/* Output (with associated amp) states */
+	u8 hsl_enabled, hsr_enabled;
+	u8 earpiece_enabled;
+	u8 predrivel_enabled, predriver_enabled;
+	u8 carkitl_enabled, carkitr_enabled;
 };
 
 /*
@@ -173,12 +175,47 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
 static int twl4030_write(struct snd_soc_codec *codec,
 			unsigned int reg, unsigned int value)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
+	int write_to_reg = 0;
+
 	twl4030_write_reg_cache(codec, reg, value);
-	if (likely(reg < TWL4030_REG_SW_SHADOW))
-		return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
-					    reg);
-	else
-		return 0;
+	if (likely(reg < TWL4030_REG_SW_SHADOW)) {
+		/* Decide if the given register can be written */
+		switch (reg) {
+		case TWL4030_REG_EAR_CTL:
+			if (twl4030->earpiece_enabled)
+				write_to_reg = 1;
+			break;
+		case TWL4030_REG_PREDL_CTL:
+			if (twl4030->predrivel_enabled)
+				write_to_reg = 1;
+			break;
+		case TWL4030_REG_PREDR_CTL:
+			if (twl4030->predriver_enabled)
+				write_to_reg = 1;
+			break;
+		case TWL4030_REG_PRECKL_CTL:
+			if (twl4030->carkitl_enabled)
+				write_to_reg = 1;
+			break;
+		case TWL4030_REG_PRECKR_CTL:
+			if (twl4030->carkitr_enabled)
+				write_to_reg = 1;
+			break;
+		case TWL4030_REG_HS_GAIN_SET:
+			if (twl4030->hsl_enabled || twl4030->hsr_enabled)
+				write_to_reg = 1;
+			break;
+		default:
+			/* All other register can be written */
+			write_to_reg = 1;
+			break;
+		}
+		if (write_to_reg)
+			return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+						    value, reg);
+	}
+	return 0;
 }
 
 static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
@@ -525,26 +562,26 @@ static int micpath_event(struct snd_soc_dapm_widget *w,
  * Output PGA builder:
  * Handle the muting and unmuting of the given output (turning off the
  * amplifier associated with the output pin)
- * On mute bypass the reg_cache and mute the volume
- * On unmute: restore the register content
+ * On mute bypass the reg_cache and write 0 to the register
+ * On unmute: restore the register content from the reg_cache
  * Outputs handled in this way:  Earpiece, PreDrivL/R, CarkitL/R
  */
 #define TWL4030_OUTPUT_PGA(pin_name, reg, mask)				\
 static int pin_name##pga_event(struct snd_soc_dapm_widget *w,		\
 		struct snd_kcontrol *kcontrol, int event)		\
 {									\
-	u8 reg_val;							\
+	struct twl4030_priv *twl4030 = w->codec->private_data;		\
 									\
 	switch (event) {						\
 	case SND_SOC_DAPM_POST_PMU:					\
+		twl4030->pin_name##_enabled = 1;			\
 		twl4030_write(w->codec, reg,				\
 			twl4030_read_reg_cache(w->codec, reg));		\
 		break;							\
 	case SND_SOC_DAPM_POST_PMD:					\
-		reg_val = twl4030_read_reg_cache(w->codec, reg);	\
-		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,	\
-					reg_val & (~mask),		\
-					reg);				\
+		twl4030->pin_name##_enabled = 0;			\
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,		\
+					0, reg);			\
 		break;							\
 	}								\
 	return 0;							\
@@ -664,7 +701,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 		/* Headset ramp-up according to the TRM */
 		hs_pop |= TWL4030_VMID_EN;
 		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-		twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
+		/* Actually write to the register */
+		twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					hs_gain,
+					TWL4030_REG_HS_GAIN_SET);
 		hs_pop |= TWL4030_RAMP_EN;
 		twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
 		/* Wait ramp delay time + 1, so the VMID can settle */
-- 
cgit v1.2.3


From d5021ec9fc32edc6f512c2375923d757e9825f6a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 22 Mar 2010 12:06:30 +0000
Subject: ASoC: Add a notifier for jack status changes

Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.

Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/soc.h  |  6 ++++++
 sound/soc/soc-jack.c | 38 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 44 insertions(+)

(limited to 'sound/soc')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index b8bac6ae6244..80dfac162723 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -15,6 +15,7 @@
 
 #include <linux/platform_device.h>
 #include <linux/types.h>
+#include <linux/notifier.h>
 #include <linux/workqueue.h>
 #include <linux/interrupt.h>
 #include <linux/kernel.h>
@@ -261,6 +262,10 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
 int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 			  struct snd_soc_jack_pin *pins);
+void snd_soc_jack_notifier_register(struct snd_soc_jack *jack,
+				    struct notifier_block *nb);
+void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack,
+				      struct notifier_block *nb);
 #ifdef CONFIG_GPIOLIB
 int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios);
@@ -364,6 +369,7 @@ struct snd_soc_jack {
 	struct snd_soc_card *card;
 	struct list_head pins;
 	int status;
+	struct blocking_notifier_head notifier;
 };
 
 /* SoC PCM stream information */
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 3c07a94c2e30..f8fd22cc70bc 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
 {
 	jack->card = card;
 	INIT_LIST_HEAD(&jack->pins);
+	BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier);
 
 	return snd_jack_new(card->codec->card, id, type, &jack->jack);
 }
@@ -93,6 +94,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 			snd_soc_dapm_disable_pin(codec, pin->pin);
 	}
 
+	/* Report before the DAPM sync to help users updating micbias status */
+	blocking_notifier_call_chain(&jack->notifier, status, NULL);
+
 	snd_soc_dapm_sync(codec);
 
 	snd_jack_report(jack->jack, status);
@@ -143,6 +147,40 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 }
 EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins);
 
+/**
+ * snd_soc_jack_notifier_register - Register a notifier for jack status
+ *
+ * @jack:  ASoC jack
+ * @nb:    Notifier block to register
+ *
+ * Register for notification of the current status of the jack.  Note
+ * that it is not possible to report additional jack events in the
+ * callback from the notifier, this is intended to support
+ * applications such as enabling electrical detection only when a
+ * mechanical detection event has occurred.
+ */
+void snd_soc_jack_notifier_register(struct snd_soc_jack *jack,
+				    struct notifier_block *nb)
+{
+	blocking_notifier_chain_register(&jack->notifier, nb);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_register);
+
+/**
+ * snd_soc_jack_notifier_unregister - Unregister a notifier for jack status
+ *
+ * @jack:  ASoC jack
+ * @nb:    Notifier block to unregister
+ *
+ * Stop notifying for status changes.
+ */
+void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack,
+				      struct notifier_block *nb)
+{
+	blocking_notifier_chain_unregister(&jack->notifier, nb);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister);
+
 #ifdef CONFIG_GPIOLIB
 /* gpio detect */
 static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
-- 
cgit v1.2.3


From 5b9e87ccccf77f46c006c2cf0988a66d0f3f310d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 22 Mar 2010 13:36:13 +0000
Subject: ASoC: Allow force enabled pins to be disabled

Some systems, such as those with mechanical jack detection, may wish
to force enable a pin (typically mic bias) only some of the time.
Support such systems by having disable_pin() also coveer force enabled
pins.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/soc-dapm.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 611284194000..476dbe6dabad 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1266,6 +1266,9 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
 		if (!strcmp(w->name, pin)) {
 			pr_debug("dapm: %s: pin %s\n", codec->name, pin);
 			w->connected = status;
+			/* Allow disabling of forced pins */
+			if (status == 0)
+				w->force = 0;
 			return 0;
 		}
 	}
-- 
cgit v1.2.3


From 2f14430af52fff08ed1987a3cee612e133a6b82f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 22 Mar 2010 13:48:36 +0000
Subject: ASoC: Move WM8350 microphone detection bias managment out of driver

Allow machines to control exactly when the bias is turned on and off.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8350.c  | 2 --
 sound/soc/imx/wm1133-ev1.c | 2 ++
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ff6c10155ee6..fb1005edee79 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1475,8 +1475,6 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
 	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
 	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_1, WM8350_MIC_DET_ENA);
 
-	snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
-
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c
index cec8bae7ac18..a6e7d9497639 100644
--- a/sound/soc/imx/wm1133-ev1.c
+++ b/sound/soc/imx/wm1133-ev1.c
@@ -234,6 +234,8 @@ static int wm1133_ev1_init(struct snd_soc_codec *codec)
 	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
 			       SND_JACK_BTN_0);
 
+	snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From f06bce9c8c54354ea4ff96852bd92e10204c8cfb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 22 Mar 2010 15:30:30 +0000
Subject: ASoC: Allow disabling of WM835x jack detection

If no report is specified then disable detection. Note that we don't
disable the slow clock, though the power consumption from it should
be negligable. That should be reference counted, ideally through DAPM.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8350.c | 24 ++++++++++++++++++------
 1 file changed, 18 insertions(+), 6 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index fb1005edee79..8ac92f5e4f91 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1393,7 +1393,8 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
  * @jack:   jack to report detection events on
  * @report: value to report
  *
- * Enables the headphone jack detection of the WM8350.
+ * Enables the headphone jack detection of the WM8350.  If no report
+ * is specified then detection is disabled.
  */
 int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 			  struct snd_soc_jack *jack, int report)
@@ -1422,8 +1423,12 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 		return -EINVAL;
 	}
 
-	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
-	wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
+	if (report) {
+		wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+		wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
+	} else {
+		wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, ena);
+	}
 
 	/* Sync status */
 	wm8350_hp_jack_handler(irq + wm8350->irq_base, priv);
@@ -1459,7 +1464,8 @@ static irqreturn_t wm8350_mic_handler(int irq, void *data)
  * @detect_report: value to report when presence detected
  * @short_report:  value to report when microphone short detected
  *
- * Enables the microphone jack detection of the WM8350.
+ * Enables the microphone jack detection of the WM8350.  If both reports
+ * are specified as zero then detection is disabled.
  */
 int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
 			   struct snd_soc_jack *jack,
@@ -1472,8 +1478,14 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
 	priv->mic.report = detect_report;
 	priv->mic.short_report = short_report;
 
-	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
-	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_1, WM8350_MIC_DET_ENA);
+	if (detect_report || short_report) {
+		wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+		wm8350_set_bits(wm8350, WM8350_POWER_MGMT_1,
+				WM8350_MIC_DET_ENA);
+	} else {
+		wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_1,
+				  WM8350_MIC_DET_ENA);
+	}
 
 	return 0;
 }
-- 
cgit v1.2.3


From 69266866a5790080d7fe80094b28d670ff8aa765 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 22 Mar 2010 16:37:01 +0000
Subject: ASoC: Allow WM8903 mic detect disable and don't force bias on

Don't force enable the microphone bias on WM8903 when doing jack
detection, and don't force enable microphone bias. This allows
platforms to only enable microphone detection when a jack has been
inserted.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8903.c | 23 ++++++++++++-----------
 1 file changed, 12 insertions(+), 11 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b5427b47d6fd..134b17532f23 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1465,7 +1465,7 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		      int det, int shrt)
 {
 	struct wm8903_priv *wm8903 = codec->private_data;
-	int irq_mask = 0;
+	int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT;
 
 	dev_dbg(codec->dev, "Enabling microphone detection: %x %x\n",
 		det, shrt);
@@ -1485,16 +1485,17 @@ int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 			    WM8903_MICDET_EINT | WM8903_MICSHRT_EINT,
 			    irq_mask);
 
-	/* Enable mic detection, this may not have been set through
-	 * platform data (eg, if the defaults are OK). */
-	snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
-			    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
-	snd_soc_update_bits(codec, WM8903_MIC_BIAS_CONTROL_0,
-			    WM8903_MICDET_ENA, WM8903_MICDET_ENA);
-
-	/* Force the microphone bias on; this will trigger an initial
-	 * detection. */
-	snd_soc_dapm_force_enable_pin(codec, "Mic Bias");
+	if (det && shrt) {
+		/* Enable mic detection, this may not have been set through
+		 * platform data (eg, if the defaults are OK). */
+		snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
+				    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+		snd_soc_update_bits(codec, WM8903_MIC_BIAS_CONTROL_0,
+				    WM8903_MICDET_ENA, WM8903_MICDET_ENA);
+	} else {
+		snd_soc_update_bits(codec, WM8903_MIC_BIAS_CONTROL_0,
+				    WM8903_MICDET_ENA, 0);
+	}
 
 	return 0;
 }
-- 
cgit v1.2.3


From 778a76e2dbdb896d005849e9e74518d6aba85671 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 22 Mar 2010 22:05:10 +0000
Subject: ASoC: Implement WM8994 DAI tristate support

This also adds the first DAI operation for AIF3 so fill out the ID and
the ops for that too.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8994.c | 38 ++++++++++++++++++++++++++++++++++++++
 1 file changed, 38 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 29f3771c33a4..d01d3091fe81 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3332,6 +3332,36 @@ static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute)
 	return 0;
 }
 
+static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int reg, val, mask;
+
+	switch (codec_dai->id) {
+	case 1:
+		reg = WM8994_AIF1_MASTER_SLAVE;
+		mask = WM8994_AIF1_TRI;
+		break;
+	case 2:
+		reg = WM8994_AIF2_MASTER_SLAVE;
+		mask = WM8994_AIF2_TRI;
+		break;
+	case 3:
+		reg = WM8994_POWER_MANAGEMENT_6;
+		mask = WM8994_AIF3_TRI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (tristate)
+		val = mask;
+	else
+		val = 0;
+
+	return snd_soc_update_bits(codec, reg, mask, reg);
+}
+
 #define WM8994_RATES SNDRV_PCM_RATE_8000_96000
 
 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
@@ -3343,6 +3373,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
 	.hw_params	= wm8994_hw_params,
 	.digital_mute	= wm8994_aif_mute,
 	.set_pll	= wm8994_set_fll,
+	.set_tristate	= wm8994_set_tristate,
 };
 
 static struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
@@ -3351,6 +3382,11 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
 	.hw_params	= wm8994_hw_params,
 	.digital_mute   = wm8994_aif_mute,
 	.set_pll	= wm8994_set_fll,
+	.set_tristate	= wm8994_set_tristate,
+};
+
+static struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
+	.set_tristate	= wm8994_set_tristate,
 };
 
 struct snd_soc_dai wm8994_dai[] = {
@@ -3394,6 +3430,7 @@ struct snd_soc_dai wm8994_dai[] = {
 	},
 	{
 		.name = "WM8994 AIF3",
+		.id = 3,
 		.playback = {
 			.stream_name = "AIF3 Playback",
 			.channels_min = 2,
@@ -3408,6 +3445,7 @@ struct snd_soc_dai wm8994_dai[] = {
 			.rates = WM8994_RATES,
 			.formats = WM8994_FORMATS,
 		},
+		.ops = &wm8994_aif3_dai_ops,
 	}
 };
 EXPORT_SYMBOL_GPL(wm8994_dai);
-- 
cgit v1.2.3


From 4b6316b4b16c7fb5d51df43f0371416e054e7102 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Tue, 23 Mar 2010 16:27:28 +0900
Subject: ASoC: ak4642: Add pll select support

Current ak4642 was not able to select pll.
This patch add support it.
It still expect PLL base input pin is MCKI.
see Table 5 "setting of PLL Mode" of datasheet

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ak4642.c | 40 +++++++++++++++++++++++++++++++++-------
 sound/soc/sh/fsi-ak4642.c | 10 ++++++++++
 2 files changed, 43 insertions(+), 7 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 3ef16bbc8c83..d5bd4cae73a1 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -80,12 +80,18 @@
 
 #define AK4642_CACHEREGNUM 	0x25
 
+/* MD_CTL1 */
+#define PLL3		(1 << 7)
+#define PLL2		(1 << 6)
+#define PLL1		(1 << 5)
+#define PLL0		(1 << 4)
+#define PLL_MASK	(PLL3 | PLL2 | PLL1 | PLL0)
+
 struct snd_soc_codec_device soc_codec_dev_ak4642;
 
 /* codec private data */
 struct ak4642_priv {
 	struct snd_soc_codec codec;
-	unsigned int sysclk;
 };
 
 static struct snd_soc_codec *ak4642_codec;
@@ -249,9 +255,32 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
 	int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct ak4642_priv *ak4642 = codec->private_data;
+	u8 pll;
+
+	switch (freq) {
+	case 11289600:
+		pll = PLL2;
+		break;
+	case 12288000:
+		pll = PLL2 | PLL0;
+		break;
+	case 12000000:
+		pll = PLL2 | PLL1;
+		break;
+	case 24000000:
+		pll = PLL2 | PLL1 | PLL0;
+		break;
+	case 13500000:
+		pll = PLL3 | PLL2;
+		break;
+	case 27000000:
+		pll = PLL3 | PLL2 | PLL0;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
 
-	ak4642->sysclk = freq;
 	return 0;
 }
 
@@ -342,7 +371,6 @@ static int ak4642_init(struct ak4642_priv *ak4642)
 	 *
 	 * Audio I/F Format: MSB justified (ADC & DAC)
 	 * BICK frequency at Master Mode: 64fs
-	 * Input Master Clock Select at PLL Mode: 11.2896MHz
 	 * MCKO: Enable
 	 * Sampling Frequency: 44.1kHz
 	 *
@@ -352,10 +380,8 @@ static int ak4642_init(struct ak4642_priv *ak4642)
 	 * please fix-me
 	 */
 	ak4642_write(codec, 0x01, 0x08);
-	ak4642_write(codec, 0x04, 0x4a);
 	ak4642_write(codec, 0x05, 0x27);
-	ak4642_write(codec, 0x00, 0x40);
-	ak4642_write(codec, 0x01, 0x0b);
+	ak4642_write(codec, 0x04, 0x0a);
 
 	return ret;
 
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index 5263ab18f827..c0207dc524b8 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -22,11 +22,21 @@
 #include <sound/sh_fsi.h>
 #include <../sound/soc/codecs/ak4642.h>
 
+static int fsi_ak4642_dai_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0);
+
+	return ret;
+}
+
 static struct snd_soc_dai_link fsi_dai_link = {
 	.name		= "AK4642",
 	.stream_name	= "AK4642",
 	.cpu_dai	= &fsi_soc_dai[0], /* fsi */
 	.codec_dai	= &ak4642_dai,
+	.init		= fsi_ak4642_dai_init,
 	.ops		= NULL,
 };
 
-- 
cgit v1.2.3


From 0643ce8f42e763635d8d6cf311730a6e84d4be92 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Mon, 15 Mar 2010 18:10:50 +0900
Subject: ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ak4642.c | 69 ++++++++++++++++++++++++++++++-----------------
 sound/soc/sh/fsi-ak4642.c |  4 +++
 2 files changed, 49 insertions(+), 24 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index d5bd4cae73a1..3452bd752739 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -80,6 +80,17 @@
 
 #define AK4642_CACHEREGNUM 	0x25
 
+/* PW_MGMT2 */
+#define HPMTN		(1 << 6)
+#define PMHPL		(1 << 5)
+#define PMHPR		(1 << 4)
+#define MS		(1 << 3) /* master/slave select */
+#define MCKO		(1 << 1)
+#define PMPLL		(1 << 0)
+
+#define PMHP_MASK	(PMHPL | PMHPR)
+#define PMHP		PMHP_MASK
+
 /* MD_CTL1 */
 #define PLL3		(1 << 7)
 #define PLL2		(1 << 6)
@@ -87,6 +98,9 @@
 #define PLL0		(1 << 4)
 #define PLL_MASK	(PLL3 | PLL2 | PLL1 | PLL0)
 
+#define BCKO_MASK	(1 << 3)
+#define BCKO_64		BCKO_MASK
+
 struct snd_soc_codec_device soc_codec_dev_ak4642;
 
 /* codec private data */
@@ -188,9 +202,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 *
 		 * This operation came from example code of
 		 * "ASAHI KASEI AK4642" (japanese) manual p97.
-		 *
-		 * Example code use 0x39, 0x79 value for 0x01 address,
-		 * But we need MCKO (0x02) bit now
 		 */
 		ak4642_write(codec, 0x05, 0x27);
 		ak4642_write(codec, 0x0f, 0x09);
@@ -200,8 +211,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		ak4642_write(codec, 0x0a, 0x28);
 		ak4642_write(codec, 0x0d, 0x28);
 		ak4642_write(codec, 0x00, 0x64);
-		ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */
-		ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */
+		snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK,	PMHP);
+		snd_soc_update_bits(codec, PW_MGMT2, HPMTN,	HPMTN);
 	} else {
 		/*
 		 * start stereo input
@@ -238,8 +249,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
 
 	if (is_play) {
 		/* stop headphone output */
-		ak4642_write(codec, 0x01, 0x3b);
-		ak4642_write(codec, 0x01, 0x0b);
+		snd_soc_update_bits(codec, PW_MGMT2, HPMTN,	0);
+		snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK,	0);
 		ak4642_write(codec, 0x00, 0x40);
 		ak4642_write(codec, 0x0e, 0x11);
 		ak4642_write(codec, 0x0f, 0x08);
@@ -284,10 +295,37 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
+static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 data;
+	u8 bcko;
+
+	data = MCKO | PMPLL; /* use MCKO */
+	bcko = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		data |= MS;
+		bcko = BCKO_64;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+	snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
+
+	return 0;
+}
+
 static struct snd_soc_dai_ops ak4642_dai_ops = {
 	.startup	= ak4642_dai_startup,
 	.shutdown	= ak4642_dai_shutdown,
 	.set_sysclk	= ak4642_dai_set_sysclk,
+	.set_fmt	= ak4642_dai_set_fmt,
 };
 
 struct snd_soc_dai ak4642_dai = {
@@ -366,23 +404,6 @@ static int ak4642_init(struct ak4642_priv *ak4642)
 		goto reg_cache_err;
 	}
 
-	/*
-	 * clock setting
-	 *
-	 * Audio I/F Format: MSB justified (ADC & DAC)
-	 * BICK frequency at Master Mode: 64fs
-	 * MCKO: Enable
-	 * Sampling Frequency: 44.1kHz
-	 *
-	 * This operation came from example code of
-	 * "ASAHI KASEI AK4642" (japanese) manual p89.
-	 *
-	 * please fix-me
-	 */
-	ak4642_write(codec, 0x01, 0x08);
-	ak4642_write(codec, 0x05, 0x27);
-	ak4642_write(codec, 0x04, 0x0a);
-
 	return ret;
 
 reg_cache_err:
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index c0207dc524b8..be018542314e 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -26,6 +26,10 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec)
 {
 	int ret;
 
+	ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
 	ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0);
 
 	return ret;
-- 
cgit v1.2.3


From 1ad747ca9b6f97f895e0a6ccd447b158aeaa568d Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Tue, 23 Mar 2010 16:27:38 +0900
Subject: ASoC: ak4642: Add enhanced sampling rate

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ak4642.c | 68 +++++++++++++++++++++++++++++++++++++++++++----
 1 file changed, 63 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 3452bd752739..de1809dc8d91 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -101,6 +101,13 @@
 #define BCKO_MASK	(1 << 3)
 #define BCKO_64		BCKO_MASK
 
+/* MD_CTL2 */
+#define FS0		(1 << 0)
+#define FS1		(1 << 1)
+#define FS2		(1 << 2)
+#define FS3		(1 << 5)
+#define FS_MASK		(FS0 | FS1 | FS2 | FS3)
+
 struct snd_soc_codec_device soc_codec_dev_ak4642;
 
 /* codec private data */
@@ -196,14 +203,12 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 *
 		 * PLL, Master Mode
 		 * Audio I/F Format :MSB justified (ADC & DAC)
-		 * Sampling Frequency: 44.1kHz
-		 * Digital Volume: −8dB
+		 * Digital Volume: -8dB
 		 * Bass Boost Level : Middle
 		 *
 		 * This operation came from example code of
 		 * "ASAHI KASEI AK4642" (japanese) manual p97.
 		 */
-		ak4642_write(codec, 0x05, 0x27);
 		ak4642_write(codec, 0x0f, 0x09);
 		ak4642_write(codec, 0x0e, 0x19);
 		ak4642_write(codec, 0x09, 0x91);
@@ -219,7 +224,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 *
 		 * PLL Master Mode
 		 * Audio I/F Format:MSB justified (ADC & DAC)
-		 * Sampling Frequency:44.1kHz
 		 * Pre MIC AMP:+20dB
 		 * MIC Power On
 		 * ALC setting:Refer to Table 35
@@ -228,7 +232,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 * This operation came from example code of
 		 * "ASAHI KASEI AK4642" (japanese) manual p94.
 		 */
-		ak4642_write(codec, 0x05, 0x27);
 		ak4642_write(codec, 0x02, 0x05);
 		ak4642_write(codec, 0x06, 0x3c);
 		ak4642_write(codec, 0x08, 0xe1);
@@ -321,11 +324,65 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
+static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 rate;
+
+	switch (params_rate(params)) {
+	case 7350:
+		rate = FS2;
+		break;
+	case 8000:
+		rate = 0;
+		break;
+	case 11025:
+		rate = FS2 | FS0;
+		break;
+	case 12000:
+		rate = FS0;
+		break;
+	case 14700:
+		rate = FS2 | FS1;
+		break;
+	case 16000:
+		rate = FS1;
+		break;
+	case 22050:
+		rate = FS2 | FS1 | FS0;
+		break;
+	case 24000:
+		rate = FS1 | FS0;
+		break;
+	case 29400:
+		rate = FS3 | FS2 | FS1;
+		break;
+	case 32000:
+		rate = FS3 | FS1;
+		break;
+	case 44100:
+		rate = FS3 | FS2 | FS1 | FS0;
+		break;
+	case 48000:
+		rate = FS3 | FS1 | FS0;
+		break;
+	default:
+		return -EINVAL;
+		break;
+	}
+	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+
+	return 0;
+}
+
 static struct snd_soc_dai_ops ak4642_dai_ops = {
 	.startup	= ak4642_dai_startup,
 	.shutdown	= ak4642_dai_shutdown,
 	.set_sysclk	= ak4642_dai_set_sysclk,
 	.set_fmt	= ak4642_dai_set_fmt,
+	.hw_params	= ak4642_dai_hw_params,
 };
 
 struct snd_soc_dai ak4642_dai = {
@@ -343,6 +400,7 @@ struct snd_soc_dai ak4642_dai = {
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE },
 	.ops = &ak4642_dai_ops,
+	.symmetric_rates = 1,
 };
 EXPORT_SYMBOL_GPL(ak4642_dai);
 
-- 
cgit v1.2.3


From feb58cffca772097a849b413fb7954f87e9e9e71 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Wed, 24 Mar 2010 15:27:24 +0900
Subject: ASoC: fsi: ensures process inside master lock

Bit operation for fsi_master should be done inside master lock.
But soft-reset/interrupt operation were outside of it.
This patch modify this problem.
It still allow to INT_ST outside-operation on fsi_interrupt,
but it is not problem.
Because this register doesn't need the bit operation.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/sh/fsi.c | 22 +++++++++++-----------
 1 file changed, 11 insertions(+), 11 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 993abb730dfa..db91349b6806 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -79,6 +79,12 @@
 #define INT_A_IN	(1 << 4)
 #define INT_A_OUT	(1 << 0)
 
+/* SOFT_RST */
+#define PBSR		(1 << 12) /* Port B Software Reset */
+#define PASR		(1 <<  8) /* Port A Software Reset */
+#define IR		(1 <<  4) /* Interrupt Reset */
+#define FSISR		(1 <<  0) /* Software Reset */
+
 #define FSI_RATES SNDRV_PCM_RATE_8000_96000
 
 #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
@@ -372,18 +378,13 @@ static void fsi_irq_init(struct fsi_priv *fsi, int is_play)
 
 static void fsi_soft_all_reset(struct fsi_master *master)
 {
-	u32 status = fsi_master_read(master, SOFT_RST);
-
 	/* port AB reset */
-	status &= 0x000000ff;
-	fsi_master_write(master, SOFT_RST, status);
+	fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0);
 	mdelay(10);
 
 	/* soft reset */
-	status &= 0x000000f0;
-	fsi_master_write(master, SOFT_RST, status);
-	status |= 0x00000001;
-	fsi_master_write(master, SOFT_RST, status);
+	fsi_master_mask_set(master, SOFT_RST, FSISR, 0);
+	fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR);
 	mdelay(10);
 }
 
@@ -558,12 +559,11 @@ static int fsi_data_pop(struct fsi_priv *fsi, int startup)
 static irqreturn_t fsi_interrupt(int irq, void *data)
 {
 	struct fsi_master *master = data;
-	u32 status = fsi_master_read(master, SOFT_RST) & ~0x00000010;
 	u32 int_st = fsi_master_read(master, INT_ST);
 
 	/* clear irq status */
-	fsi_master_write(master, SOFT_RST, status);
-	fsi_master_write(master, SOFT_RST, status | 0x00000010);
+	fsi_master_mask_set(master, SOFT_RST, IR, 0);
+	fsi_master_mask_set(master, SOFT_RST, IR, IR);
 
 	if (int_st & INT_A_OUT)
 		fsi_data_push(&master->fsia, 0);
-- 
cgit v1.2.3


From 10ea76cc25b6738eb2f7c58b3312e1ebc61f753e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Tue, 23 Mar 2010 11:47:54 +0900
Subject: ASoC: fsi: IRQ related process had be united

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/sh/fsi.c | 53 +++++++++++++++++++++++++++++++++++++++--------------
 1 file changed, 39 insertions(+), 14 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index db91349b6806..7c295df6e855 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -322,7 +322,7 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play)
 /************************************************************************
 
 
-		ctrl function
+		irq function
 
 
 ************************************************************************/
@@ -344,6 +344,35 @@ static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
 	fsi_master_mask_set(master, IEMSK, data, 0);
 }
 
+static u32 fsi_irq_get_status(struct fsi_master *master)
+{
+	return fsi_master_read(master, INT_ST);
+}
+
+static void fsi_irq_clear_all_status(struct fsi_master *master)
+{
+	fsi_master_write(master, INT_ST, 0x0000000);
+}
+
+static void fsi_irq_clear_status(struct fsi_priv *fsi)
+{
+	u32 data = 0;
+	struct fsi_master *master = fsi_get_master(fsi);
+
+	data |= fsi_port_ab_io_bit(fsi, 0);
+	data |= fsi_port_ab_io_bit(fsi, 1);
+
+	/* clear interrupt factor */
+	fsi_master_mask_set(master, INT_ST, data, 0);
+}
+
+/************************************************************************
+
+
+		ctrl function
+
+
+************************************************************************/
 static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable)
 {
 	u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4);
@@ -355,25 +384,17 @@ static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable)
 		fsi_master_mask_set(master, CLK_RST, val, 0);
 }
 
-static void fsi_irq_init(struct fsi_priv *fsi, int is_play)
+static void fsi_fifo_init(struct fsi_priv *fsi, int is_play)
 {
-	u32 data;
 	u32 ctrl;
 
-	data = fsi_port_ab_io_bit(fsi, is_play);
 	ctrl = is_play ? DOFF_CTL : DIFF_CTL;
 
-	/* set IMSK */
-	fsi_irq_disable(fsi, is_play);
-
 	/* set interrupt generation factor */
 	fsi_reg_write(fsi, ctrl, IRQ_HALF);
 
 	/* clear FIFO */
 	fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR);
-
-	/* clear interrupt factor */
-	fsi_master_mask_set(fsi_get_master(fsi), INT_ST, data, 0);
 }
 
 static void fsi_soft_all_reset(struct fsi_master *master)
@@ -559,7 +580,7 @@ static int fsi_data_pop(struct fsi_priv *fsi, int startup)
 static irqreturn_t fsi_interrupt(int irq, void *data)
 {
 	struct fsi_master *master = data;
-	u32 int_st = fsi_master_read(master, INT_ST);
+	u32 int_st = fsi_irq_get_status(master);
 
 	/* clear irq status */
 	fsi_master_mask_set(master, SOFT_RST, IR, 0);
@@ -574,7 +595,7 @@ static irqreturn_t fsi_interrupt(int irq, void *data)
 	if (int_st & INT_B_IN)
 		fsi_data_pop(&master->fsib, 0);
 
-	fsi_master_write(master, INT_ST, 0x0000000);
+	fsi_irq_clear_all_status(master);
 
 	return IRQ_HANDLED;
 }
@@ -699,8 +720,12 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 	if (is_master)
 		fsi_clk_ctrl(fsi, 1);
 
-	/* irq setting */
-	fsi_irq_init(fsi, is_play);
+	/* irq clear */
+	fsi_irq_disable(fsi, is_play);
+	fsi_irq_clear_status(fsi);
+
+	/* fifo init */
+	fsi_fifo_init(fsi, is_play);
 
 	return ret;
 }
-- 
cgit v1.2.3


From 4a942b457ee239eab04db8dd4d4c12ef6dc4a152 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Thu, 25 Mar 2010 19:15:51 +0900
Subject: ASoC: fsi: Add FIFO size calculate

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/sh/fsi.c | 70 +++++++++++++++++++++++++++++++++---------------------
 1 file changed, 43 insertions(+), 27 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 7c295df6e855..ae888651a77a 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -46,8 +46,9 @@
 #define MUTE		0x020C
 #define CLK_RST		0x0210
 #define SOFT_RST	0x0214
+#define FIFO_SZ		0x0218
 #define MREG_START	INT_ST
-#define MREG_END	SOFT_RST
+#define MREG_END	FIFO_SZ
 
 /* DO_FMT */
 /* DI_FMT */
@@ -85,6 +86,11 @@
 #define IR		(1 <<  4) /* Interrupt Reset */
 #define FSISR		(1 <<  0) /* Software Reset */
 
+/* FIFO_SZ */
+#define OUT_SZ_MASK	0x7
+#define BO_SZ_SHIFT	8
+#define AO_SZ_SHIFT	0
+
 #define FSI_RATES SNDRV_PCM_RATE_8000_96000
 
 #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
@@ -384,9 +390,42 @@ static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable)
 		fsi_master_mask_set(master, CLK_RST, val, 0);
 }
 
-static void fsi_fifo_init(struct fsi_priv *fsi, int is_play)
+static void fsi_fifo_init(struct fsi_priv *fsi,
+			  int is_play,
+			  struct snd_soc_dai *dai)
 {
-	u32 ctrl;
+	struct fsi_master *master = fsi_get_master(fsi);
+	u32 ctrl, shift, i;
+
+	/* get on-chip RAM capacity */
+	shift = fsi_master_read(master, FIFO_SZ);
+	shift >>= fsi_is_port_a(fsi) ? AO_SZ_SHIFT : BO_SZ_SHIFT;
+	shift &= OUT_SZ_MASK;
+	fsi->fifo_max = 256 << shift;
+	dev_dbg(dai->dev, "fifo = %d words\n", fsi->fifo_max);
+
+	/*
+	 * The maximum number of sample data varies depending
+	 * on the number of channels selected for the format.
+	 *
+	 * FIFOs are used in 4-channel units in 3-channel mode
+	 * and in 8-channel units in 5- to 7-channel mode
+	 * meaning that more FIFOs than the required size of DPRAM
+	 * are used.
+	 *
+	 * ex) if 256 words of DP-RAM is connected
+	 * 1 channel:  256 (256 x 1 = 256)
+	 * 2 channels: 128 (128 x 2 = 256)
+	 * 3 channels:  64 ( 64 x 3 = 192)
+	 * 4 channels:  64 ( 64 x 4 = 256)
+	 * 5 channels:  32 ( 32 x 5 = 160)
+	 * 6 channels:  32 ( 32 x 6 = 192)
+	 * 7 channels:  32 ( 32 x 7 = 224)
+	 * 8 channels:  32 ( 32 x 8 = 256)
+	 */
+	for (i = 1; i < fsi->chan; i <<= 1)
+		fsi->fifo_max >>= 1;
+	dev_dbg(dai->dev, "%d channel %d store\n", fsi->chan, fsi->fifo_max);
 
 	ctrl = is_play ? DOFF_CTL : DIFF_CTL;
 
@@ -689,29 +728,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 		dev_err(dai->dev, "unknown format.\n");
 		return -EINVAL;
 	}
-
-	switch (fsi->chan) {
-	case 1:
-		fsi->fifo_max = 256;
-		break;
-	case 2:
-		fsi->fifo_max = 128;
-		break;
-	case 3:
-	case 4:
-		fsi->fifo_max = 64;
-		break;
-	case 5:
-	case 6:
-	case 7:
-	case 8:
-		fsi->fifo_max = 32;
-		break;
-	default:
-		dev_err(dai->dev, "channel size error.\n");
-		return -EINVAL;
-	}
-
 	fsi_reg_write(fsi, reg, data);
 
 	/*
@@ -725,7 +741,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 	fsi_irq_clear_status(fsi);
 
 	/* fifo init */
-	fsi_fifo_init(fsi, is_play);
+	fsi_fifo_init(fsi, is_play, dai);
 
 	return ret;
 }
-- 
cgit v1.2.3


From cc780d380a004b58c139570b037d0e3b897bb2be Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Date: Thu, 25 Mar 2010 19:15:53 +0900
Subject: ASoC: fsi: Add FSI2 device support

ARM-SHMOBILE series have FIFO-buffered serial interface 2 (FSI2)
device which is advanced version of FSI.
This patch add simple support for it.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/sh/Kconfig |  3 +--
 sound/soc/sh/fsi.c   | 56 +++++++++++++++++++++++++++++++++++++++++++---------
 2 files changed, 48 insertions(+), 11 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index f07f6d8b93e1..a1d14bc3c76f 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -1,5 +1,5 @@
 menu "SoC Audio support for SuperH"
-	depends on SUPERH
+	depends on SUPERH || ARCH_SHMOBILE
 
 config SND_SOC_PCM_SH7760
 	tristate "SoC Audio support for Renesas SH7760"
@@ -22,7 +22,6 @@ config SND_SOC_SH4_SSI
 
 config SND_SOC_SH4_FSI
 	tristate "SH4 FSI support"
-	depends on CPU_SUBTYPE_SH7724
 	help
 	  This option enables FSI sound support
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index ae888651a77a..f14bbb0410c1 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -40,6 +40,10 @@
 #define MUTE_ST		0x0028
 #define REG_END		MUTE_ST
 
+
+#define CPU_INT_ST	0x01F4
+#define CPU_IEMSK	0x01F8
+#define CPU_IMSK	0x01FC
 #define INT_ST		0x0200
 #define IEMSK		0x0204
 #define IMSK		0x0208
@@ -47,7 +51,7 @@
 #define CLK_RST		0x0210
 #define SOFT_RST	0x0214
 #define FIFO_SZ		0x0218
-#define MREG_START	INT_ST
+#define MREG_START	CPU_INT_ST
 #define MREG_END	FIFO_SZ
 
 /* DO_FMT */
@@ -116,11 +120,18 @@ struct fsi_priv {
 	int periods;
 };
 
+struct fsi_regs {
+	u32 int_st;
+	u32 iemsk;
+	u32 imsk;
+};
+
 struct fsi_master {
 	void __iomem *base;
 	int irq;
 	struct fsi_priv fsia;
 	struct fsi_priv fsib;
+	struct fsi_regs *regs;
 	struct sh_fsi_platform_info *info;
 	spinlock_t lock;
 };
@@ -337,8 +348,8 @@ static void fsi_irq_enable(struct fsi_priv *fsi, int is_play)
 	u32 data = fsi_port_ab_io_bit(fsi, is_play);
 	struct fsi_master *master = fsi_get_master(fsi);
 
-	fsi_master_mask_set(master, IMSK,  data, data);
-	fsi_master_mask_set(master, IEMSK, data, data);
+	fsi_master_mask_set(master, master->regs->imsk,  data, data);
+	fsi_master_mask_set(master, master->regs->iemsk, data, data);
 }
 
 static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
@@ -346,18 +357,18 @@ static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
 	u32 data = fsi_port_ab_io_bit(fsi, is_play);
 	struct fsi_master *master = fsi_get_master(fsi);
 
-	fsi_master_mask_set(master, IMSK,  data, 0);
-	fsi_master_mask_set(master, IEMSK, data, 0);
+	fsi_master_mask_set(master, master->regs->imsk,  data, 0);
+	fsi_master_mask_set(master, master->regs->iemsk, data, 0);
 }
 
 static u32 fsi_irq_get_status(struct fsi_master *master)
 {
-	return fsi_master_read(master, INT_ST);
+	return fsi_master_read(master, master->regs->int_st);
 }
 
 static void fsi_irq_clear_all_status(struct fsi_master *master)
 {
-	fsi_master_write(master, INT_ST, 0x0000000);
+	fsi_master_write(master, master->regs->int_st, 0x0000000);
 }
 
 static void fsi_irq_clear_status(struct fsi_priv *fsi)
@@ -369,7 +380,7 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi)
 	data |= fsi_port_ab_io_bit(fsi, 1);
 
 	/* clear interrupt factor */
-	fsi_master_mask_set(master, INT_ST, data, 0);
+	fsi_master_mask_set(master, master->regs->int_st, data, 0);
 }
 
 /************************************************************************
@@ -953,6 +964,7 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform);
 static int fsi_probe(struct platform_device *pdev)
 {
 	struct fsi_master *master;
+	const struct platform_device_id	*id_entry;
 	struct resource *res;
 	unsigned int irq;
 	int ret;
@@ -962,6 +974,12 @@ static int fsi_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 
+	id_entry = pdev->id_entry;
+	if (!id_entry) {
+		dev_err(&pdev->dev, "unknown fsi device\n");
+		return -ENODEV;
+	}
+
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	irq = platform_get_irq(pdev, 0);
 	if (!res || (int)irq <= 0) {
@@ -990,6 +1008,7 @@ static int fsi_probe(struct platform_device *pdev)
 	master->fsia.master	= master;
 	master->fsib.base	= master->base + 0x40;
 	master->fsib.master	= master;
+	master->regs		= (struct fsi_regs *)id_entry->driver_data;
 	spin_lock_init(&master->lock);
 
 	pm_runtime_enable(&pdev->dev);
@@ -1002,7 +1021,8 @@ static int fsi_probe(struct platform_device *pdev)
 
 	fsi_soft_all_reset(master);
 
-	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master);
+	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
+			  id_entry->name, master);
 	if (ret) {
 		dev_err(&pdev->dev, "irq request err\n");
 		goto exit_iounmap;
@@ -1069,6 +1089,23 @@ static struct dev_pm_ops fsi_pm_ops = {
 	.runtime_resume		= fsi_runtime_nop,
 };
 
+static struct fsi_regs fsi_regs = {
+	.int_st	= INT_ST,
+	.iemsk	= IEMSK,
+	.imsk	= IMSK,
+};
+
+static struct fsi_regs fsi2_regs = {
+	.int_st	= CPU_INT_ST,
+	.iemsk	= CPU_IEMSK,
+	.imsk	= CPU_IMSK,
+};
+
+static struct platform_device_id fsi_id_table[] = {
+	{ "sh_fsi",	(kernel_ulong_t)&fsi_regs },
+	{ "sh_fsi2",	(kernel_ulong_t)&fsi2_regs },
+};
+
 static struct platform_driver fsi_driver = {
 	.driver 	= {
 		.name	= "sh_fsi",
@@ -1076,6 +1113,7 @@ static struct platform_driver fsi_driver = {
 	},
 	.probe		= fsi_probe,
 	.remove		= fsi_remove,
+	.id_table	= fsi_id_table,
 };
 
 static int __init fsi_mobile_init(void)
-- 
cgit v1.2.3


From c3635c78e500a52c9fcd55de381a72928d9e054d Mon Sep 17 00:00:00 2001
From: Linus Walleij <linus.walleij@stericsson.com>
Date: Fri, 26 Mar 2010 16:44:01 -0700
Subject: DMAENGINE: generic slave control v2

Convert the device_terminate_all() operation on the
DMA engine to a generic device_control() operation
which can now optionally support also pausing and
resuming DMA on a certain channel. Implemented for the
COH 901 318 DMAC as an example.

[dan.j.williams@intel.com: update for timberdale]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Maciej Sosnowski <maciej.sosnowski@intel.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Pavel Machek <pavel@ucw.cz>
Cc: Li Yang <leoli@freescale.com>
Cc: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Haavard Skinnemoen <haavard.skinnemoen@atmel.com>
Cc: Magnus Damm <damm@opensource.se>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Roland Dreier <rdreier@cisco.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
---
 arch/arm/mach-u300/include/mach/coh901318.h | 14 ----------
 drivers/dma/at_hdmac.c                      | 10 +++++--
 drivers/dma/coh901318.c                     | 42 +++++++++++++++++++----------
 drivers/dma/dmaengine.c                     |  2 +-
 drivers/dma/dw_dmac.c                       | 10 +++++--
 drivers/dma/fsldma.c                        | 13 ++++++---
 drivers/dma/ipu/ipu_idmac.c                 | 21 ++++++++++-----
 drivers/dma/shdma.c                         | 12 ++++++---
 drivers/dma/timb_dma.c                      |  9 +++++--
 drivers/dma/txx9dmac.c                      | 10 +++++--
 drivers/mmc/host/atmel-mci.c                |  2 +-
 drivers/serial/sh-sci.c                     |  2 +-
 drivers/video/mx3fb.c                       |  3 ++-
 include/linux/dmaengine.h                   | 18 +++++++++++--
 sound/soc/txx9/txx9aclc.c                   |  6 ++---
 15 files changed, 117 insertions(+), 57 deletions(-)

(limited to 'sound/soc')

diff --git a/arch/arm/mach-u300/include/mach/coh901318.h b/arch/arm/mach-u300/include/mach/coh901318.h
index b8155b4e5ffa..43ec040e765b 100644
--- a/arch/arm/mach-u300/include/mach/coh901318.h
+++ b/arch/arm/mach-u300/include/mach/coh901318.h
@@ -109,20 +109,6 @@ struct coh901318_platform {
  */
 u32 coh901318_get_bytes_left(struct dma_chan *chan);
 
-/**
- * coh901318_stop() - Stops dma transfer
- * @chan: dma channel handle
- * return 0 on success otherwise negative value
- */
-void coh901318_stop(struct dma_chan *chan);
-
-/**
- * coh901318_continue() - Resumes a stopped dma transfer
- * @chan: dma channel handle
- * return 0 on success otherwise negative value
- */
-void coh901318_continue(struct dma_chan *chan);
-
 /**
  * coh901318_filter_id() - DMA channel filter function
  * @chan: dma channel handle
diff --git a/drivers/dma/at_hdmac.c b/drivers/dma/at_hdmac.c
index efc1a61ca231..f9143cf9e50a 100644
--- a/drivers/dma/at_hdmac.c
+++ b/drivers/dma/at_hdmac.c
@@ -759,13 +759,17 @@ err_desc_get:
 	return NULL;
 }
 
-static void atc_terminate_all(struct dma_chan *chan)
+static int atc_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct at_dma_chan	*atchan = to_at_dma_chan(chan);
 	struct at_dma		*atdma = to_at_dma(chan->device);
 	struct at_desc		*desc, *_desc;
 	LIST_HEAD(list);
 
+	/* Only supports DMA_TERMINATE_ALL */
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
+
 	/*
 	 * This is only called when something went wrong elsewhere, so
 	 * we don't really care about the data. Just disable the
@@ -789,6 +793,8 @@ static void atc_terminate_all(struct dma_chan *chan)
 	/* Flush all pending and queued descriptors */
 	list_for_each_entry_safe(desc, _desc, &list, desc_node)
 		atc_chain_complete(atchan, desc);
+
+	return 0;
 }
 
 /**
@@ -1091,7 +1097,7 @@ static int __init at_dma_probe(struct platform_device *pdev)
 
 	if (dma_has_cap(DMA_SLAVE, atdma->dma_common.cap_mask)) {
 		atdma->dma_common.device_prep_slave_sg = atc_prep_slave_sg;
-		atdma->dma_common.device_terminate_all = atc_terminate_all;
+		atdma->dma_common.device_control = atc_control;
 	}
 
 	dma_writel(atdma, EN, AT_DMA_ENABLE);
diff --git a/drivers/dma/coh901318.c b/drivers/dma/coh901318.c
index f636c4a87c7f..53c54e034aa3 100644
--- a/drivers/dma/coh901318.c
+++ b/drivers/dma/coh901318.c
@@ -506,10 +506,11 @@ u32 coh901318_get_bytes_left(struct dma_chan *chan)
 EXPORT_SYMBOL(coh901318_get_bytes_left);
 
 
-/* Stops a transfer without losing data. Enables power save.
-   Use this function in conjunction with coh901318_continue(..)
-*/
-void coh901318_stop(struct dma_chan *chan)
+/*
+ * Pauses a transfer without losing data. Enables power save.
+ * Use this function in conjunction with coh901318_resume.
+ */
+static void coh901318_pause(struct dma_chan *chan)
 {
 	u32 val;
 	unsigned long flags;
@@ -550,12 +551,11 @@ void coh901318_stop(struct dma_chan *chan)
 
 	spin_unlock_irqrestore(&cohc->lock, flags);
 }
-EXPORT_SYMBOL(coh901318_stop);
 
-/* Continues a transfer that has been stopped via 300_dma_stop(..).
+/* Resumes a transfer that has been stopped via 300_dma_stop(..).
    Power save is handled.
 */
-void coh901318_continue(struct dma_chan *chan)
+static void coh901318_resume(struct dma_chan *chan)
 {
 	u32 val;
 	unsigned long flags;
@@ -581,7 +581,6 @@ void coh901318_continue(struct dma_chan *chan)
 
 	spin_unlock_irqrestore(&cohc->lock, flags);
 }
-EXPORT_SYMBOL(coh901318_continue);
 
 bool coh901318_filter_id(struct dma_chan *chan, void *chan_id)
 {
@@ -945,7 +944,7 @@ coh901318_free_chan_resources(struct dma_chan *chan)
 
 	spin_unlock_irqrestore(&cohc->lock, flags);
 
-	chan->device->device_terminate_all(chan);
+	chan->device->device_control(chan, DMA_TERMINATE_ALL);
 }
 
 
@@ -1179,16 +1178,29 @@ coh901318_issue_pending(struct dma_chan *chan)
 	spin_unlock_irqrestore(&cohc->lock, flags);
 }
 
-static void
-coh901318_terminate_all(struct dma_chan *chan)
+static int
+coh901318_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	unsigned long flags;
 	struct coh901318_chan *cohc = to_coh901318_chan(chan);
 	struct coh901318_desc *cohd;
 	void __iomem *virtbase = cohc->base->virtbase;
 
-	coh901318_stop(chan);
+	if (cmd == DMA_PAUSE) {
+		coh901318_pause(chan);
+		return 0;
+	}
+
+	if (cmd == DMA_RESUME) {
+		coh901318_resume(chan);
+		return 0;
+	}
+
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
 
+	/* The remainder of this function terminates the transfer */
+	coh901318_pause(chan);
 	spin_lock_irqsave(&cohc->lock, flags);
 
 	/* Clear any pending BE or TC interrupt */
@@ -1227,6 +1239,8 @@ coh901318_terminate_all(struct dma_chan *chan)
 	cohc->busy = 0;
 
 	spin_unlock_irqrestore(&cohc->lock, flags);
+
+	return 0;
 }
 void coh901318_base_init(struct dma_device *dma, const int *pick_chans,
 			 struct coh901318_base *base)
@@ -1344,7 +1358,7 @@ static int __init coh901318_probe(struct platform_device *pdev)
 	base->dma_slave.device_prep_slave_sg = coh901318_prep_slave_sg;
 	base->dma_slave.device_is_tx_complete = coh901318_is_tx_complete;
 	base->dma_slave.device_issue_pending = coh901318_issue_pending;
-	base->dma_slave.device_terminate_all = coh901318_terminate_all;
+	base->dma_slave.device_control = coh901318_control;
 	base->dma_slave.dev = &pdev->dev;
 
 	err = dma_async_device_register(&base->dma_slave);
@@ -1364,7 +1378,7 @@ static int __init coh901318_probe(struct platform_device *pdev)
 	base->dma_memcpy.device_prep_dma_memcpy = coh901318_prep_memcpy;
 	base->dma_memcpy.device_is_tx_complete = coh901318_is_tx_complete;
 	base->dma_memcpy.device_issue_pending = coh901318_issue_pending;
-	base->dma_memcpy.device_terminate_all = coh901318_terminate_all;
+	base->dma_memcpy.device_control = coh901318_control;
 	base->dma_memcpy.dev = &pdev->dev;
 	/*
 	 * This controller can only access address at even 32bit boundaries,
diff --git a/drivers/dma/dmaengine.c b/drivers/dma/dmaengine.c
index 87399cafce37..ffc4ee9c5e21 100644
--- a/drivers/dma/dmaengine.c
+++ b/drivers/dma/dmaengine.c
@@ -694,7 +694,7 @@ int dma_async_device_register(struct dma_device *device)
 	BUG_ON(dma_has_cap(DMA_SLAVE, device->cap_mask) &&
 		!device->device_prep_slave_sg);
 	BUG_ON(dma_has_cap(DMA_SLAVE, device->cap_mask) &&
-		!device->device_terminate_all);
+		!device->device_control);
 
 	BUG_ON(!device->device_alloc_chan_resources);
 	BUG_ON(!device->device_free_chan_resources);
diff --git a/drivers/dma/dw_dmac.c b/drivers/dma/dw_dmac.c
index d28369f7afd2..8a6b85f61176 100644
--- a/drivers/dma/dw_dmac.c
+++ b/drivers/dma/dw_dmac.c
@@ -781,13 +781,17 @@ err_desc_get:
 	return NULL;
 }
 
-static void dwc_terminate_all(struct dma_chan *chan)
+static int dwc_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct dw_dma_chan	*dwc = to_dw_dma_chan(chan);
 	struct dw_dma		*dw = to_dw_dma(chan->device);
 	struct dw_desc		*desc, *_desc;
 	LIST_HEAD(list);
 
+	/* Only supports DMA_TERMINATE_ALL */
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
+
 	/*
 	 * This is only called when something went wrong elsewhere, so
 	 * we don't really care about the data. Just disable the
@@ -810,6 +814,8 @@ static void dwc_terminate_all(struct dma_chan *chan)
 	/* Flush all pending and queued descriptors */
 	list_for_each_entry_safe(desc, _desc, &list, desc_node)
 		dwc_descriptor_complete(dwc, desc);
+
+	return 0;
 }
 
 static enum dma_status
@@ -1338,7 +1344,7 @@ static int __init dw_probe(struct platform_device *pdev)
 	dw->dma.device_prep_dma_memcpy = dwc_prep_dma_memcpy;
 
 	dw->dma.device_prep_slave_sg = dwc_prep_slave_sg;
-	dw->dma.device_terminate_all = dwc_terminate_all;
+	dw->dma.device_control = dwc_control;
 
 	dw->dma.device_is_tx_complete = dwc_is_tx_complete;
 	dw->dma.device_issue_pending = dwc_issue_pending;
diff --git a/drivers/dma/fsldma.c b/drivers/dma/fsldma.c
index bbb4be5a3ff4..714fc46e7695 100644
--- a/drivers/dma/fsldma.c
+++ b/drivers/dma/fsldma.c
@@ -774,13 +774,18 @@ fail:
 	return NULL;
 }
 
-static void fsl_dma_device_terminate_all(struct dma_chan *dchan)
+static int fsl_dma_device_control(struct dma_chan *dchan,
+				  enum dma_ctrl_cmd cmd)
 {
 	struct fsldma_chan *chan;
 	unsigned long flags;
 
+	/* Only supports DMA_TERMINATE_ALL */
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
+
 	if (!dchan)
-		return;
+		return -EINVAL;
 
 	chan = to_fsl_chan(dchan);
 
@@ -794,6 +799,8 @@ static void fsl_dma_device_terminate_all(struct dma_chan *dchan)
 	fsldma_free_desc_list(chan, &chan->ld_running);
 
 	spin_unlock_irqrestore(&chan->desc_lock, flags);
+
+	return 0;
 }
 
 /**
@@ -1332,7 +1339,7 @@ static int __devinit fsldma_of_probe(struct of_device *op,
 	fdev->common.device_is_tx_complete = fsl_dma_is_complete;
 	fdev->common.device_issue_pending = fsl_dma_memcpy_issue_pending;
 	fdev->common.device_prep_slave_sg = fsl_dma_prep_slave_sg;
-	fdev->common.device_terminate_all = fsl_dma_device_terminate_all;
+	fdev->common.device_control = fsl_dma_device_control;
 	fdev->common.dev = &op->dev;
 
 	dev_set_drvdata(&op->dev, fdev);
diff --git a/drivers/dma/ipu/ipu_idmac.c b/drivers/dma/ipu/ipu_idmac.c
index 2a446397c884..39e7fb2a90e3 100644
--- a/drivers/dma/ipu/ipu_idmac.c
+++ b/drivers/dma/ipu/ipu_idmac.c
@@ -1472,13 +1472,17 @@ static void idmac_issue_pending(struct dma_chan *chan)
 	 */
 }
 
-static void __idmac_terminate_all(struct dma_chan *chan)
+static int __idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct idmac_channel *ichan = to_idmac_chan(chan);
 	struct idmac *idmac = to_idmac(chan->device);
 	unsigned long flags;
 	int i;
 
+	/* Only supports DMA_TERMINATE_ALL */
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
+
 	ipu_disable_channel(idmac, ichan,
 			    ichan->status >= IPU_CHANNEL_ENABLED);
 
@@ -1505,17 +1509,22 @@ static void __idmac_terminate_all(struct dma_chan *chan)
 	tasklet_enable(&to_ipu(idmac)->tasklet);
 
 	ichan->status = IPU_CHANNEL_INITIALIZED;
+
+	return 0;
 }
 
-static void idmac_terminate_all(struct dma_chan *chan)
+static int idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct idmac_channel *ichan = to_idmac_chan(chan);
+	int ret;
 
 	mutex_lock(&ichan->chan_mutex);
 
-	__idmac_terminate_all(chan);
+	ret = __idmac_control(chan, cmd);
 
 	mutex_unlock(&ichan->chan_mutex);
+
+	return ret;
 }
 
 #ifdef DEBUG
@@ -1607,7 +1616,7 @@ static void idmac_free_chan_resources(struct dma_chan *chan)
 
 	mutex_lock(&ichan->chan_mutex);
 
-	__idmac_terminate_all(chan);
+	__idmac_control(chan, DMA_TERMINATE_ALL);
 
 	if (ichan->status > IPU_CHANNEL_FREE) {
 #ifdef DEBUG
@@ -1669,7 +1678,7 @@ static int __init ipu_idmac_init(struct ipu *ipu)
 
 	/* Compulsory for DMA_SLAVE fields */
 	dma->device_prep_slave_sg		= idmac_prep_slave_sg;
-	dma->device_terminate_all		= idmac_terminate_all;
+	dma->device_control			= idmac_control;
 
 	INIT_LIST_HEAD(&dma->channels);
 	for (i = 0; i < IPU_CHANNELS_NUM; i++) {
@@ -1703,7 +1712,7 @@ static void __exit ipu_idmac_exit(struct ipu *ipu)
 	for (i = 0; i < IPU_CHANNELS_NUM; i++) {
 		struct idmac_channel *ichan = ipu->channel + i;
 
-		idmac_terminate_all(&ichan->dma_chan);
+		idmac_control(&ichan->dma_chan, DMA_TERMINATE_ALL);
 		idmac_prep_slave_sg(&ichan->dma_chan, NULL, 0, DMA_NONE, 0);
 	}
 
diff --git a/drivers/dma/shdma.c b/drivers/dma/shdma.c
index 5d17e09cb625..ce28c1e22825 100644
--- a/drivers/dma/shdma.c
+++ b/drivers/dma/shdma.c
@@ -580,12 +580,16 @@ static struct dma_async_tx_descriptor *sh_dmae_prep_slave_sg(
 			       direction, flags);
 }
 
-static void sh_dmae_terminate_all(struct dma_chan *chan)
+static int sh_dmae_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct sh_dmae_chan *sh_chan = to_sh_chan(chan);
 
+	/* Only supports DMA_TERMINATE_ALL */
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
+
 	if (!chan)
-		return;
+		return -EINVAL;
 
 	dmae_halt(sh_chan);
 
@@ -601,6 +605,8 @@ static void sh_dmae_terminate_all(struct dma_chan *chan)
 	spin_unlock_bh(&sh_chan->desc_lock);
 
 	sh_dmae_chan_ld_cleanup(sh_chan, true);
+
+	return 0;
 }
 
 static dma_async_tx_callback __ld_cleanup(struct sh_dmae_chan *sh_chan, bool all)
@@ -1029,7 +1035,7 @@ static int __init sh_dmae_probe(struct platform_device *pdev)
 
 	/* Compulsory for DMA_SLAVE fields */
 	shdev->common.device_prep_slave_sg = sh_dmae_prep_slave_sg;
-	shdev->common.device_terminate_all = sh_dmae_terminate_all;
+	shdev->common.device_control = sh_dmae_control;
 
 	shdev->common.dev = &pdev->dev;
 	/* Default transfer size of 32 bytes requires 32-byte alignment */
diff --git a/drivers/dma/timb_dma.c b/drivers/dma/timb_dma.c
index 145f1c23408f..7c06471ef863 100644
--- a/drivers/dma/timb_dma.c
+++ b/drivers/dma/timb_dma.c
@@ -613,7 +613,7 @@ static struct dma_async_tx_descriptor *td_prep_slave_sg(struct dma_chan *chan,
 	return &td_desc->txd;
 }
 
-static void td_terminate_all(struct dma_chan *chan)
+static int td_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct timb_dma_chan *td_chan =
 		container_of(chan, struct timb_dma_chan, chan);
@@ -621,6 +621,9 @@ static void td_terminate_all(struct dma_chan *chan)
 
 	dev_dbg(chan2dev(chan), "%s: Entry\n", __func__);
 
+	if (cmd != DMA_TERMINATE_ALL)
+		return -ENXIO;
+
 	/* first the easy part, put the queue into the free list */
 	spin_lock_bh(&td_chan->lock);
 	list_for_each_entry_safe(td_desc, _td_desc, &td_chan->queue,
@@ -630,6 +633,8 @@ static void td_terminate_all(struct dma_chan *chan)
 	/* now tear down the runnning */
 	__td_finish(td_chan);
 	spin_unlock_bh(&td_chan->lock);
+
+	return 0;
 }
 
 static void td_tasklet(unsigned long data)
@@ -743,7 +748,7 @@ static int __devinit td_probe(struct platform_device *pdev)
 	dma_cap_set(DMA_SLAVE, td->dma.cap_mask);
 	dma_cap_set(DMA_PRIVATE, td->dma.cap_mask);
 	td->dma.device_prep_slave_sg = td_prep_slave_sg;
-	td->dma.device_terminate_all = td_terminate_all;
+	td->dma.device_control = td_control;
 
 	td->dma.dev = &pdev->dev;
 
diff --git a/drivers/dma/txx9dmac.c b/drivers/dma/txx9dmac.c
index 3ebc61067e54..e528e15f44ab 100644
--- a/drivers/dma/txx9dmac.c
+++ b/drivers/dma/txx9dmac.c
@@ -938,12 +938,16 @@ txx9dmac_prep_slave_sg(struct dma_chan *chan, struct scatterlist *sgl,
 	return &first->txd;
 }
 
-static void txx9dmac_terminate_all(struct dma_chan *chan)
+static int txx9dmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 {
 	struct txx9dmac_chan *dc = to_txx9dmac_chan(chan);
 	struct txx9dmac_desc *desc, *_desc;
 	LIST_HEAD(list);
 
+	/* Only supports DMA_TERMINATE_ALL */
+	if (cmd != DMA_TERMINATE_ALL)
+		return -EINVAL;
+
 	dev_vdbg(chan2dev(chan), "terminate_all\n");
 	spin_lock_bh(&dc->lock);
 
@@ -958,6 +962,8 @@ static void txx9dmac_terminate_all(struct dma_chan *chan)
 	/* Flush all pending and queued descriptors */
 	list_for_each_entry_safe(desc, _desc, &list, desc_node)
 		txx9dmac_descriptor_complete(dc, desc);
+
+	return 0;
 }
 
 static enum dma_status
@@ -1153,7 +1159,7 @@ static int __init txx9dmac_chan_probe(struct platform_device *pdev)
 	dc->dma.dev = &pdev->dev;
 	dc->dma.device_alloc_chan_resources = txx9dmac_alloc_chan_resources;
 	dc->dma.device_free_chan_resources = txx9dmac_free_chan_resources;
-	dc->dma.device_terminate_all = txx9dmac_terminate_all;
+	dc->dma.device_control = txx9dmac_control;
 	dc->dma.device_is_tx_complete = txx9dmac_is_tx_complete;
 	dc->dma.device_issue_pending = txx9dmac_issue_pending;
 	if (pdata && pdata->memcpy_chan == ch) {
diff --git a/drivers/mmc/host/atmel-mci.c b/drivers/mmc/host/atmel-mci.c
index 8072128e933b..ae6d24ba4f08 100644
--- a/drivers/mmc/host/atmel-mci.c
+++ b/drivers/mmc/host/atmel-mci.c
@@ -578,7 +578,7 @@ static void atmci_stop_dma(struct atmel_mci *host)
 	struct dma_chan *chan = host->data_chan;
 
 	if (chan) {
-		chan->device->device_terminate_all(chan);
+		chan->device->device_control(chan, DMA_TERMINATE_ALL);
 		atmci_dma_cleanup(host);
 	} else {
 		/* Data transfer was stopped by the interrupt handler */
diff --git a/drivers/serial/sh-sci.c b/drivers/serial/sh-sci.c
index f7b9aff88f4a..690988237971 100644
--- a/drivers/serial/sh-sci.c
+++ b/drivers/serial/sh-sci.c
@@ -1087,7 +1087,7 @@ static void work_fn_rx(struct work_struct *work)
 		unsigned long flags;
 		int count;
 
-		chan->device->device_terminate_all(chan);
+		chan->device->device_control(chan, DMA_TERMINATE_ALL);
 		dev_dbg(port->dev, "Read %u bytes with cookie %d\n",
 			sh_desc->partial, sh_desc->cookie);
 
diff --git a/drivers/video/mx3fb.c b/drivers/video/mx3fb.c
index 772ba3f45e6f..3aa50bc276eb 100644
--- a/drivers/video/mx3fb.c
+++ b/drivers/video/mx3fb.c
@@ -387,7 +387,8 @@ static void sdc_disable_channel(struct mx3fb_info *mx3_fbi)
 
 	spin_unlock_irqrestore(&mx3fb->lock, flags);
 
-	mx3_fbi->txd->chan->device->device_terminate_all(mx3_fbi->txd->chan);
+	mx3_fbi->txd->chan->device->device_control(mx3_fbi->txd->chan,
+						   DMA_TERMINATE_ALL);
 	mx3_fbi->txd = NULL;
 	mx3_fbi->cookie = -EINVAL;
 }
diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h
index 20ea12c86fd0..0731802f876f 100644
--- a/include/linux/dmaengine.h
+++ b/include/linux/dmaengine.h
@@ -106,6 +106,19 @@ enum dma_ctrl_flags {
 	DMA_PREP_FENCE = (1 << 9),
 };
 
+/**
+ * enum dma_ctrl_cmd - DMA operations that can optionally be exercised
+ * on a running channel.
+ * @DMA_TERMINATE_ALL: terminate all ongoing transfers
+ * @DMA_PAUSE: pause ongoing transfers
+ * @DMA_RESUME: resume paused transfer
+ */
+enum dma_ctrl_cmd {
+	DMA_TERMINATE_ALL,
+	DMA_PAUSE,
+	DMA_RESUME,
+};
+
 /**
  * enum sum_check_bits - bit position of pq_check_flags
  */
@@ -261,7 +274,8 @@ struct dma_async_tx_descriptor {
  * @device_prep_dma_memset: prepares a memset operation
  * @device_prep_dma_interrupt: prepares an end of chain interrupt operation
  * @device_prep_slave_sg: prepares a slave dma operation
- * @device_terminate_all: terminate all pending operations
+ * @device_control: manipulate all pending operations on a channel, returns
+ *	zero or error code
  * @device_is_tx_complete: poll for transaction completion
  * @device_issue_pending: push pending transactions to hardware
  */
@@ -313,7 +327,7 @@ struct dma_device {
 		struct dma_chan *chan, struct scatterlist *sgl,
 		unsigned int sg_len, enum dma_data_direction direction,
 		unsigned long flags);
-	void (*device_terminate_all)(struct dma_chan *chan);
+	int (*device_control)(struct dma_chan *chan, enum dma_ctrl_cmd cmd);
 
 	enum dma_status (*device_is_tx_complete)(struct dma_chan *chan,
 			dma_cookie_t cookie, dma_cookie_t *last,
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index efed64b8b026..b35d00706c0e 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -159,7 +159,7 @@ static void txx9aclc_dma_tasklet(unsigned long data)
 		void __iomem *base = drvdata->base;
 
 		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
-		chan->device->device_terminate_all(chan);
+		chan->device->device_control(chan, DMA_TERMINATE_ALL);
 		/* first time */
 		for (i = 0; i < NR_DMA_CHAIN; i++) {
 			desc = txx9aclc_dma_submit(dmadata,
@@ -267,7 +267,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
 	struct dma_chan *chan = dmadata->dma_chan;
 
 	dmadata->frag_count = -1;
-	chan->device->device_terminate_all(chan);
+	chan->device->device_control(chan, DMA_TERMINATE_ALL);
 	return 0;
 }
 
@@ -396,7 +396,7 @@ static int txx9aclc_pcm_remove(struct platform_device *pdev)
 		struct dma_chan *chan = dmadata->dma_chan;
 		if (chan) {
 			dmadata->frag_count = -1;
-			chan->device->device_terminate_all(chan);
+			chan->device->device_control(chan, DMA_TERMINATE_ALL);
 			dma_release_channel(chan);
 		}
 		dev->dmadata[i].dma_chan = NULL;
-- 
cgit v1.2.3


From 1b132ea03e169ce5eca75b00224a78a6782740e8 Mon Sep 17 00:00:00 2001
From: Stephen Rothwell <sfr@canb.auug.org.au>
Date: Mon, 29 Mar 2010 15:32:18 +1100
Subject: ASoC: update for removeal of slab.h from percpu.h

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/ad193x.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index d034464e26ff..e10820670752 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -12,6 +12,7 @@
 #include <linux/device.h>
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
+#include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-- 
cgit v1.2.3


From 68b40cc40af45dd969c50fdbac6ab5b99c91255b Mon Sep 17 00:00:00 2001
From: Stephen Rothwell <sfr@canb.auug.org.au>
Date: Mon, 29 Mar 2010 17:55:51 +1100
Subject: ASoC: TWL6040: use of kzalloc/kfree requires the include of slab.h

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/twl6040.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 0e4dce7de052..108c51a513c8 100755
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -27,6 +27,7 @@
 #include <linux/i2c.h>
 #include <linux/gpio.h>
 #include <linux/platform_device.h>
+#include <linux/slab.h>
 #include <linux/i2c/twl.h>
 
 #include <sound/core.h>
-- 
cgit v1.2.3


From d01e14a6b96a3384b6806fac1a769ca3be0022bb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 29 Mar 2010 09:31:57 +0200
Subject: ASoC: Fix file permission of soc/codecs/twl6040.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/twl6040.c | 0
 1 file changed, 0 insertions(+), 0 deletions(-)
 mode change 100755 => 100644 sound/soc/codecs/twl6040.c

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
old mode 100755
new mode 100644
-- 
cgit v1.2.3


From 923a00427a141a6d1c83f1b151221a2458b2de8d Mon Sep 17 00:00:00 2001
From: Tejun Heo <tj@kernel.org>
Date: Tue, 30 Mar 2010 02:52:29 +0900
Subject: ASoC: update gfp/slab.h includes

Implicit slab.h inclusion via percpu.h is about to go away.  Make sure
gfp.h or slab.h is included as necessary.

Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/cq93vc.c        | 1 +
 sound/soc/davinci/davinci-vcif.c | 1 +
 2 files changed, 2 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 513297441369..8f19b9310645 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -27,6 +27,7 @@
 #include <linux/pm.h>
 #include <linux/platform_device.h>
 #include <linux/device.h>
+#include <linux/slab.h>
 #include <linux/clk.h>
 #include <linux/mfd/davinci_voicecodec.h>
 
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 54b91e1768c4..9aa980d38231 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -24,6 +24,7 @@
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/delay.h>
+#include <linux/slab.h>
 #include <linux/io.h>
 #include <linux/mfd/davinci_voicecodec.h>
 
-- 
cgit v1.2.3


From 9dd7b79a86c53a097463ee1582c0bc6c4b83b770 Mon Sep 17 00:00:00 2001
From: Barry Song <21cnbao@gmail.com>
Date: Mon, 29 Mar 2010 11:16:00 +0800
Subject: ASoC: ad193x: move codec register/unregister to bus probe/remove

The way i've factored out the bus probe and removal functions so
that there's no code in the individual I2C and SPI functions means
that the register() and unregister() functions could just be squashed
into the bus_probe() and bus_remove() functions.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ad193x.c | 102 ++++++++++++++++++++--------------------------
 1 file changed, 44 insertions(+), 58 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index e10820670752..4bfd66bc2624 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -35,8 +35,6 @@ static const u8 ad193x_reg[AD193X_NUM_REGS] = {
 
 static struct snd_soc_codec *ad193x_codec;
 struct snd_soc_codec_device soc_codec_dev_ad193x;
-static int ad193x_register(struct ad193x_priv *ad193x, int bus_type);
-static void ad193x_unregister(struct ad193x_priv *ad193x);
 
 /*
  * AD193X volume/mute/de-emphasis etc. controls
@@ -290,69 +288,23 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
 {
 	struct snd_soc_codec *codec;
 	struct ad193x_priv *ad193x;
+	int ret;
+
+	if (ad193x_codec) {
+		dev_err(dev, "Another ad193x is registered\n");
+		return -EINVAL;
+	}
 
 	ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL);
 	if (ad193x == NULL)
 		return -ENOMEM;
 
-	codec = &ad193x->codec;
-	codec->control_data = ctrl_data;
-	codec->dev = dev;
-
 	dev_set_drvdata(dev, ad193x);
 
-	return ad193x_register(ad193x, bus_type);
-}
-
-static int ad193x_bus_remove(struct device *dev)
-{
-	struct ad193x_priv *ad193x = dev_get_drvdata(dev);
-
-	ad193x_unregister(ad193x);
-	return 0;
-}
-
-static struct snd_soc_dai_ops ad193x_dai_ops = {
-	.hw_params = ad193x_hw_params,
-	.digital_mute = ad193x_mute,
-	.set_tdm_slot = ad193x_set_tdm_slot,
-	.set_fmt = ad193x_set_dai_fmt,
-};
-
-/* codec DAI instance */
-struct snd_soc_dai ad193x_dai = {
-	.name = "AD193X",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 2,
-		.channels_max = 8,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 2,
-		.channels_max = 4,
-		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
-	},
-	.ops = &ad193x_dai_ops,
-};
-EXPORT_SYMBOL_GPL(ad193x_dai);
-
-static int ad193x_register(struct ad193x_priv *ad193x, int bus_type)
-{
-	int ret;
-	struct snd_soc_codec *codec = &ad193x->codec;
-
-	if (ad193x_codec) {
-		dev_err(codec->dev, "Another ad193x is registered\n");
-		return -EINVAL;
-	}
-
+	codec = &ad193x->codec;
 	mutex_init(&codec->mutex);
+	codec->control_data = ctrl_data;
+	codec->dev = dev;
 	codec->private_data = ad193x;
 	codec->reg_cache = ad193x->reg_cache;
 	codec->reg_cache_size = AD193X_NUM_REGS;
@@ -413,14 +365,48 @@ static int ad193x_register(struct ad193x_priv *ad193x, int bus_type)
 	return 0;
 }
 
-static void ad193x_unregister(struct ad193x_priv *ad193x)
+static int ad193x_bus_remove(struct device *dev)
 {
+	struct ad193x_priv *ad193x = dev_get_drvdata(dev);
+
 	snd_soc_unregister_dai(&ad193x_dai);
 	snd_soc_unregister_codec(&ad193x->codec);
 	kfree(ad193x);
 	ad193x_codec = NULL;
+
+	return 0;
 }
 
+static struct snd_soc_dai_ops ad193x_dai_ops = {
+	.hw_params = ad193x_hw_params,
+	.digital_mute = ad193x_mute,
+	.set_tdm_slot = ad193x_set_tdm_slot,
+	.set_fmt = ad193x_set_dai_fmt,
+};
+
+/* codec DAI instance */
+struct snd_soc_dai ad193x_dai = {
+	.name = "AD193X",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 4,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.ops = &ad193x_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ad193x_dai);
+
 static int ad193x_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-- 
cgit v1.2.3


From 8876698406147986a9a7748586a54c4b14514c0e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 29 Mar 2010 20:57:12 +0100
Subject: ASoC: Implement interrupt based WM8994 microphone detection

Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.

Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 137 +++++++++++++++++++++++++++++++++++++++++++++-
 sound/soc/codecs/wm8994.h |   3 +
 2 files changed, 137 insertions(+), 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index bdda0936a404..fc3dce814924 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -61,6 +61,12 @@ static int wm8994_retune_mobile_base[] = {
 
 #define WM8994_REG_CACHE_SIZE  0x621
 
+struct wm8994_micdet {
+	struct snd_soc_jack *jack;
+	int det;
+	int shrt;
+};
+
 /* codec private data */
 struct wm8994_priv {
 	struct wm_hubs_data hubs;
@@ -86,6 +92,8 @@ struct wm8994_priv {
 	int retune_mobile_cfg[WM8994_NUM_EQ];
 	struct soc_enum retune_mobile_enum;
 
+	struct wm8994_micdet micdet[2];
+
 	struct wm8994_pdata *pdata;
 };
 
@@ -3702,6 +3710,96 @@ struct snd_soc_codec_device soc_codec_dev_wm8994 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8994);
 
+/**
+ * wm8994_mic_detect - Enable microphone detection via the WM8994 IRQ
+ *
+ * @codec:   WM8994 codec
+ * @jack:    jack to report detection events on
+ * @micbias: microphone bias to detect on
+ * @det:     value to report for presence detection
+ * @shrt:    value to report for short detection
+ *
+ * Enable microphone detection via IRQ on the WM8994.  If GPIOs are
+ * being used to bring out signals to the processor then only platform
+ * data configuration is needed for WM8903 and processor GPIOs should
+ * be configured using snd_soc_jack_add_gpios() instead.
+ *
+ * Configuration of detection levels is available via the micbias1_lvl
+ * and micbias2_lvl platform data members.
+ */
+int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
+		      int micbias, int det, int shrt)
+{
+	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_micdet *micdet;
+	int reg;
+
+	switch (micbias) {
+	case 1:
+		micdet = &wm8994->micdet[0];
+		break;
+	case 2:
+		micdet = &wm8994->micdet[1];
+		break;
+	default:
+		return -EINVAL;
+	}	
+
+	dev_dbg(codec->dev, "Configuring microphone detection on %d: %x %x\n",
+		micbias, det, shrt);
+
+	/* Store the configuration */
+	micdet->jack = jack;
+	micdet->det = det;
+	micdet->shrt = shrt;
+
+	/* If either of the jacks is set up then enable detection */
+	if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
+		reg = WM8994_MICD_ENA;
+	else 
+		reg = 0;
+
+	snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm8994_mic_detect);
+
+static irqreturn_t wm8994_mic_irq(int irq, void *data)
+{
+	struct wm8994_priv *priv = data;
+	struct snd_soc_codec *codec = &priv->codec;
+	int reg;
+	int report;
+
+	reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2);
+	if (reg < 0) {
+		dev_err(codec->dev, "Failed to read microphone status: %d\n",
+			reg);
+		return IRQ_HANDLED;
+	}
+
+	dev_dbg(codec->dev, "Microphone status: %x\n", reg);
+
+	report = 0;
+	if (reg & WM8994_MIC1_DET_STS)
+		report |= priv->micdet[0].det;
+	if (reg & WM8994_MIC1_SHRT_STS)
+		report |= priv->micdet[0].shrt;
+	snd_soc_jack_report(priv->micdet[0].jack, report,
+			    priv->micdet[0].det | priv->micdet[0].shrt);
+
+	report = 0;
+	if (reg & WM8994_MIC2_DET_STS)
+		report |= priv->micdet[1].det;
+	if (reg & WM8994_MIC2_SHRT_STS)
+		report |= priv->micdet[1].shrt;
+	snd_soc_jack_report(priv->micdet[1].jack, report,
+			    priv->micdet[1].det | priv->micdet[1].shrt);
+
+	return IRQ_HANDLED;
+}
+
 static int wm8994_codec_probe(struct platform_device *pdev)
 {
 	int ret;
@@ -3774,6 +3872,30 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	}
 			   
 
+	ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC1_DET,
+				 wm8994_mic_irq, "Mic 1 detect", wm8994);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to request Mic1 detect IRQ: %d\n", ret);
+
+	ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT,
+				 wm8994_mic_irq, "Mic 1 short", wm8994);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to request Mic1 short IRQ: %d\n", ret);
+
+	ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC2_DET,
+				 wm8994_mic_irq, "Mic 2 detect", wm8994);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to request Mic2 detect IRQ: %d\n", ret);
+
+	ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT,
+				 wm8994_mic_irq, "Mic 2 short", wm8994);
+	if (ret != 0)
+		dev_warn(&pdev->dev,
+			 "Failed to request Mic2 short IRQ: %d\n", ret);
+
 	/* Remember if AIFnLRCLK is configured as a GPIO.  This should be
 	 * configured on init - if a system wants to do this dynamically
 	 * at runtime we can deal with that then.
@@ -3781,7 +3903,7 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_1);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to read GPIO1 state: %d\n", ret);
-		goto err;
+		goto err_irq;
 	}
 	if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) {
 		wm8994->lrclk_shared[0] = 1;
@@ -3793,7 +3915,7 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	ret = wm8994_reg_read(codec->control_data, WM8994_GPIO_6);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to read GPIO6 state: %d\n", ret);
-		goto err;
+		goto err_irq;
 	}
 	if ((ret & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) {
 		wm8994->lrclk_shared[1] = 1;
@@ -3843,7 +3965,7 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		goto err;
+		goto err_irq;
 	}
 
 	ret = snd_soc_register_dais(wm8994_dai, ARRAY_SIZE(wm8994_dai));
@@ -3858,6 +3980,11 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 
 err_codec:
 	snd_soc_unregister_codec(codec);
+err_irq:
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
 err:
 	kfree(wm8994);
 	return ret;
@@ -3871,6 +3998,10 @@ static int __devexit wm8994_codec_remove(struct platform_device *pdev)
 	wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dais(wm8994_dai, ARRAY_SIZE(wm8994_dai));
 	snd_soc_unregister_codec(&wm8994->codec);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
 	kfree(wm8994);
 	wm8994_codec = NULL;
 
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 0a5e1424dea0..79d5915ae4b3 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -23,4 +23,7 @@ extern struct snd_soc_dai wm8994_dai[];
 #define WM8994_FLL1 1
 #define WM8994_FLL2 2
 
+int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
+		      int micbias, int det, int shrt);
+
 #endif
-- 
cgit v1.2.3


From 6ca0c22ef8a4e988e2487d25964d55e6c37c5785 Mon Sep 17 00:00:00 2001
From: Marek Vasut <marek.vasut@gmail.com>
Date: Thu, 8 Apr 2010 20:48:51 +0200
Subject: ASoC: WM8750: Convert to new API

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.

Also, this patch fixes the Jive and Spitz machine.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8750.c       | 295 +++++++++++++++++++---------------------
 sound/soc/pxa/spitz.c           |  43 +++++-
 sound/soc/s3c24xx/jive_wm8750.c |   5 -
 3 files changed, 175 insertions(+), 168 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index ee084083a49d..4bbc512c0836 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -29,11 +29,6 @@
 
 #include "wm8750.h"
 
-/* codec private data */
-struct wm8750_priv {
-	unsigned int sysclk;
-};
-
 /*
  * wm8750 register cache
  * We can't read the WM8750 register space when we
@@ -53,6 +48,13 @@ static const u16 wm8750_reg[] = {
 	0x0079, 0x0079, 0x0079,          /* 40 */
 };
 
+/* codec private data */
+struct wm8750_priv {
+	unsigned int sysclk;
+	struct snd_soc_codec codec;
+	u16 reg_cache[ARRAY_SIZE(wm8750_reg)];
+};
+
 #define wm8750_reset(c)	snd_soc_write(c, WM8750_RESET, 0)
 
 /*
@@ -695,25 +697,90 @@ static int wm8750_resume(struct platform_device *pdev)
 	return 0;
 }
 
+static struct snd_soc_codec *wm8750_codec;
+
+static int wm8750_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (!wm8750_codec) {
+		dev_err(&pdev->dev, "WM8750 codec not yet registered\n");
+		return -EINVAL;
+	}
+
+	socdev->card->codec = wm8750_codec;
+	codec = wm8750_codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8750: failed to create pcms\n");
+		goto err;
+	}
+
+	snd_soc_add_controls(codec, wm8750_snd_controls,
+				ARRAY_SIZE(wm8750_snd_controls));
+	wm8750_add_widgets(codec);
+
+	return 0;
+
+err:
+	return ret;
+}
+
+/* power down chip */
+static int wm8750_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8750 = {
+	.probe		= wm8750_probe,
+	.remove		= wm8750_remove,
+	.suspend	= wm8750_suspend,
+	.resume		= wm8750_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
+
 /*
  * initialise the WM8750 driver
  * register the mixer and dsp interfaces with the kernel
  */
-static int wm8750_init(struct snd_soc_device *socdev,
-		       enum snd_soc_control_type control)
+static int wm8750_register(struct wm8750_priv *wm8750,
+			enum snd_soc_control_type control)
 {
-	struct snd_soc_codec *codec = socdev->card->codec;
+	struct snd_soc_codec *codec = &wm8750->codec;
 	int reg, ret = 0;
 
+	if (wm8750_codec) {
+		dev_err(codec->dev, "Multiple WM8750 devices not supported\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
 	codec->name = "WM8750";
 	codec->owner = THIS_MODULE;
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	codec->set_bias_level = wm8750_set_bias_level;
 	codec->dai = &wm8750_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = ARRAY_SIZE(wm8750_reg);
-	codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL);
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
+	codec->private_data = wm8750;
+	codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1;
+	codec->reg_cache = &wm8750->reg_cache;
+	codec->private_data = wm8750;
+
+	memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache));
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
 	if (ret < 0) {
@@ -727,13 +794,6 @@ static int wm8750_init(struct snd_soc_device *socdev,
 		goto err;
 	}
 
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8750: failed to create pcms\n");
-		goto err;
-	}
-
 	/* charge output caps */
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -755,19 +815,37 @@ static int wm8750_init(struct snd_soc_device *socdev,
 	reg = snd_soc_read(codec, WM8750_RINVOL);
 	snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100);
 
-	snd_soc_add_controls(codec, wm8750_snd_controls,
-				ARRAY_SIZE(wm8750_snd_controls));
-	wm8750_add_widgets(codec);
-	return ret;
+	wm8750_codec = codec;
 
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dais(&wm8750_dai, 1);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
 err:
-	kfree(codec->reg_cache);
+	kfree(wm8750);
 	return ret;
 }
 
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8750_socdev;
+static void wm8750_unregister(struct wm8750_priv *wm8750)
+{
+	wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dais(&wm8750_dai, 1);
+	snd_soc_unregister_codec(&wm8750->codec);
+	kfree(wm8750);
+	wm8750_codec = NULL;
+}
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 
@@ -781,24 +859,26 @@ static struct snd_soc_device *wm8750_socdev;
 static int wm8750_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
+	struct snd_soc_codec *codec;
+	struct wm8750_priv *wm8750;
 
-	i2c_set_clientdata(i2c, codec);
+	wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
+	if (wm8750 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8750->codec;
 	codec->control_data = i2c;
+	i2c_set_clientdata(i2c, wm8750);
 
-	ret = wm8750_init(socdev, SND_SOC_I2C);
-	if (ret < 0)
-		pr_err("failed to initialise WM8750\n");
+	codec->dev = &i2c->dev;
 
-	return ret;
+	return wm8750_register(wm8750, SND_SOC_I2C);
 }
 
 static int wm8750_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+	struct wm8750_priv *wm8750 = i2c_get_clientdata(client);
+	wm8750_unregister(wm8750);
 	return 0;
 }
 
@@ -817,66 +897,31 @@ static struct i2c_driver wm8750_i2c_driver = {
 	.remove =   wm8750_i2c_remove,
 	.id_table = wm8750_i2c_id,
 };
-
-static int wm8750_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8750_setup_data *setup)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&wm8750_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8750", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	return 0;
-
-err_driver:
-	i2c_del_driver(&wm8750_i2c_driver);
-	return -ENODEV;
-}
 #endif
 
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8750_spi_probe(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
+	struct snd_soc_codec *codec;
+	struct wm8750_priv *wm8750;
 
+	wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
+	if (wm8750 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8750->codec;
 	codec->control_data = spi;
+	codec->dev = &spi->dev;
 
-	ret = wm8750_init(socdev, SND_SOC_SPI);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8750\n");
+	dev_set_drvdata(&spi->dev, wm8750);
 
-	return ret;
+	return wm8750_register(wm8750, SND_SOC_SPI);
 }
 
 static int __devexit wm8750_spi_remove(struct spi_device *spi)
 {
+	struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev);
+	wm8750_unregister(wm8750);
 	return 0;
 }
 
@@ -891,93 +936,31 @@ static struct spi_driver wm8750_spi_driver = {
 };
 #endif
 
-static int wm8750_probe(struct platform_device *pdev)
+static int __init wm8750_modinit(void)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8750_setup_data *setup = socdev->codec_data;
-	struct snd_soc_codec *codec;
-	struct wm8750_priv *wm8750;
 	int ret;
-
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
-	if (wm8750 == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
-
-	codec->private_data = wm8750;
-	socdev->card->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8750_socdev = socdev;
-
-	ret = -ENODEV;
-
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		ret = wm8750_add_i2c_device(pdev, setup);
-	}
+	ret = i2c_add_driver(&wm8750_i2c_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8750 I2C driver: %d\n", ret);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		ret = spi_register_driver(&wm8750_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
+	ret = spi_register_driver(&wm8750_spi_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8750 SPI driver: %d\n", ret);
 #endif
-
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
-	return ret;
+	return 0;
 }
+module_init(wm8750_modinit);
 
-/* power down chip */
-static int wm8750_remove(struct platform_device *pdev)
+static void __exit wm8750_exit(void)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	if (codec->control_data)
-		wm8750_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_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8750_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
 	spi_unregister_driver(&wm8750_spi_driver);
 #endif
-	kfree(codec->private_data);
-	kfree(codec);
-
-	return 0;
-}
-
-struct snd_soc_codec_device soc_codec_dev_wm8750 = {
-	.probe = 	wm8750_probe,
-	.remove = 	wm8750_remove,
-	.suspend = 	wm8750_suspend,
-	.resume =	wm8750_resume,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
-
-static int __init wm8750_modinit(void)
-{
-	return snd_soc_register_dai(&wm8750_dai);
-}
-module_init(wm8750_modinit);
-
-static void __exit wm8750_exit(void)
-{
-	snd_soc_unregister_dai(&wm8750_dai);
 }
 module_exit(wm8750_exit);
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c4cd2acaacb4..1941a357e8c4 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -322,19 +322,44 @@ static struct snd_soc_card snd_soc_spitz = {
 	.num_links = 1,
 };
 
-/* spitz audio private data */
-static struct wm8750_setup_data spitz_wm8750_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
-
 /* spitz audio subsystem */
 static struct snd_soc_device spitz_snd_devdata = {
 	.card = &snd_soc_spitz,
 	.codec_dev = &soc_codec_dev_wm8750,
-	.codec_data = &spitz_wm8750_setup,
 };
 
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8750 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8750_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8750", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+		(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
 static struct platform_device *spitz_snd_device;
 
 static int __init spitz_init(void)
@@ -344,6 +369,10 @@ static int __init spitz_init(void)
 	if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
 		return -ENODEV;
 
+	ret = wm8750_i2c_setup();
+	if (ret != 0)
+		return ret;
+
 	spitz_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!spitz_snd_device)
 		return -ENOMEM;
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 59dc2c6b56d9..97d8ff3196be 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -152,15 +152,10 @@ static struct snd_soc_card snd_soc_machine_jive = {
 	.num_links	= 1,
 };
 
-/* jive audio private data */
-static struct wm8750_setup_data jive_wm8750_setup = {
-};
-
 /* jive audio subsystem */
 static struct snd_soc_device jive_snd_devdata = {
 	.card		= &snd_soc_machine_jive,
 	.codec_dev	= &soc_codec_dev_wm8750,
-	.codec_data	= &jive_wm8750_setup,
 };
 
 static struct platform_device *jive_snd_device;
-- 
cgit v1.2.3


From e135443e2133473f44aa439ad333554dad14c21a Mon Sep 17 00:00:00 2001
From: Bill Gatliff <bgat@billgatliff.com>
Date: Fri, 9 Apr 2010 18:08:08 +0100
Subject: ASoC: Use SNDRV_PCM_RATE_8000_96000 macro for WM8731

Signed-off-by: Bill Gatliff <bgat@billgatliff.com>
Acked-by: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5a2619dbf283..d5e1c5b681e3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -389,11 +389,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define WM8731_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-		SNDRV_PCM_RATE_96000)
+#define WM8731_RATES SNDRV_PCM_RATE_8000_96000
 
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
-- 
cgit v1.2.3


From d21e0f4cd16656f71207683ee27465600ad21625 Mon Sep 17 00:00:00 2001
From: Marek Vasut <marek.vasut@gmail.com>
Date: Mon, 5 Apr 2010 06:13:38 +0200
Subject: ASoC: Zipit Z2 WM8750 ASoC driver

This patch adds support for sound through the WM8750 codec on Zipit Z2.
Also, this patch incorporates support for detecting headset jack
insertion through the jack detection API.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig  |   8 ++
 sound/soc/pxa/Makefile |   2 +
 sound/soc/pxa/z2.c     | 246 +++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 256 insertions(+)
 create mode 100644 sound/soc/pxa/z2.c

(limited to 'sound/soc')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 376e14a9c273..495a36fba360 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -42,6 +42,14 @@ config SND_PXA2XX_SOC_SPITZ
 	  Say Y if you want to add support for SoC audio on Sharp
 	  Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
 
+config SND_PXA2XX_SOC_Z2
+	tristate "SoC Audio support for Zipit Z2"
+	depends on SND_PXA2XX_SOC && MACH_ZIPIT2
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8750
+	help
+	  Say Y if you want to add support for SoC audio on Zipit Z2.
+
 config SND_PXA2XX_SOC_POODLE
 	tristate "SoC Audio support for Poodle"
 	depends on SND_PXA2XX_SOC && MACH_POODLE
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index f3e08fd40ca2..caa03d8f4789 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
 snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-z2-objs := z2.o
 snd-soc-imote2-objs := imote2.o
 snd-soc-raumfeld-objs := raumfeld.o
 
@@ -36,6 +37,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
+obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
new file mode 100644
index 000000000000..4e4d2fa8ddc5
--- /dev/null
+++ b/sound/soc/pxa/z2.c
@@ -0,0 +1,246 @@
+/*
+ * linux/sound/soc/pxa/z2.c
+ *
+ * SoC Audio driver for Aeronix Zipit Z2
+ *
+ * Copyright (C) 2009 Ken McGuire <kenm@desertweyr.com>
+ * Copyright (C) 2010 Marek Vasut <marek.vasut@gmail.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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+#include <mach/z2.h>
+
+#include "../codecs/wm8750.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+static struct snd_soc_card snd_soc_z2;
+
+static int z2_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin	= "Mic Jack",
+		.mask	= SND_JACK_MICROPHONE,
+	},
+	{
+		.pin	= "Headphone Jack",
+		.mask	= SND_JACK_HEADPHONE,
+	},
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.gpio		= GPIO37_ZIPITZ2_HEADSET_DETECT,
+		.name		= "hsdet-gpio",
+		.report		= SND_JACK_HEADSET,
+		.debounce_time	= 200,
+	},
+};
+
+/* z2 machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+
+	/* headset is a mic and mono headphone */
+	SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Z2 machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* headphone connected to LOUT1, ROUT1 */
+	{"Headphone Jack", NULL, "LOUT1"},
+	{"Headphone Jack", NULL, "ROUT1"},
+
+	/* ext speaker connected to LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+	/* mic is connected to R input 2 - with bias */
+	{"RINPUT2", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic Jack"},
+};
+
+/*
+ * Logic for a wm8750 as connected on a Z2 Device
+ */
+static int z2_wm8750_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	/* NC codec pins */
+	snd_soc_dapm_disable_pin(codec, "LINPUT3");
+	snd_soc_dapm_disable_pin(codec, "RINPUT3");
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "MONO");
+
+	/* Add z2 specific widgets */
+	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+				 ARRAY_SIZE(wm8750_dapm_widgets));
+
+	/* Set up z2 specific audio paths */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		goto err;
+
+	/* Jack detection API stuff */
+	ret = snd_soc_jack_new(&snd_soc_z2, "Headset Jack", SND_JACK_HEADSET,
+				&hs_jack);
+	if (ret)
+		goto err;
+
+	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+				hs_jack_pins);
+	if (ret)
+		goto err;
+
+	ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+	if (ret)
+		goto err;
+
+	return 0;
+
+err:
+	return ret;
+}
+
+static struct snd_soc_ops z2_ops = {
+	.hw_params = z2_hw_params,
+};
+
+/* z2 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link z2_dai = {
+	.name		= "wm8750",
+	.stream_name	= "WM8750",
+	.cpu_dai	= &pxa_i2s_dai,
+	.codec_dai	= &wm8750_dai,
+	.init		= z2_wm8750_init,
+	.ops		= &z2_ops,
+};
+
+/* z2 audio machine driver */
+static struct snd_soc_card snd_soc_z2 = {
+	.name		= "Z2",
+	.platform	= &pxa2xx_soc_platform,
+	.dai_link	= &z2_dai,
+	.num_links	= 1,
+};
+
+/* z2 audio subsystem */
+static struct snd_soc_device z2_snd_devdata = {
+	.card		= &snd_soc_z2,
+	.codec_dev	= &soc_codec_dev_wm8750,
+};
+
+static struct platform_device *z2_snd_device;
+
+static int __init z2_init(void)
+{
+	int ret;
+
+	if (!machine_is_zipit2())
+		return -ENODEV;
+
+	z2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!z2_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(z2_snd_device, &z2_snd_devdata);
+	z2_snd_devdata.dev = &z2_snd_device->dev;
+	ret = platform_device_add(z2_snd_device);
+
+	if (ret)
+		platform_device_put(z2_snd_device);
+
+	return ret;
+}
+
+static void __exit z2_exit(void)
+{
+	platform_device_unregister(z2_snd_device);
+}
+
+module_init(z2_init);
+module_exit(z2_exit);
+
+MODULE_AUTHOR("Ken McGuire <kenm@desertweyr.com>, "
+		"Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC ZipitZ2");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From b2c812e22de88bb79c290c0e718280f10b64a48d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 14 Apr 2010 15:35:19 +0900
Subject: ASoC: Add indirection for CODEC private data

One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.

To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/soc.h            | 13 ++++++++++++-
 sound/soc/codecs/ad1836.c      |  2 +-
 sound/soc/codecs/ad193x.c      |  2 +-
 sound/soc/codecs/ak4104.c      |  2 +-
 sound/soc/codecs/ak4535.c      | 10 +++++-----
 sound/soc/codecs/ak4642.c      |  2 +-
 sound/soc/codecs/ak4671.c      |  2 +-
 sound/soc/codecs/cs4270.c      | 20 ++++++++++----------
 sound/soc/codecs/cx20442.c     |  2 +-
 sound/soc/codecs/da7210.c      |  2 +-
 sound/soc/codecs/ssm2602.c     | 12 ++++++------
 sound/soc/codecs/stac9766.c    |  2 +-
 sound/soc/codecs/tlv320aic3x.c | 10 +++++-----
 sound/soc/codecs/tlv320dac33.c | 36 ++++++++++++++++++------------------
 sound/soc/codecs/twl4030.c     | 32 ++++++++++++++++----------------
 sound/soc/codecs/uda134x.c     | 16 ++++++++--------
 sound/soc/codecs/uda1380.c     |  4 ++--
 sound/soc/codecs/wm8350.c      | 28 ++++++++++++++--------------
 sound/soc/codecs/wm8400.c      | 16 ++++++++--------
 sound/soc/codecs/wm8523.c      | 10 +++++-----
 sound/soc/codecs/wm8580.c      |  4 ++--
 sound/soc/codecs/wm8711.c      |  6 +++---
 sound/soc/codecs/wm8731.c      | 10 +++++-----
 sound/soc/codecs/wm8750.c      |  7 +++----
 sound/soc/codecs/wm8753.c      |  8 ++++----
 sound/soc/codecs/wm8776.c      |  6 +++---
 sound/soc/codecs/wm8900.c      | 10 +++++-----
 sound/soc/codecs/wm8903.c      | 20 ++++++++++----------
 sound/soc/codecs/wm8904.c      | 42 +++++++++++++++++++++---------------------
 sound/soc/codecs/wm8940.c      |  4 ++--
 sound/soc/codecs/wm8955.c      | 16 ++++++++--------
 sound/soc/codecs/wm8960.c      |  6 +++---
 sound/soc/codecs/wm8961.c      |  6 +++---
 sound/soc/codecs/wm8971.c      | 12 ++++++------
 sound/soc/codecs/wm8974.c      |  2 +-
 sound/soc/codecs/wm8978.c      | 12 ++++++------
 sound/soc/codecs/wm8988.c      |  8 ++++----
 sound/soc/codecs/wm8990.c      |  8 ++++----
 sound/soc/codecs/wm8993.c      | 24 ++++++++++++------------
 sound/soc/codecs/wm8994.c      | 34 +++++++++++++++++-----------------
 sound/soc/codecs/wm9081.c      | 18 +++++++++---------
 sound/soc/codecs/wm9713.c      | 13 +++++++------
 sound/soc/codecs/wm_hubs.c     |  6 +++---
 43 files changed, 258 insertions(+), 247 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4ab3dad4a9c9..01e9c66ddc6d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -414,7 +414,7 @@ struct snd_soc_codec {
 	struct snd_ac97 *ac97;  /* for ad-hoc ac97 devices */
 	unsigned int active;
 	unsigned int pcm_devs;
-	void *private_data;
+	void *drvdata;
 
 	/* codec IO */
 	void *control_data; /* codec control (i2c/3wire) data */
@@ -597,6 +597,17 @@ static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
 	return codec->write(codec, reg, val);
 }
 
+static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec,
+					     void *data)
+{
+	codec->drvdata = data;
+}
+
+static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec)
+{
+	return codec->drvdata;
+}
+
 #include <sound/soc-dai.h>
 
 #endif
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 3c80137d5938..48e1272892f9 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -277,7 +277,7 @@ static int ad1836_register(struct ad1836_priv *ad1836)
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
-	codec->private_data = ad1836;
+	snd_soc_codec_set_drvdata(codec, ad1836);
 	codec->reg_cache = ad1836->reg_cache;
 	codec->reg_cache_size = AD1836_NUM_REGS;
 	codec->name = "AD1836";
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 4bfd66bc2624..7ed787e2e519 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -305,7 +305,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
 	mutex_init(&codec->mutex);
 	codec->control_data = ctrl_data;
 	codec->dev = dev;
-	codec->private_data = ad193x;
+	snd_soc_codec_set_drvdata(codec, ad193x);
 	codec->reg_cache = ad193x->reg_cache;
 	codec->reg_cache_size = AD193X_NUM_REGS;
 	codec->name = "AD193X";
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index b68d99fb6af0..962d74682122 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -221,7 +221,7 @@ static int ak4104_spi_probe(struct spi_device *spi)
 	codec->owner = THIS_MODULE;
 	codec->dai = &ak4104_dai;
 	codec->num_dai = 1;
-	codec->private_data = ak4104;
+	snd_soc_codec_set_drvdata(codec, ak4104);
 	codec->control_data = spi;
 	codec->reg_cache = ak4104->reg_cache;
 	codec->reg_cache_size = AK4104_NUM_REGS;
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index ff966567e2ba..2f6221b2ebb5 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -301,7 +301,7 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct ak4535_priv *ak4535 = codec->private_data;
+	struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
 
 	ak4535->sysclk = freq;
 	return 0;
@@ -314,7 +314,7 @@ static int ak4535_hw_params(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->card->codec;
-	struct ak4535_priv *ak4535 = codec->private_data;
+	struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
 	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
 	int rate = params_rate(params), fs = 256;
 
@@ -599,7 +599,7 @@ static int ak4535_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	}
 
-	codec->private_data = ak4535;
+	snd_soc_codec_set_drvdata(codec, ak4535);
 	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -616,7 +616,7 @@ static int ak4535_probe(struct platform_device *pdev)
 #endif
 
 	if (ret != 0) {
-		kfree(codec->private_data);
+		kfree(snd_soc_codec_get_drvdata(codec));
 		kfree(codec);
 	}
 	return ret;
@@ -638,7 +638,7 @@ static int ak4535_remove(struct platform_device *pdev)
 		i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&ak4535_i2c_driver);
 #endif
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index de1809dc8d91..1254a7ac561e 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -431,7 +431,7 @@ static int ak4642_init(struct ak4642_priv *ak4642)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data	= ak4642;
+	snd_soc_codec_set_drvdata(codec, ak4642);
 	codec->name		= "AK4642";
 	codec->owner		= THIS_MODULE;
 	codec->read		= ak4642_read_reg_cache;
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 82fca284d007..18fd4475413c 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -701,7 +701,7 @@ static int ak4671_register(struct ak4671_priv *ak4671,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = ak4671;
+	snd_soc_codec_set_drvdata(codec,  ak4671);
 	codec->name = "AK4671";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index dfbeb2db61b3..a7f09f33c560 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -210,7 +210,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	unsigned int rates = 0;
 	unsigned int rate_min = -1;
 	unsigned int rate_max = 0;
@@ -269,7 +269,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
 	/* set DAI format */
@@ -411,7 +411,7 @@ static int cs4270_hw_params(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->card->codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 	unsigned int i;
 	unsigned int rate;
@@ -490,7 +490,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int reg6;
 
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
@@ -523,7 +523,7 @@ static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int left = !ucontrol->value.integer.value[0];
 	int right = !ucontrol->value.integer.value[1];
 
@@ -599,7 +599,7 @@ static int cs4270_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = cs4270_codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
 
 	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
@@ -656,7 +656,7 @@ static int cs4270_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = cs4270_codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 
 	snd_soc_free_pcms(socdev);
 	regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies);
@@ -729,7 +729,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	codec->owner = THIS_MODULE;
 	codec->dai = &cs4270_dai;
 	codec->num_dai = 1;
-	codec->private_data = cs4270;
+	snd_soc_codec_set_drvdata(codec, cs4270);
 	codec->control_data = i2c_client;
 	codec->read = cs4270_read_reg_cache;
 	codec->write = cs4270_i2c_write;
@@ -842,7 +842,7 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id);
 static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg)
 {
 	struct snd_soc_codec *codec = cs4270_codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int reg, ret;
 
 	reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
@@ -862,7 +862,7 @@ static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg)
 static int cs4270_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_codec *codec = cs4270_codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c_client = codec->control_data;
 	int reg;
 
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index e000cdfec1ec..203e51f91489 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -386,7 +386,7 @@ static int cx20442_register(struct cx20442_priv *cx20442)
 
 	codec->name = "CX20442";
 	codec->owner = THIS_MODULE;
-	codec->private_data = cx20442;
+	snd_soc_codec_set_drvdata(codec, cx20442);
 
 	codec->dai = &cx20442_dai;
 	codec->num_dai = 1;
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 35b881638658..89cea32f9424 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -460,7 +460,7 @@ static int da7210_init(struct da7210_priv *da7210)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data	= da7210;
+	snd_soc_codec_set_drvdata(codec, da7210);
 	codec->name		= "DA7210";
 	codec->owner		= THIS_MODULE;
 	codec->read		= da7210_read;
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 942f5dc30801..99ddec7df614 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -277,7 +277,7 @@ static int ssm2602_hw_params(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->card->codec;
-	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = codec->control_data;
 	u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
 	int i = get_coeff(ssm2602->sysclk, params_rate(params));
@@ -322,7 +322,7 @@ static int ssm2602_startup(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->card->codec;
-	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
 
@@ -373,7 +373,7 @@ static void ssm2602_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->card->codec;
-	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
 	/* deactivate */
 	if (!codec->active)
@@ -401,7 +401,7 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	switch (freq) {
 	case 11289600:
 	case 12000000:
@@ -726,7 +726,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	}
 
-	codec->private_data = ssm2602;
+	snd_soc_codec_set_drvdata(codec, ssm2602);
 	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -759,7 +759,7 @@ static int ssm2602_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&ssm2602_i2c_driver);
 #endif
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 81b8c9dfe7fc..a1c7ca869d0e 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -409,7 +409,7 @@ reset_err:
 pcm_err:
 	snd_soc_free_ac97_codec(codec);
 codec_err:
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 cache_err:
 	kfree(socdev->card->codec);
 	socdev->card->codec = NULL;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e4b946a19ea3..556123b4059c 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -763,7 +763,7 @@ static int aic3x_hw_params(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->card->codec;
-	struct aic3x_priv *aic3x = codec->private_data;
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
 	u16 d, pll_d = 1;
@@ -930,7 +930,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct aic3x_priv *aic3x = codec->private_data;
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 
 	aic3x->sysclk = freq;
 	return 0;
@@ -940,7 +940,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct aic3x_priv *aic3x = codec->private_data;
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	u8 iface_areg, iface_breg;
 	int delay = 0;
 
@@ -994,7 +994,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 				enum snd_soc_bias_level level)
 {
-	struct aic3x_priv *aic3x = codec->private_data;
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	u8 reg;
 
 	switch (level) {
@@ -1338,7 +1338,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 
 	codec = &aic3x->codec;
 	codec->dev = &i2c->dev;
-	codec->private_data = aic3x;
+	snd_soc_codec_set_drvdata(codec, aic3x);
 	codec->control_data = i2c;
 	codec->hw_write = (hw_write_t) i2c_master_send;
 
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index ee2e1e36d995..824bb354ebc9 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -167,7 +167,7 @@ static inline void dac33_write_reg_cache(struct snd_soc_codec *codec,
 static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 		      u8 *value)
 {
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int val;
 
 	*value = reg & 0xff;
@@ -192,7 +192,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
 static int dac33_write(struct snd_soc_codec *codec, unsigned int reg,
 		       unsigned int value)
 {
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	u8 data[2];
 	int ret = 0;
 
@@ -219,7 +219,7 @@ static int dac33_write(struct snd_soc_codec *codec, unsigned int reg,
 static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg,
 		       unsigned int value)
 {
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	mutex_lock(&dac33->mutex);
@@ -233,7 +233,7 @@ static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg,
 static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg,
 		       unsigned int value)
 {
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	u8 data[3];
 	int ret = 0;
 
@@ -265,7 +265,7 @@ static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg,
 
 static void dac33_restore_regs(struct snd_soc_codec *codec)
 {
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	u8 *cache = codec->reg_cache;
 	u8 data[2];
 	int i, ret;
@@ -319,7 +319,7 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
 
 static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 {
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	mutex_lock(&dac33->mutex);
@@ -366,7 +366,7 @@ static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	ucontrol->value.integer.value[0] = dac33->nsample;
 
@@ -377,7 +377,7 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
 	if (dac33->nsample == ucontrol->value.integer.value[0])
@@ -396,7 +396,7 @@ static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	ucontrol->value.integer.value[0] = dac33->fifo_mode;
 
@@ -407,7 +407,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
 	if (dac33->fifo_mode == ucontrol->value.integer.value[0])
@@ -630,7 +630,7 @@ static void dac33_work(struct work_struct *work)
 static irqreturn_t dac33_interrupt_handler(int irq, void *dev)
 {
 	struct snd_soc_codec *codec = dev;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	queue_work(dac33->dac33_wq, &dac33->work);
 
@@ -697,7 +697,7 @@ static int dac33_prepare_chip(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->card->codec;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned int oscset, ratioset, pwr_ctrl, reg_tmp;
 	u8 aictrl_a, aictrl_b, fifoctrl_a;
 
@@ -883,7 +883,7 @@ static void dac33_calculate_times(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->card->codec;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned int nsample_limit;
 
 	/* Number of samples (16bit, stereo) in one period */
@@ -936,7 +936,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
 	switch (cmd) {
@@ -967,7 +967,7 @@ static int dac33_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	u8 ioc_reg, asrcb_reg;
 
 	ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL);
@@ -997,7 +997,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct tlv320dac33_priv *dac33 = codec->private_data;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	u8 aictrl_a, aictrl_b;
 
 	aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A);
@@ -1088,7 +1088,7 @@ static int dac33_soc_probe(struct platform_device *pdev)
 
 	codec = tlv320dac33_codec;
 	socdev->card->codec = codec;
-	dac33 = codec->private_data;
+	dac33 = snd_soc_codec_get_drvdata(codec);
 
 	/* Power up the codec */
 	dac33_hard_power(codec, 1);
@@ -1209,7 +1209,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 		return -ENOMEM;
 
 	codec = &dac33->codec;
-	codec->private_data = dac33;
+	snd_soc_codec_set_drvdata(codec, dac33);
 	codec->control_data = client;
 
 	mutex_init(&codec->mutex);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index bf59b8a4d1d7..4eb5a2e59f3c 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -175,7 +175,7 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
 static int twl4030_write(struct snd_soc_codec *codec,
 			unsigned int reg, unsigned int value)
 {
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	int write_to_reg = 0;
 
 	twl4030_write_reg_cache(codec, reg, value);
@@ -220,7 +220,7 @@ static int twl4030_write(struct snd_soc_codec *codec,
 
 static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 {
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	int mode;
 
 	if (enable == twl4030->codec_powered)
@@ -258,7 +258,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	int status;
 
 	if (enable == twl4030->apll_enabled)
@@ -279,7 +279,7 @@ static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 
 static void twl4030_power_up(struct snd_soc_codec *codec)
 {
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 anamicl, regmisc1, byte;
 	int i = 0;
 
@@ -570,7 +570,7 @@ static int micpath_event(struct snd_soc_dapm_widget *w,
 static int pin_name##pga_event(struct snd_soc_dapm_widget *w,		\
 		struct snd_kcontrol *kcontrol, int event)		\
 {									\
-	struct twl4030_priv *twl4030 = w->codec->private_data;		\
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); \
 									\
 	switch (event) {						\
 	case SND_SOC_DAPM_POST_PMU:					\
@@ -678,7 +678,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 	struct twl4030_setup_data *setup = socdev->codec_data;
 
 	unsigned char hs_gain, hs_pop;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	/* Base values for ramp delay calculation: 2^19 - 2^26 */
 	unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304,
 				    8388608, 16777216, 33554432, 67108864};
@@ -741,7 +741,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 static int headsetlpga_event(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *kcontrol, int event)
 {
-	struct twl4030_priv *twl4030 = w->codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
@@ -765,7 +765,7 @@ static int headsetlpga_event(struct snd_soc_dapm_widget *w,
 static int headsetrpga_event(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *kcontrol, int event)
 {
-	struct twl4030_priv *twl4030 = w->codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
@@ -957,7 +957,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned short val;
 	unsigned short mask, bitmask;
@@ -1627,7 +1627,7 @@ static int twl4030_startup(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->card->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
 	if (twl4030->master_substream) {
 		twl4030->slave_substream = substream;
@@ -1658,7 +1658,7 @@ static void twl4030_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->card->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
 	if (twl4030->master_substream == substream)
 		twl4030->master_substream = twl4030->slave_substream;
@@ -1684,7 +1684,7 @@ static int twl4030_hw_params(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->card->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 mode, old_mode, format, old_format;
 
 	 /* If the substream has 4 channel, do the necessary setup */
@@ -1804,7 +1804,7 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 19200000:
@@ -1919,7 +1919,7 @@ static int twl4030_voice_startup(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->card->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	u8 mode;
 
 	/* If the system master clock is not 26MHz, the voice PCM interface is
@@ -2164,7 +2164,7 @@ static int twl4030_soc_probe(struct platform_device *pdev)
 	BUG_ON(!twl4030_codec);
 
 	codec = twl4030_codec;
-	twl4030 = codec->private_data;
+	twl4030 = snd_soc_codec_get_drvdata(codec);
 	socdev->card->codec = codec;
 
 	/* Configuration for headset ramp delay from setup data */
@@ -2227,7 +2227,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 	}
 
 	codec = &twl4030->codec;
-	codec->private_data = twl4030;
+	snd_soc_codec_set_drvdata(codec, twl4030);
 	codec->dev = &pdev->dev;
 	twl4030_dai[0].dev = &pdev->dev;
 	twl4030_dai[1].dev = &pdev->dev;
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 3e99fe5131dd..20deaca8e107 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -174,7 +174,7 @@ static int uda134x_startup(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->card->codec;
-	struct uda134x_priv *uda134x = codec->private_data;
+	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 	struct snd_pcm_runtime *master_runtime;
 
 	if (uda134x->master_substream) {
@@ -207,7 +207,7 @@ static void uda134x_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->card->codec;
-	struct uda134x_priv *uda134x = codec->private_data;
+	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 
 	if (uda134x->master_substream == substream)
 		uda134x->master_substream = uda134x->slave_substream;
@@ -222,7 +222,7 @@ static int uda134x_hw_params(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->card->codec;
-	struct uda134x_priv *uda134x = codec->private_data;
+	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 	u8 hw_params;
 
 	if (substream == uda134x->slave_substream) {
@@ -294,7 +294,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				  int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct uda134x_priv *uda134x = codec->private_data;
+	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 
 	pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
 		 clk_id, freq, dir);
@@ -316,7 +316,7 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			       unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct uda134x_priv *uda134x = codec->private_data;
+	struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec);
 
 	pr_debug("%s fmt: %08X\n", __func__, fmt);
 
@@ -503,7 +503,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
 	if (uda134x == NULL)
 		goto priv_err;
-	codec->private_data = uda134x;
+	snd_soc_codec_set_drvdata(codec, uda134x);
 
 	codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
 				   GFP_KERNEL);
@@ -567,7 +567,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 pcm_err:
 	kfree(codec->reg_cache);
 reg_err:
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 priv_err:
 	kfree(codec);
 	return ret;
@@ -585,7 +585,7 @@ static int uda134x_soc_remove(struct platform_device *pdev)
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 	kfree(codec->reg_cache);
 	kfree(codec);
 
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 9cd0a66b7663..ecba686563aa 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -476,7 +476,7 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct uda1380_priv *uda1380 = codec->private_data;
+	struct uda1380_priv *uda1380 = snd_soc_codec_get_drvdata(codec);
 	int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
 
 	switch (cmd) {
@@ -774,7 +774,7 @@ static int uda1380_register(struct uda1380_priv *uda1380)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = uda1380;
+	snd_soc_codec_set_drvdata(codec, uda1380);
 	codec->name = "UDA1380";
 	codec->owner = THIS_MODULE;
 	codec->read = uda1380_read_reg_cache;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 8ac92f5e4f91..145467d37054 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -95,7 +95,7 @@ static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
  */
 static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
 {
-	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out1 = &wm8350_data->out1;
 	struct wm8350 *wm8350 = codec->control_data;
 	int left_complete = 0, right_complete = 0;
@@ -161,7 +161,7 @@ static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
  */
 static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
 {
-	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out2 = &wm8350_data->out2;
 	struct wm8350 *wm8350 = codec->control_data;
 	int left_complete = 0, right_complete = 0;
@@ -231,7 +231,7 @@ static void wm8350_pga_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec =
 	    container_of(work, struct snd_soc_codec, delayed_work.work);
-	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out1 = &wm8350_data->out1,
 	    *out2 = &wm8350_data->out2;
 	int i, out1_complete, out2_complete;
@@ -278,7 +278,7 @@ static int pga_event(struct snd_soc_dapm_widget *w,
 		     struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out;
 
 	switch (w->shift) {
@@ -323,7 +323,7 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_data *wm8350_priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out = NULL;
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
@@ -366,7 +366,7 @@ static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_data *wm8350_priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_output *out1 = &wm8350_priv->out1;
 	struct wm8350_output *out2 = &wm8350_priv->out2;
 	struct soc_mixer_control *mc =
@@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct wm8350 *wm8350 = codec->control_data;
-	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct _fll_div fll_div;
 	int ret = 0;
 	u16 fll_1, fll_4;
@@ -1160,7 +1160,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
 	struct wm8350 *wm8350 = codec->control_data;
-	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350_audio_platform_data *platform =
 		wm8350->codec.platform_data;
 	u16 pm1;
@@ -1399,7 +1399,7 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
 int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 			  struct snd_soc_jack *jack, int report)
 {
-	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350 *wm8350 = codec->control_data;
 	int irq;
 	int ena;
@@ -1471,7 +1471,7 @@ int wm8350_mic_jack_detect(struct snd_soc_codec *codec,
 			   struct snd_soc_jack *jack,
 			   int detect_report, int short_report)
 {
-	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350 *wm8350 = codec->control_data;
 
 	priv->mic.jack = jack;
@@ -1508,7 +1508,7 @@ static int wm8350_probe(struct platform_device *pdev)
 	socdev->card->codec = wm8350_codec;
 	codec = socdev->card->codec;
 	wm8350 = codec->control_data;
-	priv = codec->private_data;
+	priv = snd_soc_codec_get_drvdata(codec);
 
 	/* Enable the codec */
 	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
@@ -1579,7 +1579,7 @@ static int wm8350_remove(struct platform_device *pdev)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8350 *wm8350 = codec->control_data;
-	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
@@ -1698,7 +1698,7 @@ static __devinit int wm8350_codec_probe(struct platform_device *pdev)
 	codec->dai = &wm8350_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = WM8350_MAX_REGISTER;
-	codec->private_data = priv;
+	snd_soc_codec_set_drvdata(codec, priv);
 	codec->control_data = wm8350;
 
 	/* Put the codec into reset if it wasn't already */
@@ -1730,7 +1730,7 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev)
 {
 	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = wm8350->codec.codec;
-	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 
 	snd_soc_unregister_dai(&wm8350_dai);
 	snd_soc_unregister_codec(codec);
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index b432f4d4a324..a7506ae2b8cc 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -76,7 +76,7 @@ struct wm8400_priv {
 static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
 				       unsigned int reg)
 {
-	struct wm8400_priv *wm8400 = codec->private_data;
+	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
 
 	if (reg == WM8400_INTDRIVBITS)
 		return wm8400->fake_register;
@@ -90,7 +90,7 @@ static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
 static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
 	unsigned int value)
 {
-	struct wm8400_priv *wm8400 = codec->private_data;
+	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
 
 	if (reg == WM8400_INTDRIVBITS) {
 		wm8400->fake_register = value;
@@ -101,7 +101,7 @@ static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
 
 static void wm8400_codec_reset(struct snd_soc_codec *codec)
 {
-	struct wm8400_priv *wm8400 = codec->private_data;
+	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
 
 	wm8400_reset_codec_reg_cache(wm8400->wm8400);
 }
@@ -925,7 +925,7 @@ static int wm8400_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8400_priv *wm8400 = codec->private_data;
+	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
 
 	wm8400->sysclk = freq;
 	return 0;
@@ -1014,7 +1014,7 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 			      unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8400_priv *wm8400 = codec->private_data;
+	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
 	struct fll_factors factors;
 	int ret;
 	u16 reg;
@@ -1203,7 +1203,7 @@ static int wm8400_mute(struct snd_soc_dai *dai, int mute)
 static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	struct wm8400_priv *wm8400 = codec->private_data;
+	struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec);
 	u16 val;
 	int ret;
 
@@ -1466,7 +1466,7 @@ static int wm8400_codec_probe(struct platform_device *dev)
 		return -ENOMEM;
 
 	codec = &priv->codec;
-	codec->private_data = priv;
+	snd_soc_codec_set_drvdata(codec, priv);
 	codec->control_data = dev_get_drvdata(&dev->dev);
 	priv->wm8400 = dev_get_drvdata(&dev->dev);
 
@@ -1529,7 +1529,7 @@ err:
 
 static int __exit wm8400_codec_remove(struct platform_device *dev)
 {
-	struct wm8400_priv *priv = wm8400_codec->private_data;
+	struct wm8400_priv *priv = snd_soc_codec_get_drvdata(wm8400_codec);
 	u16 reg;
 
 	snd_soc_unregister_dai(&wm8400_dai);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index d3a61d7ea0c5..2cdffb0b2a80 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -137,7 +137,7 @@ static int wm8523_startup(struct snd_pcm_substream *substream,
 			  struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8523_priv *wm8523 = codec->private_data;
+	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
 
 	/* The set of sample rates that can be supported depends on the
 	 * MCLK supplied to the CODEC - enforce this.
@@ -163,7 +163,7 @@ static int wm8523_hw_params(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->card->codec;
-	struct wm8523_priv *wm8523 = codec->private_data;
+	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
 	int i;
 	u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1);
 	u16 aifctrl2 = snd_soc_read(codec, WM8523_AIF_CTRL2);
@@ -210,7 +210,7 @@ static int wm8523_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8523_priv *wm8523 = codec->private_data;
+	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
 	int i;
 
@@ -317,7 +317,7 @@ static int wm8523_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int wm8523_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	struct wm8523_priv *wm8523 = codec->private_data;
+	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
 	int ret, i;
 
 	switch (level) {
@@ -488,7 +488,7 @@ static int wm8523_register(struct wm8523_priv *wm8523,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8523;
+	snd_soc_codec_set_drvdata(codec, wm8523);
 	codec->name = "WM8523";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d077df6f5e75..94b3f4c4d64e 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -411,7 +411,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 {
 	int offset;
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8580_priv *wm8580 = codec->private_data;
+	struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec);
 	struct pll_state *state;
 	struct _pll_div pll_div;
 	unsigned int reg;
@@ -839,7 +839,7 @@ static int wm8580_register(struct wm8580_priv *wm8580,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8580;
+	snd_soc_codec_set_drvdata(codec, wm8580);
 	codec->name = "WM8580";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 24a35603bcf7..f350c5ecd093 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -162,7 +162,7 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8711_priv *wm8711 = codec->private_data;
+	struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc;
 	int i = get_coeff(wm8711->sysclk, params_rate(params));
 	u16 srate = (coeff_div[i].sr << 2) |
@@ -226,7 +226,7 @@ static int wm8711_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8711_priv *wm8711 = codec->private_data;
+	struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -445,7 +445,7 @@ static int wm8711_register(struct wm8711_priv *wm8711,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8711;
+	snd_soc_codec_set_drvdata(codec, wm8711);
 	codec->name = "WM8711";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index d5e1c5b681e3..63f90c4ad93a 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -224,7 +224,7 @@ static int wm8731_hw_params(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->card->codec;
-	struct wm8731_priv *wm8731 = codec->private_data;
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3;
 	int i = get_coeff(wm8731->sysclk, params_rate(params));
 	u16 srate = (coeff_div[i].sr << 2) |
@@ -291,7 +291,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8731_priv *wm8731 = codec->private_data;
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -427,7 +427,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8731_priv *wm8731 = codec->private_data;
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
 	snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -440,7 +440,7 @@ static int wm8731_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8731_priv *wm8731 = codec->private_data;
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -535,7 +535,7 @@ static int wm8731_register(struct wm8731_priv *wm8731,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8731;
+	snd_soc_codec_set_drvdata(codec, wm8731);
 	codec->name = "WM8731";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 4bbc512c0836..60d7cf88bf1d 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -482,7 +482,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8750_priv *wm8750 = codec->private_data;
+	struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -561,7 +561,7 @@ static int wm8750_pcm_hw_params(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->card->codec;
-	struct wm8750_priv *wm8750 = codec->private_data;
+	struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3;
 	u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0;
 	int coeff = get_coeff(wm8750->sysclk, params_rate(params));
@@ -775,10 +775,9 @@ static int wm8750_register(struct wm8750_priv *wm8750,
 	codec->set_bias_level = wm8750_set_bias_level;
 	codec->dai = &wm8750_dai;
 	codec->num_dai = 1;
-	codec->private_data = wm8750;
 	codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1;
 	codec->reg_cache = &wm8750->reg_cache;
-	codec->private_data = wm8750;
+	snd_soc_codec_set_drvdata(codec, wm8750);
 
 	memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache));
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index c2444e7c8480..f9e56a1dfde1 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -850,7 +850,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8753_priv *wm8753 = codec->private_data;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -913,7 +913,7 @@ static int wm8753_pcm_hw_params(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->card->codec;
-	struct wm8753_priv *wm8753 = codec->private_data;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
 
@@ -1147,7 +1147,7 @@ static int wm8753_i2s_hw_params(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->card->codec;
-	struct wm8753_priv *wm8753 = codec->private_data;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
 	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
 	int coeff;
@@ -1645,7 +1645,7 @@ static int wm8753_register(struct wm8753_priv *wm8753)
 	codec->num_dai = 2;
 	codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache) + 1;
 	codec->reg_cache = &wm8753->reg_cache;
-	codec->private_data = wm8753;
+	snd_soc_codec_set_drvdata(codec, wm8753);
 
 	memcpy(codec->reg_cache, wm8753_reg, sizeof(wm8753->reg_cache));
 	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 44e7d9d82f87..694600969d67 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -226,7 +226,7 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8776_priv *wm8776 = codec->private_data;
+	struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec);
 	int iface_reg, iface;
 	int ratio_shift, master;
 	int i;
@@ -303,7 +303,7 @@ static int wm8776_set_sysclk(struct snd_soc_dai *dai,
 			     int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8776_priv *wm8776 = codec->private_data;
+	struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec);
 
 	BUG_ON(dai->id >= ARRAY_SIZE(wm8776->sysclk));
 
@@ -490,7 +490,7 @@ static int wm8776_register(struct wm8776_priv *wm8776,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8776;
+	snd_soc_codec_set_drvdata(codec, wm8776);
 	codec->name = "WM8776";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index dbc368c08263..19a6b25988c8 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -744,7 +744,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 static int wm8900_set_fll(struct snd_soc_codec *codec,
 	int fll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	struct wm8900_priv *wm8900 = codec->private_data;
+	struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec);
 	struct _fll_div fll_div;
 	unsigned int reg;
 
@@ -1131,7 +1131,7 @@ static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8900_priv *wm8900 = codec->private_data;
+	struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec);
 	int fll_out = wm8900->fll_out;
 	int fll_in  = wm8900->fll_in;
 	int ret;
@@ -1155,7 +1155,7 @@ static int wm8900_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8900_priv *wm8900 = codec->private_data;
+	struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec);
 	u16 *cache;
 	int i, ret;
 
@@ -1205,7 +1205,7 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	codec = &wm8900->codec;
-	codec->private_data = wm8900;
+	snd_soc_codec_set_drvdata(codec, wm8900);
 	codec->reg_cache = &wm8900->reg_cache[0];
 	codec->reg_cache_size = WM8900_MAXREG;
 
@@ -1304,7 +1304,7 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client)
 	wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF);
 
 	wm8900_dai.dev = NULL;
-	kfree(wm8900_codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(wm8900_codec));
 	wm8900_codec = NULL;
 
 	return 0;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 134b17532f23..5a2510c39c10 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -252,7 +252,7 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
 {
 	u16 reg[5];
 	struct i2c_client *i2c = codec->control_data;
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
 	BUG_ON(start > 48);
 
@@ -420,7 +420,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
 	struct snd_soc_codec *codec = widget->codec;
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = codec->control_data;
 	u16 reg;
 	int ret;
@@ -1001,7 +1001,7 @@ static int wm8903_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
 	wm8903->sysclk = freq;
 
@@ -1229,7 +1229,7 @@ static int wm8903_startup(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->card->codec;
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
 
@@ -1265,7 +1265,7 @@ static void wm8903_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->card->codec;
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		wm8903->playback_active--;
@@ -1285,7 +1285,7 @@ static int wm8903_hw_params(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->card->codec;
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = codec->control_data;
 	int fs = params_rate(params);
 	int bclk;
@@ -1464,7 +1464,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		      int det, int shrt)
 {
-	struct wm8903_priv *wm8903 = codec->private_data;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT;
 
 	dev_dbg(codec->dev, "Enabling microphone detection: %x %x\n",
@@ -1674,7 +1674,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 	codec->num_dai = 1;
 	codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache);
 	codec->reg_cache = &wm8903->reg_cache[0];
-	codec->private_data = wm8903;
+	snd_soc_codec_set_drvdata(codec, wm8903);
 	codec->volatile_register = wm8903_volatile_register;
 	init_completion(&wm8903->wseq);
 
@@ -1812,7 +1812,7 @@ err:
 static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	struct wm8903_priv *priv = codec->private_data;
+	struct wm8903_priv *priv = snd_soc_codec_get_drvdata(codec);
 
 	snd_soc_unregister_dai(&wm8903_dai);
 	snd_soc_unregister_codec(codec);
@@ -1822,7 +1822,7 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 	if (client->irq)
 		free_irq(client->irq, priv);
 
-	kfree(codec->private_data);
+	kfree(priv);
 
 	wm8903_codec = NULL;
 	wm8903_dai.dev = NULL;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index c5b50d73f2d5..a022ca7e7585 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -612,7 +612,7 @@ static int wm8904_reset(struct snd_soc_codec *codec)
 
 static int wm8904_configure_clocking(struct snd_soc_codec *codec)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	unsigned int clock0, clock2, rate;
 
 	/* Gate the clock while we're updating to avoid misclocking */
@@ -668,7 +668,7 @@ static int wm8904_configure_clocking(struct snd_soc_codec *codec)
 
 static void wm8904_set_drc(struct snd_soc_codec *codec)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	struct wm8904_pdata *pdata = wm8904->pdata;
 	int save, i;
 
@@ -688,7 +688,7 @@ static int wm8904_put_drc_enum(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8904_priv *wm8904 = codec->private_data;	
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);	
 	struct wm8904_pdata *pdata = wm8904->pdata;
 	int value = ucontrol->value.integer.value[0];
 
@@ -706,7 +706,7 @@ static int wm8904_get_drc_enum(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
 	ucontrol->value.enumerated.item[0] = wm8904->drc_cfg;
 
@@ -715,7 +715,7 @@ static int wm8904_get_drc_enum(struct snd_kcontrol *kcontrol,
 
 static void wm8904_set_retune_mobile(struct snd_soc_codec *codec)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	struct wm8904_pdata *pdata = wm8904->pdata;
 	int best, best_val, save, i, cfg;
 
@@ -759,7 +759,7 @@ static int wm8904_put_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8904_priv *wm8904 = codec->private_data;	
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);	
 	struct wm8904_pdata *pdata = wm8904->pdata;
 	int value = ucontrol->value.integer.value[0];
 
@@ -777,7 +777,7 @@ static int wm8904_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
 	ucontrol->value.enumerated.item[0] = wm8904->retune_mobile_cfg;
 
@@ -788,7 +788,7 @@ static int deemph_settings[] = { 0, 32000, 44100, 48000 };
 
 static int wm8904_set_deemph(struct snd_soc_codec *codec)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int val, i, best;
 
 	/* If we're using deemphasis select the nearest available sample 
@@ -817,7 +817,7 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
 	return wm8904->deemph;
 }
@@ -826,7 +826,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int deemph = ucontrol->value.enumerated.item[0];
 
 	if (deemph > 1)
@@ -942,7 +942,7 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
@@ -980,7 +980,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int reg, val;
 	int dcs_mask;
 	int dcs_l, dcs_r;
@@ -1428,7 +1428,7 @@ static const struct snd_soc_dapm_route wm8912_intercon[] = {
 
 static int wm8904_add_widgets(struct snd_soc_codec *codec)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
 	snd_soc_dapm_new_controls(codec, wm8904_core_dapm_widgets,
 				  ARRAY_SIZE(wm8904_core_dapm_widgets));
@@ -1542,7 +1542,7 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int ret, i, best, best_val, cur_val;
 	unsigned int aif1 = 0;
 	unsigned int aif2 = 0;
@@ -1669,7 +1669,7 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 			     unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8904_priv *priv = codec->private_data;
+	struct wm8904_priv *priv = snd_soc_codec_get_drvdata(codec);
 
 	switch (clk_id) {
 	case WM8904_CLK_MCLK:
@@ -1785,7 +1785,7 @@ static int wm8904_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 			       unsigned int rx_mask, int slots, int slot_width)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int aif1 = 0;
 
 	/* Don't need to validate anything if we're turning off TDM */
@@ -1942,7 +1942,7 @@ static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
 			  unsigned int Fref, unsigned int Fout)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	struct _fll_div fll_div;
 	int ret, val;
 	int clock2, fll1;
@@ -2094,7 +2094,7 @@ static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 
 static void wm8904_sync_cache(struct snd_soc_codec *codec)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int i;
 
 	if (!codec->cache_sync)
@@ -2121,7 +2121,7 @@ static void wm8904_sync_cache(struct snd_soc_codec *codec)
 static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	struct wm8904_priv *wm8904 = codec->private_data;
+	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (level) {
@@ -2394,7 +2394,7 @@ static int wm8904_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	wm8904_handle_pdata(codec->private_data);
+	wm8904_handle_pdata(snd_soc_codec_get_drvdata(codec));
 
 	wm8904_add_widgets(codec);
 
@@ -2439,7 +2439,7 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8904;
+	snd_soc_codec_set_drvdata(codec, wm8904);
 	codec->name = "WM8904";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 31e39ffd1d8e..b2f12828630c 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -580,7 +580,7 @@ static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8940_priv *wm8940 = codec->private_data;
+	struct wm8940_priv *wm8940 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -772,7 +772,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8940;
+	snd_soc_codec_set_drvdata(codec, wm8940);
 	codec->name = "WM8940";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 615dab2b62ef..5f0bde56cf0d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -234,7 +234,7 @@ static struct {
 
 static int wm8955_configure_clocking(struct snd_soc_codec *codec)
 {
-	struct wm8955_priv *wm8955 = codec->private_data;
+	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 	int i, ret, val;
 	int clocking = 0;
 	int srate = 0;
@@ -352,7 +352,7 @@ static int deemph_settings[] = { 0, 32000, 44100, 48000 };
 
 static int wm8955_set_deemph(struct snd_soc_codec *codec)
 {
-	struct wm8955_priv *wm8955 = codec->private_data;
+	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 	int val, i, best;
 
 	/* If we're using deemphasis select the nearest available sample
@@ -381,7 +381,7 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8955_priv *wm8955 = codec->private_data;
+	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 
 	return wm8955->deemph;
 }
@@ -390,7 +390,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8955_priv *wm8955 = codec->private_data;
+	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 	int deemph = ucontrol->value.enumerated.item[0];
 
 	if (deemph > 1)
@@ -597,7 +597,7 @@ static int wm8955_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8955_priv *wm8955 = codec->private_data;
+	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 	int wl;
 
@@ -646,7 +646,7 @@ static int wm8955_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 			     unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8955_priv *priv = codec->private_data;
+	struct wm8955_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int div;
 
 	switch (clk_id) {
@@ -769,7 +769,7 @@ static int wm8955_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 static int wm8955_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	struct wm8955_priv *wm8955 = codec->private_data;
+	struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
 	int ret, i;
 
 	switch (level) {
@@ -970,7 +970,7 @@ static int wm8955_register(struct wm8955_priv *wm8955,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8955;
+	snd_soc_codec_set_drvdata(codec, wm8955);
 	codec->name = "WM8955";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index c2960d3ec6df..eba3ac013119 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -331,7 +331,7 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = {
 static int wm8960_add_widgets(struct snd_soc_codec *codec)
 {
 	struct wm8960_data *pdata = codec->dev->platform_data;
-	struct wm8960_priv *wm8960 = codec->private_data;
+	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_widget *w;
 
 	snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
@@ -534,7 +534,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
 static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
 					 enum snd_soc_bias_level level)
 {
-	struct wm8960_priv *wm8960 = codec->private_data;
+	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 	int reg;
 
 	switch (level) {
@@ -915,7 +915,7 @@ static int wm8960_register(struct wm8960_priv *wm8960,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8960;
+	snd_soc_codec_set_drvdata(codec, wm8960);
 	codec->name = "WM8960";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index d2342c5e0425..bab393feb945 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -630,7 +630,7 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8961_priv *wm8961 = codec->private_data;
+	struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec);
 	int i, best, target, fs;
 	u16 reg;
 
@@ -721,7 +721,7 @@ static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 			     int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8961_priv *wm8961 = codec->private_data;
+	struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec);
 	u16 reg = snd_soc_read(codec, WM8961_CLOCKING1);
 
 	if (freq > 33000000) {
@@ -1064,7 +1064,7 @@ static int wm8961_register(struct wm8961_priv *wm8961)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8961;
+	snd_soc_codec_set_drvdata(codec, wm8961);
 	codec->name = "WM8961";
 	codec->owner = THIS_MODULE;
 	codec->dai = &wm8961_dai;
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index d9540d55fc89..95c9ec3862d5 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -414,7 +414,7 @@ static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8971_priv *wm8971 = codec->private_data;
+	struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -493,7 +493,7 @@ static int wm8971_pcm_hw_params(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->card->codec;
-	struct wm8971_priv *wm8971 = codec->private_data;
+	struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3;
 	u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0;
 	int coeff = get_coeff(wm8971->sysclk, params_rate(params));
@@ -819,7 +819,7 @@ static int wm8971_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	}
 
-	codec->private_data = wm8971;
+	snd_soc_codec_set_drvdata(codec, wm8971);
 	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -829,7 +829,7 @@ static int wm8971_probe(struct platform_device *pdev)
 	INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work);
 	wm8971_workq = create_workqueue("wm8971");
 	if (wm8971_workq == NULL) {
-		kfree(codec->private_data);
+		kfree(snd_soc_codec_get_drvdata(codec));
 		kfree(codec);
 		return -ENOMEM;
 	}
@@ -843,7 +843,7 @@ static int wm8971_probe(struct platform_device *pdev)
 
 	if (ret != 0) {
 		destroy_workqueue(wm8971_workq);
-		kfree(codec->private_data);
+		kfree(snd_soc_codec_get_drvdata(codec));
 		kfree(codec);
 	}
 
@@ -866,7 +866,7 @@ static int wm8971_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8971_i2c_driver);
 #endif
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index ee637af4737a..1e4da32c4078 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -676,7 +676,7 @@ static __devinit int wm8974_register(struct wm8974_priv *wm8974)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8974;
+	snd_soc_codec_set_drvdata(codec, wm8974);
 	codec->name = "WM8974";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 28bb59ea6ea1..6866a6a80fec 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -438,7 +438,7 @@ static int wm8978_enum_mclk(unsigned int f_out, unsigned int f_mclk,
  */
 static int wm8978_configure_pll(struct snd_soc_codec *codec)
 {
-	struct wm8978_priv *wm8978 = codec->private_data;
+	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 	struct wm8978_pll_div pll_div;
 	unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk,
 		f_256fs = wm8978->f_256fs;
@@ -534,7 +534,7 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 				 int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8978_priv *wm8978 = codec->private_data;
+	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
 	switch (div_id) {
@@ -579,7 +579,7 @@ static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
 				 unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8978_priv *wm8978 = codec->private_data;
+	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
 	dev_dbg(codec->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq);
@@ -691,7 +691,7 @@ static int wm8978_hw_params(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->card->codec;
-	struct wm8978_priv *wm8978 = codec->private_data;
+	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 	/* Word length mask = 0x60 */
 	u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60;
 	/* Sampling rate mask = 0xe (for filters) */
@@ -911,7 +911,7 @@ static int wm8978_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8978_priv *wm8978 = codec->private_data;
+	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
 	int i;
 	u16 *cache = codec->reg_cache;
 
@@ -1019,7 +1019,7 @@ static __devinit int wm8978_register(struct wm8978_priv *wm8978)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8978;
+	snd_soc_codec_set_drvdata(codec, wm8978);
 	codec->name = "WM8978";
 	codec->owner = THIS_MODULE;
 	codec->bias_level = SND_SOC_BIAS_OFF;
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 2862e4dced27..563cf898dafc 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -494,7 +494,7 @@ static int wm8988_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8988_priv *wm8988 = codec->private_data;
+	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
 
 	switch (freq) {
 	case 11289600:
@@ -584,7 +584,7 @@ static int wm8988_pcm_startup(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8988_priv *wm8988 = codec->private_data;
+	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
 
 	/* The set of sample rates that can be supported depends on the
 	 * MCLK supplied to the CODEC - enforce this.
@@ -609,7 +609,7 @@ static int wm8988_pcm_hw_params(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->card->codec;
-	struct wm8988_priv *wm8988 = codec->private_data;
+	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
 	u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3;
 	u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180;
 	int coeff;
@@ -832,7 +832,7 @@ static int wm8988_register(struct wm8988_priv *wm8988,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8988;
+	snd_soc_codec_set_drvdata(codec, wm8988);
 	codec->name = "WM8988";
 	codec->owner = THIS_MODULE;
 	codec->dai = &wm8988_dai;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 056b787b6ee0..9a9528e9044e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1011,7 +1011,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8990_priv *wm8990 = codec->private_data;
+	struct wm8990_priv *wm8990 = snd_soc_codec_get_drvdata(codec);
 
 	wm8990->sysclk = freq;
 	return 0;
@@ -1523,7 +1523,7 @@ static int wm8990_probe(struct platform_device *pdev)
 		return -ENOMEM;
 	}
 
-	codec->private_data = wm8990;
+	snd_soc_codec_set_drvdata(codec, wm8990);
 	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -1540,7 +1540,7 @@ static int wm8990_probe(struct platform_device *pdev)
 #endif
 
 	if (ret != 0) {
-		kfree(codec->private_data);
+		kfree(snd_soc_codec_get_drvdata(codec));
 		kfree(codec);
 	}
 	return ret;
@@ -1560,7 +1560,7 @@ static int wm8990_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8990_i2c_driver);
 #endif
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index bf022f68b84f..e739c5008b1b 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -370,7 +370,7 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
 			  unsigned int Fref, unsigned int Fout)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	u16 reg1, reg4, reg5;
 	struct _fll_div fll_div;
 	int ret;
@@ -457,7 +457,7 @@ static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
 
 static int configure_clock(struct snd_soc_codec *codec)
 {
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	unsigned int reg;
 
 	/* This should be done on init() for bypass paths */
@@ -716,7 +716,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
 	struct snd_soc_codec *codec = widget->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	/* Turn it off if we're using the main output mixer */
@@ -948,7 +948,7 @@ static void wm8993_cache_restore(struct snd_soc_codec *codec)
 static int wm8993_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (level) {
@@ -1046,7 +1046,7 @@ static int wm8993_set_sysclk(struct snd_soc_dai *codec_dai,
 			     int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 
 	switch (clk_id) {
 	case WM8993_SYSCLK_MCLK:
@@ -1066,7 +1066,7 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai,
 			      unsigned int fmt)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	unsigned int aif1 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_1);
 	unsigned int aif4 = snd_soc_read(codec, WM8993_AUDIO_INTERFACE_4);
 
@@ -1162,7 +1162,7 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int ret, i, best, best_val, cur_val;
 	unsigned int clocking1, clocking3, aif1, aif4;
 
@@ -1327,7 +1327,7 @@ static int wm8993_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 			       unsigned int rx_mask, int slots, int slot_width)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int aif1 = 0;
 	int aif2 = 0;
 
@@ -1430,7 +1430,7 @@ static int wm8993_probe(struct platform_device *pdev)
 
 	socdev->card->codec = wm8993_codec;
 	codec = wm8993_codec;
-	wm8993 = codec->private_data;
+	wm8993 = snd_soc_codec_get_drvdata(codec);
 
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -1477,7 +1477,7 @@ static int wm8993_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int fll_fout = wm8993->fll_fout;
 	int fll_fref  = wm8993->fll_fref;
 	int ret;
@@ -1501,7 +1501,7 @@ static int wm8993_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8993_priv *wm8993 = codec->private_data;
+	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1570,7 +1570,7 @@ static int wm8993_i2c_probe(struct i2c_client *i2c,
 	codec->set_bias_level = wm8993_set_bias_level;
 	codec->dai = &wm8993_dai;
 	codec->num_dai = 1;
-	codec->private_data = wm8993;
+	snd_soc_codec_set_drvdata(codec, wm8993);
 
 	wm8993->hubs_data.hp_startup_mode = 1;
 	wm8993->hubs_data.dcs_codes = -2;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index e00201e0820e..29c4cfccd6b9 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1703,7 +1703,7 @@ static int wm8994_volatile(unsigned int reg)
 static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg,
 	unsigned int value)
 {
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
 	BUG_ON(reg > WM8994_MAX_REGISTER);
 
@@ -1728,7 +1728,7 @@ static unsigned int wm8994_read(struct snd_soc_codec *codec,
 
 static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 {
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int rate;
 	int reg1 = 0;
 	int offset;
@@ -1780,7 +1780,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 
 static int configure_clock(struct snd_soc_codec *codec)
 {
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int old, new;
 
 	/* Bring up the AIF clocks first */
@@ -1877,7 +1877,7 @@ static int wm8994_put_drc_sw(struct snd_kcontrol *kcontrol,
 
 static void wm8994_set_drc(struct snd_soc_codec *codec, int drc)
 {
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994_pdata *pdata = wm8994->pdata;
 	int base = wm8994_drc_base[drc];
 	int cfg = wm8994->drc_cfg[drc];
@@ -1913,7 +1913,7 @@ static int wm8994_put_drc_enum(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8994_priv *wm8994 = codec->private_data;	
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);	
 	struct wm8994_pdata *pdata = wm8994->pdata;
 	int drc = wm8994_get_drc(kcontrol->id.name);
 	int value = ucontrol->value.integer.value[0];
@@ -1935,7 +1935,7 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int drc = wm8994_get_drc(kcontrol->id.name);
 
 	ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
@@ -1945,7 +1945,7 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
 
 static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block)
 {
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994_pdata *pdata = wm8994->pdata;
 	int base = wm8994_retune_mobile_base[block];
 	int iface, best, best_val, save, i, cfg;
@@ -2016,7 +2016,7 @@ static int wm8994_put_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8994_priv *wm8994 = codec->private_data;	
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);	
 	struct wm8994_pdata *pdata = wm8994->pdata;
 	int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 	int value = ucontrol->value.integer.value[0];
@@ -2038,7 +2038,7 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
 	ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
@@ -2819,7 +2819,7 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
 			  unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int reg_offset, ret;
 	struct fll_div fll;
 	u16 reg, aif1, aif2;
@@ -2915,7 +2915,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
 	switch (dai->id) {
 	case 1:
@@ -3181,7 +3181,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int aif1_reg;
 	int bclk_reg;
 	int lrclk_reg;
@@ -3468,7 +3468,7 @@ static int wm8994_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
 
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
@@ -3489,7 +3489,7 @@ static int wm8994_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	u16 *reg_cache = codec->reg_cache;
 	int i, ret;
 
@@ -3684,7 +3684,7 @@ static int wm8994_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	wm8994_handle_pdata(codec->private_data);
+	wm8994_handle_pdata(snd_soc_codec_get_drvdata(codec));
 
 	wm_hubs_add_analogue_controls(codec);
 	snd_soc_add_controls(codec, wm8994_snd_controls,
@@ -3735,7 +3735,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8994);
 int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		      int micbias, int det, int shrt)
 {
-	struct wm8994_priv *wm8994 = codec->private_data;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994_micdet *micdet;
 	int reg;
 
@@ -3830,7 +3830,7 @@ static int wm8994_codec_probe(struct platform_device *pdev)
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm8994;
+	snd_soc_codec_set_drvdata(codec, wm8994);
 	codec->control_data = dev_get_drvdata(pdev->dev.parent);
 	codec->name = "WM8994";
 	codec->owner = THIS_MODULE;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index c468497314ba..87a87487d3f9 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -520,7 +520,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
 			  unsigned int Fref, unsigned int Fout)
 {
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 	u16 reg1, reg4, reg5;
 	struct _fll_div fll_div;
 	int ret;
@@ -606,7 +606,7 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
 
 static int configure_clock(struct snd_soc_codec *codec)
 {
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 	int new_sysclk, i, target;
 	unsigned int reg;
 	int ret = 0;
@@ -701,7 +701,7 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 
 	/* This should be done on init() for bypass paths */
 	switch (wm9081->sysclk_source) {
@@ -872,7 +872,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
 			      unsigned int fmt)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 	unsigned int aif2 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_2);
 
 	aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV |
@@ -964,7 +964,7 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 	int ret, i, best, best_val, cur_val;
 	unsigned int clk_ctrl2, aif1, aif2, aif3, aif4;
 
@@ -1138,7 +1138,7 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
 			     int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 
 	switch (clk_id) {
 	case WM9081_SYSCLK_MCLK:
@@ -1158,7 +1158,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm9081_priv *wm9081 = codec->private_data;
+	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 	unsigned int aif1 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_1);
 
 	aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK);
@@ -1241,7 +1241,7 @@ static int wm9081_probe(struct platform_device *pdev)
 
 	socdev->card->codec = wm9081_codec;
 	codec = wm9081_codec;
-	wm9081 = codec->private_data;
+	wm9081 = snd_soc_codec_get_drvdata(codec);
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -1338,7 +1338,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	codec->private_data = wm9081;
+	snd_soc_codec_set_drvdata(codec, wm9081);
 	codec->name = "WM9081";
 	codec->owner = THIS_MODULE;
 	codec->dai = &wm9081_dai;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index ceb86b4ddb25..d2244c93f581 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -763,7 +763,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source)
 static int wm9713_set_pll(struct snd_soc_codec *codec,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	struct wm9713_priv *wm9713 = codec->private_data;
+	struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
 	u16 reg, reg2;
 	struct _pll_div pll_div;
 
@@ -1174,7 +1174,7 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm9713_priv *wm9713 = codec->private_data;
+	struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
 	u16 *cache = codec->reg_cache;
 
@@ -1227,8 +1227,9 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	codec->reg_cache_size = sizeof(wm9713_reg);
 	codec->reg_cache_step = 2;
 
-	codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL);
-	if (codec->private_data == NULL) {
+	snd_soc_codec_set_drvdata(codec, kzalloc(sizeof(struct wm9713_priv),
+						 GFP_KERNEL));
+	if (snd_soc_codec_get_drvdata(codec) == NULL) {
 		ret = -ENOMEM;
 		goto priv_err;
 	}
@@ -1279,7 +1280,7 @@ pcm_err:
 	snd_soc_free_ac97_codec(codec);
 
 codec_err:
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 
 priv_err:
 	kfree(codec->reg_cache);
@@ -1301,7 +1302,7 @@ static int wm9713_soc_remove(struct platform_device *pdev)
 	snd_soc_dapm_free(socdev);
 	snd_soc_free_pcms(socdev);
 	snd_soc_free_ac97_codec(codec);
-	kfree(codec->private_data);
+	kfree(snd_soc_codec_get_drvdata(codec));
 	kfree(codec->reg_cache);
 	kfree(codec);
 	return 0;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e1f225a3ac46..defdbd08e135 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,7 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
  */
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
-	struct wm_hubs_data *hubs = codec->private_data;
+	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
 	/* Set for 32 series updates */
@@ -154,7 +154,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm_hubs_data *hubs = codec->private_data;
+	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
@@ -327,7 +327,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w,
 			   struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct wm_hubs_data *hubs = codec->private_data;
+	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 
 	switch (event) {
 	case SND_SOC_DAPM_PRE_PMU:
-- 
cgit v1.2.3


From d4a8ca246146099e9068fc87cd2bc565896bc186 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 20 Apr 2010 08:20:31 +0200
Subject: ASoC: missing conversions to snd_soc_codec_*_drvdata()

Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files
in the previous commit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/tlv320aic26.c | 14 +++++++-------
 sound/soc/codecs/twl4030.c     |  2 +-
 sound/soc/codecs/twl6040.c     | 20 ++++++++++----------
 3 files changed, 18 insertions(+), 18 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 357b609196e3..5856f7aae4d9 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -48,7 +48,7 @@ struct aic26 {
 static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
 				   unsigned int reg)
 {
-	struct aic26 *aic26 = codec->private_data;
+	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 	u16 *cache = codec->reg_cache;
 	u16 cmd, value;
 	u8 buffer[2];
@@ -92,7 +92,7 @@ static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
 static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
 			   unsigned int value)
 {
-	struct aic26 *aic26 = codec->private_data;
+	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 	u16 *cache = codec->reg_cache;
 	u16 cmd;
 	u8 buffer[4];
@@ -131,7 +131,7 @@ static int aic26_hw_params(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->card->codec;
-	struct aic26 *aic26 = codec->private_data;
+	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 	int fsref, divisor, wlen, pval, jval, dval, qval;
 	u16 reg;
 
@@ -198,7 +198,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 static int aic26_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct aic26 *aic26 = codec->private_data;
+	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 	u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
 
 	dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
@@ -217,7 +217,7 @@ static int aic26_set_sysclk(struct snd_soc_dai *codec_dai,
 			    int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct aic26 *aic26 = codec->private_data;
+	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 
 	dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
 		" freq=%i, dir=%i)\n",
@@ -234,7 +234,7 @@ static int aic26_set_sysclk(struct snd_soc_dai *codec_dai,
 static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct aic26 *aic26 = codec->private_data;
+	struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec);
 
 	dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
 		codec_dai, fmt);
@@ -430,7 +430,7 @@ static int aic26_spi_probe(struct spi_device *spi)
 	/* Setup what we can in the codec structure so that the register
 	 * access functions will work as expected.  More will be filled
 	 * out when it is probed by the SoC CODEC part of this driver */
-	aic26->codec.private_data = aic26;
+	snd_soc_codec_set_drvdata(&aic26->codec, aic26);
 	aic26->codec.name = "aic26";
 	aic26->codec.owner = THIS_MODULE;
 	aic26->codec.dai = &aic26_dai;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 4eb5a2e59f3c..2e025a3a2618 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2001,7 +2001,7 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct twl4030_priv *twl4030 = codec->private_data;
+	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 
 	if (freq != 26000000) {
 		dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice"
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 108c51a513c8..ead6d4b43b25 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -364,7 +364,7 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
 			struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
 	if (SND_SOC_DAPM_EVENT_ON(event))
 		priv->non_lp++;
@@ -378,7 +378,7 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
 static irqreturn_t twl6040_naudint_handler(int irq, void *data)
 {
 	struct snd_soc_codec *codec = data;
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	u8 intid;
 
 	twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID);
@@ -636,7 +636,7 @@ static int twl6040_add_widgets(struct snd_soc_codec *codec)
 static int twl6040_power_up_completion(struct snd_soc_codec *codec,
 					int naudint)
 {
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	int time_left;
 	u8 intid;
 
@@ -660,7 +660,7 @@ static int twl6040_power_up_completion(struct snd_soc_codec *codec,
 static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 				enum snd_soc_bias_level level)
 {
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	int audpwron = priv->audpwron;
 	int naudint = priv->naudint;
 	int ret;
@@ -753,7 +753,7 @@ static int twl6040_startup(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->card->codec;
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
 	if (!priv->sysclk) {
 		dev_err(codec->dev,
@@ -786,7 +786,7 @@ static int twl6040_hw_params(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->card->codec;
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	u8 lppllctl;
 	int rate;
 
@@ -822,7 +822,7 @@ static int twl6040_trigger(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->card->codec;
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -849,7 +849,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct twl6040_data *priv = codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec);
 	u8 hppllctl, lppllctl;
 
 	hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL);
@@ -1095,7 +1095,7 @@ static int __devinit twl6040_codec_probe(struct platform_device *pdev)
 	codec->read = twl6040_read_reg_cache;
 	codec->write = twl6040_write;
 	codec->set_bias_level = twl6040_set_bias_level;
-	codec->private_data = priv;
+	snd_soc_codec_set_drvdata(codec, priv);
 	codec->dai = &twl6040_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = ARRAY_SIZE(twl6040_reg);
@@ -1183,7 +1183,7 @@ cache_err:
 
 static int __devexit twl6040_codec_remove(struct platform_device *pdev)
 {
-	struct twl6040_data *priv = twl6040_codec->private_data;
+	struct twl6040_data *priv = snd_soc_codec_get_drvdata(twl6040_codec);
 	int audpwron = priv->audpwron;
 	int naudint = priv->naudint;
 
-- 
cgit v1.2.3


From 4f6f22d7bef77dfb6b27eaed4240784339c546e6 Mon Sep 17 00:00:00 2001
From: Phil Carmody <ext-phil.2.carmody@nokia.com>
Date: Wed, 14 Apr 2010 17:03:13 +0300
Subject: ASoC: da7210: Fencepost error in reg cache read

An index equal to the array size may not be accessed.

Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/da7210.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 89cea32f9424..b9cbaeb09e05 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -181,7 +181,7 @@ static const u8 da7210_reg[] = {
 static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg)
 {
 	u8 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(da7210_reg));
+	BUG_ON(reg >= ARRAY_SIZE(da7210_reg));
 	return cache[reg];
 }
 
-- 
cgit v1.2.3


From 136ff2a272ad4bee33bf85f8c490ff8a2dd08f96 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 20 Apr 2010 12:56:18 +0900
Subject: ASoC: Support FLL input clock selection on WM8994

The WM8994 FLL can be clocked from one of four inputs, the two MCLKs and
the LRCLK and BCLK of the AIF associated with the FLL. Allow all four
inputs to be used rather than defaulting to MCLK1.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 17 +++++++++++++++--
 sound/soc/codecs/wm8994.h |  5 +++++
 2 files changed, 20 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 29c4cfccd6b9..a27b2ff769d9 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2843,6 +2843,16 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
 		return -EINVAL;
 	}
 
+	switch (src) {
+	case WM8994_FLL_SRC_MCLK1:
+	case WM8994_FLL_SRC_MCLK2:
+	case WM8994_FLL_SRC_LRCLK:
+	case WM8994_FLL_SRC_BCLK:
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	/* Are we changing anything? */
 	if (wm8994->fll[id].src == src &&
 	    wm8994->fll[id].in == freq_in && wm8994->fll[id].out == freq_out)
@@ -2883,8 +2893,10 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
 				    fll.n << WM8994_FLL1_N_SHIFT);
 
 	snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
-			    WM8994_FLL1_REFCLK_DIV_MASK,
-			    fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT);
+			    WM8994_FLL1_REFCLK_DIV_MASK |
+			    WM8994_FLL1_REFCLK_SRC_MASK,
+			    (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) |
+			    (src - 1));
 
 	/* Enable (with fractional mode if required) */
 	if (freq_out) {
@@ -2899,6 +2911,7 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
 
 	wm8994->fll[id].in = freq_in;
 	wm8994->fll[id].out = freq_out;
+	wm8994->fll[id].src = src;
 
 	/* Enable any gated AIF clocks */
 	snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 79d5915ae4b3..7072dc539354 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -23,6 +23,11 @@ extern struct snd_soc_dai wm8994_dai[];
 #define WM8994_FLL1 1
 #define WM8994_FLL2 2
 
+#define WM8994_FLL_SRC_MCLK1  1
+#define WM8994_FLL_SRC_MCLK2  2
+#define WM8994_FLL_SRC_LRCLK  3
+#define WM8994_FLL_SRC_BCLK   4
+
 int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 		      int micbias, int det, int shrt);
 
-- 
cgit v1.2.3


From 7d48a6acbcf5325a2551e948bd1557a46dec4c76 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 20 Apr 2010 13:36:11 +0900
Subject: ASoC: Set full range of WM8994 FLL Fratio values

Use all the available Fratio values when configuring the WM8994 FLL, not
just 0 and 3, following more complete characterisation of the device
performance.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 11 ++++++++++-
 1 file changed, 10 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index a27b2ff769d9..8d3ea9b24267 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2784,9 +2784,18 @@ static int wm8994_get_fll_config(struct fll_div *fll,
 
 	if (freq_in > 1000000) {
 		fll->fll_fratio = 0;
-	} else {
+	} else if (freq_in > 256000) {
+		fll->fll_fratio = 1;
+		freq_in *= 2;
+	} else if (freq_in > 128000) {
+		fll->fll_fratio = 2;
+		freq_in *= 4;
+	} else if (freq_in > 64000) {
 		fll->fll_fratio = 3;
 		freq_in *= 8;
+	} else {
+		fll->fll_fratio = 4;
+		freq_in *= 16;
 	}
 	pr_debug("FLL_FRATIO=%d, Fref=%dHz\n", fll->fll_fratio, freq_in);
 
-- 
cgit v1.2.3


From ee839a2127a4f74505c5597966919ca57413f4de Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 20 Apr 2010 13:57:08 +0900
Subject: ASoC: Tone down debugging for WM8994 class W

It's a little verbose during path changes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 14 +++++++-------
 1 file changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 8d3ea9b24267..eefa2427b102 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2189,13 +2189,13 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 	/* Only support direct DAC->headphone paths */
 	reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_1);
 	if (!(reg & WM8994_DAC1L_TO_HPOUT1L)) {
-		dev_dbg(codec->dev, "HPL connected to output mixer\n");
+		dev_vdbg(codec->dev, "HPL connected to output mixer\n");
 		enable = 0;
 	}
 
 	reg = snd_soc_read(codec, WM8994_OUTPUT_MIXER_2);
 	if (!(reg & WM8994_DAC1R_TO_HPOUT1R)) {
-		dev_dbg(codec->dev, "HPR connected to output mixer\n");
+		dev_vdbg(codec->dev, "HPR connected to output mixer\n");
 		enable = 0;
 	}
 
@@ -2203,26 +2203,26 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec)
 	reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING);
 	switch (reg) {
 	case WM8994_AIF2DACL_TO_DAC1L:
-		dev_dbg(codec->dev, "Class W source AIF2DAC\n");
+		dev_vdbg(codec->dev, "Class W source AIF2DAC\n");
 		source = 2 << WM8994_CP_DYN_SRC_SEL_SHIFT;
 		break;
 	case WM8994_AIF1DAC2L_TO_DAC1L:
-		dev_dbg(codec->dev, "Class W source AIF1DAC2\n");
+		dev_vdbg(codec->dev, "Class W source AIF1DAC2\n");
 		source = 1 << WM8994_CP_DYN_SRC_SEL_SHIFT;
 		break;
 	case WM8994_AIF1DAC1L_TO_DAC1L:
-		dev_dbg(codec->dev, "Class W source AIF1DAC1\n");
+		dev_vdbg(codec->dev, "Class W source AIF1DAC1\n");
 		source = 0 << WM8994_CP_DYN_SRC_SEL_SHIFT;
 		break;
 	default:
-		dev_dbg(codec->dev, "DAC mixer setting: %x\n", reg);
+		dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg);
 		enable = 0;
 		break;
 	}
 
 	reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING);
 	if (reg_r != reg) {
-		dev_dbg(codec->dev, "Left and right DAC mixers different\n");
+		dev_vdbg(codec->dev, "Left and right DAC mixers different\n");
 		enable = 0;
 	}
 
-- 
cgit v1.2.3


From 7add84aa779b4f7ca39484a82a11e01d7acd4dca Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 22 Apr 2010 02:29:01 +0900
Subject: ASoC: Allow unspecified source when stopping WM8994 FLLs

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index eefa2427b102..cdfbfb633f30 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2853,6 +2853,11 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
 	}
 
 	switch (src) {
+	case 0:
+		/* Allow no source specification when stopping */
+		if (freq_out)
+			return -EINVAL;
+		break;
 	case WM8994_FLL_SRC_MCLK1:
 	case WM8994_FLL_SRC_MCLK2:
 	case WM8994_FLL_SRC_LRCLK:
-- 
cgit v1.2.3


From d6bdc0f7fe561a4b08b280c2b2add2db3747e64b Mon Sep 17 00:00:00 2001
From: Barry Song <21cnbao@gmail.com>
Date: Wed, 21 Apr 2010 17:36:48 +0800
Subject: ASoC: ad193x: fix wrong register setting in ad193x_set_dai_fmt

Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ad193x.c | 48 ++++++++++++++++++++++++-----------------------
 1 file changed, 25 insertions(+), 23 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 7ed787e2e519..45dfc90540b7 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -163,9 +163,10 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	int adc_reg, dac_reg;
+	int adc_reg1, adc_reg2, dac_reg;
 
-	adc_reg = snd_soc_read(codec, AD193X_ADC_CTRL2);
+	adc_reg1 = snd_soc_read(codec, AD193X_ADC_CTRL1);
+	adc_reg2 = snd_soc_read(codec, AD193X_ADC_CTRL2);
 	dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1);
 
 	/* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S
@@ -173,12 +174,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	 */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		adc_reg &= ~AD193X_ADC_SERFMT_MASK;
-		adc_reg |= AD193X_ADC_SERFMT_TDM;
+		adc_reg1 &= ~AD193X_ADC_SERFMT_MASK;
+		adc_reg1 |= AD193X_ADC_SERFMT_TDM;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
-		adc_reg &= ~AD193X_ADC_SERFMT_MASK;
-		adc_reg |= AD193X_ADC_SERFMT_AUX;
+		adc_reg1 &= ~AD193X_ADC_SERFMT_MASK;
+		adc_reg1 |= AD193X_ADC_SERFMT_AUX;
 		break;
 	default:
 		return -EINVAL;
@@ -186,27 +187,27 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
-		adc_reg &= ~AD193X_ADC_LEFT_HIGH;
-		adc_reg &= ~AD193X_ADC_BCLK_INV;
+		adc_reg2 &= ~AD193X_ADC_LEFT_HIGH;
+		adc_reg2 &= ~AD193X_ADC_BCLK_INV;
 		dac_reg &= ~AD193X_DAC_LEFT_HIGH;
 		dac_reg &= ~AD193X_DAC_BCLK_INV;
 		break;
 	case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */
-		adc_reg |= AD193X_ADC_LEFT_HIGH;
-		adc_reg &= ~AD193X_ADC_BCLK_INV;
+		adc_reg2 |= AD193X_ADC_LEFT_HIGH;
+		adc_reg2 &= ~AD193X_ADC_BCLK_INV;
 		dac_reg |= AD193X_DAC_LEFT_HIGH;
 		dac_reg &= ~AD193X_DAC_BCLK_INV;
 		break;
 	case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */
-		adc_reg &= ~AD193X_ADC_LEFT_HIGH;
-		adc_reg |= AD193X_ADC_BCLK_INV;
+		adc_reg2 &= ~AD193X_ADC_LEFT_HIGH;
+		adc_reg2 |= AD193X_ADC_BCLK_INV;
 		dac_reg &= ~AD193X_DAC_LEFT_HIGH;
 		dac_reg |= AD193X_DAC_BCLK_INV;
 		break;
 
 	case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */
-		adc_reg |= AD193X_ADC_LEFT_HIGH;
-		adc_reg |= AD193X_ADC_BCLK_INV;
+		adc_reg2 |= AD193X_ADC_LEFT_HIGH;
+		adc_reg2 |= AD193X_ADC_BCLK_INV;
 		dac_reg |= AD193X_DAC_LEFT_HIGH;
 		dac_reg |= AD193X_DAC_BCLK_INV;
 		break;
@@ -216,26 +217,26 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */
-		adc_reg |= AD193X_ADC_LCR_MASTER;
-		adc_reg |= AD193X_ADC_BCLK_MASTER;
+		adc_reg2 |= AD193X_ADC_LCR_MASTER;
+		adc_reg2 |= AD193X_ADC_BCLK_MASTER;
 		dac_reg |= AD193X_DAC_LCR_MASTER;
 		dac_reg |= AD193X_DAC_BCLK_MASTER;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */
-		adc_reg |= AD193X_ADC_LCR_MASTER;
-		adc_reg &= ~AD193X_ADC_BCLK_MASTER;
+		adc_reg2 |= AD193X_ADC_LCR_MASTER;
+		adc_reg2 &= ~AD193X_ADC_BCLK_MASTER;
 		dac_reg |= AD193X_DAC_LCR_MASTER;
 		dac_reg &= ~AD193X_DAC_BCLK_MASTER;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
-		adc_reg &= ~AD193X_ADC_LCR_MASTER;
-		adc_reg |= AD193X_ADC_BCLK_MASTER;
+		adc_reg2 &= ~AD193X_ADC_LCR_MASTER;
+		adc_reg2 |= AD193X_ADC_BCLK_MASTER;
 		dac_reg &= ~AD193X_DAC_LCR_MASTER;
 		dac_reg |= AD193X_DAC_BCLK_MASTER;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */
-		adc_reg &= ~AD193X_ADC_LCR_MASTER;
-		adc_reg &= ~AD193X_ADC_BCLK_MASTER;
+		adc_reg2 &= ~AD193X_ADC_LCR_MASTER;
+		adc_reg2 &= ~AD193X_ADC_BCLK_MASTER;
 		dac_reg &= ~AD193X_DAC_LCR_MASTER;
 		dac_reg &= ~AD193X_DAC_BCLK_MASTER;
 		break;
@@ -243,7 +244,8 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg);
+	snd_soc_write(codec, AD193X_ADC_CTRL1, adc_reg1);
+	snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg2);
 	snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg);
 
 	return 0;
-- 
cgit v1.2.3


From ba0a24e7387bb56425da07ed858ada37896ed36a Mon Sep 17 00:00:00 2001
From: Barry Song <21cnbao@gmail.com>
Date: Wed, 21 Apr 2010 17:36:49 +0800
Subject: ASoC: ad193x: fix typo, delete redundant space

Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ad193x.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 45dfc90540b7..c8ca1142b2f4 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -46,13 +46,13 @@ static const struct soc_enum ad193x_deemp_enum =
 
 static const struct snd_kcontrol_new ad193x_snd_controls[] = {
 	/* DAC volume control */
-	SOC_DOUBLE_R("DAC1  Volume", AD193X_DAC_L1_VOL,
+	SOC_DOUBLE_R("DAC1 Volume", AD193X_DAC_L1_VOL,
 			AD193X_DAC_R1_VOL, 0, 0xFF, 1),
-	SOC_DOUBLE_R("DAC2  Volume", AD193X_DAC_L2_VOL,
+	SOC_DOUBLE_R("DAC2 Volume", AD193X_DAC_L2_VOL,
 			AD193X_DAC_R2_VOL, 0, 0xFF, 1),
-	SOC_DOUBLE_R("DAC3  Volume", AD193X_DAC_L3_VOL,
+	SOC_DOUBLE_R("DAC3 Volume", AD193X_DAC_L3_VOL,
 			AD193X_DAC_R3_VOL, 0, 0xFF, 1),
-	SOC_DOUBLE_R("DAC4  Volume", AD193X_DAC_L4_VOL,
+	SOC_DOUBLE_R("DAC4 Volume", AD193X_DAC_L4_VOL,
 			AD193X_DAC_R4_VOL, 0, 0xFF, 1),
 
 	/* ADC switch control */
-- 
cgit v1.2.3


From 3a278a0c65c924950c4744ba451a110998998044 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 29 Mar 2010 20:31:14 +0100
Subject: ASoC: Allow reporting of NULL jacks

Follow the core jack implementation and allow reporting on the status
of NULL jacks, avoiding the need to check in detection implementations.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/soc-jack.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index f8fd22cc70bc..29159e1781d0 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -64,10 +64,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	int enable;
 	int oldstatus;
 
-	if (!jack) {
-		WARN_ON_ONCE(!jack);
+	if (!jack)
 		return;
-	}
+
 	codec = jack->card->codec;
 
 	mutex_lock(&codec->mutex);
-- 
cgit v1.2.3


From f4d593280652025f797d432e90f49d49b9334689 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 23 Apr 2010 10:09:57 +0300
Subject: ASoC: tlv320dac33: Fix for early interrupt in FIFO Mode1

Alarm threshold interrupt is triggered right after the
playback start.
This interrupt is recieved during the first burst period,
and caused the state machine to write additional nSample
command, which has to be avoided.
To fix this issue move the DAC33 interrupt unmasking
after we configured the PREFILL register with a small
delay.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 824bb354ebc9..520377bdb61c 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -557,9 +557,13 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
 		dac33_write16(codec, DAC33_NSAMPLE_MSB,
-				DAC33_THRREG(dac33->nsample));
+			DAC33_THRREG(dac33->nsample + dac33->alarm_threshold));
 		dac33_write16(codec, DAC33_PREFILL_MSB,
 				DAC33_THRREG(dac33->alarm_threshold));
+		/* Enable Alarm Threshold IRQ with a delay */
+		udelay(SAMPLES_TO_US(dac33->burst_rate,
+				     dac33->alarm_threshold));
+		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT);
 		break;
 	case DAC33_FIFO_MODE7:
 		dac33_write16(codec, DAC33_PREFILL_MSB,
@@ -782,7 +786,6 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 	case DAC33_FIFO_MODE1:
 		dac33_write(codec, DAC33_FIFO_IRQ_MODE_B,
 			    DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL));
-		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT);
 		break;
 	case DAC33_FIFO_MODE7:
 		/* Disable all interrupts */
-- 
cgit v1.2.3


From 55abb59c9a03d3aceda61e97b5ff19d9274fc611 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 23 Apr 2010 10:09:58 +0300
Subject: ASoC: tlv320dac33: Skip calculations in FIFO Bypass mode

There is no need for calculations for FIFO bypass mode.
Just in case set the nsample maximum limit, which
has been done in the calculation phase.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 520377bdb61c..b101cbd95740 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -889,6 +889,10 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 	unsigned int nsample_limit;
 
+	/* In bypass mode we don't need to calculate */
+	if (!dac33->fifo_mode)
+		return;
+
 	/* Number of samples (16bit, stereo) in one period */
 	dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4;
 
@@ -1244,6 +1248,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 	dac33->keep_bclk = pdata->keep_bclk;
 	dac33->irq = client->irq;
 	dac33->nsample = NSAMPLE_MAX;
+	dac33->nsample_max = NSAMPLE_MAX;
 	/* Disable FIFO use by default */
 	dac33->fifo_mode = DAC33_FIFO_BYPASS;
 
-- 
cgit v1.2.3


From 4260393e71f1e99c2f4fcde10fcb98fe00188d21 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 23 Apr 2010 10:09:59 +0300
Subject: ASoC: tlv320dac33: Change magic numbers used in Mode7

Upper and Lower threshold values are used as magic
numbers. Replace them with defines for later use.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index b101cbd95740..fa1994262c98 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -50,6 +50,9 @@
 
 #define LATENCY_TIME_MS		20
 
+#define MODE7_LTHR		10
+#define MODE7_UTHR		(DAC33_BUFFER_SIZE_SAMPLES - 10)
+
 static struct snd_soc_codec *tlv320dac33_codec;
 
 enum dac33_state {
@@ -567,7 +570,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 		break;
 	case DAC33_FIFO_MODE7:
 		dac33_write16(codec, DAC33_PREFILL_MSB,
-				DAC33_THRREG(10));
+				DAC33_THRREG(MODE7_LTHR));
 		break;
 	default:
 		dev_warn(codec->dev, "Unhandled FIFO mode: %d\n",
@@ -867,10 +870,8 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		 * Configure the threshold levels, and leave 10 sample space
 		 * at the bottom, and also at the top of the FIFO
 		 */
-		dac33_write16(codec, DAC33_UTHR_MSB,
-			DAC33_THRREG(DAC33_BUFFER_SIZE_SAMPLES - 10));
-		dac33_write16(codec, DAC33_LTHR_MSB,
-			DAC33_THRREG(10));
+		dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR));
+		dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
 		break;
 	default:
 		break;
-- 
cgit v1.2.3


From 76f471274dc9acacd521f151ac9171fd7bbc34c3 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 23 Apr 2010 10:10:00 +0300
Subject: ASoC: tlv320dac33: Calculate the interface speed during bursts

When the DAC33 FIFO is in use the dai interface is running in
much higher speed than the sampling frequency.
Calculate the rate based on the internal base frequency and
the bclk divider.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index fa1994262c98..a8eb19b9b6c5 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -53,6 +53,8 @@
 #define MODE7_LTHR		10
 #define MODE7_UTHR		(DAC33_BUFFER_SIZE_SAMPLES - 10)
 
+#define BURST_BASEFREQ_HZ	49152000
+
 static struct snd_soc_codec *tlv320dac33_codec;
 
 enum dac33_state {
@@ -95,6 +97,7 @@ struct tlv320dac33_priv {
 	enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
 	unsigned int nsample;		/* burst read amount from host */
 	u8 burst_bclkdiv;		/* BCLK divider value in burst mode */
+	unsigned int burst_rate;	/* Interface speed in Burst modes */
 
 	int keep_bclk;			/* Keep the BCLK continuously running
 					 * in FIFO modes */
@@ -1246,6 +1249,8 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 
 	dac33->power_gpio = pdata->power_gpio;
 	dac33->burst_bclkdiv = pdata->burst_bclkdiv;
+	/* Pre calculate the burst rate */
+	dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32;
 	dac33->keep_bclk = pdata->keep_bclk;
 	dac33->irq = client->irq;
 	dac33->nsample = NSAMPLE_MAX;
-- 
cgit v1.2.3


From f57d2cfaad0d6858d700b5671e01cf3aba6ef779 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 23 Apr 2010 10:10:01 +0300
Subject: ASoC: tlv320dac33: FIFO caused delay reporting

Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.

Bypass mode: FIFO is bypassed, report 0 as delay

Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.

Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.

Phase1: when we received the alarm threshold, but our workqueue has
        not been executed (safeguard phase). Just count the played out
        samples since ts1 and subtract it from the alarm threshold
        value.
Phase2: During nSample burst (after writing to nSample register), count
        the played out samples since ts1, count the samples received
        since ts2 (in a burst). Estimate the FIFO depth using these and
        alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
        samples since ts1. Estimate the FIFO depth using the nSample
        configuration and the alarm threshold value.

Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received

Interrupts are coming when DAC33 FIFO depth reaches upper threshold.

Phase1: Draining phase (after the burst), counting the played out
        samples since ts1, and subtract it from the upper threshold
        value.
Phase2: During burst operation. Using the pre calculated time needed to
        play out samples from the buffer during the drain period (from
        upper to lower threshold), move the time window to cover the
        estimated time from the burst start to the current time.
        Calculate the samples played out since lower threshold and also
        the samples received during the same time.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 222 ++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 217 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index a8eb19b9b6c5..3eddaec728c1 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -55,6 +55,13 @@
 
 #define BURST_BASEFREQ_HZ	49152000
 
+#define SAMPLES_TO_US(rate, samples) \
+	(1000000000 / ((rate * 1000) / samples))
+
+#define US_TO_SAMPLES(rate, us) \
+	(rate / (1000000 / us))
+
+
 static struct snd_soc_codec *tlv320dac33_codec;
 
 enum dac33_state {
@@ -101,6 +108,14 @@ struct tlv320dac33_priv {
 
 	int keep_bclk;			/* Keep the BCLK continuously running
 					 * in FIFO modes */
+	spinlock_t lock;
+	unsigned long long t_stamp1;	/* Time stamp for FIFO modes to */
+	unsigned long long t_stamp2;	/* calculate the FIFO caused delay */
+
+	unsigned int mode1_us_burst;	/* Time to burst read n number of
+					 * samples */
+	unsigned int mode7_us_to_lthr;	/* Time to reach lthr from uthr */
+
 	enum dac33_state state;
 };
 
@@ -390,10 +405,14 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	if (ucontrol->value.integer.value[0] < dac33->nsample_min ||
-	    ucontrol->value.integer.value[0] > dac33->nsample_max)
+	    ucontrol->value.integer.value[0] > dac33->nsample_max) {
 		ret = -EINVAL;
-	else
+	} else {
 		dac33->nsample = ucontrol->value.integer.value[0];
+		/* Re calculate the burst time */
+		dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
+						      dac33->nsample);
+	}
 
 	return ret;
 }
@@ -564,6 +583,13 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 	case DAC33_FIFO_MODE1:
 		dac33_write16(codec, DAC33_NSAMPLE_MSB,
 			DAC33_THRREG(dac33->nsample + dac33->alarm_threshold));
+
+		/* Take the timestamps */
+		spin_lock_irq(&dac33->lock);
+		dac33->t_stamp2 = ktime_to_us(ktime_get());
+		dac33->t_stamp1 = dac33->t_stamp2;
+		spin_unlock_irq(&dac33->lock);
+
 		dac33_write16(codec, DAC33_PREFILL_MSB,
 				DAC33_THRREG(dac33->alarm_threshold));
 		/* Enable Alarm Threshold IRQ with a delay */
@@ -572,8 +598,18 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT);
 		break;
 	case DAC33_FIFO_MODE7:
+		/* Take the timestamp */
+		spin_lock_irq(&dac33->lock);
+		dac33->t_stamp1 = ktime_to_us(ktime_get());
+		/* Move back the timestamp with drain time */
+		dac33->t_stamp1 -= dac33->mode7_us_to_lthr;
+		spin_unlock_irq(&dac33->lock);
+
 		dac33_write16(codec, DAC33_PREFILL_MSB,
 				DAC33_THRREG(MODE7_LTHR));
+
+		/* Enable Upper Threshold IRQ */
+		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MUT);
 		break;
 	default:
 		dev_warn(codec->dev, "Unhandled FIFO mode: %d\n",
@@ -590,6 +626,11 @@ static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33)
 
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
+		/* Take the timestamp */
+		spin_lock_irq(&dac33->lock);
+		dac33->t_stamp2 = ktime_to_us(ktime_get());
+		spin_unlock_irq(&dac33->lock);
+
 		dac33_write16(codec, DAC33_NSAMPLE_MSB,
 				DAC33_THRREG(dac33->nsample));
 		break;
@@ -642,7 +683,13 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev)
 	struct snd_soc_codec *codec = dev;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
-	queue_work(dac33->dac33_wq, &dac33->work);
+	spin_lock(&dac33->lock);
+	dac33->t_stamp1 = ktime_to_us(ktime_get());
+	spin_unlock(&dac33->lock);
+
+	/* Do not schedule the workqueue in Mode7 */
+	if (dac33->fifo_mode != DAC33_FIFO_MODE7)
+		queue_work(dac33->dac33_wq, &dac33->work);
 
 	return IRQ_HANDLED;
 }
@@ -794,8 +841,8 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 			    DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL));
 		break;
 	case DAC33_FIFO_MODE7:
-		/* Disable all interrupts */
-		dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0);
+		dac33_write(codec, DAC33_FIFO_IRQ_MODE_A,
+			DAC33_UTM(DAC33_FIFO_IRQ_MODE_LEVEL));
 		break;
 	default:
 		/* in FIFO bypass mode, the interrupts are not used */
@@ -930,6 +977,24 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 
 	if (dac33->nsample > dac33->nsample_max)
 		dac33->nsample = dac33->nsample_max;
+
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_MODE1:
+		dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
+						      dac33->nsample);
+		dac33->t_stamp1 = 0;
+		dac33->t_stamp2 = 0;
+		break;
+	case DAC33_FIFO_MODE7:
+		dac33->mode7_us_to_lthr =
+					SAMPLES_TO_US(substream->runtime->rate,
+						MODE7_UTHR - MODE7_LTHR + 1);
+		dac33->t_stamp1 = 0;
+		break;
+	default:
+		break;
+	}
+
 }
 
 static int dac33_pcm_prepare(struct snd_pcm_substream *substream,
@@ -974,6 +1039,151 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static snd_pcm_sframes_t dac33_dai_delay(
+			struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+	unsigned long long t0, t1, t_now;
+	unsigned int time_delta;
+	int samples_out, samples_in, samples;
+	snd_pcm_sframes_t delay = 0;
+
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_BYPASS:
+		break;
+	case DAC33_FIFO_MODE1:
+		spin_lock(&dac33->lock);
+		t0 = dac33->t_stamp1;
+		t1 = dac33->t_stamp2;
+		spin_unlock(&dac33->lock);
+		t_now = ktime_to_us(ktime_get());
+
+		/* We have not started to fill the FIFO yet, delay is 0 */
+		if (!t1)
+			goto out;
+
+		if (t0 > t1) {
+			/*
+			 * Phase 1:
+			 * After Alarm threshold, and before nSample write
+			 */
+			time_delta = t_now - t0;
+			samples_out = time_delta ? US_TO_SAMPLES(
+						substream->runtime->rate,
+						time_delta) : 0;
+
+			if (likely(dac33->alarm_threshold > samples_out))
+				delay = dac33->alarm_threshold - samples_out;
+			else
+				delay = 0;
+		} else if ((t_now - t1) <= dac33->mode1_us_burst) {
+			/*
+			 * Phase 2:
+			 * After nSample write (during burst operation)
+			 */
+			time_delta = t_now - t0;
+			samples_out = time_delta ? US_TO_SAMPLES(
+						substream->runtime->rate,
+						time_delta) : 0;
+
+			time_delta = t_now - t1;
+			samples_in = time_delta ? US_TO_SAMPLES(
+						dac33->burst_rate,
+						time_delta) : 0;
+
+			samples = dac33->alarm_threshold;
+			samples += (samples_in - samples_out);
+
+			if (likely(samples > 0))
+				delay = samples;
+			else
+				delay = 0;
+		} else {
+			/*
+			 * Phase 3:
+			 * After burst operation, before next alarm threshold
+			 */
+			time_delta = t_now - t0;
+			samples_out = time_delta ? US_TO_SAMPLES(
+						substream->runtime->rate,
+						time_delta) : 0;
+
+			samples_in = dac33->nsample;
+			samples = dac33->alarm_threshold;
+			samples += (samples_in - samples_out);
+
+			if (likely(samples > 0))
+				delay = samples > DAC33_BUFFER_SIZE_SAMPLES ?
+					DAC33_BUFFER_SIZE_SAMPLES : samples;
+			else
+				delay = 0;
+		}
+		break;
+	case DAC33_FIFO_MODE7:
+		spin_lock(&dac33->lock);
+		t0 = dac33->t_stamp1;
+		spin_unlock(&dac33->lock);
+		t_now = ktime_to_us(ktime_get());
+
+		/* We have not started to fill the FIFO yet, delay is 0 */
+		if (!t0)
+			goto out;
+
+		if (t_now <= t0) {
+			/*
+			 * Either the timestamps are messed or equal. Report
+			 * maximum delay
+			 */
+			delay = MODE7_UTHR;
+			goto out;
+		}
+
+		time_delta = t_now - t0;
+		if (time_delta <= dac33->mode7_us_to_lthr) {
+			/*
+			* Phase 1:
+			* After burst (draining phase)
+			*/
+			samples_out = US_TO_SAMPLES(
+					substream->runtime->rate,
+					time_delta);
+
+			if (likely(MODE7_UTHR > samples_out))
+				delay = MODE7_UTHR - samples_out;
+			else
+				delay = 0;
+		} else {
+			/*
+			* Phase 2:
+			* During burst operation
+			*/
+			time_delta = time_delta - dac33->mode7_us_to_lthr;
+
+			samples_out = US_TO_SAMPLES(
+					substream->runtime->rate,
+					time_delta);
+			samples_in = US_TO_SAMPLES(
+					dac33->burst_rate,
+					time_delta);
+			delay = MODE7_LTHR + samples_in - samples_out;
+
+			if (unlikely(delay > MODE7_UTHR))
+				delay = MODE7_UTHR;
+		}
+		break;
+	default:
+		dev_warn(codec->dev, "Unhandled FIFO mode: %d\n",
+							dac33->fifo_mode);
+		break;
+	}
+out:
+	return delay;
+}
+
 static int dac33_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		int clk_id, unsigned int freq, int dir)
 {
@@ -1185,6 +1395,7 @@ static struct snd_soc_dai_ops dac33_dai_ops = {
 	.hw_params	= dac33_hw_params,
 	.prepare	= dac33_pcm_prepare,
 	.trigger	= dac33_pcm_trigger,
+	.delay		= dac33_dai_delay,
 	.set_sysclk	= dac33_set_dai_sysclk,
 	.set_fmt	= dac33_set_dai_fmt,
 };
@@ -1225,6 +1436,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 
 	mutex_init(&codec->mutex);
 	mutex_init(&dac33->mutex);
+	spin_lock_init(&dac33->lock);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-- 
cgit v1.2.3


From 759512fbacfbb6fd9b7f77f9806fb4995775848f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 23 Apr 2010 17:39:23 +0100
Subject: ASoC: Correct inversion of speaker mixer PCM switch

Reported-by: Anti Sullin <anti.sullin@artecdesign.ee>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8974.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 1e4da32c4078..270eb956294f 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -180,7 +180,7 @@ SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0),
 static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0),
 SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0),
-SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 0),
 };
 
 /* Mono Output Mixer */
-- 
cgit v1.2.3


From 5e5e2bef287b96ab29d2d93e89ecf0888323d7bf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 25 Apr 2010 12:20:30 +0100
Subject: ASoC: Warn on low WM8994 AIFCLK

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8994.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index cdfbfb633f30..f04a116e678b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1769,6 +1769,11 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 		dev_dbg(codec->dev, "Dividing AIF%d clock to %dHz\n",
 			aif + 1, rate);
 	}
+
+	if (rate && rate < 3000000)
+		dev_warn(codec->dev, "AIF%dCLK is %dHz, should be >=3MHz for optimal performance\n",
+			 aif + 1, rate);
+
 	wm8994->aifclk[aif] = rate;
 
 	snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset,
-- 
cgit v1.2.3


From b28528a124d0235d84e6c9b3edeedd70f8767740 Mon Sep 17 00:00:00 2001
From: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Date: Mon, 26 Apr 2010 14:56:57 +0400
Subject: ASoC: UDA134X: Add UDA1345 CODEC support

This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/uda134x.h    |  1 +
 sound/soc/codecs/uda134x.c | 13 +++++++++++++
 2 files changed, 14 insertions(+)

(limited to 'sound/soc')

diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
index 475ef8bb7dcd..509efb050176 100644
--- a/include/sound/uda134x.h
+++ b/include/sound/uda134x.h
@@ -21,6 +21,7 @@ struct uda134x_platform_data {
 #define UDA134X_UDA1340 1
 #define UDA134X_UDA1341 2
 #define UDA134X_UDA1344 3
+#define UDA134X_UDA1345 4
 };
 
 #endif /* _UDA134X_H */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 20deaca8e107..2f4d7287fa3c 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -431,6 +431,14 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
 };
 
+static const struct snd_kcontrol_new uda1345_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
 static struct snd_soc_dai_ops uda134x_dai_ops = {
 	.startup	= uda134x_startup,
 	.shutdown	= uda134x_shutdown,
@@ -486,6 +494,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 	case UDA134X_UDA1340:
 	case UDA134X_UDA1341:
 	case UDA134X_UDA1344:
+	case UDA134X_UDA1345:
 		break;
 	default:
 		printk(KERN_ERR "UDA134X SoC codec: "
@@ -551,6 +560,10 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		ret = snd_soc_add_controls(codec, uda1341_snd_controls,
 					ARRAY_SIZE(uda1341_snd_controls));
 	break;
+	case UDA134X_UDA1345:
+		ret = snd_soc_add_controls(codec, uda1345_snd_controls,
+					ARRAY_SIZE(uda1345_snd_controls));
+	break;
 	default:
 		printk(KERN_ERR "%s unknown codec type: %d",
 			__func__, pd->model);
-- 
cgit v1.2.3


From c6de6e03009efaa56baeb98da8bbcce0c6c889a6 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Mon, 26 Apr 2010 15:49:11 +0300
Subject: ASoC: tlv320aic3x: Remove unused version string

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320aic3x.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 556123b4059c..260a2bef72c0 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -49,8 +49,6 @@
 
 #include "tlv320aic3x.h"
 
-#define AIC3X_VERSION "0.2"
-
 /* codec private data */
 struct aic3x_priv {
 	struct snd_soc_codec codec;
-- 
cgit v1.2.3


From d3235c4ac17b7f2a9c64cc3ebc71e23d1d1fa466 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Mon, 26 Apr 2010 15:49:12 +0300
Subject: ASoC: tlv320aic3x: Remove needless power off from
 aic3x_set_bias_level

These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320aic3x.c | 34 ----------------------------------
 1 file changed, 34 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 260a2bef72c0..19602eca4924 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1012,41 +1012,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 		 * all power is driven by DAPM system,
 		 * so output power is safe if bypass was set
 		 */
-		if (aic3x->master) {
-			/* disable pll */
-			reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
-			aic3x_write(codec, AIC3X_PLL_PROGA_REG,
-				    reg & ~PLL_ENABLE);
-		}
-		break;
 	case SND_SOC_BIAS_OFF:
-		/* force all power off */
-		reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL);
-		aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON);
-		reg = aic3x_read_reg_cache(codec, LINE1R_2_RADC_CTRL);
-		aic3x_write(codec, LINE1R_2_RADC_CTRL, reg & ~RADC_PWR_ON);
-
-		reg = aic3x_read_reg_cache(codec, DAC_PWR);
-		aic3x_write(codec, DAC_PWR, reg & ~(LDAC_PWR_ON | RDAC_PWR_ON));
-
-		reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL);
-		aic3x_write(codec, HPLOUT_CTRL, reg & ~HPLOUT_PWR_ON);
-		reg = aic3x_read_reg_cache(codec, HPROUT_CTRL);
-		aic3x_write(codec, HPROUT_CTRL, reg & ~HPROUT_PWR_ON);
-
-		reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL);
-		aic3x_write(codec, HPLCOM_CTRL, reg & ~HPLCOM_PWR_ON);
-		reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL);
-		aic3x_write(codec, HPRCOM_CTRL, reg & ~HPRCOM_PWR_ON);
-
-		reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL);
-		aic3x_write(codec, MONOLOPM_CTRL, reg & ~MONOLOPM_PWR_ON);
-
-		reg = aic3x_read_reg_cache(codec, LLOPM_CTRL);
-		aic3x_write(codec, LLOPM_CTRL, reg & ~LLOPM_PWR_ON);
-		reg = aic3x_read_reg_cache(codec, RLOPM_CTRL);
-		aic3x_write(codec, RLOPM_CTRL, reg & ~RLOPM_PWR_ON);
-
 		if (aic3x->master) {
 			/* disable pll */
 			reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
-- 
cgit v1.2.3


From db13802e51f29c40b404038485121a3973c0947d Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Mon, 26 Apr 2010 15:49:13 +0300
Subject: ASoC: tlv320aic3x: Change bias management semantics

Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320aic3x.c | 10 +++-------
 1 file changed, 3 insertions(+), 7 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 19602eca4924..6b74ad808a09 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -997,7 +997,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		/* all power is driven by DAPM system */
+		break;
+	case SND_SOC_BIAS_PREPARE:
 		if (aic3x->master) {
 			/* enable pll */
 			reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
@@ -1005,13 +1006,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
 				    reg | PLL_ENABLE);
 		}
 		break;
-	case SND_SOC_BIAS_PREPARE:
-		break;
 	case SND_SOC_BIAS_STANDBY:
-		/*
-		 * all power is driven by DAPM system,
-		 * so output power is safe if bypass was set
-		 */
+		/* fall through and disable pll */
 	case SND_SOC_BIAS_OFF:
 		if (aic3x->master) {
 			/* disable pll */
-- 
cgit v1.2.3


From 07779fdd1a236145b5f5dc6916c6b84d9712b305 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Mon, 26 Apr 2010 15:49:14 +0300
Subject: ASoC: tlv320aic3x: Add basic regulator support

This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320aic3x.c | 37 +++++++++++++++++++++++++++++++++++++
 1 file changed, 37 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 6b74ad808a09..584bc1e67f76 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -38,6 +38,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/regulator/consumer.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -49,9 +50,18 @@
 
 #include "tlv320aic3x.h"
 
+#define AIC3X_NUM_SUPPLIES	4
+static const char *aic3x_supply_names[AIC3X_NUM_SUPPLIES] = {
+	"IOVDD",	/* I/O Voltage */
+	"DVDD",		/* Digital Core Voltage */
+	"AVDD",		/* Analog DAC Voltage */
+	"DRVDD",	/* ADC Analog and Output Driver Voltage */
+};
+
 /* codec private data */
 struct aic3x_priv {
 	struct snd_soc_codec codec;
+	struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES];
 	unsigned int sysclk;
 	int master;
 };
@@ -1268,6 +1278,9 @@ static int aic3x_unregister(struct aic3x_priv *aic3x)
 	snd_soc_unregister_dai(&aic3x_dai);
 	snd_soc_unregister_codec(&aic3x->codec);
 
+	regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
+	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
+
 	kfree(aic3x);
 	aic3x_codec = NULL;
 
@@ -1289,6 +1302,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 {
 	struct snd_soc_codec *codec;
 	struct aic3x_priv *aic3x;
+	int ret, i;
 
 	aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
 	if (aic3x == NULL) {
@@ -1304,7 +1318,30 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, aic3x);
 
+	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
+		aic3x->supplies[i].supply = aic3x_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies),
+				 aic3x->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err_get;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies),
+				    aic3x->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_enable;
+	}
+
 	return aic3x_register(codec);
+
+err_enable:
+	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
+err_get:
+	kfree(aic3x);
+	return ret;
 }
 
 static int aic3x_i2c_remove(struct i2c_client *client)
-- 
cgit v1.2.3


From dde3a7e9cb187e25deeac0269733116d4840f91e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 27 Apr 2010 20:01:56 +0100
Subject: ASoC: Remove redundant WM8960 SYSCLKSEL clkdiv option

The SYSCLK source is automatically managed when configuring the PLL.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8960.c | 4 ----
 sound/soc/codecs/wm8960.h | 1 -
 2 files changed, 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index eba3ac013119..50b2376680c1 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -738,10 +738,6 @@ static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 	u16 reg;
 
 	switch (div_id) {
-	case WM8960_SYSCLKSEL:
-		reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1fe;
-		snd_soc_write(codec, WM8960_CLOCK1, reg | div);
-		break;
 	case WM8960_SYSCLKDIV:
 		reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1f9;
 		snd_soc_write(codec, WM8960_CLOCK1, reg | div);
diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h
index d67bfe1300da..a5ef65481b86 100644
--- a/sound/soc/codecs/wm8960.h
+++ b/sound/soc/codecs/wm8960.h
@@ -76,7 +76,6 @@
 #define WM8960_OPCLKDIV			2
 #define WM8960_DCLKDIV			3
 #define WM8960_TOCLKSEL			4
-#define WM8960_SYSCLKSEL		5
 
 #define WM8960_SYSCLK_DIV_1		(0 << 1)
 #define WM8960_SYSCLK_DIV_2		(2 << 1)
-- 
cgit v1.2.3


From 1beb91f004e0efe83b933ca6c84a8b9935f4cf53 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@slimlogic.co.uk>
Date: Mon, 22 Mar 2010 19:30:54 +0000
Subject: ASoC: pandora - update DAPM pins

Remove bogus TWL4030 pins.

Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/omap3pandora.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index de10f76baded..87ce842fa2e8 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -188,8 +188,6 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
 	int ret;
 
 	/* All TWL4030 output pins are floating */
-	snd_soc_dapm_nc_pin(codec, "OUTL");
-	snd_soc_dapm_nc_pin(codec, "OUTR");
 	snd_soc_dapm_nc_pin(codec, "EARPIECE");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
-- 
cgit v1.2.3


From 1849235876b046e26a07e33972906bd23fbb8705 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@slimlogic.co.uk>
Date: Mon, 22 Mar 2010 19:35:06 +0000
Subject: ASoC: zoom2 - update DAPM pins

Remove bogus twl4030 pins

Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/zoom2.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
index f90a2ac888cf..50a94ee76ecc 100644
--- a/sound/soc/omap/zoom2.c
+++ b/sound/soc/omap/zoom2.c
@@ -181,9 +181,6 @@ static int zoom2_twl4030_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
 	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
 	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
-
-	snd_soc_dapm_nc_pin(codec, "OUTL");
-	snd_soc_dapm_nc_pin(codec, "OUTR");
 	snd_soc_dapm_nc_pin(codec, "EARPIECE");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
 	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
-- 
cgit v1.2.3


From cf134d5bfb19cdee922b95738ce3cfe86c0e8f7a Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@slimlogic.co.uk>
Date: Fri, 26 Mar 2010 20:05:54 +0000
Subject: ASoC: tlv320dac33 - disable regulators at i2c remove()

Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 3eddaec728c1..54b2a0508a11 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1584,6 +1584,7 @@ static int __devexit dac33_i2c_remove(struct i2c_client *client)
 	if (dac33->irq >= 0)
 		free_irq(dac33->irq, &dac33->codec);
 
+	regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
 	regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies);
 
 	destroy_workqueue(dac33->dac33_wq);
-- 
cgit v1.2.3


From 39b8eab7e7fe429d8d57f18c0ebdb7c25df55f5c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 28 Apr 2010 18:36:10 +0100
Subject: ASoC: Add WM9090 amplifier driver

The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.

Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control.  The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/wm9090.h    |  28 ++
 sound/soc/codecs/Kconfig  |   4 +
 sound/soc/codecs/Makefile |   2 +
 sound/soc/codecs/wm9090.c | 773 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm9090.h | 715 ++++++++++++++++++++++++++++++++++++++++++
 5 files changed, 1522 insertions(+)
 create mode 100644 include/sound/wm9090.h
 create mode 100644 sound/soc/codecs/wm9090.c
 create mode 100644 sound/soc/codecs/wm9090.h

(limited to 'sound/soc')

diff --git a/include/sound/wm9090.h b/include/sound/wm9090.h
new file mode 100644
index 000000000000..3718928cde1a
--- /dev/null
+++ b/include/sound/wm9090.h
@@ -0,0 +1,28 @@
+/*
+ * linux/sound/wm9090.h -- Platform data for WM9090
+ *
+ * Copyright 2009, 2010 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM9090_H
+#define __LINUX_SND_WM9090_H
+
+struct wm9090_platform_data {
+	/* Line inputs 1 & 2 can optionally be differential */
+	unsigned int lin1_diff:1;
+	unsigned int lin2_diff:1;
+
+	/* AGC configuration.  This is intended to protect the speaker
+	 * against overdriving and will therefore depend on the
+	 * hardware setup with incorrect runtime configuration
+	 * potentially causing hardware damage.
+	 */
+	unsigned int agc_ena:1;
+	u16 agc[3];
+};
+
+#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bc0ab47e156b..31ac5538fe7e 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -66,6 +66,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8993 if I2C
 	select SND_SOC_WM8994 if MFD_WM8994
 	select SND_SOC_WM9081 if I2C
+	select SND_SOC_WM9090 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
@@ -277,3 +278,6 @@ config SND_SOC_TPA6130A2
 
 config SND_SOC_WM2000
 	tristate
+
+config SND_SOC_WM9090
+	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 337904167358..91429eab0707 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -61,6 +61,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o
 snd-soc-max9877-objs := max9877.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-wm2000-objs := wm2000.o
+snd-soc-wm9090-objs := wm9090.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
@@ -125,3 +126,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
 obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o
+obj-$(CONFIG_SND_SOC_WM9090)	+= snd-soc-wm9090.o
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
new file mode 100644
index 000000000000..1592250daec0
--- /dev/null
+++ b/sound/soc/codecs/wm9090.c
@@ -0,0 +1,773 @@
+/*
+ * ALSA SoC WM9090 driver
+ *
+ * Copyright 2009, 2010 Wolfson Microelectronics
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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/module.h>
+#include <linux/errno.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/wm9090.h>
+
+#include "wm9090.h"
+
+static struct snd_soc_codec *wm9090_codec;
+
+static const u16 wm9090_reg_defaults[] = {
+	0x9093,     /* R0   - Software Reset */
+	0x0006,     /* R1   - Power Management (1) */
+	0x6000,     /* R2   - Power Management (2) */
+	0x0000,     /* R3   - Power Management (3) */
+	0x0000,     /* R4 */
+	0x0000,     /* R5 */
+	0x01C0,     /* R6   - Clocking 1 */
+	0x0000,     /* R7 */
+	0x0000,     /* R8 */
+	0x0000,     /* R9 */
+	0x0000,     /* R10 */
+	0x0000,     /* R11 */
+	0x0000,     /* R12 */
+	0x0000,     /* R13 */
+	0x0000,     /* R14 */
+	0x0000,     /* R15 */
+	0x0000,     /* R16 */
+	0x0000,     /* R17 */
+	0x0000,     /* R18 */
+	0x0000,     /* R19 */
+	0x0000,     /* R20 */
+	0x0000,     /* R21 */
+	0x0003,     /* R22  - IN1 Line Control */
+	0x0003,     /* R23  - IN2 Line Control */
+	0x0083,     /* R24  - IN1 Line Input A Volume */
+	0x0083,     /* R25  - IN1  Line Input B Volume */
+	0x0083,     /* R26  - IN2 Line Input A Volume */
+	0x0083,     /* R27  - IN2 Line Input B Volume */
+	0x002D,     /* R28  - Left Output Volume */
+	0x002D,     /* R29  - Right Output Volume */
+	0x0000,     /* R30 */
+	0x0000,     /* R31 */
+	0x0000,     /* R32 */
+	0x0000,     /* R33 */
+	0x0100,     /* R34  - SPKMIXL Attenuation */
+	0x0000,     /* R35 */
+	0x0010,     /* R36  - SPKOUT Mixers */
+	0x0140,     /* R37  - ClassD3 */
+	0x0039,     /* R38  - Speaker Volume Left */
+	0x0000,     /* R39 */
+	0x0000,     /* R40 */
+	0x0000,     /* R41 */
+	0x0000,     /* R42 */
+	0x0000,     /* R43 */
+	0x0000,     /* R44 */
+	0x0000,     /* R45  - Output Mixer1 */
+	0x0000,     /* R46  - Output Mixer2 */
+	0x0100,     /* R47  - Output Mixer3 */
+	0x0100,     /* R48  - Output Mixer4 */
+	0x0000,     /* R49 */
+	0x0000,     /* R50 */
+	0x0000,     /* R51 */
+	0x0000,     /* R52 */
+	0x0000,     /* R53 */
+	0x0000,     /* R54  - Speaker Mixer */
+	0x0000,     /* R55 */
+	0x0000,     /* R56 */
+	0x000D,     /* R57  - AntiPOP2 */
+	0x0000,     /* R58 */
+	0x0000,     /* R59 */
+	0x0000,     /* R60 */
+	0x0000,     /* R61 */
+	0x0000,     /* R62 */
+	0x0000,     /* R63 */
+	0x0000,     /* R64 */
+	0x0000,     /* R65 */
+	0x0000,     /* R66 */
+	0x0000,     /* R67 */
+	0x0000,     /* R68 */
+	0x0000,     /* R69 */
+	0x0000,     /* R70  - Write Sequencer 0 */
+	0x0000,     /* R71  - Write Sequencer 1 */
+	0x0000,     /* R72  - Write Sequencer 2 */
+	0x0000,     /* R73  - Write Sequencer 3 */
+	0x0000,     /* R74  - Write Sequencer 4 */
+	0x0000,     /* R75  - Write Sequencer 5 */
+	0x1F25,     /* R76  - Charge Pump 1 */
+	0x0000,     /* R77 */
+	0x0000,     /* R78 */
+	0x0000,     /* R79 */
+	0x0000,     /* R80 */
+	0x0000,     /* R81 */
+	0x0000,     /* R82 */
+	0x0000,     /* R83 */
+	0x0000,     /* R84  - DC Servo 0 */
+	0x054A,     /* R85  - DC Servo 1 */
+	0x0000,     /* R86 */
+	0x0000,     /* R87  - DC Servo 3 */
+	0x0000,     /* R88  - DC Servo Readback 0 */
+	0x0000,     /* R89  - DC Servo Readback 1 */
+	0x0000,     /* R90  - DC Servo Readback 2 */
+	0x0000,     /* R91 */
+	0x0000,     /* R92 */
+	0x0000,     /* R93 */
+	0x0000,     /* R94 */
+	0x0000,     /* R95 */
+	0x0100,     /* R96  - Analogue HP 0 */
+	0x0000,     /* R97 */
+	0x8640,     /* R98  - AGC Control 0 */
+	0xC000,     /* R99  - AGC Control 1 */
+	0x0200,     /* R100 - AGC Control 2 */
+};
+
+/* This struct is used to save the context */
+struct wm9090_priv {
+	/* We're not really registering as a CODEC since ASoC core
+	 * does not yet support multiple CODECs but having the CODEC
+	 * structure means we can reuse some of the ASoC core
+	 * features.
+	 */
+	struct snd_soc_codec codec;
+	struct mutex mutex;
+	u16 reg_cache[WM9090_MAX_REGISTER + 1];
+	struct wm9090_platform_data pdata;
+};
+
+static int wm9090_volatile(unsigned int reg)
+{
+	switch (reg) {
+	case WM9090_SOFTWARE_RESET:
+	case WM9090_DC_SERVO_0:
+	case WM9090_DC_SERVO_READBACK_0:
+	case WM9090_DC_SERVO_READBACK_1:
+	case WM9090_DC_SERVO_READBACK_2:
+		return 1;
+
+	default:
+		return 0;
+	}
+}
+
+static void wait_for_dc_servo(struct snd_soc_codec *codec)
+{
+	unsigned int reg;
+	int count = 0;
+
+	dev_dbg(codec->dev, "Waiting for DC servo...\n");
+	do {
+		count++;
+		msleep(1);
+		reg = snd_soc_read(codec, WM9090_DC_SERVO_READBACK_0);
+		dev_dbg(codec->dev, "DC servo status: %x\n", reg);
+	} while ((reg & WM9090_DCS_CAL_COMPLETE_MASK)
+		 != WM9090_DCS_CAL_COMPLETE_MASK && count < 1000);
+
+	if ((reg & WM9090_DCS_CAL_COMPLETE_MASK)
+	    != WM9090_DCS_CAL_COMPLETE_MASK)
+		dev_err(codec->dev, "Timed out waiting for DC Servo\n");
+}
+
+static const unsigned int in_tlv[] = {
+	TLV_DB_RANGE_HEAD(6),
+	0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
+	1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0),
+	4, 6, TLV_DB_SCALE_ITEM(600, 600, 0),
+};
+static const unsigned int mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0),
+	3, 3, TLV_DB_SCALE_ITEM(0, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+static const unsigned int spkboost_tlv[] = {
+	TLV_DB_RANGE_HEAD(7),
+	0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
+	7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
+};
+
+static const struct snd_kcontrol_new wm9090_controls[] = {
+SOC_SINGLE_TLV("IN1A Volume", WM9090_IN1_LINE_INPUT_A_VOLUME, 0, 6, 0,
+	       in_tlv),
+SOC_SINGLE("IN1A Switch", WM9090_IN1_LINE_INPUT_A_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN1A ZC Switch", WM9090_IN1_LINE_INPUT_A_VOLUME, 6, 1, 0),
+
+SOC_SINGLE_TLV("IN2A Volume", WM9090_IN2_LINE_INPUT_A_VOLUME, 0, 6, 0,
+	       in_tlv),
+SOC_SINGLE("IN2A Switch", WM9090_IN2_LINE_INPUT_A_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN2A ZC Switch", WM9090_IN2_LINE_INPUT_A_VOLUME, 6, 1, 0),
+
+SOC_SINGLE("MIXOUTL Switch", WM9090_OUTPUT_MIXER3, 8, 1, 1),
+SOC_SINGLE_TLV("MIXOUTL IN1A Volume", WM9090_OUTPUT_MIXER3, 6, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("MIXOUTL IN2A Volume", WM9090_OUTPUT_MIXER3, 2, 3, 1,
+	       mix_tlv),
+
+SOC_SINGLE("MIXOUTR Switch", WM9090_OUTPUT_MIXER4, 8, 1, 1),
+SOC_SINGLE_TLV("MIXOUTR IN1A Volume", WM9090_OUTPUT_MIXER4, 6, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("MIXOUTR IN2A Volume", WM9090_OUTPUT_MIXER4, 2, 3, 1,
+	       mix_tlv),
+
+SOC_SINGLE("SPKMIX Switch", WM9090_SPKMIXL_ATTENUATION, 8, 1, 1),
+SOC_SINGLE_TLV("SPKMIX IN1A Volume", WM9090_SPKMIXL_ATTENUATION, 6, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("SPKMIX IN2A Volume", WM9090_SPKMIXL_ATTENUATION, 2, 3, 1,
+	       mix_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Volume", WM9090_LEFT_OUTPUT_VOLUME,
+		 WM9090_RIGHT_OUTPUT_VOLUME, 0, 63, 0, out_tlv),
+SOC_DOUBLE_R("Headphone Switch", WM9090_LEFT_OUTPUT_VOLUME,
+	     WM9090_RIGHT_OUTPUT_VOLUME, 6, 1, 1),
+SOC_DOUBLE_R("Headphone ZC Switch", WM9090_LEFT_OUTPUT_VOLUME,
+	     WM9090_RIGHT_OUTPUT_VOLUME, 7, 1, 0),
+
+SOC_SINGLE_TLV("Speaker Volume", WM9090_SPEAKER_VOLUME_LEFT, 0, 63, 0,
+	       out_tlv),
+SOC_SINGLE("Speaker Switch", WM9090_SPEAKER_VOLUME_LEFT, 6, 1, 1),
+SOC_SINGLE("Speaker ZC Switch", WM9090_SPEAKER_VOLUME_LEFT, 7, 1, 0),
+SOC_SINGLE_TLV("Speaker Boost Volume", WM9090_CLASSD3, 3, 7, 0, spkboost_tlv),
+};
+
+static const struct snd_kcontrol_new wm9090_in1_se_controls[] = {
+SOC_SINGLE_TLV("IN1B Volume", WM9090_IN1_LINE_INPUT_B_VOLUME, 0, 6, 0,
+	       in_tlv),
+SOC_SINGLE("IN1B Switch", WM9090_IN1_LINE_INPUT_B_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN1B ZC Switch", WM9090_IN1_LINE_INPUT_B_VOLUME, 6, 1, 0),
+
+SOC_SINGLE_TLV("SPKMIX IN1B Volume", WM9090_SPKMIXL_ATTENUATION, 4, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("MIXOUTL IN1B Volume", WM9090_OUTPUT_MIXER3, 4, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("MIXOUTR IN1B Volume", WM9090_OUTPUT_MIXER4, 4, 3, 1,
+	       mix_tlv),
+};
+
+static const struct snd_kcontrol_new wm9090_in2_se_controls[] = {
+SOC_SINGLE_TLV("IN2B Volume", WM9090_IN2_LINE_INPUT_B_VOLUME, 0, 6, 0,
+	       in_tlv),
+SOC_SINGLE("IN2B Switch", WM9090_IN2_LINE_INPUT_B_VOLUME, 7, 1, 1),
+SOC_SINGLE("IN2B ZC Switch", WM9090_IN2_LINE_INPUT_B_VOLUME, 6, 1, 0),
+
+SOC_SINGLE_TLV("SPKMIX IN2B Volume", WM9090_SPKMIXL_ATTENUATION, 0, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("MIXOUTL IN2B Volume", WM9090_OUTPUT_MIXER3, 0, 3, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("MIXOUTR IN2B Volume", WM9090_OUTPUT_MIXER4, 0, 3, 1,
+	       mix_tlv),
+};
+
+static int hp_ev(struct snd_soc_dapm_widget *w,
+		 struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int reg = snd_soc_read(codec, WM9090_ANALOGUE_HP_0);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_update_bits(codec, WM9090_CHARGE_PUMP_1,
+				    WM9090_CP_ENA, WM9090_CP_ENA);
+
+		msleep(5);
+
+		snd_soc_update_bits(codec, WM9090_POWER_MANAGEMENT_1,
+				    WM9090_HPOUT1L_ENA | WM9090_HPOUT1R_ENA,
+				    WM9090_HPOUT1L_ENA | WM9090_HPOUT1R_ENA);
+
+		reg |= WM9090_HPOUT1L_DLY | WM9090_HPOUT1R_DLY;
+		snd_soc_write(codec, WM9090_ANALOGUE_HP_0, reg);
+
+		/* Start the DC servo.  We don't currently use the
+		 * ability to save the state since we don't have full
+		 * control of the analogue paths and they can change
+		 * DC offsets; see the WM8904 driver for an example of
+		 * doing so.
+		 */
+		snd_soc_write(codec, WM9090_DC_SERVO_0,
+			      WM9090_DCS_ENA_CHAN_0 |
+			      WM9090_DCS_ENA_CHAN_1 |
+			      WM9090_DCS_TRIG_STARTUP_1 |
+			      WM9090_DCS_TRIG_STARTUP_0);
+		wait_for_dc_servo(codec);
+
+		reg |= WM9090_HPOUT1R_OUTP | WM9090_HPOUT1R_RMV_SHORT |
+			WM9090_HPOUT1L_OUTP | WM9090_HPOUT1L_RMV_SHORT;
+		snd_soc_write(codec, WM9090_ANALOGUE_HP_0, reg);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		reg &= ~(WM9090_HPOUT1L_RMV_SHORT |
+			 WM9090_HPOUT1L_DLY |
+			 WM9090_HPOUT1L_OUTP |
+			 WM9090_HPOUT1R_RMV_SHORT |
+			 WM9090_HPOUT1R_DLY |
+			 WM9090_HPOUT1R_OUTP);
+
+		snd_soc_write(codec, WM9090_ANALOGUE_HP_0, reg);
+
+		snd_soc_write(codec, WM9090_DC_SERVO_0, 0);
+
+		snd_soc_update_bits(codec, WM9090_POWER_MANAGEMENT_1,
+				    WM9090_HPOUT1L_ENA | WM9090_HPOUT1R_ENA,
+				    0);
+
+		snd_soc_update_bits(codec, WM9090_CHARGE_PUMP_1,
+				    WM9090_CP_ENA, 0);
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new spkmix[] = {
+SOC_DAPM_SINGLE("IN1A Switch", WM9090_SPEAKER_MIXER, 6, 1, 0),
+SOC_DAPM_SINGLE("IN1B Switch", WM9090_SPEAKER_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("IN2A Switch", WM9090_SPEAKER_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("IN2B Switch", WM9090_SPEAKER_MIXER, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new spkout[] = {
+SOC_DAPM_SINGLE("Mixer Switch", WM9090_SPKOUT_MIXERS, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new mixoutl[] = {
+SOC_DAPM_SINGLE("IN1A Switch", WM9090_OUTPUT_MIXER1, 6, 1, 0),
+SOC_DAPM_SINGLE("IN1B Switch", WM9090_OUTPUT_MIXER1, 4, 1, 0),
+SOC_DAPM_SINGLE("IN2A Switch", WM9090_OUTPUT_MIXER1, 2, 1, 0),
+SOC_DAPM_SINGLE("IN2B Switch", WM9090_OUTPUT_MIXER1, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new mixoutr[] = {
+SOC_DAPM_SINGLE("IN1A Switch", WM9090_OUTPUT_MIXER2, 6, 1, 0),
+SOC_DAPM_SINGLE("IN1B Switch", WM9090_OUTPUT_MIXER2, 4, 1, 0),
+SOC_DAPM_SINGLE("IN2A Switch", WM9090_OUTPUT_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN2B Switch", WM9090_OUTPUT_MIXER2, 0, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm9090_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN1+"),
+SND_SOC_DAPM_INPUT("IN1-"),
+SND_SOC_DAPM_INPUT("IN2+"),
+SND_SOC_DAPM_INPUT("IN2-"),
+
+SND_SOC_DAPM_SUPPLY("OSC", WM9090_POWER_MANAGEMENT_1, 3, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("IN1A PGA", WM9090_POWER_MANAGEMENT_2, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("IN1B PGA", WM9090_POWER_MANAGEMENT_2, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("IN2A PGA", WM9090_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("IN2B PGA", WM9090_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
+
+SND_SOC_DAPM_MIXER("SPKMIX", WM9090_POWER_MANAGEMENT_3, 3, 0,
+		   spkmix, ARRAY_SIZE(spkmix)),
+SND_SOC_DAPM_MIXER("MIXOUTL", WM9090_POWER_MANAGEMENT_3, 5, 0,
+		   mixoutl, ARRAY_SIZE(mixoutl)),
+SND_SOC_DAPM_MIXER("MIXOUTR", WM9090_POWER_MANAGEMENT_3, 4, 0,
+		   mixoutr, ARRAY_SIZE(mixoutr)),
+
+SND_SOC_DAPM_PGA_E("HP PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+		   hp_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+SND_SOC_DAPM_PGA("SPKPGA", WM9090_POWER_MANAGEMENT_3, 8, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("SPKOUT", WM9090_POWER_MANAGEMENT_1, 12, 0,
+		   spkout, ARRAY_SIZE(spkout)),
+
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("Speaker"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{ "IN1A PGA", NULL, "IN1+" },
+	{ "IN2A PGA", NULL, "IN2+" },
+
+	{ "SPKMIX", "IN1A Switch", "IN1A PGA" },
+	{ "SPKMIX", "IN2A Switch", "IN2A PGA" },
+
+	{ "MIXOUTL", "IN1A Switch", "IN1A PGA" },
+	{ "MIXOUTL", "IN2A Switch", "IN2A PGA" },
+
+	{ "MIXOUTR", "IN1A Switch", "IN1A PGA" },
+	{ "MIXOUTR", "IN2A Switch", "IN2A PGA" },
+
+	{ "HP PGA", NULL, "OSC" },
+	{ "HP PGA", NULL, "MIXOUTL" },
+	{ "HP PGA", NULL, "MIXOUTR" },
+
+	{ "HPL", NULL, "HP PGA" },
+	{ "HPR", NULL, "HP PGA" },
+
+	{ "SPKPGA", NULL, "OSC" },
+	{ "SPKPGA", NULL, "SPKMIX" },
+
+	{ "SPKOUT", "Mixer Switch", "SPKPGA" },
+
+	{ "Speaker", NULL, "SPKOUT" },
+};
+
+static const struct snd_soc_dapm_route audio_map_in1_se[] = {
+	{ "IN1B PGA", NULL, "IN1-" },	
+
+	{ "SPKMIX", "IN1B Switch", "IN1B PGA" },
+	{ "MIXOUTL", "IN1B Switch", "IN1B PGA" },
+	{ "MIXOUTR", "IN1B Switch", "IN1B PGA" },
+};
+
+static const struct snd_soc_dapm_route audio_map_in1_diff[] = {
+	{ "IN1A PGA", NULL, "IN1-" },	
+};
+
+static const struct snd_soc_dapm_route audio_map_in2_se[] = {
+	{ "IN2B PGA", NULL, "IN2-" },	
+
+	{ "SPKMIX", "IN2B Switch", "IN2B PGA" },
+	{ "MIXOUTL", "IN2B Switch", "IN2B PGA" },
+	{ "MIXOUTR", "IN2B Switch", "IN2B PGA" },
+};
+
+static const struct snd_soc_dapm_route audio_map_in2_diff[] = {
+	{ "IN2A PGA", NULL, "IN2-" },	
+};
+
+static int wm9090_add_controls(struct snd_soc_codec *codec)
+{
+	struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	snd_soc_dapm_new_controls(codec, wm9090_dapm_widgets,
+				  ARRAY_SIZE(wm9090_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_add_controls(codec, wm9090_controls,
+			     ARRAY_SIZE(wm9090_controls));
+
+	if (wm9090->pdata.lin1_diff) {
+		snd_soc_dapm_add_routes(codec, audio_map_in1_diff,
+					ARRAY_SIZE(audio_map_in1_diff));
+	} else {
+		snd_soc_dapm_add_routes(codec, audio_map_in1_se,
+					ARRAY_SIZE(audio_map_in1_se));
+		snd_soc_add_controls(codec, wm9090_in1_se_controls,
+				     ARRAY_SIZE(wm9090_in1_se_controls));
+	}
+
+	if (wm9090->pdata.lin2_diff) {
+		snd_soc_dapm_add_routes(codec, audio_map_in2_diff,
+					ARRAY_SIZE(audio_map_in2_diff));
+	} else {
+		snd_soc_dapm_add_routes(codec, audio_map_in2_se,
+					ARRAY_SIZE(audio_map_in2_se));
+		snd_soc_add_controls(codec, wm9090_in2_se_controls,
+				     ARRAY_SIZE(wm9090_in2_se_controls));
+	}
+
+	if (wm9090->pdata.agc_ena) {
+		for (i = 0; i < ARRAY_SIZE(wm9090->pdata.agc); i++)
+			snd_soc_write(codec, WM9090_AGC_CONTROL_0 + i,
+				      wm9090->pdata.agc[i]);
+		snd_soc_update_bits(codec, WM9090_POWER_MANAGEMENT_3,
+				    WM9090_AGC_ENA, WM9090_AGC_ENA);
+	} else {
+		snd_soc_update_bits(codec, WM9090_POWER_MANAGEMENT_3,
+				    WM9090_AGC_ENA, 0);
+	}
+
+	return 0;
+
+}
+
+/*
+ * The machine driver should call this from their set_bias_level; if there
+ * isn't one then this can just be set as the set_bias_level function.
+ */
+static int wm9090_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 *reg_cache = codec->reg_cache;
+	int i, ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_update_bits(codec, WM9090_ANTIPOP2, WM9090_VMID_ENA,
+				    WM9090_VMID_ENA);
+		snd_soc_update_bits(codec, WM9090_POWER_MANAGEMENT_1,
+				    WM9090_BIAS_ENA |
+				    WM9090_VMID_RES_MASK,
+				    WM9090_BIAS_ENA |
+				    1 << WM9090_VMID_RES_SHIFT);
+		msleep(1);  /* Probably an overestimate */
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Restore the register cache */
+			for (i = 1; i < codec->reg_cache_size; i++) {
+				if (reg_cache[i] == wm9090_reg_defaults[i])
+					continue;
+				if (wm9090_volatile(i))
+					continue;
+
+				ret = snd_soc_write(codec, i, reg_cache[i]);
+				if (ret != 0)
+					dev_warn(codec->dev,
+						 "Failed to restore register %d: %d\n",
+						 i, ret);
+			}
+		}
+
+		/* We keep VMID off during standby since the combination of
+		 * ground referenced outputs and class D speaker mean that
+		 * latency is not an issue.
+		 */
+		snd_soc_update_bits(codec, WM9090_POWER_MANAGEMENT_1,
+				    WM9090_BIAS_ENA | WM9090_VMID_RES_MASK, 0);
+		snd_soc_update_bits(codec, WM9090_ANTIPOP2,
+				    WM9090_VMID_ENA, 0);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static int wm9090_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (wm9090_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm9090_codec;
+	codec = wm9090_codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+		goto pcm_err;
+	}
+
+	wm9090_add_controls(codec);
+
+	return 0;
+
+pcm_err:
+	return ret;
+}
+
+#ifdef CONFIG_PM
+static int wm9090_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm9090_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define wm9090_suspend NULL
+#define wm9090_resume NULL
+#endif
+
+static int wm9090_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9090 = {
+	.probe = 	wm9090_probe,
+	.remove = 	wm9090_remove,
+	.suspend = 	wm9090_suspend,
+	.resume =	wm9090_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9090);
+
+static int wm9090_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct wm9090_priv *wm9090;
+	struct snd_soc_codec *codec;
+	int ret;
+
+	wm9090 = kzalloc(sizeof(*wm9090), GFP_KERNEL);
+	if (wm9090 == NULL) {
+		dev_err(&i2c->dev, "Can not allocate memory\n");
+		return -ENOMEM;
+	}
+	codec = &wm9090->codec;
+
+	if (i2c->dev.platform_data)
+		memcpy(&wm9090->pdata, i2c->dev.platform_data,
+		       sizeof(wm9090->pdata));
+
+	wm9090_codec = codec;
+
+	i2c_set_clientdata(i2c, wm9090);
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->control_data = i2c;
+	snd_soc_codec_set_drvdata(codec, wm9090);
+	codec->dev = &i2c->dev;
+	codec->name = "WM9090";
+	codec->owner = THIS_MODULE;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm9090_set_bias_level,
+	codec->reg_cache_size = WM9090_MAX_REGISTER + 1;
+	codec->reg_cache = &wm9090->reg_cache;
+	codec->volatile_register = wm9090_volatile;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		goto err;
+	}
+
+	memcpy(&wm9090->reg_cache, wm9090_reg_defaults,
+	       sizeof(wm9090->reg_cache));
+
+	ret = snd_soc_read(codec, WM9090_SOFTWARE_RESET);
+	if (ret < 0)
+		goto err;
+	if (ret != wm9090_reg_defaults[WM9090_SOFTWARE_RESET]) {
+		dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", ret);
+		ret = -EINVAL;
+		goto err;
+	}
+
+	ret = snd_soc_write(codec, WM9090_SOFTWARE_RESET, 0);
+	if (ret < 0)
+		goto err;
+
+	/* Configure some defaults; they will be written out when we
+	 * bring the bias up.
+	 */
+	wm9090->reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU
+		| WM9090_IN1A_ZC;
+	wm9090->reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU
+		| WM9090_IN1B_ZC;
+	wm9090->reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU
+		| WM9090_IN2A_ZC;
+	wm9090->reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU
+		| WM9090_IN2B_ZC;
+	wm9090->reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |=
+		WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC;
+	wm9090->reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |= 
+		WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC;
+	wm9090->reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |=
+		WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC;
+
+	wm9090->reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA;
+
+	wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+		goto err_bias;
+	}
+
+	return 0;
+
+err_bias:
+	wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF);
+err:
+	kfree(wm9090);
+	i2c_set_clientdata(i2c, NULL);
+	wm9090_codec = NULL;
+
+	return ret;
+}
+
+static int wm9090_i2c_remove(struct i2c_client *i2c)
+{
+	struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c);
+	struct snd_soc_codec *codec = &wm9090->codec;
+
+	snd_soc_unregister_codec(codec);
+	wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	kfree(wm9090);
+	wm9090_codec = NULL;
+
+	return 0;
+}
+
+static const struct i2c_device_id wm9090_id[] = {
+	{ "wm9090", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm9090_id);
+
+static struct i2c_driver wm9090_i2c_driver = {
+	.driver = {
+		.name = "wm9090",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm9090_i2c_probe,
+	.remove = __devexit_p(wm9090_i2c_remove),
+	.id_table = wm9090_id,
+};
+
+static int __init wm9090_init(void)
+{
+	return i2c_add_driver(&wm9090_i2c_driver);
+}
+module_init(wm9090_init);
+
+static void __exit wm9090_exit(void)
+{
+	i2c_del_driver(&wm9090_i2c_driver);
+}
+module_exit(wm9090_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("WM9090 ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9090.h b/sound/soc/codecs/wm9090.h
new file mode 100644
index 000000000000..b08eab932a5b
--- /dev/null
+++ b/sound/soc/codecs/wm9090.h
@@ -0,0 +1,715 @@
+/*
+ * ALSA SoC WM9090 driver
+ *
+ * Copyright 2009 Wolfson Microelectronics
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.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
+ */
+
+#ifndef __WM9090_H
+#define __WM9090_H
+
+extern struct snd_soc_codec_device soc_codec_dev_wm9090;
+
+/*
+ * Register values.
+ */
+#define WM9090_SOFTWARE_RESET                   0x00
+#define WM9090_POWER_MANAGEMENT_1               0x01
+#define WM9090_POWER_MANAGEMENT_2               0x02
+#define WM9090_POWER_MANAGEMENT_3               0x03
+#define WM9090_CLOCKING_1                       0x06
+#define WM9090_IN1_LINE_CONTROL                 0x16
+#define WM9090_IN2_LINE_CONTROL                 0x17
+#define WM9090_IN1_LINE_INPUT_A_VOLUME          0x18
+#define WM9090_IN1_LINE_INPUT_B_VOLUME          0x19
+#define WM9090_IN2_LINE_INPUT_A_VOLUME          0x1A
+#define WM9090_IN2_LINE_INPUT_B_VOLUME          0x1B
+#define WM9090_LEFT_OUTPUT_VOLUME               0x1C
+#define WM9090_RIGHT_OUTPUT_VOLUME              0x1D
+#define WM9090_SPKMIXL_ATTENUATION              0x22
+#define WM9090_SPKOUT_MIXERS                    0x24
+#define WM9090_CLASSD3                          0x25
+#define WM9090_SPEAKER_VOLUME_LEFT              0x26
+#define WM9090_OUTPUT_MIXER1                    0x2D
+#define WM9090_OUTPUT_MIXER2                    0x2E
+#define WM9090_OUTPUT_MIXER3                    0x2F
+#define WM9090_OUTPUT_MIXER4                    0x30
+#define WM9090_SPEAKER_MIXER                    0x36
+#define WM9090_ANTIPOP2                         0x39
+#define WM9090_WRITE_SEQUENCER_0                0x46
+#define WM9090_WRITE_SEQUENCER_1                0x47
+#define WM9090_WRITE_SEQUENCER_2                0x48
+#define WM9090_WRITE_SEQUENCER_3                0x49
+#define WM9090_WRITE_SEQUENCER_4                0x4A
+#define WM9090_WRITE_SEQUENCER_5                0x4B
+#define WM9090_CHARGE_PUMP_1                    0x4C
+#define WM9090_DC_SERVO_0                       0x54
+#define WM9090_DC_SERVO_1                       0x55
+#define WM9090_DC_SERVO_3                       0x57
+#define WM9090_DC_SERVO_READBACK_0              0x58
+#define WM9090_DC_SERVO_READBACK_1              0x59
+#define WM9090_DC_SERVO_READBACK_2              0x5A
+#define WM9090_ANALOGUE_HP_0                    0x60
+#define WM9090_AGC_CONTROL_0                    0x62
+#define WM9090_AGC_CONTROL_1                    0x63
+#define WM9090_AGC_CONTROL_2                    0x64
+
+#define WM9090_REGISTER_COUNT                   40
+#define WM9090_MAX_REGISTER                     0x64
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Software Reset
+ */
+#define WM9090_SW_RESET_MASK                    0xFFFF  /* SW_RESET - [15:0] */
+#define WM9090_SW_RESET_SHIFT                        0  /* SW_RESET - [15:0] */
+#define WM9090_SW_RESET_WIDTH                       16  /* SW_RESET - [15:0] */
+
+/*
+ * R1 (0x01) - Power Management (1)
+ */
+#define WM9090_SPKOUTL_ENA                      0x1000  /* SPKOUTL_ENA */
+#define WM9090_SPKOUTL_ENA_MASK                 0x1000  /* SPKOUTL_ENA */
+#define WM9090_SPKOUTL_ENA_SHIFT                    12  /* SPKOUTL_ENA */
+#define WM9090_SPKOUTL_ENA_WIDTH                     1  /* SPKOUTL_ENA */
+#define WM9090_HPOUT1L_ENA                      0x0200  /* HPOUT1L_ENA */
+#define WM9090_HPOUT1L_ENA_MASK                 0x0200  /* HPOUT1L_ENA */
+#define WM9090_HPOUT1L_ENA_SHIFT                     9  /* HPOUT1L_ENA */
+#define WM9090_HPOUT1L_ENA_WIDTH                     1  /* HPOUT1L_ENA */
+#define WM9090_HPOUT1R_ENA                      0x0100  /* HPOUT1R_ENA */
+#define WM9090_HPOUT1R_ENA_MASK                 0x0100  /* HPOUT1R_ENA */
+#define WM9090_HPOUT1R_ENA_SHIFT                     8  /* HPOUT1R_ENA */
+#define WM9090_HPOUT1R_ENA_WIDTH                     1  /* HPOUT1R_ENA */
+#define WM9090_OSC_ENA                          0x0008  /* OSC_ENA */
+#define WM9090_OSC_ENA_MASK                     0x0008  /* OSC_ENA */
+#define WM9090_OSC_ENA_SHIFT                         3  /* OSC_ENA */
+#define WM9090_OSC_ENA_WIDTH                         1  /* OSC_ENA */
+#define WM9090_VMID_RES_MASK                    0x0006  /* VMID_RES - [2:1] */
+#define WM9090_VMID_RES_SHIFT                        1  /* VMID_RES - [2:1] */
+#define WM9090_VMID_RES_WIDTH                        2  /* VMID_RES - [2:1] */
+#define WM9090_BIAS_ENA                         0x0001  /* BIAS_ENA */
+#define WM9090_BIAS_ENA_MASK                    0x0001  /* BIAS_ENA */
+#define WM9090_BIAS_ENA_SHIFT                        0  /* BIAS_ENA */
+#define WM9090_BIAS_ENA_WIDTH                        1  /* BIAS_ENA */
+
+/*
+ * R2 (0x02) - Power Management (2)
+ */
+#define WM9090_TSHUT                            0x8000  /* TSHUT */
+#define WM9090_TSHUT_MASK                       0x8000  /* TSHUT */
+#define WM9090_TSHUT_SHIFT                          15  /* TSHUT */
+#define WM9090_TSHUT_WIDTH                           1  /* TSHUT */
+#define WM9090_TSHUT_ENA                        0x4000  /* TSHUT_ENA */
+#define WM9090_TSHUT_ENA_MASK                   0x4000  /* TSHUT_ENA */
+#define WM9090_TSHUT_ENA_SHIFT                      14  /* TSHUT_ENA */
+#define WM9090_TSHUT_ENA_WIDTH                       1  /* TSHUT_ENA */
+#define WM9090_TSHUT_OPDIS                      0x2000  /* TSHUT_OPDIS */
+#define WM9090_TSHUT_OPDIS_MASK                 0x2000  /* TSHUT_OPDIS */
+#define WM9090_TSHUT_OPDIS_SHIFT                    13  /* TSHUT_OPDIS */
+#define WM9090_TSHUT_OPDIS_WIDTH                     1  /* TSHUT_OPDIS */
+#define WM9090_IN1A_ENA                         0x0080  /* IN1A_ENA */
+#define WM9090_IN1A_ENA_MASK                    0x0080  /* IN1A_ENA */
+#define WM9090_IN1A_ENA_SHIFT                        7  /* IN1A_ENA */
+#define WM9090_IN1A_ENA_WIDTH                        1  /* IN1A_ENA */
+#define WM9090_IN1B_ENA                         0x0040  /* IN1B_ENA */
+#define WM9090_IN1B_ENA_MASK                    0x0040  /* IN1B_ENA */
+#define WM9090_IN1B_ENA_SHIFT                        6  /* IN1B_ENA */
+#define WM9090_IN1B_ENA_WIDTH                        1  /* IN1B_ENA */
+#define WM9090_IN2A_ENA                         0x0020  /* IN2A_ENA */
+#define WM9090_IN2A_ENA_MASK                    0x0020  /* IN2A_ENA */
+#define WM9090_IN2A_ENA_SHIFT                        5  /* IN2A_ENA */
+#define WM9090_IN2A_ENA_WIDTH                        1  /* IN2A_ENA */
+#define WM9090_IN2B_ENA                         0x0010  /* IN2B_ENA */
+#define WM9090_IN2B_ENA_MASK                    0x0010  /* IN2B_ENA */
+#define WM9090_IN2B_ENA_SHIFT                        4  /* IN2B_ENA */
+#define WM9090_IN2B_ENA_WIDTH                        1  /* IN2B_ENA */
+
+/*
+ * R3 (0x03) - Power Management (3)
+ */
+#define WM9090_AGC_ENA                          0x4000  /* AGC_ENA */
+#define WM9090_AGC_ENA_MASK                     0x4000  /* AGC_ENA */
+#define WM9090_AGC_ENA_SHIFT                        14  /* AGC_ENA */
+#define WM9090_AGC_ENA_WIDTH                         1  /* AGC_ENA */
+#define WM9090_SPKLVOL_ENA                      0x0100  /* SPKLVOL_ENA */
+#define WM9090_SPKLVOL_ENA_MASK                 0x0100  /* SPKLVOL_ENA */
+#define WM9090_SPKLVOL_ENA_SHIFT                     8  /* SPKLVOL_ENA */
+#define WM9090_SPKLVOL_ENA_WIDTH                     1  /* SPKLVOL_ENA */
+#define WM9090_MIXOUTL_ENA                      0x0020  /* MIXOUTL_ENA */
+#define WM9090_MIXOUTL_ENA_MASK                 0x0020  /* MIXOUTL_ENA */
+#define WM9090_MIXOUTL_ENA_SHIFT                     5  /* MIXOUTL_ENA */
+#define WM9090_MIXOUTL_ENA_WIDTH                     1  /* MIXOUTL_ENA */
+#define WM9090_MIXOUTR_ENA                      0x0010  /* MIXOUTR_ENA */
+#define WM9090_MIXOUTR_ENA_MASK                 0x0010  /* MIXOUTR_ENA */
+#define WM9090_MIXOUTR_ENA_SHIFT                     4  /* MIXOUTR_ENA */
+#define WM9090_MIXOUTR_ENA_WIDTH                     1  /* MIXOUTR_ENA */
+#define WM9090_SPKMIX_ENA                       0x0008  /* SPKMIX_ENA */
+#define WM9090_SPKMIX_ENA_MASK                  0x0008  /* SPKMIX_ENA */
+#define WM9090_SPKMIX_ENA_SHIFT                      3  /* SPKMIX_ENA */
+#define WM9090_SPKMIX_ENA_WIDTH                      1  /* SPKMIX_ENA */
+
+/*
+ * R6 (0x06) - Clocking 1
+ */
+#define WM9090_TOCLK_RATE                       0x8000  /* TOCLK_RATE */
+#define WM9090_TOCLK_RATE_MASK                  0x8000  /* TOCLK_RATE */
+#define WM9090_TOCLK_RATE_SHIFT                     15  /* TOCLK_RATE */
+#define WM9090_TOCLK_RATE_WIDTH                      1  /* TOCLK_RATE */
+#define WM9090_TOCLK_ENA                        0x4000  /* TOCLK_ENA */
+#define WM9090_TOCLK_ENA_MASK                   0x4000  /* TOCLK_ENA */
+#define WM9090_TOCLK_ENA_SHIFT                      14  /* TOCLK_ENA */
+#define WM9090_TOCLK_ENA_WIDTH                       1  /* TOCLK_ENA */
+
+/*
+ * R22 (0x16) - IN1 Line Control
+ */
+#define WM9090_IN1_DIFF                         0x0002  /* IN1_DIFF */
+#define WM9090_IN1_DIFF_MASK                    0x0002  /* IN1_DIFF */
+#define WM9090_IN1_DIFF_SHIFT                        1  /* IN1_DIFF */
+#define WM9090_IN1_DIFF_WIDTH                        1  /* IN1_DIFF */
+#define WM9090_IN1_CLAMP                        0x0001  /* IN1_CLAMP */
+#define WM9090_IN1_CLAMP_MASK                   0x0001  /* IN1_CLAMP */
+#define WM9090_IN1_CLAMP_SHIFT                       0  /* IN1_CLAMP */
+#define WM9090_IN1_CLAMP_WIDTH                       1  /* IN1_CLAMP */
+
+/*
+ * R23 (0x17) - IN2 Line Control
+ */
+#define WM9090_IN2_DIFF                         0x0002  /* IN2_DIFF */
+#define WM9090_IN2_DIFF_MASK                    0x0002  /* IN2_DIFF */
+#define WM9090_IN2_DIFF_SHIFT                        1  /* IN2_DIFF */
+#define WM9090_IN2_DIFF_WIDTH                        1  /* IN2_DIFF */
+#define WM9090_IN2_CLAMP                        0x0001  /* IN2_CLAMP */
+#define WM9090_IN2_CLAMP_MASK                   0x0001  /* IN2_CLAMP */
+#define WM9090_IN2_CLAMP_SHIFT                       0  /* IN2_CLAMP */
+#define WM9090_IN2_CLAMP_WIDTH                       1  /* IN2_CLAMP */
+
+/*
+ * R24 (0x18) - IN1 Line Input A Volume
+ */
+#define WM9090_IN1_VU                           0x0100  /* IN1_VU */
+#define WM9090_IN1_VU_MASK                      0x0100  /* IN1_VU */
+#define WM9090_IN1_VU_SHIFT                          8  /* IN1_VU */
+#define WM9090_IN1_VU_WIDTH                          1  /* IN1_VU */
+#define WM9090_IN1A_MUTE                        0x0080  /* IN1A_MUTE */
+#define WM9090_IN1A_MUTE_MASK                   0x0080  /* IN1A_MUTE */
+#define WM9090_IN1A_MUTE_SHIFT                       7  /* IN1A_MUTE */
+#define WM9090_IN1A_MUTE_WIDTH                       1  /* IN1A_MUTE */
+#define WM9090_IN1A_ZC                          0x0040  /* IN1A_ZC */
+#define WM9090_IN1A_ZC_MASK                     0x0040  /* IN1A_ZC */
+#define WM9090_IN1A_ZC_SHIFT                         6  /* IN1A_ZC */
+#define WM9090_IN1A_ZC_WIDTH                         1  /* IN1A_ZC */
+#define WM9090_IN1A_VOL_MASK                    0x0007  /* IN1A_VOL - [2:0] */
+#define WM9090_IN1A_VOL_SHIFT                        0  /* IN1A_VOL - [2:0] */
+#define WM9090_IN1A_VOL_WIDTH                        3  /* IN1A_VOL - [2:0] */
+
+/*
+ * R25 (0x19) - IN1  Line Input B Volume
+ */
+#define WM9090_IN1_VU                           0x0100  /* IN1_VU */
+#define WM9090_IN1_VU_MASK                      0x0100  /* IN1_VU */
+#define WM9090_IN1_VU_SHIFT                          8  /* IN1_VU */
+#define WM9090_IN1_VU_WIDTH                          1  /* IN1_VU */
+#define WM9090_IN1B_MUTE                        0x0080  /* IN1B_MUTE */
+#define WM9090_IN1B_MUTE_MASK                   0x0080  /* IN1B_MUTE */
+#define WM9090_IN1B_MUTE_SHIFT                       7  /* IN1B_MUTE */
+#define WM9090_IN1B_MUTE_WIDTH                       1  /* IN1B_MUTE */
+#define WM9090_IN1B_ZC                          0x0040  /* IN1B_ZC */
+#define WM9090_IN1B_ZC_MASK                     0x0040  /* IN1B_ZC */
+#define WM9090_IN1B_ZC_SHIFT                         6  /* IN1B_ZC */
+#define WM9090_IN1B_ZC_WIDTH                         1  /* IN1B_ZC */
+#define WM9090_IN1B_VOL_MASK                    0x0007  /* IN1B_VOL - [2:0] */
+#define WM9090_IN1B_VOL_SHIFT                        0  /* IN1B_VOL - [2:0] */
+#define WM9090_IN1B_VOL_WIDTH                        3  /* IN1B_VOL - [2:0] */
+
+/*
+ * R26 (0x1A) - IN2 Line Input A Volume
+ */
+#define WM9090_IN2_VU                           0x0100  /* IN2_VU */
+#define WM9090_IN2_VU_MASK                      0x0100  /* IN2_VU */
+#define WM9090_IN2_VU_SHIFT                          8  /* IN2_VU */
+#define WM9090_IN2_VU_WIDTH                          1  /* IN2_VU */
+#define WM9090_IN2A_MUTE                        0x0080  /* IN2A_MUTE */
+#define WM9090_IN2A_MUTE_MASK                   0x0080  /* IN2A_MUTE */
+#define WM9090_IN2A_MUTE_SHIFT                       7  /* IN2A_MUTE */
+#define WM9090_IN2A_MUTE_WIDTH                       1  /* IN2A_MUTE */
+#define WM9090_IN2A_ZC                          0x0040  /* IN2A_ZC */
+#define WM9090_IN2A_ZC_MASK                     0x0040  /* IN2A_ZC */
+#define WM9090_IN2A_ZC_SHIFT                         6  /* IN2A_ZC */
+#define WM9090_IN2A_ZC_WIDTH                         1  /* IN2A_ZC */
+#define WM9090_IN2A_VOL_MASK                    0x0007  /* IN2A_VOL - [2:0] */
+#define WM9090_IN2A_VOL_SHIFT                        0  /* IN2A_VOL - [2:0] */
+#define WM9090_IN2A_VOL_WIDTH                        3  /* IN2A_VOL - [2:0] */
+
+/*
+ * R27 (0x1B) - IN2 Line Input B Volume
+ */
+#define WM9090_IN2_VU                           0x0100  /* IN2_VU */
+#define WM9090_IN2_VU_MASK                      0x0100  /* IN2_VU */
+#define WM9090_IN2_VU_SHIFT                          8  /* IN2_VU */
+#define WM9090_IN2_VU_WIDTH                          1  /* IN2_VU */
+#define WM9090_IN2B_MUTE                        0x0080  /* IN2B_MUTE */
+#define WM9090_IN2B_MUTE_MASK                   0x0080  /* IN2B_MUTE */
+#define WM9090_IN2B_MUTE_SHIFT                       7  /* IN2B_MUTE */
+#define WM9090_IN2B_MUTE_WIDTH                       1  /* IN2B_MUTE */
+#define WM9090_IN2B_ZC                          0x0040  /* IN2B_ZC */
+#define WM9090_IN2B_ZC_MASK                     0x0040  /* IN2B_ZC */
+#define WM9090_IN2B_ZC_SHIFT                         6  /* IN2B_ZC */
+#define WM9090_IN2B_ZC_WIDTH                         1  /* IN2B_ZC */
+#define WM9090_IN2B_VOL_MASK                    0x0007  /* IN2B_VOL - [2:0] */
+#define WM9090_IN2B_VOL_SHIFT                        0  /* IN2B_VOL - [2:0] */
+#define WM9090_IN2B_VOL_WIDTH                        3  /* IN2B_VOL - [2:0] */
+
+/*
+ * R28 (0x1C) - Left Output Volume
+ */
+#define WM9090_HPOUT1_VU                        0x0100  /* HPOUT1_VU */
+#define WM9090_HPOUT1_VU_MASK                   0x0100  /* HPOUT1_VU */
+#define WM9090_HPOUT1_VU_SHIFT                       8  /* HPOUT1_VU */
+#define WM9090_HPOUT1_VU_WIDTH                       1  /* HPOUT1_VU */
+#define WM9090_HPOUT1L_ZC                       0x0080  /* HPOUT1L_ZC */
+#define WM9090_HPOUT1L_ZC_MASK                  0x0080  /* HPOUT1L_ZC */
+#define WM9090_HPOUT1L_ZC_SHIFT                      7  /* HPOUT1L_ZC */
+#define WM9090_HPOUT1L_ZC_WIDTH                      1  /* HPOUT1L_ZC */
+#define WM9090_HPOUT1L_MUTE                     0x0040  /* HPOUT1L_MUTE */
+#define WM9090_HPOUT1L_MUTE_MASK                0x0040  /* HPOUT1L_MUTE */
+#define WM9090_HPOUT1L_MUTE_SHIFT                    6  /* HPOUT1L_MUTE */
+#define WM9090_HPOUT1L_MUTE_WIDTH                    1  /* HPOUT1L_MUTE */
+#define WM9090_HPOUT1L_VOL_MASK                 0x003F  /* HPOUT1L_VOL - [5:0] */
+#define WM9090_HPOUT1L_VOL_SHIFT                     0  /* HPOUT1L_VOL - [5:0] */
+#define WM9090_HPOUT1L_VOL_WIDTH                     6  /* HPOUT1L_VOL - [5:0] */
+
+/*
+ * R29 (0x1D) - Right Output Volume
+ */
+#define WM9090_HPOUT1_VU                        0x0100  /* HPOUT1_VU */
+#define WM9090_HPOUT1_VU_MASK                   0x0100  /* HPOUT1_VU */
+#define WM9090_HPOUT1_VU_SHIFT                       8  /* HPOUT1_VU */
+#define WM9090_HPOUT1_VU_WIDTH                       1  /* HPOUT1_VU */
+#define WM9090_HPOUT1R_ZC                       0x0080  /* HPOUT1R_ZC */
+#define WM9090_HPOUT1R_ZC_MASK                  0x0080  /* HPOUT1R_ZC */
+#define WM9090_HPOUT1R_ZC_SHIFT                      7  /* HPOUT1R_ZC */
+#define WM9090_HPOUT1R_ZC_WIDTH                      1  /* HPOUT1R_ZC */
+#define WM9090_HPOUT1R_MUTE                     0x0040  /* HPOUT1R_MUTE */
+#define WM9090_HPOUT1R_MUTE_MASK                0x0040  /* HPOUT1R_MUTE */
+#define WM9090_HPOUT1R_MUTE_SHIFT                    6  /* HPOUT1R_MUTE */
+#define WM9090_HPOUT1R_MUTE_WIDTH                    1  /* HPOUT1R_MUTE */
+#define WM9090_HPOUT1R_VOL_MASK                 0x003F  /* HPOUT1R_VOL - [5:0] */
+#define WM9090_HPOUT1R_VOL_SHIFT                     0  /* HPOUT1R_VOL - [5:0] */
+#define WM9090_HPOUT1R_VOL_WIDTH                     6  /* HPOUT1R_VOL - [5:0] */
+
+/*
+ * R34 (0x22) - SPKMIXL Attenuation
+ */
+#define WM9090_SPKMIX_MUTE                      0x0100  /* SPKMIX_MUTE */
+#define WM9090_SPKMIX_MUTE_MASK                 0x0100  /* SPKMIX_MUTE */
+#define WM9090_SPKMIX_MUTE_SHIFT                     8  /* SPKMIX_MUTE */
+#define WM9090_SPKMIX_MUTE_WIDTH                     1  /* SPKMIX_MUTE */
+#define WM9090_IN1A_SPKMIX_VOL_MASK             0x00C0  /* IN1A_SPKMIX_VOL - [7:6] */
+#define WM9090_IN1A_SPKMIX_VOL_SHIFT                 6  /* IN1A_SPKMIX_VOL - [7:6] */
+#define WM9090_IN1A_SPKMIX_VOL_WIDTH                 2  /* IN1A_SPKMIX_VOL - [7:6] */
+#define WM9090_IN1B_SPKMIX_VOL_MASK             0x0030  /* IN1B_SPKMIX_VOL - [5:4] */
+#define WM9090_IN1B_SPKMIX_VOL_SHIFT                 4  /* IN1B_SPKMIX_VOL - [5:4] */
+#define WM9090_IN1B_SPKMIX_VOL_WIDTH                 2  /* IN1B_SPKMIX_VOL - [5:4] */
+#define WM9090_IN2A_SPKMIX_VOL_MASK             0x000C  /* IN2A_SPKMIX_VOL - [3:2] */
+#define WM9090_IN2A_SPKMIX_VOL_SHIFT                 2  /* IN2A_SPKMIX_VOL - [3:2] */
+#define WM9090_IN2A_SPKMIX_VOL_WIDTH                 2  /* IN2A_SPKMIX_VOL - [3:2] */
+#define WM9090_IN2B_SPKMIX_VOL_MASK             0x0003  /* IN2B_SPKMIX_VOL - [1:0] */
+#define WM9090_IN2B_SPKMIX_VOL_SHIFT                 0  /* IN2B_SPKMIX_VOL - [1:0] */
+#define WM9090_IN2B_SPKMIX_VOL_WIDTH                 2  /* IN2B_SPKMIX_VOL - [1:0] */
+
+/*
+ * R36 (0x24) - SPKOUT Mixers
+ */
+#define WM9090_SPKMIXL_TO_SPKOUTL               0x0010  /* SPKMIXL_TO_SPKOUTL */
+#define WM9090_SPKMIXL_TO_SPKOUTL_MASK          0x0010  /* SPKMIXL_TO_SPKOUTL */
+#define WM9090_SPKMIXL_TO_SPKOUTL_SHIFT              4  /* SPKMIXL_TO_SPKOUTL */
+#define WM9090_SPKMIXL_TO_SPKOUTL_WIDTH              1  /* SPKMIXL_TO_SPKOUTL */
+
+/*
+ * R37 (0x25) - ClassD3
+ */
+#define WM9090_SPKOUTL_BOOST_MASK               0x0038  /* SPKOUTL_BOOST - [5:3] */
+#define WM9090_SPKOUTL_BOOST_SHIFT                   3  /* SPKOUTL_BOOST - [5:3] */
+#define WM9090_SPKOUTL_BOOST_WIDTH                   3  /* SPKOUTL_BOOST - [5:3] */
+
+/*
+ * R38 (0x26) - Speaker Volume Left
+ */
+#define WM9090_SPKOUT_VU                        0x0100  /* SPKOUT_VU */
+#define WM9090_SPKOUT_VU_MASK                   0x0100  /* SPKOUT_VU */
+#define WM9090_SPKOUT_VU_SHIFT                       8  /* SPKOUT_VU */
+#define WM9090_SPKOUT_VU_WIDTH                       1  /* SPKOUT_VU */
+#define WM9090_SPKOUTL_ZC                       0x0080  /* SPKOUTL_ZC */
+#define WM9090_SPKOUTL_ZC_MASK                  0x0080  /* SPKOUTL_ZC */
+#define WM9090_SPKOUTL_ZC_SHIFT                      7  /* SPKOUTL_ZC */
+#define WM9090_SPKOUTL_ZC_WIDTH                      1  /* SPKOUTL_ZC */
+#define WM9090_SPKOUTL_MUTE                     0x0040  /* SPKOUTL_MUTE */
+#define WM9090_SPKOUTL_MUTE_MASK                0x0040  /* SPKOUTL_MUTE */
+#define WM9090_SPKOUTL_MUTE_SHIFT                    6  /* SPKOUTL_MUTE */
+#define WM9090_SPKOUTL_MUTE_WIDTH                    1  /* SPKOUTL_MUTE */
+#define WM9090_SPKOUTL_VOL_MASK                 0x003F  /* SPKOUTL_VOL - [5:0] */
+#define WM9090_SPKOUTL_VOL_SHIFT                     0  /* SPKOUTL_VOL - [5:0] */
+#define WM9090_SPKOUTL_VOL_WIDTH                     6  /* SPKOUTL_VOL - [5:0] */
+
+/*
+ * R45 (0x2D) - Output Mixer1
+ */
+#define WM9090_IN1A_TO_MIXOUTL                  0x0040  /* IN1A_TO_MIXOUTL */
+#define WM9090_IN1A_TO_MIXOUTL_MASK             0x0040  /* IN1A_TO_MIXOUTL */
+#define WM9090_IN1A_TO_MIXOUTL_SHIFT                 6  /* IN1A_TO_MIXOUTL */
+#define WM9090_IN1A_TO_MIXOUTL_WIDTH                 1  /* IN1A_TO_MIXOUTL */
+#define WM9090_IN2A_TO_MIXOUTL                  0x0004  /* IN2A_TO_MIXOUTL */
+#define WM9090_IN2A_TO_MIXOUTL_MASK             0x0004  /* IN2A_TO_MIXOUTL */
+#define WM9090_IN2A_TO_MIXOUTL_SHIFT                 2  /* IN2A_TO_MIXOUTL */
+#define WM9090_IN2A_TO_MIXOUTL_WIDTH                 1  /* IN2A_TO_MIXOUTL */
+
+/*
+ * R46 (0x2E) - Output Mixer2
+ */
+#define WM9090_IN1A_TO_MIXOUTR                  0x0040  /* IN1A_TO_MIXOUTR */
+#define WM9090_IN1A_TO_MIXOUTR_MASK             0x0040  /* IN1A_TO_MIXOUTR */
+#define WM9090_IN1A_TO_MIXOUTR_SHIFT                 6  /* IN1A_TO_MIXOUTR */
+#define WM9090_IN1A_TO_MIXOUTR_WIDTH                 1  /* IN1A_TO_MIXOUTR */
+#define WM9090_IN1B_TO_MIXOUTR                  0x0010  /* IN1B_TO_MIXOUTR */
+#define WM9090_IN1B_TO_MIXOUTR_MASK             0x0010  /* IN1B_TO_MIXOUTR */
+#define WM9090_IN1B_TO_MIXOUTR_SHIFT                 4  /* IN1B_TO_MIXOUTR */
+#define WM9090_IN1B_TO_MIXOUTR_WIDTH                 1  /* IN1B_TO_MIXOUTR */
+#define WM9090_IN2A_TO_MIXOUTR                  0x0004  /* IN2A_TO_MIXOUTR */
+#define WM9090_IN2A_TO_MIXOUTR_MASK             0x0004  /* IN2A_TO_MIXOUTR */
+#define WM9090_IN2A_TO_MIXOUTR_SHIFT                 2  /* IN2A_TO_MIXOUTR */
+#define WM9090_IN2A_TO_MIXOUTR_WIDTH                 1  /* IN2A_TO_MIXOUTR */
+#define WM9090_IN2B_TO_MIXOUTR                  0x0001  /* IN2B_TO_MIXOUTR */
+#define WM9090_IN2B_TO_MIXOUTR_MASK             0x0001  /* IN2B_TO_MIXOUTR */
+#define WM9090_IN2B_TO_MIXOUTR_SHIFT                 0  /* IN2B_TO_MIXOUTR */
+#define WM9090_IN2B_TO_MIXOUTR_WIDTH                 1  /* IN2B_TO_MIXOUTR */
+
+/*
+ * R47 (0x2F) - Output Mixer3
+ */
+#define WM9090_MIXOUTL_MUTE                     0x0100  /* MIXOUTL_MUTE */
+#define WM9090_MIXOUTL_MUTE_MASK                0x0100  /* MIXOUTL_MUTE */
+#define WM9090_MIXOUTL_MUTE_SHIFT                    8  /* MIXOUTL_MUTE */
+#define WM9090_MIXOUTL_MUTE_WIDTH                    1  /* MIXOUTL_MUTE */
+#define WM9090_IN1A_MIXOUTL_VOL_MASK            0x00C0  /* IN1A_MIXOUTL_VOL - [7:6] */
+#define WM9090_IN1A_MIXOUTL_VOL_SHIFT                6  /* IN1A_MIXOUTL_VOL - [7:6] */
+#define WM9090_IN1A_MIXOUTL_VOL_WIDTH                2  /* IN1A_MIXOUTL_VOL - [7:6] */
+#define WM9090_IN2A_MIXOUTL_VOL_MASK            0x000C  /* IN2A_MIXOUTL_VOL - [3:2] */
+#define WM9090_IN2A_MIXOUTL_VOL_SHIFT                2  /* IN2A_MIXOUTL_VOL - [3:2] */
+#define WM9090_IN2A_MIXOUTL_VOL_WIDTH                2  /* IN2A_MIXOUTL_VOL - [3:2] */
+
+/*
+ * R48 (0x30) - Output Mixer4
+ */
+#define WM9090_MIXOUTR_MUTE                     0x0100  /* MIXOUTR_MUTE */
+#define WM9090_MIXOUTR_MUTE_MASK                0x0100  /* MIXOUTR_MUTE */
+#define WM9090_MIXOUTR_MUTE_SHIFT                    8  /* MIXOUTR_MUTE */
+#define WM9090_MIXOUTR_MUTE_WIDTH                    1  /* MIXOUTR_MUTE */
+#define WM9090_IN1A_MIXOUTR_VOL_MASK            0x00C0  /* IN1A_MIXOUTR_VOL - [7:6] */
+#define WM9090_IN1A_MIXOUTR_VOL_SHIFT                6  /* IN1A_MIXOUTR_VOL - [7:6] */
+#define WM9090_IN1A_MIXOUTR_VOL_WIDTH                2  /* IN1A_MIXOUTR_VOL - [7:6] */
+#define WM9090_IN1B_MIXOUTR_VOL_MASK            0x0030  /* IN1B_MIXOUTR_VOL - [5:4] */
+#define WM9090_IN1B_MIXOUTR_VOL_SHIFT                4  /* IN1B_MIXOUTR_VOL - [5:4] */
+#define WM9090_IN1B_MIXOUTR_VOL_WIDTH                2  /* IN1B_MIXOUTR_VOL - [5:4] */
+#define WM9090_IN2A_MIXOUTR_VOL_MASK            0x000C  /* IN2A_MIXOUTR_VOL - [3:2] */
+#define WM9090_IN2A_MIXOUTR_VOL_SHIFT                2  /* IN2A_MIXOUTR_VOL - [3:2] */
+#define WM9090_IN2A_MIXOUTR_VOL_WIDTH                2  /* IN2A_MIXOUTR_VOL - [3:2] */
+#define WM9090_IN2B_MIXOUTR_VOL_MASK            0x0003  /* IN2B_MIXOUTR_VOL - [1:0] */
+#define WM9090_IN2B_MIXOUTR_VOL_SHIFT                0  /* IN2B_MIXOUTR_VOL - [1:0] */
+#define WM9090_IN2B_MIXOUTR_VOL_WIDTH                2  /* IN2B_MIXOUTR_VOL - [1:0] */
+
+/*
+ * R54 (0x36) - Speaker Mixer
+ */
+#define WM9090_IN1A_TO_SPKMIX                   0x0040  /* IN1A_TO_SPKMIX */
+#define WM9090_IN1A_TO_SPKMIX_MASK              0x0040  /* IN1A_TO_SPKMIX */
+#define WM9090_IN1A_TO_SPKMIX_SHIFT                  6  /* IN1A_TO_SPKMIX */
+#define WM9090_IN1A_TO_SPKMIX_WIDTH                  1  /* IN1A_TO_SPKMIX */
+#define WM9090_IN1B_TO_SPKMIX                   0x0010  /* IN1B_TO_SPKMIX */
+#define WM9090_IN1B_TO_SPKMIX_MASK              0x0010  /* IN1B_TO_SPKMIX */
+#define WM9090_IN1B_TO_SPKMIX_SHIFT                  4  /* IN1B_TO_SPKMIX */
+#define WM9090_IN1B_TO_SPKMIX_WIDTH                  1  /* IN1B_TO_SPKMIX */
+#define WM9090_IN2A_TO_SPKMIX                   0x0004  /* IN2A_TO_SPKMIX */
+#define WM9090_IN2A_TO_SPKMIX_MASK              0x0004  /* IN2A_TO_SPKMIX */
+#define WM9090_IN2A_TO_SPKMIX_SHIFT                  2  /* IN2A_TO_SPKMIX */
+#define WM9090_IN2A_TO_SPKMIX_WIDTH                  1  /* IN2A_TO_SPKMIX */
+#define WM9090_IN2B_TO_SPKMIX                   0x0001  /* IN2B_TO_SPKMIX */
+#define WM9090_IN2B_TO_SPKMIX_MASK              0x0001  /* IN2B_TO_SPKMIX */
+#define WM9090_IN2B_TO_SPKMIX_SHIFT                  0  /* IN2B_TO_SPKMIX */
+#define WM9090_IN2B_TO_SPKMIX_WIDTH                  1  /* IN2B_TO_SPKMIX */
+
+/*
+ * R57 (0x39) - AntiPOP2
+ */
+#define WM9090_VMID_BUF_ENA                     0x0008  /* VMID_BUF_ENA */
+#define WM9090_VMID_BUF_ENA_MASK                0x0008  /* VMID_BUF_ENA */
+#define WM9090_VMID_BUF_ENA_SHIFT                    3  /* VMID_BUF_ENA */
+#define WM9090_VMID_BUF_ENA_WIDTH                    1  /* VMID_BUF_ENA */
+#define WM9090_VMID_ENA                         0x0001  /* VMID_ENA */
+#define WM9090_VMID_ENA_MASK                    0x0001  /* VMID_ENA */
+#define WM9090_VMID_ENA_SHIFT                        0  /* VMID_ENA */
+#define WM9090_VMID_ENA_WIDTH                        1  /* VMID_ENA */
+
+/*
+ * R70 (0x46) - Write Sequencer 0
+ */
+#define WM9090_WSEQ_ENA                         0x0100  /* WSEQ_ENA */
+#define WM9090_WSEQ_ENA_MASK                    0x0100  /* WSEQ_ENA */
+#define WM9090_WSEQ_ENA_SHIFT                        8  /* WSEQ_ENA */
+#define WM9090_WSEQ_ENA_WIDTH                        1  /* WSEQ_ENA */
+#define WM9090_WSEQ_WRITE_INDEX_MASK            0x000F  /* WSEQ_WRITE_INDEX - [3:0] */
+#define WM9090_WSEQ_WRITE_INDEX_SHIFT                0  /* WSEQ_WRITE_INDEX - [3:0] */
+#define WM9090_WSEQ_WRITE_INDEX_WIDTH                4  /* WSEQ_WRITE_INDEX - [3:0] */
+
+/*
+ * R71 (0x47) - Write Sequencer 1
+ */
+#define WM9090_WSEQ_DATA_WIDTH_MASK             0x7000  /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM9090_WSEQ_DATA_WIDTH_SHIFT                12  /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM9090_WSEQ_DATA_WIDTH_WIDTH                 3  /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM9090_WSEQ_DATA_START_MASK             0x0F00  /* WSEQ_DATA_START - [11:8] */
+#define WM9090_WSEQ_DATA_START_SHIFT                 8  /* WSEQ_DATA_START - [11:8] */
+#define WM9090_WSEQ_DATA_START_WIDTH                 4  /* WSEQ_DATA_START - [11:8] */
+#define WM9090_WSEQ_ADDR_MASK                   0x00FF  /* WSEQ_ADDR - [7:0] */
+#define WM9090_WSEQ_ADDR_SHIFT                       0  /* WSEQ_ADDR - [7:0] */
+#define WM9090_WSEQ_ADDR_WIDTH                       8  /* WSEQ_ADDR - [7:0] */
+
+/*
+ * R72 (0x48) - Write Sequencer 2
+ */
+#define WM9090_WSEQ_EOS                         0x4000  /* WSEQ_EOS */
+#define WM9090_WSEQ_EOS_MASK                    0x4000  /* WSEQ_EOS */
+#define WM9090_WSEQ_EOS_SHIFT                       14  /* WSEQ_EOS */
+#define WM9090_WSEQ_EOS_WIDTH                        1  /* WSEQ_EOS */
+#define WM9090_WSEQ_DELAY_MASK                  0x0F00  /* WSEQ_DELAY - [11:8] */
+#define WM9090_WSEQ_DELAY_SHIFT                      8  /* WSEQ_DELAY - [11:8] */
+#define WM9090_WSEQ_DELAY_WIDTH                      4  /* WSEQ_DELAY - [11:8] */
+#define WM9090_WSEQ_DATA_MASK                   0x00FF  /* WSEQ_DATA - [7:0] */
+#define WM9090_WSEQ_DATA_SHIFT                       0  /* WSEQ_DATA - [7:0] */
+#define WM9090_WSEQ_DATA_WIDTH                       8  /* WSEQ_DATA - [7:0] */
+
+/*
+ * R73 (0x49) - Write Sequencer 3
+ */
+#define WM9090_WSEQ_ABORT                       0x0200  /* WSEQ_ABORT */
+#define WM9090_WSEQ_ABORT_MASK                  0x0200  /* WSEQ_ABORT */
+#define WM9090_WSEQ_ABORT_SHIFT                      9  /* WSEQ_ABORT */
+#define WM9090_WSEQ_ABORT_WIDTH                      1  /* WSEQ_ABORT */
+#define WM9090_WSEQ_START                       0x0100  /* WSEQ_START */
+#define WM9090_WSEQ_START_MASK                  0x0100  /* WSEQ_START */
+#define WM9090_WSEQ_START_SHIFT                      8  /* WSEQ_START */
+#define WM9090_WSEQ_START_WIDTH                      1  /* WSEQ_START */
+#define WM9090_WSEQ_START_INDEX_MASK            0x003F  /* WSEQ_START_INDEX - [5:0] */
+#define WM9090_WSEQ_START_INDEX_SHIFT                0  /* WSEQ_START_INDEX - [5:0] */
+#define WM9090_WSEQ_START_INDEX_WIDTH                6  /* WSEQ_START_INDEX - [5:0] */
+
+/*
+ * R74 (0x4A) - Write Sequencer 4
+ */
+#define WM9090_WSEQ_BUSY                        0x0001  /* WSEQ_BUSY */
+#define WM9090_WSEQ_BUSY_MASK                   0x0001  /* WSEQ_BUSY */
+#define WM9090_WSEQ_BUSY_SHIFT                       0  /* WSEQ_BUSY */
+#define WM9090_WSEQ_BUSY_WIDTH                       1  /* WSEQ_BUSY */
+
+/*
+ * R75 (0x4B) - Write Sequencer 5
+ */
+#define WM9090_WSEQ_CURRENT_INDEX_MASK          0x003F  /* WSEQ_CURRENT_INDEX - [5:0] */
+#define WM9090_WSEQ_CURRENT_INDEX_SHIFT              0  /* WSEQ_CURRENT_INDEX - [5:0] */
+#define WM9090_WSEQ_CURRENT_INDEX_WIDTH              6  /* WSEQ_CURRENT_INDEX - [5:0] */
+
+/*
+ * R76 (0x4C) - Charge Pump 1
+ */
+#define WM9090_CP_ENA                           0x8000  /* CP_ENA */
+#define WM9090_CP_ENA_MASK                      0x8000  /* CP_ENA */
+#define WM9090_CP_ENA_SHIFT                         15  /* CP_ENA */
+#define WM9090_CP_ENA_WIDTH                          1  /* CP_ENA */
+
+/*
+ * R84 (0x54) - DC Servo 0
+ */
+#define WM9090_DCS_TRIG_SINGLE_1                0x2000  /* DCS_TRIG_SINGLE_1 */
+#define WM9090_DCS_TRIG_SINGLE_1_MASK           0x2000  /* DCS_TRIG_SINGLE_1 */
+#define WM9090_DCS_TRIG_SINGLE_1_SHIFT              13  /* DCS_TRIG_SINGLE_1 */
+#define WM9090_DCS_TRIG_SINGLE_1_WIDTH               1  /* DCS_TRIG_SINGLE_1 */
+#define WM9090_DCS_TRIG_SINGLE_0                0x1000  /* DCS_TRIG_SINGLE_0 */
+#define WM9090_DCS_TRIG_SINGLE_0_MASK           0x1000  /* DCS_TRIG_SINGLE_0 */
+#define WM9090_DCS_TRIG_SINGLE_0_SHIFT              12  /* DCS_TRIG_SINGLE_0 */
+#define WM9090_DCS_TRIG_SINGLE_0_WIDTH               1  /* DCS_TRIG_SINGLE_0 */
+#define WM9090_DCS_TRIG_SERIES_1                0x0200  /* DCS_TRIG_SERIES_1 */
+#define WM9090_DCS_TRIG_SERIES_1_MASK           0x0200  /* DCS_TRIG_SERIES_1 */
+#define WM9090_DCS_TRIG_SERIES_1_SHIFT               9  /* DCS_TRIG_SERIES_1 */
+#define WM9090_DCS_TRIG_SERIES_1_WIDTH               1  /* DCS_TRIG_SERIES_1 */
+#define WM9090_DCS_TRIG_SERIES_0                0x0100  /* DCS_TRIG_SERIES_0 */
+#define WM9090_DCS_TRIG_SERIES_0_MASK           0x0100  /* DCS_TRIG_SERIES_0 */
+#define WM9090_DCS_TRIG_SERIES_0_SHIFT               8  /* DCS_TRIG_SERIES_0 */
+#define WM9090_DCS_TRIG_SERIES_0_WIDTH               1  /* DCS_TRIG_SERIES_0 */
+#define WM9090_DCS_TRIG_STARTUP_1               0x0020  /* DCS_TRIG_STARTUP_1 */
+#define WM9090_DCS_TRIG_STARTUP_1_MASK          0x0020  /* DCS_TRIG_STARTUP_1 */
+#define WM9090_DCS_TRIG_STARTUP_1_SHIFT              5  /* DCS_TRIG_STARTUP_1 */
+#define WM9090_DCS_TRIG_STARTUP_1_WIDTH              1  /* DCS_TRIG_STARTUP_1 */
+#define WM9090_DCS_TRIG_STARTUP_0               0x0010  /* DCS_TRIG_STARTUP_0 */
+#define WM9090_DCS_TRIG_STARTUP_0_MASK          0x0010  /* DCS_TRIG_STARTUP_0 */
+#define WM9090_DCS_TRIG_STARTUP_0_SHIFT              4  /* DCS_TRIG_STARTUP_0 */
+#define WM9090_DCS_TRIG_STARTUP_0_WIDTH              1  /* DCS_TRIG_STARTUP_0 */
+#define WM9090_DCS_TRIG_DAC_WR_1                0x0008  /* DCS_TRIG_DAC_WR_1 */
+#define WM9090_DCS_TRIG_DAC_WR_1_MASK           0x0008  /* DCS_TRIG_DAC_WR_1 */
+#define WM9090_DCS_TRIG_DAC_WR_1_SHIFT               3  /* DCS_TRIG_DAC_WR_1 */
+#define WM9090_DCS_TRIG_DAC_WR_1_WIDTH               1  /* DCS_TRIG_DAC_WR_1 */
+#define WM9090_DCS_TRIG_DAC_WR_0                0x0004  /* DCS_TRIG_DAC_WR_0 */
+#define WM9090_DCS_TRIG_DAC_WR_0_MASK           0x0004  /* DCS_TRIG_DAC_WR_0 */
+#define WM9090_DCS_TRIG_DAC_WR_0_SHIFT               2  /* DCS_TRIG_DAC_WR_0 */
+#define WM9090_DCS_TRIG_DAC_WR_0_WIDTH               1  /* DCS_TRIG_DAC_WR_0 */
+#define WM9090_DCS_ENA_CHAN_1                   0x0002  /* DCS_ENA_CHAN_1 */
+#define WM9090_DCS_ENA_CHAN_1_MASK              0x0002  /* DCS_ENA_CHAN_1 */
+#define WM9090_DCS_ENA_CHAN_1_SHIFT                  1  /* DCS_ENA_CHAN_1 */
+#define WM9090_DCS_ENA_CHAN_1_WIDTH                  1  /* DCS_ENA_CHAN_1 */
+#define WM9090_DCS_ENA_CHAN_0                   0x0001  /* DCS_ENA_CHAN_0 */
+#define WM9090_DCS_ENA_CHAN_0_MASK              0x0001  /* DCS_ENA_CHAN_0 */
+#define WM9090_DCS_ENA_CHAN_0_SHIFT                  0  /* DCS_ENA_CHAN_0 */
+#define WM9090_DCS_ENA_CHAN_0_WIDTH                  1  /* DCS_ENA_CHAN_0 */
+
+/*
+ * R85 (0x55) - DC Servo 1
+ */
+#define WM9090_DCS_SERIES_NO_01_MASK            0x0FE0  /* DCS_SERIES_NO_01 - [11:5] */
+#define WM9090_DCS_SERIES_NO_01_SHIFT                5  /* DCS_SERIES_NO_01 - [11:5] */
+#define WM9090_DCS_SERIES_NO_01_WIDTH                7  /* DCS_SERIES_NO_01 - [11:5] */
+#define WM9090_DCS_TIMER_PERIOD_01_MASK         0x000F  /* DCS_TIMER_PERIOD_01 - [3:0] */
+#define WM9090_DCS_TIMER_PERIOD_01_SHIFT             0  /* DCS_TIMER_PERIOD_01 - [3:0] */
+#define WM9090_DCS_TIMER_PERIOD_01_WIDTH             4  /* DCS_TIMER_PERIOD_01 - [3:0] */
+
+/*
+ * R87 (0x57) - DC Servo 3
+ */
+#define WM9090_DCS_DAC_WR_VAL_1_MASK            0xFF00  /* DCS_DAC_WR_VAL_1 - [15:8] */
+#define WM9090_DCS_DAC_WR_VAL_1_SHIFT                8  /* DCS_DAC_WR_VAL_1 - [15:8] */
+#define WM9090_DCS_DAC_WR_VAL_1_WIDTH                8  /* DCS_DAC_WR_VAL_1 - [15:8] */
+#define WM9090_DCS_DAC_WR_VAL_0_MASK            0x00FF  /* DCS_DAC_WR_VAL_0 - [7:0] */
+#define WM9090_DCS_DAC_WR_VAL_0_SHIFT                0  /* DCS_DAC_WR_VAL_0 - [7:0] */
+#define WM9090_DCS_DAC_WR_VAL_0_WIDTH                8  /* DCS_DAC_WR_VAL_0 - [7:0] */
+
+/*
+ * R88 (0x58) - DC Servo Readback 0
+ */
+#define WM9090_DCS_CAL_COMPLETE_MASK            0x0300  /* DCS_CAL_COMPLETE - [9:8] */
+#define WM9090_DCS_CAL_COMPLETE_SHIFT                8  /* DCS_CAL_COMPLETE - [9:8] */
+#define WM9090_DCS_CAL_COMPLETE_WIDTH                2  /* DCS_CAL_COMPLETE - [9:8] */
+#define WM9090_DCS_DAC_WR_COMPLETE_MASK         0x0030  /* DCS_DAC_WR_COMPLETE - [5:4] */
+#define WM9090_DCS_DAC_WR_COMPLETE_SHIFT             4  /* DCS_DAC_WR_COMPLETE - [5:4] */
+#define WM9090_DCS_DAC_WR_COMPLETE_WIDTH             2  /* DCS_DAC_WR_COMPLETE - [5:4] */
+#define WM9090_DCS_STARTUP_COMPLETE_MASK        0x0003  /* DCS_STARTUP_COMPLETE - [1:0] */
+#define WM9090_DCS_STARTUP_COMPLETE_SHIFT            0  /* DCS_STARTUP_COMPLETE - [1:0] */
+#define WM9090_DCS_STARTUP_COMPLETE_WIDTH            2  /* DCS_STARTUP_COMPLETE - [1:0] */
+
+/*
+ * R89 (0x59) - DC Servo Readback 1
+ */
+#define WM9090_DCS_DAC_WR_VAL_1_RD_MASK         0x00FF  /* DCS_DAC_WR_VAL_1_RD - [7:0] */
+#define WM9090_DCS_DAC_WR_VAL_1_RD_SHIFT             0  /* DCS_DAC_WR_VAL_1_RD - [7:0] */
+#define WM9090_DCS_DAC_WR_VAL_1_RD_WIDTH             8  /* DCS_DAC_WR_VAL_1_RD - [7:0] */
+
+/*
+ * R90 (0x5A) - DC Servo Readback 2
+ */
+#define WM9090_DCS_DAC_WR_VAL_0_RD_MASK         0x00FF  /* DCS_DAC_WR_VAL_0_RD - [7:0] */
+#define WM9090_DCS_DAC_WR_VAL_0_RD_SHIFT             0  /* DCS_DAC_WR_VAL_0_RD - [7:0] */
+#define WM9090_DCS_DAC_WR_VAL_0_RD_WIDTH             8  /* DCS_DAC_WR_VAL_0_RD - [7:0] */
+
+/*
+ * R96 (0x60) - Analogue HP 0
+ */
+#define WM9090_HPOUT1L_RMV_SHORT                0x0080  /* HPOUT1L_RMV_SHORT */
+#define WM9090_HPOUT1L_RMV_SHORT_MASK           0x0080  /* HPOUT1L_RMV_SHORT */
+#define WM9090_HPOUT1L_RMV_SHORT_SHIFT               7  /* HPOUT1L_RMV_SHORT */
+#define WM9090_HPOUT1L_RMV_SHORT_WIDTH               1  /* HPOUT1L_RMV_SHORT */
+#define WM9090_HPOUT1L_OUTP                     0x0040  /* HPOUT1L_OUTP */
+#define WM9090_HPOUT1L_OUTP_MASK                0x0040  /* HPOUT1L_OUTP */
+#define WM9090_HPOUT1L_OUTP_SHIFT                    6  /* HPOUT1L_OUTP */
+#define WM9090_HPOUT1L_OUTP_WIDTH                    1  /* HPOUT1L_OUTP */
+#define WM9090_HPOUT1L_DLY                      0x0020  /* HPOUT1L_DLY */
+#define WM9090_HPOUT1L_DLY_MASK                 0x0020  /* HPOUT1L_DLY */
+#define WM9090_HPOUT1L_DLY_SHIFT                     5  /* HPOUT1L_DLY */
+#define WM9090_HPOUT1L_DLY_WIDTH                     1  /* HPOUT1L_DLY */
+#define WM9090_HPOUT1R_RMV_SHORT                0x0008  /* HPOUT1R_RMV_SHORT */
+#define WM9090_HPOUT1R_RMV_SHORT_MASK           0x0008  /* HPOUT1R_RMV_SHORT */
+#define WM9090_HPOUT1R_RMV_SHORT_SHIFT               3  /* HPOUT1R_RMV_SHORT */
+#define WM9090_HPOUT1R_RMV_SHORT_WIDTH               1  /* HPOUT1R_RMV_SHORT */
+#define WM9090_HPOUT1R_OUTP                     0x0004  /* HPOUT1R_OUTP */
+#define WM9090_HPOUT1R_OUTP_MASK                0x0004  /* HPOUT1R_OUTP */
+#define WM9090_HPOUT1R_OUTP_SHIFT                    2  /* HPOUT1R_OUTP */
+#define WM9090_HPOUT1R_OUTP_WIDTH                    1  /* HPOUT1R_OUTP */
+#define WM9090_HPOUT1R_DLY                      0x0002  /* HPOUT1R_DLY */
+#define WM9090_HPOUT1R_DLY_MASK                 0x0002  /* HPOUT1R_DLY */
+#define WM9090_HPOUT1R_DLY_SHIFT                     1  /* HPOUT1R_DLY */
+#define WM9090_HPOUT1R_DLY_WIDTH                     1  /* HPOUT1R_DLY */
+
+/*
+ * R98 (0x62) - AGC Control 0
+ */
+#define WM9090_AGC_CLIP_ENA                     0x8000  /* AGC_CLIP_ENA */
+#define WM9090_AGC_CLIP_ENA_MASK                0x8000  /* AGC_CLIP_ENA */
+#define WM9090_AGC_CLIP_ENA_SHIFT                   15  /* AGC_CLIP_ENA */
+#define WM9090_AGC_CLIP_ENA_WIDTH                    1  /* AGC_CLIP_ENA */
+#define WM9090_AGC_CLIP_THR_MASK                0x0F00  /* AGC_CLIP_THR - [11:8] */
+#define WM9090_AGC_CLIP_THR_SHIFT                    8  /* AGC_CLIP_THR - [11:8] */
+#define WM9090_AGC_CLIP_THR_WIDTH                    4  /* AGC_CLIP_THR - [11:8] */
+#define WM9090_AGC_CLIP_ATK_MASK                0x0070  /* AGC_CLIP_ATK - [6:4] */
+#define WM9090_AGC_CLIP_ATK_SHIFT                    4  /* AGC_CLIP_ATK - [6:4] */
+#define WM9090_AGC_CLIP_ATK_WIDTH                    3  /* AGC_CLIP_ATK - [6:4] */
+#define WM9090_AGC_CLIP_DCY_MASK                0x0007  /* AGC_CLIP_DCY - [2:0] */
+#define WM9090_AGC_CLIP_DCY_SHIFT                    0  /* AGC_CLIP_DCY - [2:0] */
+#define WM9090_AGC_CLIP_DCY_WIDTH                    3  /* AGC_CLIP_DCY - [2:0] */
+
+/*
+ * R99 (0x63) - AGC Control 1
+ */
+#define WM9090_AGC_PWR_ENA                      0x8000  /* AGC_PWR_ENA */
+#define WM9090_AGC_PWR_ENA_MASK                 0x8000  /* AGC_PWR_ENA */
+#define WM9090_AGC_PWR_ENA_SHIFT                    15  /* AGC_PWR_ENA */
+#define WM9090_AGC_PWR_ENA_WIDTH                     1  /* AGC_PWR_ENA */
+#define WM9090_AGC_PWR_AVG                      0x1000  /* AGC_PWR_AVG */
+#define WM9090_AGC_PWR_AVG_MASK                 0x1000  /* AGC_PWR_AVG */
+#define WM9090_AGC_PWR_AVG_SHIFT                    12  /* AGC_PWR_AVG */
+#define WM9090_AGC_PWR_AVG_WIDTH                     1  /* AGC_PWR_AVG */
+#define WM9090_AGC_PWR_THR_MASK                 0x0F00  /* AGC_PWR_THR - [11:8] */
+#define WM9090_AGC_PWR_THR_SHIFT                     8  /* AGC_PWR_THR - [11:8] */
+#define WM9090_AGC_PWR_THR_WIDTH                     4  /* AGC_PWR_THR - [11:8] */
+#define WM9090_AGC_PWR_ATK_MASK                 0x0070  /* AGC_PWR_ATK - [6:4] */
+#define WM9090_AGC_PWR_ATK_SHIFT                     4  /* AGC_PWR_ATK - [6:4] */
+#define WM9090_AGC_PWR_ATK_WIDTH                     3  /* AGC_PWR_ATK - [6:4] */
+#define WM9090_AGC_PWR_DCY_MASK                 0x0007  /* AGC_PWR_DCY - [2:0] */
+#define WM9090_AGC_PWR_DCY_SHIFT                     0  /* AGC_PWR_DCY - [2:0] */
+#define WM9090_AGC_PWR_DCY_WIDTH                     3  /* AGC_PWR_DCY - [2:0] */
+
+/*
+ * R100 (0x64) - AGC Control 2
+ */
+#define WM9090_AGC_RAMP                         0x0100  /* AGC_RAMP */
+#define WM9090_AGC_RAMP_MASK                    0x0100  /* AGC_RAMP */
+#define WM9090_AGC_RAMP_SHIFT                        8  /* AGC_RAMP */
+#define WM9090_AGC_RAMP_WIDTH                        1  /* AGC_RAMP */
+#define WM9090_AGC_MINGAIN_MASK                 0x003F  /* AGC_MINGAIN - [5:0] */
+#define WM9090_AGC_MINGAIN_SHIFT                     0  /* AGC_MINGAIN - [5:0] */
+#define WM9090_AGC_MINGAIN_WIDTH                     6  /* AGC_MINGAIN - [5:0] */
+
+#endif
-- 
cgit v1.2.3


From b0b4ce38a535ed3de5ec6fdd4f3c34435a1c1d1e Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert@linux-m68k.org>
Date: Thu, 8 Apr 2010 20:52:00 +0200
Subject: MIPS: TXx9: Add missing MODULE_ALIAS definitions for TXx9 platform
 devices

This enables autoloading of the TXx9 sound driver on RBTX4927.

Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
To: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Cc: Linux MIPS Mailing List <linux-mips@linux-mips.org>
Patchwork: http://patchwork.linux-mips.org/patch/1101/
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
---
 drivers/dma/txx9dmac.c            | 2 ++
 sound/soc/txx9/txx9aclc-ac97.c    | 1 +
 sound/soc/txx9/txx9aclc-generic.c | 1 +
 3 files changed, 4 insertions(+)

(limited to 'sound/soc')

diff --git a/drivers/dma/txx9dmac.c b/drivers/dma/txx9dmac.c
index 3ebc61067e54..75fcf1ac8bb7 100644
--- a/drivers/dma/txx9dmac.c
+++ b/drivers/dma/txx9dmac.c
@@ -1359,3 +1359,5 @@ module_exit(txx9dmac_exit);
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("TXx9 DMA Controller driver");
 MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_ALIAS("platform:txx9dmac");
+MODULE_ALIAS("platform:txx9dmac-chan");
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 612e18b4bf4e..0ec20b68e8cb 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -254,3 +254,4 @@ module_exit(txx9aclc_ac97_exit);
 MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
 MODULE_DESCRIPTION("TXx9 ACLC AC97 driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:txx9aclc-ac97");
diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c
index 3175de9a92cb..95b17f731aec 100644
--- a/sound/soc/txx9/txx9aclc-generic.c
+++ b/sound/soc/txx9/txx9aclc-generic.c
@@ -96,3 +96,4 @@ module_exit(txx9aclc_generic_exit);
 MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
 MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:txx9aclc-generic");
-- 
cgit v1.2.3


From 7b4c734eead5ef0b1c95ec336ddd28e58e648676 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 29 Apr 2010 10:58:08 +0300
Subject: ASoC: TWL4030: AIF/APLL fix in DAPM domain

This patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.

We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */

/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */

/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */

The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).

The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/twl4030.c | 82 +++++++++++++++++++++++++++++++++-------------
 1 file changed, 60 insertions(+), 22 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 2e025a3a2618..12931f6d445b 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -123,6 +123,8 @@ struct twl4030_priv {
 	struct snd_soc_codec codec;
 
 	unsigned int codec_powered;
+
+	/* reference counts of AIF/APLL users */
 	unsigned int apll_enabled;
 
 	struct snd_pcm_substream *master_substream;
@@ -259,22 +261,22 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-	int status;
-
-	if (enable == twl4030->apll_enabled)
-		return;
+	int status = -1;
 
-	if (enable)
-		/* Enable PLL */
-		status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL);
-	else
-		/* Disable PLL */
-		status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL);
+	if (enable) {
+		twl4030->apll_enabled++;
+		if (twl4030->apll_enabled == 1)
+			status = twl4030_codec_enable_resource(
+							TWL4030_CODEC_RES_APLL);
+	} else {
+		twl4030->apll_enabled--;
+		if (!twl4030->apll_enabled)
+			status = twl4030_codec_disable_resource(
+							TWL4030_CODEC_RES_APLL);
+	}
 
 	if (status >= 0)
 		twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
-
-	twl4030->apll_enabled = enable;
 }
 
 static void twl4030_power_up(struct snd_soc_codec *codec)
@@ -672,6 +674,31 @@ static int apll_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int aif_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	u8 audio_if;
+
+	audio_if = twl4030_read_reg_cache(w->codec, TWL4030_REG_AUDIO_IF);
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		/* Enable AIF */
+		/* enable the PLL before we use it to clock the DAI */
+		twl4030_apll_enable(w->codec, 1);
+
+		twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
+						audio_if | TWL4030_AIF_EN);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* disable the DAI before we stop it's source PLL */
+		twl4030_write(w->codec, TWL4030_REG_AUDIO_IF,
+						audio_if &  ~TWL4030_AIF_EN);
+		twl4030_apll_enable(w->codec, 0);
+		break;
+	}
+	return 0;
+}
+
 static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 {
 	struct snd_soc_device *socdev = codec->socdev;
@@ -1180,6 +1207,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HFR"),
 	SND_SOC_DAPM_OUTPUT("VIBRA"),
 
+	/* AIF and APLL clocks for running DAIs (including loopback) */
+	SND_SOC_DAPM_OUTPUT("Virtual HiFi OUT"),
+	SND_SOC_DAPM_INPUT("Virtual HiFi IN"),
+	SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"),
+
 	/* DACs */
 	SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
 			SND_SOC_NOPM, 0, 0),
@@ -1243,7 +1275,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event,
 			    SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
 
-	SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF Enable", SND_SOC_NOPM, 0, 0, aif_event,
+			    SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
 
 	/* Output MIXER controls */
 	/* Earpiece */
@@ -1373,10 +1406,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Digital Voice Playback Mixer", NULL, "DAC Voice"},
 
 	/* Supply for the digital part (APLL) */
-	{"Digital R1 Playback Mixer", NULL, "APLL Enable"},
-	{"Digital L1 Playback Mixer", NULL, "APLL Enable"},
-	{"Digital R2 Playback Mixer", NULL, "APLL Enable"},
-	{"Digital L2 Playback Mixer", NULL, "APLL Enable"},
 	{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
 
 	{"Digital R1 Playback Mixer", NULL, "AIF Enable"},
@@ -1450,6 +1479,14 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Vibra Mux", "AudioR2", "DAC Right2"},
 
 	/* outputs */
+	/* Must be always connected (for AIF and APLL) */
+	{"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"},
+	{"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"},
+	/* Must be always connected (for APLL) */
+	{"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
+	/* Physical outputs */
 	{"OUTL", NULL, "Analog L2 Playback Mixer"},
 	{"OUTR", NULL, "Analog R2 Playback Mixer"},
 	{"EARPIECE", NULL, "Earpiece PGA"},
@@ -1465,6 +1502,12 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"VIBRA", NULL, "Vibra Route"},
 
 	/* Capture path */
+	/* Must be always connected (for AIF and APLL) */
+	{"ADC Virtual Left1", NULL, "Virtual HiFi IN"},
+	{"ADC Virtual Right1", NULL, "Virtual HiFi IN"},
+	{"ADC Virtual Left2", NULL, "Virtual HiFi IN"},
+	{"ADC Virtual Right2", NULL, "Virtual HiFi IN"},
+	/* Physical inputs */
 	{"Analog Left", "Main Mic Capture Switch", "MAINMIC"},
 	{"Analog Left", "Headset Mic Capture Switch", "HSMIC"},
 	{"Analog Left", "AUXL Capture Switch", "AUXL"},
@@ -1497,11 +1540,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
 	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
 
-	{"ADC Virtual Left1", NULL, "APLL Enable"},
-	{"ADC Virtual Right1", NULL, "APLL Enable"},
-	{"ADC Virtual Left2", NULL, "APLL Enable"},
-	{"ADC Virtual Right2", NULL, "APLL Enable"},
-
 	{"ADC Virtual Left1", NULL, "AIF Enable"},
 	{"ADC Virtual Right1", NULL, "AIF Enable"},
 	{"ADC Virtual Left2", NULL, "AIF Enable"},
-- 
cgit v1.2.3


From 1b7c9afbfbfde93d4da89dcebfd2314f7d79c064 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 29 Apr 2010 10:58:09 +0300
Subject: ASoC: TWL4030: Remove OUTL/R outputs

OUTL/R are leftovers from the original driver, and they
are no longer needed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/twl4030.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 12931f6d445b..b717a03dfacf 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1194,8 +1194,6 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("DIGIMIC1"),
 
 	/* Outputs */
-	SND_SOC_DAPM_OUTPUT("OUTL"),
-	SND_SOC_DAPM_OUTPUT("OUTR"),
 	SND_SOC_DAPM_OUTPUT("EARPIECE"),
 	SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
 	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
@@ -1487,8 +1485,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 	/* Must be always connected (for APLL) */
 	{"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
 	/* Physical outputs */
-	{"OUTL", NULL, "Analog L2 Playback Mixer"},
-	{"OUTR", NULL, "Analog R2 Playback Mixer"},
 	{"EARPIECE", NULL, "Earpiece PGA"},
 	{"PREDRIVEL", NULL, "PredriveL PGA"},
 	{"PREDRIVER", NULL, "PredriveR PGA"},
-- 
cgit v1.2.3


From ef909d67299498010f07889bd0980c829ae78990 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 30 Apr 2010 14:59:33 +0300
Subject: ASoC: tlv320dac33: Optimize power up, and restore

On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 106 +++++++++++++++--------------------------
 1 file changed, 39 insertions(+), 67 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 54b2a0508a11..329a97f6e0f8 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -284,45 +284,49 @@ static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg,
 	return ret;
 }
 
-static void dac33_restore_regs(struct snd_soc_codec *codec)
+static void dac33_init_chip(struct snd_soc_codec *codec)
 {
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-	u8 *cache = codec->reg_cache;
-	u8 data[2];
-	int i, ret;
 
-	if (!dac33->chip_power)
+	if (unlikely(!dac33->chip_power))
 		return;
 
-	for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) {
-		data[0] = i;
-		data[1] = cache[i];
-		/* Skip the read only registers */
-		if ((i >= DAC33_INT_OSC_STATUS &&
-				i <= DAC33_INT_OSC_FREQ_RAT_READ_B) ||
-		    (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) ||
-		    i == DAC33_DAC_STATUS_FLAGS ||
-		    i == DAC33_SRC_EST_REF_CLK_RATIO_A ||
-		    i == DAC33_SRC_EST_REF_CLK_RATIO_B)
-			continue;
-		ret = codec->hw_write(codec->control_data, data, 2);
-		if (ret != 2)
-			dev_err(codec->dev, "Write failed (%d)\n", ret);
-	}
-	for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) {
-		data[0] = i;
-		data[1] = cache[i];
-		ret = codec->hw_write(codec->control_data, data, 2);
-		if (ret != 2)
-			dev_err(codec->dev, "Write failed (%d)\n", ret);
-	}
-	for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) {
-		data[0] = i;
-		data[1] = cache[i];
-		ret = codec->hw_write(codec->control_data, data, 2);
-		if (ret != 2)
-			dev_err(codec->dev, "Write failed (%d)\n", ret);
-	}
+	/* 44-46: DAC Control Registers */
+	/* A : DAC sample rate Fsref/1.5 */
+	dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
+	/* B : DAC src=normal, not muted */
+	dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT |
+					     DAC33_DACSRCL_LEFT);
+	/* C : (defaults) */
+	dac33_write(codec, DAC33_DAC_CTRL_C, 0x00);
+
+	/* 64-65 : L&R DAC power control
+	 Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/
+	dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
+	dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
+
+	/* 73 : volume soft stepping control,
+	 clock source = internal osc (?) */
+	dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN);
+
+	/* 66 : LOP/LOM Modes */
+	dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff);
+
+	/* 68 : LOM inverted from LOP */
+	dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2));
+
+	dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB);
+
+	/* Restore only selected registers (gains mostly) */
+	dac33_write(codec, DAC33_LDAC_DIG_VOL_CTRL,
+		    dac33_read_reg_cache(codec, DAC33_LDAC_DIG_VOL_CTRL));
+	dac33_write(codec, DAC33_RDAC_DIG_VOL_CTRL,
+		    dac33_read_reg_cache(codec, DAC33_RDAC_DIG_VOL_CTRL));
+
+	dac33_write(codec, DAC33_LINEL_TO_LLO_VOL,
+		    dac33_read_reg_cache(codec, DAC33_LINEL_TO_LLO_VOL));
+	dac33_write(codec, DAC33_LINER_TO_RLO_VOL,
+		    dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
 }
 
 static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
@@ -358,8 +362,7 @@ static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 
 		dac33->chip_power = 1;
 
-		/* Restore registers */
-		dac33_restore_regs(codec);
+		dac33_init_chip(codec);
 
 		dac33_soft_power(codec, 1);
 	} else {
@@ -1269,35 +1272,6 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static void dac33_init_chip(struct snd_soc_codec *codec)
-{
-	/* 44-46: DAC Control Registers */
-	/* A : DAC sample rate Fsref/1.5 */
-	dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
-	/* B : DAC src=normal, not muted */
-	dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT |
-					     DAC33_DACSRCL_LEFT);
-	/* C : (defaults) */
-	dac33_write(codec, DAC33_DAC_CTRL_C, 0x00);
-
-	/* 64-65 : L&R DAC power control
-	 Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/
-	dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
-	dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
-
-	/* 73 : volume soft stepping control,
-	 clock source = internal osc (?) */
-	dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN);
-
-	/* 66 : LOP/LOM Modes */
-	dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff);
-
-	/* 68 : LOM inverted from LOP */
-	dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2));
-
-	dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB);
-}
-
 static int dac33_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1313,8 +1287,6 @@ static int dac33_soc_probe(struct platform_device *pdev)
 
 	/* Power up the codec */
 	dac33_hard_power(codec, 1);
-	/* Set default configuration */
-	dac33_init_chip(codec);
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-- 
cgit v1.2.3


From 239fe55c7fe17d67403cb1e9222fcaea84248974 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 30 Apr 2010 14:59:34 +0300
Subject: ASoC: tlv320dac33: Revised module loading, and DAC33 ID read

Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 37 ++++++++++++++++++-------------------
 1 file changed, 18 insertions(+), 19 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 329a97f6e0f8..9944721a055c 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -329,6 +329,15 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 		    dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
 }
 
+static inline void dac33_read_id(struct snd_soc_codec *codec)
+{
+	u8 reg;
+
+	dac33_read(codec, DAC33_DEVICE_ID_MSB, &reg);
+	dac33_read(codec, DAC33_DEVICE_ID_LSB, &reg);
+	dac33_read(codec, DAC33_DEVICE_REV_ID, &reg);
+}
+
 static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
 {
 	u8 reg;
@@ -1285,9 +1294,6 @@ static int dac33_soc_probe(struct platform_device *pdev)
 	socdev->card->codec = codec;
 	dac33 = snd_soc_codec_get_drvdata(codec);
 
-	/* Power up the codec */
-	dac33_hard_power(codec, 1);
-
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -1307,9 +1313,6 @@ static int dac33_soc_probe(struct platform_device *pdev)
 	/* power on device */
 	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	/* Bias level configuration has enabled regulator an extra time */
-	regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
-
 	return 0;
 
 pcm_err:
@@ -1459,8 +1462,6 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 			goto error_gpio;
 		}
 		gpio_direction_output(dac33->power_gpio, 0);
-	} else {
-		dac33->chip_power = 1;
 	}
 
 	/* Check if the IRQ number is valid and request it */
@@ -1498,12 +1499,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 		goto err_get;
 	}
 
-	ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies),
-				    dac33->supplies);
+	/* Read the tlv320dac33 ID registers */
+	ret = dac33_hard_power(codec, 1);
 	if (ret != 0) {
-		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
-		goto err_enable;
+		dev_err(codec->dev, "Failed to power up codec: %d\n", ret);
+		goto error_codec;
 	}
+	dac33_read_id(codec);
+	dac33_hard_power(codec, 0);
 
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
@@ -1518,14 +1521,9 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 		goto error_codec;
 	}
 
-	/* Shut down the codec for now */
-	dac33_hard_power(codec, 0);
-
 	return ret;
 
 error_codec:
-	regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
-err_enable:
 	regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies);
 err_get:
 	if (dac33->irq >= 0) {
@@ -1549,14 +1547,15 @@ static int __devexit dac33_i2c_remove(struct i2c_client *client)
 	struct tlv320dac33_priv *dac33;
 
 	dac33 = i2c_get_clientdata(client);
-	dac33_hard_power(&dac33->codec, 0);
+
+	if (unlikely(dac33->chip_power))
+		dac33_hard_power(&dac33->codec, 0);
 
 	if (dac33->power_gpio >= 0)
 		gpio_free(dac33->power_gpio);
 	if (dac33->irq >= 0)
 		free_irq(dac33->irq, &dac33->codec);
 
-	regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
 	regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies);
 
 	destroy_workqueue(dac33->dac33_wq);
-- 
cgit v1.2.3


From 0b61d2b9f2f78fc55faaedcc37f622ffd4103d14 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 30 Apr 2010 14:59:35 +0300
Subject: ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private
 structure

As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.

The substream must be easily available in other places than
pcm_* callbacks.

Manage a pointer in _startup, and _shutdown for this.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 28 ++++++++++++++++++++++++++++
 1 file changed, 28 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 9944721a055c..50d152215abd 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -91,6 +91,7 @@ struct tlv320dac33_priv {
 	struct work_struct work;
 	struct snd_soc_codec codec;
 	struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES];
+	struct snd_pcm_substream *substream;
 	int power_gpio;
 	int chip_power;
 	int irq;
@@ -720,6 +721,31 @@ static void dac33_oscwait(struct snd_soc_codec *codec)
 			"internal oscillator calibration failed\n");
 }
 
+static int dac33_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+	/* Stream started, save the substream pointer */
+	dac33->substream = substream;
+
+	return 0;
+}
+
+static void dac33_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+	dac33->substream = NULL;
+}
+
 static int dac33_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
@@ -1367,6 +1393,8 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33);
 #define DAC33_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
 
 static struct snd_soc_dai_ops dac33_dai_ops = {
+	.startup	= dac33_startup,
+	.shutdown	= dac33_shutdown,
 	.hw_params	= dac33_hw_params,
 	.prepare	= dac33_pcm_prepare,
 	.trigger	= dac33_pcm_trigger,
-- 
cgit v1.2.3


From ad05c03b1c4c1fb4db066a7bd502b674148ccd89 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 30 Apr 2010 14:59:36 +0300
Subject: ASoC: tlv320dac33: Support for turning off the codec

Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).

After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).

The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.

There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
   We need to power on the codec, and do the chip init, but we does not
   need to execute the playback related configuration
2. Playback caused  BIAS_OFF -> BIAS_ON
   We need to power on the codec, and do the chip init, and also we need
   to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
   We need to execute the playback related configuration. The codec is
   already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
   Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
   We need to execute the playback related configuration. The codec is
   still on.

Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.

Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 66 ++++++++++++++++++++++++++++--------------
 1 file changed, 45 insertions(+), 21 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 50d152215abd..68b7ccbf2e7c 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -61,6 +61,8 @@
 #define US_TO_SAMPLES(rate, us) \
 	(rate / (1000000 / us))
 
+static void dac33_calculate_times(struct snd_pcm_substream *substream);
+static int dac33_prepare_chip(struct snd_pcm_substream *substream);
 
 static struct snd_soc_codec *tlv320dac33_codec;
 
@@ -355,9 +357,17 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
 static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 {
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-	int ret;
+	int ret = 0;
 
 	mutex_lock(&dac33->mutex);
+
+	/* Safety check */
+	if (unlikely(power == dac33->chip_power)) {
+		dev_warn(codec->dev, "Trying to set the same power state: %s\n",
+			power ? "ON" : "OFF");
+		goto exit;
+	}
+
 	if (power) {
 		ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies),
 					  dac33->supplies);
@@ -371,10 +381,6 @@ static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 			gpio_set_value(dac33->power_gpio, 1);
 
 		dac33->chip_power = 1;
-
-		dac33_init_chip(codec);
-
-		dac33_soft_power(codec, 1);
 	} else {
 		dac33_soft_power(codec, 0);
 		if (dac33->power_gpio >= 0)
@@ -396,6 +402,22 @@ exit:
 	return ret;
 }
 
+static int playback_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(w->codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (likely(dac33->substream)) {
+			dac33_calculate_times(dac33->substream);
+			dac33_prepare_chip(dac33->substream);
+		}
+		break;
+	}
+	return 0;
+}
+
 static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
@@ -525,6 +547,8 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
 			 DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0),
 	SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power",
 			 DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0),
+
+	SND_SOC_DAPM_PRE("Prepare Playback", playback_event),
 };
 
 static const struct snd_soc_dapm_route audio_map[] = {
@@ -567,18 +591,18 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Coming from OFF, switch on the codec */
 			ret = dac33_hard_power(codec, 1);
 			if (ret != 0)
 				return ret;
-		}
 
-		dac33_soft_power(codec, 0);
+			dac33_init_chip(codec);
+		}
 		break;
 	case SND_SOC_BIAS_OFF:
 		ret = dac33_hard_power(codec, 0);
 		if (ret != 0)
 			return ret;
-
 		break;
 	}
 	codec->bias_level = level;
@@ -829,6 +853,16 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 	}
 
 	mutex_lock(&dac33->mutex);
+
+	if (!dac33->chip_power) {
+		/*
+		 * Chip is not powered yet.
+		 * Do the init in the dac33_set_bias_level later.
+		 */
+		mutex_unlock(&dac33->mutex);
+		return 0;
+	}
+
 	dac33_soft_power(codec, 0);
 	dac33_soft_power(codec, 1);
 
@@ -1035,15 +1069,6 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 
 }
 
-static int dac33_pcm_prepare(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	dac33_calculate_times(substream);
-	dac33_prepare_chip(substream);
-
-	return 0;
-}
-
 static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 			     struct snd_soc_dai *dai)
 {
@@ -1336,9 +1361,6 @@ static int dac33_soc_probe(struct platform_device *pdev)
 
 	dac33_add_widgets(codec);
 
-	/* power on device */
-	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 
 pcm_err:
@@ -1375,6 +1397,8 @@ static int dac33_soc_resume(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->card->codec;
 
 	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		dac33_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
 	dac33_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
@@ -1396,7 +1420,6 @@ static struct snd_soc_dai_ops dac33_dai_ops = {
 	.startup	= dac33_startup,
 	.shutdown	= dac33_shutdown,
 	.hw_params	= dac33_hw_params,
-	.prepare	= dac33_pcm_prepare,
 	.trigger	= dac33_pcm_trigger,
 	.delay		= dac33_dai_delay,
 	.set_sysclk	= dac33_set_dai_sysclk,
@@ -1450,6 +1473,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 	codec->hw_write = (hw_write_t) i2c_master_send;
 	codec->bias_level = SND_SOC_BIAS_OFF;
 	codec->set_bias_level = dac33_set_bias_level;
+	codec->idle_bias_off = 1;
 	codec->dai = &dac33_dai;
 	codec->num_dai = 1;
 	codec->reg_cache_size = ARRAY_SIZE(dac33_reg);
-- 
cgit v1.2.3


From e5e5b31e8c729b6bae569bec0790c655ee0121a1 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Tue, 4 May 2010 11:08:18 +0300
Subject: ASoC: tpa6130a2: TLV mapping for tpa6140a2

Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.

Provide different mixer control for the chips with correct
TLV mapping.

User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140

The way machine drivers are using this amplifier remained
the same.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tpa6130a2.c | 31 ++++++++++++++++++++++++++++---
 1 file changed, 28 insertions(+), 3 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 958d49c969ac..0cf3e3862e7b 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -53,6 +53,7 @@ struct tpa6130a2_data {
 	struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES];
 	int power_gpio;
 	unsigned char power_state;
+	enum tpa_model id;
 };
 
 static int tpa6130a2_i2c_read(int reg)
@@ -263,6 +264,20 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = {
 		       tpa6130_tlv),
 };
 
+static const unsigned int tpa6140_tlv[] = {
+	TLV_DB_RANGE_HEAD(3),
+	0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0),
+	9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0),
+	17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0),
+};
+
+static const struct snd_kcontrol_new tpa6140a2_controls[] = {
+	SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume",
+		       TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0,
+		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+		       tpa6140_tlv),
+};
+
 /*
  * Enable or disable channel (left or right)
  * The bit number for mute and amplifier are the same per channel:
@@ -368,13 +383,22 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 int tpa6130a2_add_controls(struct snd_soc_codec *codec)
 {
+	struct	tpa6130a2_data *data;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
 	snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets,
 				ARRAY_SIZE(tpa6130a2_dapm_widgets));
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	return snd_soc_add_controls(codec, tpa6130a2_controls,
-				ARRAY_SIZE(tpa6130a2_controls));
+	if (data->id == TPA6140A2)
+		return snd_soc_add_controls(codec, tpa6140a2_controls,
+						ARRAY_SIZE(tpa6140a2_controls));
+	else
+		return snd_soc_add_controls(codec, tpa6130a2_controls,
+						ARRAY_SIZE(tpa6130a2_controls));
 
 }
 EXPORT_SYMBOL_GPL(tpa6130a2_add_controls);
@@ -407,6 +431,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 
 	pdata = client->dev.platform_data;
 	data->power_gpio = pdata->power_gpio;
+	data->id = pdata->id;
 
 	mutex_init(&data->mutex);
 
@@ -425,7 +450,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 		gpio_direction_output(data->power_gpio, 0);
 	}
 
-	switch (pdata->id) {
+	switch (data->id) {
 	case TPA6130A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
-- 
cgit v1.2.3


From 985d8c4c9eb18bd7b295c73a5ab1fc990e5ed182 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 3 May 2010 16:25:52 +0100
Subject: ASoC: Add debug output tracing all cache register writes

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/soc-cache.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 9dfe9a58a314..472af38188c1 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -44,6 +44,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	ret = codec->hw_write(codec->control_data, data, 2);
 	if (ret == 2)
 		return 0;
@@ -112,6 +114,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	ret = codec->hw_write(codec->control_data, data, 2);
 	if (ret == 2)
 		return 0;
@@ -171,6 +175,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	if (codec->hw_write(codec->control_data, data, 2) == 2)
 		return 0;
 	else
@@ -205,6 +211,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	if (codec->hw_write(codec->control_data, data, 3) == 3)
 		return 0;
 	else
@@ -362,6 +370,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	ret = codec->hw_write(codec->control_data, data, 3);
 	if (ret == 3)
 		return 0;
@@ -472,6 +482,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	ret = codec->hw_write(codec->control_data, data, 4);
 	if (ret == 4)
 		return 0;
-- 
cgit v1.2.3


From d07e7ce9b6fd1a4ff0a6b40c74fbecdc584a135b Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:55:21 +0900
Subject: ASoC: S3C: I2Sv2: Move defines closer to driver

The header for I2Sv2
   linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
contains only controller specific definitions and nothing
SoC specific. So, it could be moved to sound/soc/s3c24xx/

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 .../plat-samsung/include/plat/regs-s3c2412-iis.h   | 82 ----------------------
 sound/soc/s3c24xx/regs-i2s-v2.h                    | 82 ++++++++++++++++++++++
 sound/soc/s3c24xx/s3c-i2s-v2.c                     |  3 +-
 sound/soc/s3c24xx/s3c2412-i2s.c                    |  3 +-
 sound/soc/s3c24xx/s3c64xx-i2s.c                    |  2 +-
 5 files changed, 85 insertions(+), 87 deletions(-)
 delete mode 100644 arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h
 create mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h

(limited to 'sound/soc')

diff --git a/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h
deleted file mode 100644
index abf2fbc2eb2f..000000000000
--- a/arch/arm/plat-samsung/include/plat/regs-s3c2412-iis.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h
- *
- * Copyright 2007 Simtec Electronics <linux@simtec.co.uk>
- *	http://armlinux.simtec.co.uk/
- *
- * 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.
- *
- * S3C2412 IIS register definition
-*/
-
-#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H
-#define __ASM_ARCH_REGS_S3C2412_IIS_H
-
-#define S3C2412_IISCON			(0x00)
-#define S3C2412_IISMOD			(0x04)
-#define S3C2412_IISFIC			(0x08)
-#define S3C2412_IISPSR			(0x0C)
-#define S3C2412_IISTXD			(0x10)
-#define S3C2412_IISRXD			(0x14)
-
-#define S3C2412_IISCON_LRINDEX		(1 << 11)
-#define S3C2412_IISCON_TXFIFO_EMPTY	(1 << 10)
-#define S3C2412_IISCON_RXFIFO_EMPTY	(1 << 9)
-#define S3C2412_IISCON_TXFIFO_FULL	(1 << 8)
-#define S3C2412_IISCON_RXFIFO_FULL	(1 << 7)
-#define S3C2412_IISCON_TXDMA_PAUSE	(1 << 6)
-#define S3C2412_IISCON_RXDMA_PAUSE	(1 << 5)
-#define S3C2412_IISCON_TXCH_PAUSE	(1 << 4)
-#define S3C2412_IISCON_RXCH_PAUSE	(1 << 3)
-#define S3C2412_IISCON_TXDMA_ACTIVE	(1 << 2)
-#define S3C2412_IISCON_RXDMA_ACTIVE	(1 << 1)
-#define S3C2412_IISCON_IIS_ACTIVE	(1 << 0)
-
-#define S3C64XX_IISMOD_BLC_16BIT	(0 << 13)
-#define S3C64XX_IISMOD_BLC_8BIT		(1 << 13)
-#define S3C64XX_IISMOD_BLC_24BIT	(2 << 13)
-#define S3C64XX_IISMOD_BLC_MASK		(3 << 13)
-
-#define S3C64XX_IISMOD_IMS_PCLK		(0 << 10)
-#define S3C64XX_IISMOD_IMS_SYSMUX	(1 << 10)
-
-#define S3C2412_IISMOD_MASTER_INTERNAL	(0 << 10)
-#define S3C2412_IISMOD_MASTER_EXTERNAL	(1 << 10)
-#define S3C2412_IISMOD_SLAVE		(2 << 10)
-#define S3C2412_IISMOD_MASTER_MASK	(3 << 10)
-#define S3C2412_IISMOD_MODE_TXONLY	(0 << 8)
-#define S3C2412_IISMOD_MODE_RXONLY	(1 << 8)
-#define S3C2412_IISMOD_MODE_TXRX	(2 << 8)
-#define S3C2412_IISMOD_MODE_MASK	(3 << 8)
-#define S3C2412_IISMOD_LR_LLOW		(0 << 7)
-#define S3C2412_IISMOD_LR_RLOW		(1 << 7)
-#define S3C2412_IISMOD_SDF_IIS		(0 << 5)
-#define S3C2412_IISMOD_SDF_MSB		(1 << 5)
-#define S3C2412_IISMOD_SDF_LSB		(2 << 5)
-#define S3C2412_IISMOD_SDF_MASK		(3 << 5)
-#define S3C2412_IISMOD_RCLK_256FS	(0 << 3)
-#define S3C2412_IISMOD_RCLK_512FS	(1 << 3)
-#define S3C2412_IISMOD_RCLK_384FS	(2 << 3)
-#define S3C2412_IISMOD_RCLK_768FS	(3 << 3)
-#define S3C2412_IISMOD_RCLK_MASK 	(3 << 3)
-#define S3C2412_IISMOD_BCLK_32FS	(0 << 1)
-#define S3C2412_IISMOD_BCLK_48FS	(1 << 1)
-#define S3C2412_IISMOD_BCLK_16FS	(2 << 1)
-#define S3C2412_IISMOD_BCLK_24FS	(3 << 1)
-#define S3C2412_IISMOD_BCLK_MASK	(3 << 1)
-#define S3C2412_IISMOD_8BIT		(1 << 0)
-
-#define S3C64XX_IISMOD_CDCLKCON		(1 << 12)
-
-#define S3C2412_IISPSR_PSREN		(1 << 15)
-
-#define S3C2412_IISFIC_TXFLUSH		(1 << 15)
-#define S3C2412_IISFIC_RXFLUSH		(1 << 7)
-#define S3C2412_IISFIC_TXCOUNT(x)	(((x) >>  8) & 0xf)
-#define S3C2412_IISFIC_RXCOUNT(x)	(((x) >>  0) & 0xf)
-
-
-
-#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */
-
diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h
new file mode 100644
index 000000000000..abf2fbc2eb2f
--- /dev/null
+++ b/sound/soc/s3c24xx/regs-i2s-v2.h
@@ -0,0 +1,82 @@
+/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h
+ *
+ * Copyright 2007 Simtec Electronics <linux@simtec.co.uk>
+ *	http://armlinux.simtec.co.uk/
+ *
+ * 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.
+ *
+ * S3C2412 IIS register definition
+*/
+
+#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H
+#define __ASM_ARCH_REGS_S3C2412_IIS_H
+
+#define S3C2412_IISCON			(0x00)
+#define S3C2412_IISMOD			(0x04)
+#define S3C2412_IISFIC			(0x08)
+#define S3C2412_IISPSR			(0x0C)
+#define S3C2412_IISTXD			(0x10)
+#define S3C2412_IISRXD			(0x14)
+
+#define S3C2412_IISCON_LRINDEX		(1 << 11)
+#define S3C2412_IISCON_TXFIFO_EMPTY	(1 << 10)
+#define S3C2412_IISCON_RXFIFO_EMPTY	(1 << 9)
+#define S3C2412_IISCON_TXFIFO_FULL	(1 << 8)
+#define S3C2412_IISCON_RXFIFO_FULL	(1 << 7)
+#define S3C2412_IISCON_TXDMA_PAUSE	(1 << 6)
+#define S3C2412_IISCON_RXDMA_PAUSE	(1 << 5)
+#define S3C2412_IISCON_TXCH_PAUSE	(1 << 4)
+#define S3C2412_IISCON_RXCH_PAUSE	(1 << 3)
+#define S3C2412_IISCON_TXDMA_ACTIVE	(1 << 2)
+#define S3C2412_IISCON_RXDMA_ACTIVE	(1 << 1)
+#define S3C2412_IISCON_IIS_ACTIVE	(1 << 0)
+
+#define S3C64XX_IISMOD_BLC_16BIT	(0 << 13)
+#define S3C64XX_IISMOD_BLC_8BIT		(1 << 13)
+#define S3C64XX_IISMOD_BLC_24BIT	(2 << 13)
+#define S3C64XX_IISMOD_BLC_MASK		(3 << 13)
+
+#define S3C64XX_IISMOD_IMS_PCLK		(0 << 10)
+#define S3C64XX_IISMOD_IMS_SYSMUX	(1 << 10)
+
+#define S3C2412_IISMOD_MASTER_INTERNAL	(0 << 10)
+#define S3C2412_IISMOD_MASTER_EXTERNAL	(1 << 10)
+#define S3C2412_IISMOD_SLAVE		(2 << 10)
+#define S3C2412_IISMOD_MASTER_MASK	(3 << 10)
+#define S3C2412_IISMOD_MODE_TXONLY	(0 << 8)
+#define S3C2412_IISMOD_MODE_RXONLY	(1 << 8)
+#define S3C2412_IISMOD_MODE_TXRX	(2 << 8)
+#define S3C2412_IISMOD_MODE_MASK	(3 << 8)
+#define S3C2412_IISMOD_LR_LLOW		(0 << 7)
+#define S3C2412_IISMOD_LR_RLOW		(1 << 7)
+#define S3C2412_IISMOD_SDF_IIS		(0 << 5)
+#define S3C2412_IISMOD_SDF_MSB		(1 << 5)
+#define S3C2412_IISMOD_SDF_LSB		(2 << 5)
+#define S3C2412_IISMOD_SDF_MASK		(3 << 5)
+#define S3C2412_IISMOD_RCLK_256FS	(0 << 3)
+#define S3C2412_IISMOD_RCLK_512FS	(1 << 3)
+#define S3C2412_IISMOD_RCLK_384FS	(2 << 3)
+#define S3C2412_IISMOD_RCLK_768FS	(3 << 3)
+#define S3C2412_IISMOD_RCLK_MASK 	(3 << 3)
+#define S3C2412_IISMOD_BCLK_32FS	(0 << 1)
+#define S3C2412_IISMOD_BCLK_48FS	(1 << 1)
+#define S3C2412_IISMOD_BCLK_16FS	(2 << 1)
+#define S3C2412_IISMOD_BCLK_24FS	(3 << 1)
+#define S3C2412_IISMOD_BCLK_MASK	(3 << 1)
+#define S3C2412_IISMOD_8BIT		(1 << 0)
+
+#define S3C64XX_IISMOD_CDCLKCON		(1 << 12)
+
+#define S3C2412_IISPSR_PSREN		(1 << 15)
+
+#define S3C2412_IISFIC_TXFLUSH		(1 << 15)
+#define S3C2412_IISFIC_RXFLUSH		(1 << 7)
+#define S3C2412_IISFIC_TXCOUNT(x)	(((x) >>  8) & 0xf)
+#define S3C2412_IISFIC_RXCOUNT(x)	(((x) >>  0) & 0xf)
+
+
+
+#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */
+
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 865f93143bf1..cb3652ed057d 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -24,10 +24,9 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <plat/regs-s3c2412-iis.h>
-
 #include <mach/dma.h>
 
+#include "regs-i2s-v2.h"
 #include "s3c-i2s-v2.h"
 #include "s3c-dma.h"
 
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index f3148f98b419..7134465693e3 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -32,12 +32,11 @@
 #include <sound/soc.h>
 #include <mach/hardware.h>
 
-#include <plat/regs-s3c2412-iis.h>
-
 #include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
 #include "s3c-dma.h"
+#include "regs-i2s-v2.h"
 #include "s3c2412-i2s.h"
 
 #define S3C2412_I2S_DEBUG 0
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index ab1fa159d3ae..b7e951fe30a6 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -18,7 +18,6 @@
 
 #include <sound/soc.h>
 
-#include <plat/regs-s3c2412-iis.h>
 #include <mach/gpio-bank-d.h>
 #include <mach/gpio-bank-e.h>
 #include <plat/gpio-cfg.h>
@@ -27,6 +26,7 @@
 #include <mach/dma.h>
 
 #include "s3c-dma.h"
+#include "regs-i2s-v2.h"
 #include "s3c64xx-i2s.h"
 
 /* The value should be set to maximum of the total number
-- 
cgit v1.2.3


From b720d56294187ea4167a20e59c4ea0bb4f064132 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:03 +0900
Subject: ASoC: SAMSUNG: I2S: Add bit definitions

Define more bit definitions in the order of mainline
support for the SoC.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/regs-i2s-v2.h | 42 +++++++++++++++++++++++++++++++++++++++--
 1 file changed, 40 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h
index abf2fbc2eb2f..f3d90289e7be 100644
--- a/sound/soc/s3c24xx/regs-i2s-v2.h
+++ b/sound/soc/s3c24xx/regs-i2s-v2.h
@@ -20,6 +20,24 @@
 #define S3C2412_IISTXD			(0x10)
 #define S3C2412_IISRXD			(0x14)
 
+#define S5PC1XX_IISFICS		0x18
+#define S5PC1XX_IISTXDS		0x1C
+
+#define S5PC1XX_IISCON_SW_RST		(1 << 31)
+#define S5PC1XX_IISCON_FRXOFSTATUS	(1 << 26)
+#define S5PC1XX_IISCON_FRXORINTEN	(1 << 25)
+#define S5PC1XX_IISCON_FTXSURSTAT	(1 << 24)
+#define S5PC1XX_IISCON_FTXSURINTEN	(1 << 23)
+#define S5PC1XX_IISCON_TXSDMAPAUSE	(1 << 20)
+#define S5PC1XX_IISCON_TXSDMACTIVE	(1 << 18)
+
+#define S3C64XX_IISCON_FTXURSTATUS	(1 << 17)
+#define S3C64XX_IISCON_FTXURINTEN	(1 << 16)
+#define S3C64XX_IISCON_TXFIFO2_EMPTY	(1 << 15)
+#define S3C64XX_IISCON_TXFIFO1_EMPTY	(1 << 14)
+#define S3C64XX_IISCON_TXFIFO2_FULL	(1 << 13)
+#define S3C64XX_IISCON_TXFIFO1_FULL	(1 << 12)
+
 #define S3C2412_IISCON_LRINDEX		(1 << 11)
 #define S3C2412_IISCON_TXFIFO_EMPTY	(1 << 10)
 #define S3C2412_IISCON_RXFIFO_EMPTY	(1 << 9)
@@ -33,6 +51,23 @@
 #define S3C2412_IISCON_RXDMA_ACTIVE	(1 << 1)
 #define S3C2412_IISCON_IIS_ACTIVE	(1 << 0)
 
+#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT	(0 << 30)
+#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN	(1 << 30)
+#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT	(2 << 30)
+#define S5PC1XX_IISMOD_OPCLK_PCLK	(3 << 30)
+#define S5PC1XX_IISMOD_OPCLK_MASK	(3 << 30)
+#define S5PC1XX_IISMOD_TXS_IDMA		(1 << 28) /* Sec_TXFIFO use I-DMA */
+#define S5PC1XX_IISMOD_BLCS_MASK	0x3
+#define S5PC1XX_IISMOD_BLCS_SHIFT	26
+#define S5PC1XX_IISMOD_BLCP_MASK	0x3
+#define S5PC1XX_IISMOD_BLCP_SHIFT	24
+
+#define S3C64XX_IISMOD_C2DD_HHALF	(1 << 21) /* Discard Higher-half */
+#define S3C64XX_IISMOD_C2DD_LHALF	(1 << 20) /* Discard Lower-half */
+#define S3C64XX_IISMOD_C1DD_HHALF	(1 << 19)
+#define S3C64XX_IISMOD_C1DD_LHALF	(1 << 18)
+#define S3C64XX_IISMOD_DC2_EN		(1 << 17)
+#define S3C64XX_IISMOD_DC1_EN		(1 << 16)
 #define S3C64XX_IISMOD_BLC_16BIT	(0 << 13)
 #define S3C64XX_IISMOD_BLC_8BIT		(1 << 13)
 #define S3C64XX_IISMOD_BLC_24BIT	(2 << 13)
@@ -71,12 +106,15 @@
 
 #define S3C2412_IISPSR_PSREN		(1 << 15)
 
+#define S3C64XX_IISFIC_TX2COUNT(x)	(((x) >>  24) & 0xf)
+#define S3C64XX_IISFIC_TX1COUNT(x)	(((x) >>  16) & 0xf)
+
 #define S3C2412_IISFIC_TXFLUSH		(1 << 15)
 #define S3C2412_IISFIC_RXFLUSH		(1 << 7)
 #define S3C2412_IISFIC_TXCOUNT(x)	(((x) >>  8) & 0xf)
 #define S3C2412_IISFIC_RXCOUNT(x)	(((x) >>  0) & 0xf)
 
-
+#define S5PC1XX_IISFICS_TXFLUSH		(1 << 15)
+#define S5PC1XX_IISFICS_TXCOUNT(x)	(((x) >>  8) & 0x7f)
 
 #endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */
-
-- 
cgit v1.2.3


From ce76f9fd34d4d39fbda330f14b327273551e2f54 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:27 +0900
Subject: ASoC: S3C2412: I2S: Debug IMS field

The IMS field of s3c2412/13 is essentially the same as that of s3c64xx.
That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit
selects source clock for signal generation.
For that reason, remove improper defines for IISMOD[11:10] field mask
and define two 1bit fields that can be set independent of each other.
As a consequence, corresponding fields for PLAT_S3C64XX too get to use
these new defines.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/regs-i2s-v2.h |  6 ++----
 sound/soc/s3c24xx/s3c-i2s-v2.c  | 25 ++-----------------------
 sound/soc/s3c24xx/s3c2412-i2s.c |  8 ++------
 3 files changed, 6 insertions(+), 33 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h
index f3d90289e7be..484858a5f33b 100644
--- a/sound/soc/s3c24xx/regs-i2s-v2.h
+++ b/sound/soc/s3c24xx/regs-i2s-v2.h
@@ -76,10 +76,8 @@
 #define S3C64XX_IISMOD_IMS_PCLK		(0 << 10)
 #define S3C64XX_IISMOD_IMS_SYSMUX	(1 << 10)
 
-#define S3C2412_IISMOD_MASTER_INTERNAL	(0 << 10)
-#define S3C2412_IISMOD_MASTER_EXTERNAL	(1 << 10)
-#define S3C2412_IISMOD_SLAVE		(2 << 10)
-#define S3C2412_IISMOD_MASTER_MASK	(3 << 10)
+#define S3C2412_IISMOD_IMS_SYSMUX	(1 << 10)
+#define S3C2412_IISMOD_SLAVE		(1 << 11)
 #define S3C2412_IISMOD_MODE_TXONLY	(0 << 8)
 #define S3C2412_IISMOD_MODE_RXONLY	(1 << 8)
 #define S3C2412_IISMOD_MODE_TXRX	(2 << 8)
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index cb3652ed057d..b8803a182cc1 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -265,35 +265,14 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 	iismod = readl(i2s->regs + S3C2412_IISMOD);
 	pr_debug("hw_params r: IISMOD: %x \n", iismod);
 
-#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
-#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK
-#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE
-#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL
-#endif
-
-#if defined(CONFIG_PLAT_S3C64XX)
-/* From Rev1.1 datasheet, we have two master and two slave modes:
- * IMS[11:10]:
- *	00 = master mode, fed from PCLK
- *	01 = master mode, fed from CLKAUDIO
- *	10 = slave mode, using PCLK
- *	11 = slave mode, using I2SCLK
- */
-#define IISMOD_MASTER_MASK (1 << 11)
-#define IISMOD_SLAVE (1 << 11)
-#define IISMOD_MASTER (0 << 11)
-#endif
-
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
 		i2s->master = 0;
-		iismod &= ~IISMOD_MASTER_MASK;
-		iismod |= IISMOD_SLAVE;
+		iismod |= S3C2412_IISMOD_SLAVE;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		i2s->master = 1;
-		iismod &= ~IISMOD_MASTER_MASK;
-		iismod |= IISMOD_MASTER;
+		iismod &= ~S3C2412_IISMOD_SLAVE;
 		break;
 	default:
 		pr_err("unknwon master/slave format\n");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 7134465693e3..cf0c15482c34 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -78,14 +78,10 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 
 	switch (clk_id) {
 	case S3C2412_CLKSRC_PCLK:
-		s3c2412_i2s.master = 1;
-		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
-		iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
+		iismod &= ~S3C2412_IISMOD_IMS_SYSMUX;
 		break;
 	case S3C2412_CLKSRC_I2SCLK:
-		s3c2412_i2s.master = 0;
-		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
-		iismod |= S3C2412_IISMOD_MASTER_EXTERNAL;
+		iismod |= S3C2412_IISMOD_IMS_SYSMUX;
 		break;
 	default:
 		return -EINVAL;
-- 
cgit v1.2.3


From d79696ff443eb16a1a13bd7db030a25fee9f44a7 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:34 +0900
Subject: ASoC: S3C2412: I2S: Return correct source clock

Until now, s3c2412_get_iisclk would return NULL since iis_clk was never
initialized.
Return appropriate pointer as per the selection made for source clock.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c2412-i2s.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index cf0c15482c34..ce2daacc4eb9 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -94,7 +94,13 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 
 struct clk *s3c2412_get_iisclk(void)
 {
-	return s3c2412_i2s.iis_clk;
+	struct s3c_i2sv2_info *i2s = &s3c2412_i2s;
+	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+	if (iismod & S3C2412_IISMOD_IMS_SYSMUX)
+		return i2s->iis_cclk;
+	else
+		return i2s->iis_pclk;
 }
 EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
 
-- 
cgit v1.2.3


From 21a7ad08e274ce7b669b165d65e0bab8391552a6 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:39 +0900
Subject: ASoC: S3C: I2Sv2: Discard redundant field iis_clk

No need to keep redundant field iis_clk in s3c_i2sv2_info.
iis_cclk and iis_pclk is all we need.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.h | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
index b094d3c23cbe..ea56467f5adc 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.h
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -49,7 +49,6 @@ struct s3c_i2sv2_info {
 
 	struct clk	*iis_pclk;
 	struct clk	*iis_cclk;
-	struct clk	*iis_clk;
 
 	unsigned char	 master;
 
-- 
cgit v1.2.3


From 5728242789d3538abcfdcb28e01798801efd9988 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:45 +0900
Subject: ASoC: S3C: I2Sv2: Unify i2s_get_clock callback

Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock
doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock
and discard other two copies. Also, switch the users to make calls to the
newly defined and generic s3c_i2sv2_get_clock

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/jive_wm8750.c |  2 +-
 sound/soc/s3c24xx/s3c-i2s-v2.c  | 12 ++++++++++++
 sound/soc/s3c24xx/s3c-i2s-v2.h  |  2 ++
 sound/soc/s3c24xx/s3c2412-i2s.c | 13 -------------
 sound/soc/s3c24xx/s3c2412-i2s.h |  2 --
 sound/soc/s3c24xx/s3c64xx-i2s.c | 12 ------------
 sound/soc/s3c24xx/s3c64xx-i2s.h |  2 --
 7 files changed, 15 insertions(+), 30 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 97d8ff3196be..8c108b121c10 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -70,7 +70,7 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
-				s3c2412_get_iisclk());
+				s3c_i2sv2_get_clock(cpu_dai));
 
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index b8803a182cc1..ac153ca9c628 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -516,6 +516,18 @@ static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream,
 	return delay;
 }
 
+struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+	if (iismod & S3C2412_IISMOD_IMS_SYSMUX)
+		return i2s->iis_cclk;
+	else
+		return i2s->iis_pclk;
+}
+EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock);
+
 /* default table of all avaialable root fs divisors */
 static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
 
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
index ea56467f5adc..39a6db6a565c 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.h
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -60,6 +60,8 @@ struct s3c_i2sv2_info {
 	u32		 suspend_iispsr;
 };
 
+extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai);
+
 struct s3c_i2sv2_rate_calc {
 	unsigned int	clk_div;	/* for prescaler */
 	unsigned int	fs_div;		/* for root frame clock */
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ce2daacc4eb9..6b884d75235e 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -91,19 +91,6 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-
-struct clk *s3c2412_get_iisclk(void)
-{
-	struct s3c_i2sv2_info *i2s = &s3c2412_i2s;
-	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-	if (iismod & S3C2412_IISMOD_IMS_SYSMUX)
-		return i2s->iis_cclk;
-	else
-		return i2s->iis_pclk;
-}
-EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
-
 static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
 {
 	return cpu_dai->private_data;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index 60cac002a830..0b5686b4d5c3 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -24,8 +24,6 @@
 #define S3C2412_CLKSRC_PCLK	S3C_I2SV2_CLKSRC_PCLK
 #define S3C2412_CLKSRC_I2SCLK	S3C_I2SV2_CLKSRC_AUDIOBUS
 
-extern struct clk *s3c2412_get_iisclk(void);
-
 extern struct snd_soc_dai s3c2412_i2s_dai;
 
 #endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index b7e951fe30a6..97327ef09de6 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -91,18 +91,6 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-	if (iismod & S3C64XX_IISMOD_IMS_SYSMUX)
-		return i2s->iis_cclk;
-	else
-		return i2s->iis_pclk;
-}
-EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
-
 static int s3c64xx_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index 53d2a0a0df36..f27ed50e4d82 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -38,6 +38,4 @@ struct clk;
 
 extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
-extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai);
-
 #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
-- 
cgit v1.2.3


From d47ef9c79dd46c1b33696f01b6cf744b77150379 Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:51 +0900
Subject: ASoC: S3C64XX: I2S: Use s3c2412 defines

Now that the fields are defined for s3c2412, use them and avoid having
multiple copies of same defines.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/regs-i2s-v2.h | 3 ---
 sound/soc/s3c24xx/s3c64xx-i2s.c | 4 ++--
 2 files changed, 2 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h
index 484858a5f33b..5e5e5680580b 100644
--- a/sound/soc/s3c24xx/regs-i2s-v2.h
+++ b/sound/soc/s3c24xx/regs-i2s-v2.h
@@ -73,9 +73,6 @@
 #define S3C64XX_IISMOD_BLC_24BIT	(2 << 13)
 #define S3C64XX_IISMOD_BLC_MASK		(3 << 13)
 
-#define S3C64XX_IISMOD_IMS_PCLK		(0 << 10)
-#define S3C64XX_IISMOD_IMS_SYSMUX	(1 << 10)
-
 #define S3C2412_IISMOD_IMS_SYSMUX	(1 << 10)
 #define S3C2412_IISMOD_SLAVE		(1 << 11)
 #define S3C2412_IISMOD_MODE_TXONLY	(0 << 8)
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 97327ef09de6..6d49bd93717b 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -62,11 +62,11 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 
 	switch (clk_id) {
 	case S3C64XX_CLKSRC_PCLK:
-		iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX;
+		iismod &= ~S3C2412_IISMOD_IMS_SYSMUX;
 		break;
 
 	case S3C64XX_CLKSRC_MUX:
-		iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
+		iismod |= S3C2412_IISMOD_IMS_SYSMUX;
 		break;
 
 	case S3C64XX_CLKSRC_CDCLK:
-- 
cgit v1.2.3


From 9e991a4bf3290b0787d33fa72144f6eec656f20d Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:56:56 +0900
Subject: ASoC: S3C: I2Sv2: New field for controller feature

In order to make s3c-i2s-v2.c manage controllers with minor
quirks and variation in features, we define a per-block flag
that indicates the availability/lack of a particular feature
to the s3c-i2s-v2.c

While adding support for new SoCs' I2S, check for the blocks
of older SoCs that have similar feature and set the flag for
that feature.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.h | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
index 39a6db6a565c..766f43a13d8b 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.h
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -29,10 +29,16 @@
 #define S3C_I2SV2_CLKSRC_AUDIOBUS	1
 #define S3C_I2SV2_CLKSRC_CDCLK		2
 
+/* Set this flag for I2S controllers that have the bit IISMOD[12]
+ * bridge/break RCLK signal and external Xi2sCDCLK pin.
+ */
+#define S3C_FEATURE_CDCLKCON	(1 << 0)
+
 /**
  * struct s3c_i2sv2_info - S3C I2S-V2 information
  * @dev: The parent device passed to use from the probe.
  * @regs: The pointer to the device registe block.
+ * @feature: Set of bit-flags indicating features of the controller.
  * @master: True if the I2S core is the I2S bit clock master.
  * @dma_playback: DMA information for playback channel.
  * @dma_capture: DMA information for capture channel.
@@ -47,6 +53,8 @@ struct s3c_i2sv2_info {
 	struct device	*dev;
 	void __iomem	*regs;
 
+	u32		feature;
+
 	struct clk	*iis_pclk;
 	struct clk	*iis_cclk;
 
-- 
cgit v1.2.3


From 8a7c251871eff512fb380ea0e86c77ed901520fe Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:57:05 +0900
Subject: ASoC: S3C: I2S: Move set_sysclk to common code

Now that we can specify feature of a particular controller, we can
avoid multiple copies of same code by defining the CDCLKCON bit
feature in controller specific code and detecting that flag in the
code common to all controllers.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/s3c-i2s-v2.c  | 47 +++++++++++++++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c2412-i2s.c | 27 -----------------------
 sound/soc/s3c24xx/s3c64xx-i2s.c | 43 +++----------------------------------
 3 files changed, 50 insertions(+), 67 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index ac153ca9c628..13311c8cf965 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -342,6 +342,52 @@ static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream,
 
 	writel(iismod, i2s->regs + S3C2412_IISMOD);
 	pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
+
+	return 0;
+}
+
+static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+	pr_debug("Entered %s\n", __func__);
+	pr_debug("%s r: IISMOD: %x\n", __func__, iismod);
+
+	switch (clk_id) {
+	case S3C_I2SV2_CLKSRC_PCLK:
+		iismod &= ~S3C2412_IISMOD_IMS_SYSMUX;
+		break;
+
+	case S3C_I2SV2_CLKSRC_AUDIOBUS:
+		iismod |= S3C2412_IISMOD_IMS_SYSMUX;
+		break;
+
+	case S3C_I2SV2_CLKSRC_CDCLK:
+		/* Error if controller doesn't have the CDCLKCON bit */
+		if (!(i2s->feature & S3C_FEATURE_CDCLKCON))
+			return -EINVAL;
+
+		switch (dir) {
+		case SND_SOC_CLOCK_IN:
+			iismod |= S3C64XX_IISMOD_CDCLKCON;
+			break;
+		case SND_SOC_CLOCK_OUT:
+			iismod &= ~S3C64XX_IISMOD_CDCLKCON;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s w: IISMOD: %x\n", __func__, iismod);
+
 	return 0;
 }
 
@@ -714,6 +760,7 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 		ops->hw_params = s3c_i2sv2_hw_params;
 	ops->set_fmt = s3c2412_i2s_set_fmt;
 	ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
+	ops->set_sysclk = s3c_i2sv2_set_sysclk;
 
 	/* Allow overriding by (for example) IISv4 */
 	if (!ops->delay)
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 6b884d75235e..709adef9d043 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -65,32 +65,6 @@ static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = {
 
 static struct s3c_i2sv2_info s3c2412_i2s;
 
-/*
- * Set S3C2412 Clock source
- */
-static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
-				  int clk_id, unsigned int freq, int dir)
-{
-	u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-
-	pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
-	    freq, dir);
-
-	switch (clk_id) {
-	case S3C2412_CLKSRC_PCLK:
-		iismod &= ~S3C2412_IISMOD_IMS_SYSMUX;
-		break;
-	case S3C2412_CLKSRC_I2SCLK:
-		iismod |= S3C2412_IISMOD_IMS_SYSMUX;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
-	return 0;
-}
-
 static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
 {
 	return cpu_dai->private_data;
@@ -175,7 +149,6 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
-	.set_sysclk	= s3c2412_i2s_set_sysclk,
 	.hw_params	= s3c2412_i2s_hw_params,
 };
 
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 6d49bd93717b..1d85cb85a7d2 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -54,43 +54,6 @@ static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
 	return cpu_dai->private_data;
 }
 
-static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
-				  int clk_id, unsigned int freq, int dir)
-{
-	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
-	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-	switch (clk_id) {
-	case S3C64XX_CLKSRC_PCLK:
-		iismod &= ~S3C2412_IISMOD_IMS_SYSMUX;
-		break;
-
-	case S3C64XX_CLKSRC_MUX:
-		iismod |= S3C2412_IISMOD_IMS_SYSMUX;
-		break;
-
-	case S3C64XX_CLKSRC_CDCLK:
-		switch (dir) {
-		case SND_SOC_CLOCK_IN:
-			iismod |= S3C64XX_IISMOD_CDCLKCON;
-			break;
-		case SND_SOC_CLOCK_OUT:
-			iismod &= ~S3C64XX_IISMOD_CDCLKCON;
-			break;
-		default:
-			return -EINVAL;
-		}
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	writel(iismod, i2s->regs + S3C2412_IISMOD);
-
-	return 0;
-}
-
 static int s3c64xx_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
@@ -115,9 +78,7 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev,
 }
 
 
-static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
-	.set_sysclk	= s3c64xx_i2s_set_sysclk,	
-};
+static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops;
 
 static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
 {
@@ -147,6 +108,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
 	dai->probe = s3c64xx_i2s_probe;
 	dai->ops = &s3c64xx_i2s_dai_ops;
 
+	i2s->feature |= S3C_FEATURE_CDCLKCON;
+
 	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
 	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
 
-- 
cgit v1.2.3


From 49100c98359a56ea4e8c9a76e3d625cdb25f25f5 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Wed, 5 May 2010 11:14:22 +0300
Subject: ASoC: omap: Add basic audio support for Nokia RX-51/N900

This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Eduardo Valentin <eduardo.valentin@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/Kconfig  |  10 ++
 sound/soc/omap/Makefile |   2 +
 sound/soc/omap/rx51.c   | 294 ++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 306 insertions(+)
 create mode 100644 sound/soc/omap/rx51.c

(limited to 'sound/soc')

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index f11963c21873..83be4a76d2bb 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -18,6 +18,16 @@ config SND_OMAP_SOC_N810
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
 
+config SND_OMAP_SOC_RX51
+	tristate "SoC Audio support for Nokia RX-51"
+	depends on SND_OMAP_SOC && MACH_NOKIA_RX51
+	select OMAP_MCBSP
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on Nokia RX-51
+	  hardware. This is also known as Nokia N900 product.
+
 config SND_OMAP_SOC_AMS_DELTA
 	tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
 	depends on SND_OMAP_SOC && MACH_AMS_DELTA
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0bc00ca14b37..3a75755f25e4 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -9,6 +9,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
+snd-soc-rx51-objs := rx51.o
 snd-soc-ams-delta-objs := ams-delta.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
@@ -22,6 +23,7 @@ snd-soc-zoom2-objs := zoom2.o
 snd-soc-igep0020-objs := igep0020.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
 obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
new file mode 100644
index 000000000000..47d831ef2dbb
--- /dev/null
+++ b/sound/soc/omap/rx51.c
@@ -0,0 +1,294 @@
+/*
+ * rx51.c  --  SoC audio for Nokia RX-51
+ *
+ * Copyright (C) 2008 - 2009 Nokia Corporation
+ *
+ * Contact: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *          Eduardo Valentin <eduardo.valentin@nokia.com>
+ *          Jarkko Nikula <jhnikula@gmail.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/delay.h>
+#include <linux/gpio.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 "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic3x.h"
+
+/*
+ * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
+ * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
+ */
+#define RX51_SPEAKER_AMP_TWL_GPIO	(192 + 7)
+
+static int rx51_spk_func;
+static int rx51_dmic_func;
+
+static void rx51_ext_control(struct snd_soc_codec *codec)
+{
+	if (rx51_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+	if (rx51_dmic_func)
+		snd_soc_dapm_enable_pin(codec, "DMic");
+	else
+		snd_soc_dapm_disable_pin(codec, "DMic");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int rx51_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+	rx51_ext_control(codec);
+
+	return 0;
+}
+
+static int rx51_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_IB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_IB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0)
+		return err;
+
+	/* Set the codec system clock for DAC and ADC */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
+				      SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops rx51_ops = {
+	.startup = rx51_startup,
+	.hw_params = rx51_hw_params,
+};
+
+static int rx51_get_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_spk_func;
+
+	return 0;
+}
+
+static int rx51_set_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_spk_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static int rx51_spk_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 1);
+	else
+		gpio_set_value(RX51_SPEAKER_AMP_TWL_GPIO, 0);
+
+	return 0;
+}
+
+static int rx51_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_dmic_func;
+
+	return 0;
+}
+
+static int rx51_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_dmic_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(codec);
+
+	return 1;
+}
+
+static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
+	SND_SOC_DAPM_MIC("DMic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Ext Spk", NULL, "HPLOUT"},
+	{"Ext Spk", NULL, "HPROUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias 2V"},
+	{"Mic Bias 2V", NULL, "DMic"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
+
+static const struct soc_enum rx51_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+};
+
+static const struct snd_kcontrol_new aic34_rx51_controls[] = {
+	SOC_ENUM_EXT("Speaker Function", rx51_enum[0],
+		     rx51_get_spk, rx51_set_spk),
+	SOC_ENUM_EXT("Input Select",  rx51_enum[1],
+		     rx51_get_input, rx51_set_input),
+};
+
+static int rx51_aic34_init(struct snd_soc_codec *codec)
+{
+	int err;
+
+	/* Set up NC codec pins */
+	snd_soc_dapm_nc_pin(codec, "MIC3L");
+	snd_soc_dapm_nc_pin(codec, "MIC3R");
+	snd_soc_dapm_nc_pin(codec, "LINE1R");
+
+	/* Add RX-51 specific controls */
+	err = snd_soc_add_controls(codec, aic34_rx51_controls,
+				   ARRAY_SIZE(aic34_rx51_controls));
+	if (err < 0)
+		return err;
+
+	/* Add RX-51 specific widgets */
+	snd_soc_dapm_new_controls(codec, aic34_dapm_widgets,
+				  ARRAY_SIZE(aic34_dapm_widgets));
+
+	/* Set up RX-51 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link rx51_dai[] = {
+	{
+		.name = "TLV320AIC34",
+		.stream_name = "AIC34",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &aic3x_dai,
+		.init = rx51_aic34_init,
+		.ops = &rx51_ops,
+	},
+};
+
+/* Audio private data */
+static struct aic3x_setup_data rx51_aic34_setup = {
+	.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
+	.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
+};
+
+/* Audio card */
+static struct snd_soc_card rx51_sound_card = {
+	.name = "RX-51",
+	.dai_link = rx51_dai,
+	.num_links = ARRAY_SIZE(rx51_dai),
+	.platform = &omap_soc_platform,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device rx51_snd_devdata = {
+	.card = &rx51_sound_card,
+	.codec_dev = &soc_codec_dev_aic3x,
+	.codec_data = &rx51_aic34_setup,
+};
+
+static struct platform_device *rx51_snd_device;
+
+static int __init rx51_soc_init(void)
+{
+	int err;
+
+	if (!machine_is_nokia_rx51())
+		return -ENODEV;
+
+	rx51_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!rx51_snd_device) {
+		err = -ENOMEM;
+		goto err1;
+	}
+
+	platform_set_drvdata(rx51_snd_device, &rx51_snd_devdata);
+	rx51_snd_devdata.dev = &rx51_snd_device->dev;
+	*(unsigned int *)rx51_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+
+	err = platform_device_add(rx51_snd_device);
+	if (err)
+		goto err2;
+
+	return 0;
+err2:
+	platform_device_put(rx51_snd_device);
+err1:
+
+	return err;
+}
+
+static void __exit rx51_soc_exit(void)
+{
+	platform_device_unregister(rx51_snd_device);
+}
+
+module_init(rx51_soc_init);
+module_exit(rx51_soc_exit);
+
+MODULE_AUTHOR("Nokia Corporation");
+MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From 5193d62f1824cdfd72b5523be2b1cdb8049225ad Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Wed, 5 May 2010 13:02:03 +0300
Subject: ASoC: tlv320aic3x: Add platform data and reset gpio handling

Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/tlv320aic3x.h    | 17 +++++++++++++++++
 sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++
 2 files changed, 42 insertions(+)
 create mode 100644 include/sound/tlv320aic3x.h

(limited to 'sound/soc')

diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h
new file mode 100644
index 000000000000..b1a5f34e5cfa
--- /dev/null
+++ b/include/sound/tlv320aic3x.h
@@ -0,0 +1,17 @@
+/*
+ * Platform data for Texas Instruments TLV320AIC3x codec
+ *
+ * Author: Jarkko Nikula <jhnikula@gmail.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.
+ */
+#ifndef __TLV320AIC3x_H__
+#define __TLV320AIC3x_H__
+
+struct aic3x_pdata {
+	int gpio_reset; /* < 0 if not used */
+};
+
+#endif
\ No newline at end of file
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 584bc1e67f76..d57372be7a96 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -38,6 +38,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/gpio.h>
 #include <linux/regulator/consumer.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
@@ -47,6 +48,7 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
+#include <sound/tlv320aic3x.h>
 
 #include "tlv320aic3x.h"
 
@@ -64,6 +66,7 @@ struct aic3x_priv {
 	struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES];
 	unsigned int sysclk;
 	int master;
+	int gpio_reset;
 };
 
 /*
@@ -1278,6 +1281,10 @@ static int aic3x_unregister(struct aic3x_priv *aic3x)
 	snd_soc_unregister_dai(&aic3x_dai);
 	snd_soc_unregister_codec(&aic3x->codec);
 
+	if (aic3x->gpio_reset >= 0) {
+		gpio_set_value(aic3x->gpio_reset, 0);
+		gpio_free(aic3x->gpio_reset);
+	}
 	regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 
@@ -1302,6 +1309,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 {
 	struct snd_soc_codec *codec;
 	struct aic3x_priv *aic3x;
+	struct aic3x_pdata *pdata = i2c->dev.platform_data;
 	int ret, i;
 
 	aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
@@ -1318,6 +1326,15 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, aic3x);
 
+	aic3x->gpio_reset = -1;
+	if (pdata && pdata->gpio_reset >= 0) {
+		ret = gpio_request(pdata->gpio_reset, "tlv320aic3x reset");
+		if (ret != 0)
+			goto err_gpio;
+		aic3x->gpio_reset = pdata->gpio_reset;
+		gpio_direction_output(aic3x->gpio_reset, 0);
+	}
+
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
 		aic3x->supplies[i].supply = aic3x_supply_names[i];
 
@@ -1335,11 +1352,19 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		goto err_enable;
 	}
 
+	if (aic3x->gpio_reset >= 0) {
+		udelay(1);
+		gpio_set_value(aic3x->gpio_reset, 1);
+	}
+
 	return aic3x_register(codec);
 
 err_enable:
 	regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
 err_get:
+	if (aic3x->gpio_reset >= 0)
+		gpio_free(aic3x->gpio_reset);
+err_gpio:
 	kfree(aic3x);
 	return ret;
 }
-- 
cgit v1.2.3


From 6f3991152f20933b77eff30413e893bf1a15e578 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 6 May 2010 10:37:18 +0300
Subject: ASoC: tpa6130a2: Support for limiting gain

Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 include/sound/tpa6130a2-plat.h |  1 +
 sound/soc/codecs/tpa6130a2.c   | 76 +++++++++++++++++++++++++++++++++++++-----
 2 files changed, 69 insertions(+), 8 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
index e29fde6b5cbe..426f62767dab 100644
--- a/include/sound/tpa6130a2-plat.h
+++ b/include/sound/tpa6130a2-plat.h
@@ -31,6 +31,7 @@ enum tpa_model {
 struct tpa6130a2_platform_data {
 	enum tpa_model id;
 	int power_gpio;
+	int limit_gain;
 };
 
 #endif
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 0cf3e3862e7b..31f67b527ca1 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -46,6 +46,9 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
 	"AVdd",
 };
 
+#define TPA6130A2_GAIN_MAX	0x3f
+#define TPA6140A2_GAIN_MAX	0x1f
+
 /* This struct is used to save the context */
 struct tpa6130a2_data {
 	struct mutex mutex;
@@ -54,6 +57,7 @@ struct tpa6130a2_data {
 	int power_gpio;
 	unsigned char power_state;
 	enum tpa_model id;
+	int gain_limit;
 };
 
 static int tpa6130a2_i2c_read(int reg)
@@ -176,6 +180,40 @@ exit:
 	return ret;
 }
 
+static int tpa6130a2_info_volsw(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct tpa6130a2_data *data;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	mutex_lock(&data->mutex);
+	switch (mc->reg) {
+	case TPA6130A2_REG_VOL_MUTE:
+		if (data->gain_limit != mc->max)
+			mc->max = data->gain_limit;
+		break;
+	default:
+		dev_err(&tpa6130a2_client->dev,
+			"Invalid register: 0x02%x\n", mc->reg);
+		goto out;
+	}
+	if (unlikely(mc->max == 1))
+		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	else
+		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = mc->max;
+out:
+	mutex_unlock(&data->mutex);
+	return 0;
+}
+
 static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
@@ -239,6 +277,15 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+#define SOC_SINGLE_EXT_TLV_TPA(xname, xreg, xshift, xmax, xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = tpa6130a2_info_volsw, \
+	.get = tpa6130a2_get_reg, .put = tpa6130a2_set_reg, \
+	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+
 /*
  * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going
  * down in gain.
@@ -258,10 +305,9 @@ static const unsigned int tpa6130_tlv[] = {
 };
 
 static const struct snd_kcontrol_new tpa6130a2_controls[] = {
-	SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume",
-		       TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0,
-		       tpa6130a2_get_reg, tpa6130a2_set_reg,
-		       tpa6130_tlv),
+	SOC_SINGLE_EXT_TLV_TPA("TPA6130A2 Headphone Playback Volume",
+			TPA6130A2_REG_VOL_MUTE, 0, TPA6130A2_GAIN_MAX, 0,
+			tpa6130_tlv),
 };
 
 static const unsigned int tpa6140_tlv[] = {
@@ -272,10 +318,9 @@ static const unsigned int tpa6140_tlv[] = {
 };
 
 static const struct snd_kcontrol_new tpa6140a2_controls[] = {
-	SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume",
-		       TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0,
-		       tpa6130a2_get_reg, tpa6130a2_set_reg,
-		       tpa6140_tlv),
+	SOC_SINGLE_EXT_TLV_TPA("TPA6140A2 Headphone Playback Volume",
+			TPA6130A2_REG_VOL_MUTE, 1, TPA6140A2_GAIN_MAX, 0,
+			tpa6140_tlv),
 };
 
 /*
@@ -454,16 +499,31 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 	case TPA6130A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
+		if (pdata->limit_gain > 0 &&
+		    pdata->limit_gain < TPA6130A2_GAIN_MAX)
+			data->gain_limit = pdata->limit_gain;
+		else
+			data->gain_limit = TPA6130A2_GAIN_MAX;
 		break;
 	case TPA6140A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6140a2_supply_names[i];;
+		if (pdata->limit_gain > 0 &&
+		    pdata->limit_gain < TPA6140A2_GAIN_MAX)
+			data->gain_limit = pdata->limit_gain;
+		else
+			data->gain_limit = TPA6140A2_GAIN_MAX;
 		break;
 	default:
 		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
 			 pdata->id);
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
+		if (pdata->limit_gain > 0 &&
+		    pdata->limit_gain < TPA6130A2_GAIN_MAX)
+			data->gain_limit = pdata->limit_gain;
+		else
+			data->gain_limit = TPA6130A2_GAIN_MAX;
 	}
 
 	ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
-- 
cgit v1.2.3


From 2f005471e2e2f2c7fa5898153387d421f7d24ad6 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 6 May 2010 12:04:25 +0300
Subject: ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC power

Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control

It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.

The codec reset values are considered safe in all environmnts.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 11 -----------
 1 file changed, 11 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 68b7ccbf2e7c..ad5e2636c944 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -303,21 +303,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 	/* C : (defaults) */
 	dac33_write(codec, DAC33_DAC_CTRL_C, 0x00);
 
-	/* 64-65 : L&R DAC power control
-	 Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/
-	dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
-	dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
-
 	/* 73 : volume soft stepping control,
 	 clock source = internal osc (?) */
 	dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN);
 
-	/* 66 : LOP/LOM Modes */
-	dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff);
-
-	/* 68 : LOM inverted from LOP */
-	dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2));
-
 	dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB);
 
 	/* Restore only selected registers (gains mostly) */
-- 
cgit v1.2.3


From 637d3847ba0ca2c7780e6521cbe75568d19ff0db Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 7 May 2010 14:05:49 +0300
Subject: ASoC: core: Support for limiting the volume

Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)

If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:

snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);

This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  2 ++
 sound/soc/soc-core.c | 39 +++++++++++++++++++++++++++++++++++++++
 2 files changed, 41 insertions(+)

(limited to 'sound/soc')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 01e9c66ddc6d..9f306f0710df 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -326,6 +326,8 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_limit_volume(struct snd_soc_codec *codec,
+	const char *name, int max);
 
 /**
  * struct snd_soc_jack_pin - Describes a pin to update based on jack detection
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d59076e2ca84..4079223203eb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2237,6 +2237,45 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 
+/**
+ * snd_soc_limit_volume - Set new limit to an existing volume control.
+ *
+ * @codec: where to look for the control
+ * @name: Name of the control
+ * @max: new maximum limit
+ *
+ * Return 0 for success, else error.
+ */
+int snd_soc_limit_volume(struct snd_soc_codec *codec,
+	const char *name, int max)
+{
+	struct snd_card *card = codec->card;
+	struct snd_kcontrol *kctl;
+	struct soc_mixer_control *mc;
+	int found = 0;
+	int ret = -EINVAL;
+
+	/* Sanity check for name and max */
+	if (unlikely(!name || max <= 0))
+		return -EINVAL;
+
+	list_for_each_entry(kctl, &card->controls, list) {
+		if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) {
+			found = 1;
+			break;
+		}
+	}
+	if (found) {
+		mc = (struct soc_mixer_control *)kctl->private_value;
+		if (max <= mc->max) {
+			mc->max = max;
+			ret = 0;
+		}
+	}
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_limit_volume);
+
 /**
  * snd_soc_dai_set_sysclk - configure DAI system or master clock.
  * @dai: DAI
-- 
cgit v1.2.3


From 826e962c4689172e25acde72eafebc063253b69d Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 7 May 2010 14:24:10 +0300
Subject: Revert "ASoC: tpa6130a2: Support for limiting gain"

This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578.

Since core has now support for limiting the volume on controls this
patch is not needed.  Furthermore, this patch actually prevents the core
to set new volume on the TPA.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/tpa6130a2-plat.h |  1 -
 sound/soc/codecs/tpa6130a2.c   | 76 +++++-------------------------------------
 2 files changed, 8 insertions(+), 69 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
index 426f62767dab..e29fde6b5cbe 100644
--- a/include/sound/tpa6130a2-plat.h
+++ b/include/sound/tpa6130a2-plat.h
@@ -31,7 +31,6 @@ enum tpa_model {
 struct tpa6130a2_platform_data {
 	enum tpa_model id;
 	int power_gpio;
-	int limit_gain;
 };
 
 #endif
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 31f67b527ca1..0cf3e3862e7b 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -46,9 +46,6 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
 	"AVdd",
 };
 
-#define TPA6130A2_GAIN_MAX	0x3f
-#define TPA6140A2_GAIN_MAX	0x1f
-
 /* This struct is used to save the context */
 struct tpa6130a2_data {
 	struct mutex mutex;
@@ -57,7 +54,6 @@ struct tpa6130a2_data {
 	int power_gpio;
 	unsigned char power_state;
 	enum tpa_model id;
-	int gain_limit;
 };
 
 static int tpa6130a2_i2c_read(int reg)
@@ -180,40 +176,6 @@ exit:
 	return ret;
 }
 
-static int tpa6130a2_info_volsw(struct snd_kcontrol *kcontrol,
-		struct snd_ctl_elem_info *uinfo)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	struct tpa6130a2_data *data;
-
-	BUG_ON(tpa6130a2_client == NULL);
-	data = i2c_get_clientdata(tpa6130a2_client);
-
-	mutex_lock(&data->mutex);
-	switch (mc->reg) {
-	case TPA6130A2_REG_VOL_MUTE:
-		if (data->gain_limit != mc->max)
-			mc->max = data->gain_limit;
-		break;
-	default:
-		dev_err(&tpa6130a2_client->dev,
-			"Invalid register: 0x02%x\n", mc->reg);
-		goto out;
-	}
-	if (unlikely(mc->max == 1))
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-	else
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = mc->max;
-out:
-	mutex_unlock(&data->mutex);
-	return 0;
-}
-
 static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
@@ -277,15 +239,6 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-#define SOC_SINGLE_EXT_TLV_TPA(xname, xreg, xshift, xmax, xinvert, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
-		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
-	.tlv.p = (tlv_array), \
-	.info = tpa6130a2_info_volsw, \
-	.get = tpa6130a2_get_reg, .put = tpa6130a2_set_reg, \
-	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
-
 /*
  * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going
  * down in gain.
@@ -305,9 +258,10 @@ static const unsigned int tpa6130_tlv[] = {
 };
 
 static const struct snd_kcontrol_new tpa6130a2_controls[] = {
-	SOC_SINGLE_EXT_TLV_TPA("TPA6130A2 Headphone Playback Volume",
-			TPA6130A2_REG_VOL_MUTE, 0, TPA6130A2_GAIN_MAX, 0,
-			tpa6130_tlv),
+	SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume",
+		       TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0,
+		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+		       tpa6130_tlv),
 };
 
 static const unsigned int tpa6140_tlv[] = {
@@ -318,9 +272,10 @@ static const unsigned int tpa6140_tlv[] = {
 };
 
 static const struct snd_kcontrol_new tpa6140a2_controls[] = {
-	SOC_SINGLE_EXT_TLV_TPA("TPA6140A2 Headphone Playback Volume",
-			TPA6130A2_REG_VOL_MUTE, 1, TPA6140A2_GAIN_MAX, 0,
-			tpa6140_tlv),
+	SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume",
+		       TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0,
+		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+		       tpa6140_tlv),
 };
 
 /*
@@ -499,31 +454,16 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 	case TPA6130A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
-		if (pdata->limit_gain > 0 &&
-		    pdata->limit_gain < TPA6130A2_GAIN_MAX)
-			data->gain_limit = pdata->limit_gain;
-		else
-			data->gain_limit = TPA6130A2_GAIN_MAX;
 		break;
 	case TPA6140A2:
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6140a2_supply_names[i];;
-		if (pdata->limit_gain > 0 &&
-		    pdata->limit_gain < TPA6140A2_GAIN_MAX)
-			data->gain_limit = pdata->limit_gain;
-		else
-			data->gain_limit = TPA6140A2_GAIN_MAX;
 		break;
 	default:
 		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
 			 pdata->id);
 		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
 			data->supplies[i].supply = tpa6130a2_supply_names[i];
-		if (pdata->limit_gain > 0 &&
-		    pdata->limit_gain < TPA6130A2_GAIN_MAX)
-			data->gain_limit = pdata->limit_gain;
-		else
-			data->gain_limit = TPA6130A2_GAIN_MAX;
 	}
 
 	ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
-- 
cgit v1.2.3


From bd843edf81dc1200f71b61d4804af0e2918e8ae3 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Fri, 7 May 2010 14:24:11 +0300
Subject: ASoC: tpa6130a2: Fix for the custom kcontrol functions

Since the functions arre only used for volume register,
change their name, and also fix them to properly
handle the cases, when via soc core the volume is
limited.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tpa6130a2.c | 18 ++++++++++--------
 1 file changed, 10 insertions(+), 8 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 0cf3e3862e7b..fa4fa33a51aa 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -176,7 +176,7 @@ exit:
 	return ret;
 }
 
-static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
+static int tpa6130a2_get_volsw(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
 	struct soc_mixer_control *mc =
@@ -184,7 +184,8 @@ static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
 	struct tpa6130a2_data *data;
 	unsigned int reg = mc->reg;
 	unsigned int shift = mc->shift;
-	unsigned int mask = mc->max;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 
 	BUG_ON(tpa6130a2_client == NULL);
@@ -197,13 +198,13 @@ static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
 
 	if (invert)
 		ucontrol->value.integer.value[0] =
-			mask - ucontrol->value.integer.value[0];
+			max - ucontrol->value.integer.value[0];
 
 	mutex_unlock(&data->mutex);
 	return 0;
 }
 
-static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
+static int tpa6130a2_put_volsw(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
 	struct soc_mixer_control *mc =
@@ -211,7 +212,8 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
 	struct tpa6130a2_data *data;
 	unsigned int reg = mc->reg;
 	unsigned int shift = mc->shift;
-	unsigned int mask = mc->max;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 	unsigned int val = (ucontrol->value.integer.value[0] & mask);
 	unsigned int val_reg;
@@ -220,7 +222,7 @@ static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
 	data = i2c_get_clientdata(tpa6130a2_client);
 
 	if (invert)
-		val = mask - val;
+		val = max - val;
 
 	mutex_lock(&data->mutex);
 
@@ -260,7 +262,7 @@ static const unsigned int tpa6130_tlv[] = {
 static const struct snd_kcontrol_new tpa6130a2_controls[] = {
 	SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume",
 		       TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0,
-		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+		       tpa6130a2_get_volsw, tpa6130a2_put_volsw,
 		       tpa6130_tlv),
 };
 
@@ -274,7 +276,7 @@ static const unsigned int tpa6140_tlv[] = {
 static const struct snd_kcontrol_new tpa6140a2_controls[] = {
 	SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume",
 		       TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0,
-		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+		       tpa6130a2_get_volsw, tpa6130a2_put_volsw,
 		       tpa6140_tlv),
 };
 
-- 
cgit v1.2.3


From af56b1c27b909e06318d26416d7b21fe6483224a Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:57:11 +0900
Subject: ASoC: S3C64XX: IISv4: Add CPU driver

Add the CPU driver for the IISv4 block found on S3C6410.
For now, the driver is almost a copy of s3c64xx-i2s.c but
it should diverge as more IISv4 specific stuff is added.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/Kconfig          |   5 +
 sound/soc/s3c24xx/Makefile         |   2 +
 sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 +++++++++++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c64xx-i2s.h    |   1 +
 4 files changed, 217 insertions(+)
 create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 15fe57e5a232..c0c7edf792fd 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -24,6 +24,11 @@ config SND_S3C64XX_SOC_I2S
 	select SND_S3C_I2SV2_SOC
 	select S3C64XX_DMA
 
+config SND_S3C64XX_SOC_I2S_V4
+	tristate
+	select SND_S3C_I2SV2_SOC
+	select S3C64XX_DMA
+
 config SND_S3C_SOC_PCM
 	tristate
 
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index df071a376fa2..81d8dc503f87 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -4,6 +4,7 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
 snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
 snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o
 snd-soc-s3c-ac97-objs := s3c-ac97.o
+snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o
 snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o
 snd-soc-s3c-pcm-objs := s3c-pcm.o
 
@@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
 obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o
 obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
 obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o
+obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o
 obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
 obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o
 
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c
new file mode 100644
index 000000000000..06db130030a1
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c
@@ -0,0 +1,209 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver
+ * Copyright (c) 2010 Samsung Electronics Co. Ltd
+ * 	Author: Jaswinder Singh <jassi.brar@samsung.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.
+ */
+
+#include <linux/clk.h>
+#include <linux/gpio.h>
+#include <linux/io.h>
+
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include <mach/gpio-bank-c.h>
+#include <mach/gpio-bank-h.h>
+#include <plat/gpio-cfg.h>
+
+#include <mach/map.h>
+#include <mach/dma.h>
+
+#include "s3c-dma.h"
+#include "regs-i2s-v2.h"
+#include "s3c64xx-i2s.h"
+
+static struct s3c2410_dma_client s3c64xx_dma_client_out = {
+	.name		= "I2Sv4 PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c64xx_dma_client_in = {
+	.name		= "I2Sv4 PCM Stereo in"
+};
+
+static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out;
+static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in;
+static struct s3c_i2sv2_info s3c64xx_i2sv4;
+
+struct snd_soc_dai s3c64xx_i2s_v4_dai;
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai);
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
+
+static int s3c64xx_i2sv4_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
+{
+	/* configure GPIO for i2s port */
+	s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0);
+	s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1);
+	s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2);
+	s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK);
+	s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK);
+	s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK);
+	s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI);
+
+	return 0;
+}
+
+static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *cpu_dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	struct s3c_dma_params *dma_data;
+	u32 iismod;
+
+	dev_dbg(cpu_dai->dev, "Entered %s\n", __func__);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_data = i2s->dma_playback;
+	else
+		dma_data = i2s->dma_capture;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod);
+
+	iismod &= ~S3C64XX_IISMOD_BLC_MASK;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		iismod |= S3C64XX_IISMOD_BLC_8BIT;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iismod |= S3C64XX_IISMOD_BLC_24BIT;
+		break;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod);
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = {
+	.hw_params	= s3c_i2sv4_hw_params,
+};
+
+static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev)
+{
+	struct s3c_i2sv2_info *i2s;
+	struct snd_soc_dai *dai;
+	int ret;
+
+	i2s = &s3c64xx_i2sv4;
+	dai = &s3c64xx_i2s_v4_dai;
+
+	if (dai->dev) {
+		dev_dbg(dai->dev, "%s: \
+			I2Sv4 instance already registered!\n", __func__);
+		return -EBUSY;
+	}
+
+	dai->dev = &pdev->dev;
+	dai->name = "s3c64xx-i2s-v4";
+	dai->id = 0;
+	dai->symmetric_rates = 1;
+	dai->playback.channels_min = 2;
+	dai->playback.channels_max = 2;
+	dai->playback.rates = S3C64XX_I2S_RATES;
+	dai->playback.formats = S3C64XX_I2S_FMTS;
+	dai->capture.channels_min = 2;
+	dai->capture.channels_max = 2;
+	dai->capture.rates = S3C64XX_I2S_RATES;
+	dai->capture.formats = S3C64XX_I2S_FMTS;
+	dai->probe = s3c64xx_i2sv4_probe;
+	dai->ops = &s3c64xx_i2sv4_dai_ops;
+
+	i2s->feature |= S3C_FEATURE_CDCLKCON;
+
+	i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in;
+	i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out;
+
+	i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX;
+	i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD;
+	i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX;
+	i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD;
+
+	i2s->dma_capture->client = &s3c64xx_dma_client_in;
+	i2s->dma_capture->dma_size = 4;
+	i2s->dma_playback->client = &s3c64xx_dma_client_out;
+	i2s->dma_playback->dma_size = 4;
+
+	i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus");
+	if (IS_ERR(i2s->iis_cclk)) {
+		dev_err(&pdev->dev, "failed to get audio-bus\n");
+		ret = PTR_ERR(i2s->iis_cclk);
+		goto err;
+	}
+
+	clk_enable(i2s->iis_cclk);
+
+	ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
+	if (ret)
+		goto err_clk;
+
+	ret = s3c_i2sv2_register_dai(dai);
+	if (ret != 0)
+		goto err_i2sv2;
+
+	return 0;
+
+err_i2sv2:
+	/* Not implemented for I2Sv2 core yet */
+err_clk:
+	clk_put(i2s->iis_cclk);
+err:
+	return ret;
+}
+
+static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev)
+{
+	dev_err(&pdev->dev, "Device removal not yet supported\n");
+	return 0;
+}
+
+static struct platform_driver s3c64xx_i2sv4_driver = {
+	.probe  = s3c64xx_i2sv4_dev_probe,
+	.remove = s3c64xx_i2sv4_dev_remove,
+	.driver = {
+		.name = "s3c64xx-iis-v4",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init s3c64xx_i2sv4_init(void)
+{
+	return platform_driver_register(&s3c64xx_i2sv4_driver);
+}
+module_init(s3c64xx_i2sv4_init);
+
+static void __exit s3c64xx_i2sv4_exit(void)
+{
+	platform_driver_unregister(&s3c64xx_i2sv4_driver);
+}
+module_exit(s3c64xx_i2sv4_exit);
+
+/* Module information */
+MODULE_AUTHOR("Jaswinder Singh, <jassi.brar@samsung.com>");
+MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index f27ed50e4d82..7a40f43d1d51 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -37,5 +37,6 @@ struct clk;
 	 SNDRV_PCM_FMTBIT_S24_LE)
 
 extern struct snd_soc_dai s3c64xx_i2s_dai[];
+extern struct snd_soc_dai s3c64xx_i2s_v4_dai;
 
 #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
-- 
cgit v1.2.3


From d0bbc24d2ab78d9efc61bc80cfc85f59626ba5af Mon Sep 17 00:00:00 2001
From: Jassi Brar <jassi.brar@samsung.com>
Date: Tue, 27 Apr 2010 15:57:17 +0900
Subject: ASoC: SMDK64XX: Switch to IISv4 CPU driver

Switch the MACHINE driver to use IISv4 CPU dai.
Remove BROKEN dependency now that we have proper CPU driver available.
Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4
controller.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/Kconfig           | 7 +++----
 sound/soc/s3c24xx/smdk64xx_wm8580.c | 6 ++----
 2 files changed, 5 insertions(+), 8 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index c0c7edf792fd..2a7cc222d098 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -64,12 +64,11 @@ config SND_S3C24XX_SOC_JIVE_WM8750
 
 config SND_S3C64XX_SOC_WM8580
 	tristate "SoC I2S Audio support for WM8580 on SMDK64XX"
-	depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410)
-	depends on BROKEN
+	depends on SND_S3C24XX_SOC && MACH_SMDK6410
 	select SND_SOC_WM8580
-	select SND_S3C64XX_SOC_I2S
+	select SND_S3C64XX_SOC_I2S_V4
 	help
-	  Sat Y if you want to add support for SoC audio on the SMDK64XX.
+	  Say Y if you want to add support for SoC audio on the SMDK6410.
 
 config SND_S3C24XX_SOC_SMDK2443_WM9710
 	tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c
index efe4901213a3..07e8e51d10d6 100644
--- a/sound/soc/s3c24xx/smdk64xx_wm8580.c
+++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c
@@ -22,8 +22,6 @@
 #include "s3c-dma.h"
 #include "s3c64xx-i2s.h"
 
-#define S3C64XX_I2S_V4 2
-
 /* SMDK64XX has a 12MHZ crystal attached to WM8580 */
 #define SMDK64XX_WM8580_FREQ 12000000
 
@@ -215,7 +213,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = {
 { /* Primary Playback i/f */
 	.name = "WM8580 PAIF RX",
 	.stream_name = "Playback",
-	.cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+	.cpu_dai = &s3c64xx_i2s_v4_dai,
 	.codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX],
 	.init = smdk64xx_wm8580_init_paifrx,
 	.ops = &smdk64xx_ops,
@@ -223,7 +221,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = {
 { /* Primary Capture i/f */
 	.name = "WM8580 PAIF TX",
 	.stream_name = "Capture",
-	.cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+	.cpu_dai = &s3c64xx_i2s_v4_dai,
 	.codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX],
 	.init = smdk64xx_wm8580_init_paiftx,
 	.ops = &smdk64xx_ops,
-- 
cgit v1.2.3


From 29e189c29d33d273424327ca4c1a74d8cdc04f48 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 20:30:00 +0100
Subject: ASoC: Remove unneeded suspend bias managment from CODEC drivers

The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ak4535.c      | 1 -
 sound/soc/codecs/cq93vc.c      | 2 +-
 sound/soc/codecs/ssm2602.c     | 1 -
 sound/soc/codecs/stac9766.c    | 3 ---
 sound/soc/codecs/tlv320aic23.c | 1 -
 sound/soc/codecs/tlv320aic3x.c | 2 +-
 sound/soc/codecs/tlv320dac33.c | 3 ---
 sound/soc/codecs/twl4030.c     | 1 -
 sound/soc/codecs/twl6040.c     | 1 -
 sound/soc/codecs/uda1380.c     | 1 -
 sound/soc/codecs/wm8350.c      | 3 ---
 sound/soc/codecs/wm8510.c      | 2 +-
 sound/soc/codecs/wm8711.c      | 2 +-
 sound/soc/codecs/wm8728.c      | 2 +-
 sound/soc/codecs/wm8731.c      | 1 -
 sound/soc/codecs/wm8903.c      | 1 -
 sound/soc/codecs/wm8940.c      | 1 -
 sound/soc/codecs/wm8960.c      | 2 +-
 sound/soc/codecs/wm8974.c      | 2 +-
 sound/soc/codecs/wm9712.c      | 3 ---
 sound/soc/codecs/wm9713.c      | 3 ---
 21 files changed, 7 insertions(+), 31 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 2f6221b2ebb5..b5917df0d2a8 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -445,7 +445,6 @@ static int ak4535_resume(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->card->codec;
 	ak4535_sync(codec);
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	ak4535_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 8f19b9310645..a320fb5a0e26 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct platform_device *pdev)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
-	cq93vc_set_bias_level(codec, codec->suspend_bias_level);
+	cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 99ddec7df614..f18c8b2d96d4 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -559,7 +559,6 @@ static int ssm2602_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	ssm2602_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index a1c7ca869d0e..e90fe295c655 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -288,9 +288,6 @@ reset:
 	}
 	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
-		stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index da589d8664d0..40bcbbad07ff 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -633,7 +633,6 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 	}
 
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index d57372be7a96..eafa164a8f86 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1143,7 +1143,7 @@ static int aic3x_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	aic3x_set_bias_level(codec, codec->suspend_bias_level);
+	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index ad5e2636c944..007fe830be46 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1386,9 +1386,6 @@ static int dac33_soc_resume(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->card->codec;
 
 	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
-		dac33_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-	dac33_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index b717a03dfacf..0fe74d1e2c5f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2181,7 +2181,6 @@ static int twl4030_soc_resume(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	twl4030_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index ead6d4b43b25..2ae442edeb9a 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -996,7 +996,6 @@ static int twl6040_resume(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	twl6040_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index ecba686563aa..2f925a27dcde 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -670,7 +670,6 @@ static int uda1380_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 	uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	uda1380_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 145467d37054..e5a48da65f82 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1336,9 +1336,6 @@ static int wm8350_resume(struct platform_device *pdev)
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
-		wm8350_set_bias_level(codec, SND_SOC_BIAS_ON);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index af8cb6995a1f..d3527001b178 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -556,7 +556,7 @@ static int wm8510_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8510_set_bias_level(codec, codec->suspend_bias_level);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index f350c5ecd093..5acd35bdfd53 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -375,7 +375,7 @@ static int wm8711_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 	wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8711_set_bias_level(codec, codec->suspend_bias_level);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 3fb653ba363a..bf56a58d0916 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -237,7 +237,7 @@ static int wm8728_resume(struct platform_device *pdev)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
-	wm8728_set_bias_level(codec, codec->suspend_bias_level);
+	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 63f90c4ad93a..c4b63538bb77 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -460,7 +460,6 @@ static int wm8731_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8731_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 5a2510c39c10..05b28cf0f96f 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1628,7 +1628,6 @@ static int wm8903_resume(struct platform_device *pdev)
 
 	/* Bring the codec back up to standby first to minimise pop/clicks */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8903_set_bias_level(codec, codec->suspend_bias_level);
 
 	/* Sync back everything else */
 	if (tmp_cache) {
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index b2f12828630c..e313ab5de85a 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -691,7 +691,6 @@ static int wm8940_resume(struct platform_device *pdev)
 	ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	if (ret)
 		goto error_ret;
-	ret = wm8940_set_bias_level(codec, codec->suspend_bias_level);
 
 error_ret:
 	return ret;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 50b2376680c1..518c84e623ab 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -823,7 +823,7 @@ static int wm8960_resume(struct platform_device *pdev)
 	}
 
 	codec->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	codec->set_bias_level(codec, codec->suspend_bias_level);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 270eb956294f..44841fec5545 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -608,7 +608,7 @@ static int wm8974_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 	wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8974_set_bias_level(codec, codec->suspend_bias_level);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index e237bf615129..dd4b6ce01ceb 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -631,9 +631,6 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 		}
 	}
 
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
-		wm9712_set_bias_level(codec, SND_SOC_BIAS_ON);
-
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index d2244c93f581..6a30b844a671 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1200,9 +1200,6 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 		}
 	}
 
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
-		wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
-
 	return ret;
 }
 
-- 
cgit v1.2.3


From 50ae8384cde9a67714ff03010493c5052690624e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 18:40:54 +0100
Subject: ASoC: Remove unused DAPM suspend flag

We now manage suspend within the main power analysis rather than by
flipping the state of widgets.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dapm.h |  1 -
 sound/soc/soc-dapm.c     | 10 ----------
 2 files changed, 11 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index d5d6ba862dfe..55c6d3d1894f 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -427,7 +427,6 @@ struct snd_soc_dapm_widget {
 	unsigned char connected:1;		/* connected codec pin */
 	unsigned char new:1;			/* cnew complete */
 	unsigned char ext:1;			/* has external widgets */
-	unsigned char suspend:1;		/* was active before suspend */
 	unsigned char force:1;			/* force state */
 
 	int (*power_check)(struct snd_soc_dapm_widget *w);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 476dbe6dabad..b30b0a255cd5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2012,18 +2012,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 				w->active = 0;
 				break;
 			case SND_SOC_DAPM_STREAM_SUSPEND:
-				if (w->active)
-					w->suspend = 1;
-				w->active = 0;
-				break;
 			case SND_SOC_DAPM_STREAM_RESUME:
-				if (w->suspend) {
-					w->active = 1;
-					w->suspend = 0;
-				}
-				break;
 			case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
-				break;
 			case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
 				break;
 			}
-- 
cgit v1.2.3


From 9949788b793826f2d19e929ac0219ad0e0891e2d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 20:24:05 +0100
Subject: ASoC: Refactor DAPM suspend handling

Instead of using stream events to handle power down during suspend
integrate the handling with the normal widget path checking by
replacing all cases where we report a connected endpoint in a path
with a function snd_soc_dapm_suspend_check() which looks at the ALSA
power state for the card and reports false if we are in a D3 state.

Since the core moves us into D3 prior to initating the suspend all
power checks during suspend will cause the widgets to be powered
down. In order to ensure that widgets are powered up on resume set
the card to D2 at the start of resume handling (ALSA API calls
require D0 so we are still protected against userspace access).

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c |  3 +++
 sound/soc/soc-dapm.c | 53 +++++++++++++++++++++++++++++-----------------------
 2 files changed, 33 insertions(+), 23 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4079223203eb..95739767ec45 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -963,6 +963,9 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	dev_dbg(socdev->dev, "starting resume work\n");
 
+	/* Bring us up into D2 so that DAPM starts enabling things */
+	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D2);
+
 	if (card->resume_pre)
 		card->resume_pre(pdev);
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b30b0a255cd5..8c8b291320a8 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -430,6 +430,23 @@ static inline void dapm_clear_walk(struct snd_soc_codec *codec)
 		p->walked = 0;
 }
 
+/* We implement power down on suspend by checking the power state of
+ * the ALSA card - when we are suspending the ALSA state for the card
+ * is set to D3.
+ */
+static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_codec *codec = widget->codec;
+
+	switch (snd_power_get_state(codec->card)) {
+	case SNDRV_CTL_POWER_D3hot:
+	case SNDRV_CTL_POWER_D3cold:
+		return 0;
+	default:
+		return 1;
+	}
+}
+
 /*
  * Recursively check for a completed path to an active or physically connected
  * output widget. Returns number of complete paths.
@@ -446,7 +463,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 	case snd_soc_dapm_adc:
 	case snd_soc_dapm_aif_out:
 		if (widget->active)
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 	default:
 		break;
 	}
@@ -454,12 +471,12 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 	if (widget->connected) {
 		/* connected pin ? */
 		if (widget->id == snd_soc_dapm_output && !widget->ext)
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 
 		/* connected jack or spk ? */
 		if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
 		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 	}
 
 	list_for_each_entry(path, &widget->sinks, list_source) {
@@ -492,7 +509,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_aif_in:
 		if (widget->active)
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 	default:
 		break;
 	}
@@ -500,16 +517,16 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 	if (widget->connected) {
 		/* connected pin ? */
 		if (widget->id == snd_soc_dapm_input && !widget->ext)
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 
 		/* connected VMID/Bias for lower pops */
 		if (widget->id == snd_soc_dapm_vmid)
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 
 		/* connected jack ? */
 		if (widget->id == snd_soc_dapm_mic ||
 		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
-			return 1;
+			return snd_soc_dapm_suspend_check(widget);
 	}
 
 	list_for_each_entry(path, &widget->sources, list_sink) {
@@ -897,22 +914,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 			if (!w->power_check)
 				continue;
 
-			/* If we're suspending then pull down all the 
-			 * power. */
-			switch (event) {
-			case SND_SOC_DAPM_STREAM_SUSPEND:
-				power = 0;
-				break;
-
-			default:
-				if (!w->force)
-					power = w->power_check(w);
-				else
-					power = 1;
-				if (power)
-					sys_power = 1;
-				break;
-			}
+			if (!w->force)
+				power = w->power_check(w);
+			else
+				power = 1;
+			if (power)
+				sys_power = 1;
 
 			if (w->power == power)
 				continue;
-- 
cgit v1.2.3


From 1547aba993c990e5a316751431328145b38e1fea Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 21:11:40 +0100
Subject: ASoC: Support leaving paths enabled over system suspend

Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.

Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.

When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dapm.h |  2 ++
 sound/soc/soc-core.c     | 32 ++++++++++++++++++++++++++++----
 sound/soc/soc-dapm.c     | 31 ++++++++++++++++++++++++++++++-
 3 files changed, 60 insertions(+), 5 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 55c6d3d1894f..66ff4c124dbd 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -341,6 +341,7 @@ int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin);
 int snd_soc_dapm_sync(struct snd_soc_codec *codec);
 int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec,
 				  const char *pin);
+int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin);
 
 /* dapm widget types */
 enum snd_soc_dapm_type {
@@ -428,6 +429,7 @@ struct snd_soc_dapm_widget {
 	unsigned char new:1;			/* cnew complete */
 	unsigned char ext:1;			/* has external widgets */
 	unsigned char force:1;			/* force state */
+	unsigned char ignore_suspend:1;         /* kept enabled over suspend */
 
 	int (*power_check)(struct snd_soc_dapm_widget *w);
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 95739767ec45..8661e5b4adb1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -927,8 +927,19 @@ static int soc_suspend(struct device *dev)
 				SND_SOC_DAPM_STREAM_SUSPEND);
 	}
 
-	if (codec_dev->suspend)
-		codec_dev->suspend(pdev, PMSG_SUSPEND);
+	/* If there are paths active then the CODEC will be held with
+	 * bias _ON and should not be suspended. */
+	if (codec_dev->suspend) {
+		switch (codec->bias_level) {
+		case SND_SOC_BIAS_STANDBY:
+		case SND_SOC_BIAS_OFF:
+			codec_dev->suspend(pdev, PMSG_SUSPEND);
+			break;
+		default:
+			dev_dbg(socdev->dev, "CODEC is on over suspend\n");
+			break;
+		}
+	}
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
@@ -975,8 +986,21 @@ static void soc_resume_deferred(struct work_struct *work)
 			cpu_dai->resume(cpu_dai);
 	}
 
-	if (codec_dev->resume)
-		codec_dev->resume(pdev);
+	/* If the CODEC was idle over suspend then it will have been
+	 * left with bias OFF or STANDBY and suspended so we must now
+	 * resume.  Otherwise the suspend was suppressed.
+	 */
+	if (codec_dev->resume) {
+		switch (codec->bias_level) {
+		case SND_SOC_BIAS_STANDBY:
+		case SND_SOC_BIAS_OFF:
+			codec_dev->resume(pdev);
+			break;
+		default:
+			dev_dbg(socdev->dev, "CODEC was on over suspend\n");
+			break;
+		}
+	}
 
 	for (i = 0; i < codec->num_dai; i++) {
 		char *stream = codec->dai[i].playback.stream_name;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8c8b291320a8..fefb6c44fc81 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -441,7 +441,9 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
 	switch (snd_power_get_state(codec->card)) {
 	case SNDRV_CTL_POWER_D3hot:
 	case SNDRV_CTL_POWER_D3cold:
-		return 0;
+		if (widget->ignore_suspend)
+			pr_debug("%s ignoring suspend\n", widget->name);
+		return widget->ignore_suspend;
 	default:
 		return 1;
 	}
@@ -2136,6 +2138,33 @@ int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
 
+/**
+ * snd_soc_dapm_ignore_suspend - ignore suspend status for DAPM endpoint
+ * @codec: audio codec
+ * @pin: audio signal pin endpoint (or start point)
+ *
+ * Mark the given endpoint or pin as ignoring suspend.  When the
+ * system is disabled a path between two endpoints flagged as ignoring
+ * suspend will not be disabled.  The path must already be enabled via
+ * normal means at suspend time, it will not be turned on if it was not
+ * already enabled.
+ */
+int snd_soc_dapm_ignore_suspend(struct snd_soc_codec *codec, const char *pin)
+{
+	struct snd_soc_dapm_widget *w;
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (!strcmp(w->name, pin)) {
+			w->ignore_suspend = 1;
+			return 0;
+		}
+	}
+
+	pr_err("Unknown DAPM pin: %s\n", pin);
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
+
 /**
  * snd_soc_dapm_free - free dapm resources
  * @socdev: SoC device
-- 
cgit v1.2.3


From 452a5fd679f8a6f215df71c41487ad4b2c7774db Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 21:18:53 +0100
Subject: ASoC: Allow active paths from the GSM modem while the GTA02 is
 suspended

Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
index dea83d30a5c9..209c25994c7e 100644
--- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -362,6 +362,14 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_disable_pin(codec, "Handset Mic");
 	snd_soc_dapm_disable_pin(codec, "Handset Spk");
 
+	/* allow audio paths from the GSM modem to run during suspend */
+	snd_soc_dapm_ignore_suspend(codec, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(codec, "GSM Line Out");
+	snd_soc_dapm_ignore_suspend(codec, "GSM Line In");
+	snd_soc_dapm_ignore_suspend(codec, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(codec, "Handset Mic");
+	snd_soc_dapm_ignore_suspend(codec, "Handset Spk");
+
 	snd_soc_dapm_sync(codec);
 
 	return 0;
-- 
cgit v1.2.3


From 3efab7dcc0f4d0087f73ff975eaa2fddc02ffc69 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 9 May 2010 13:25:43 +0100
Subject: ASoC: Allow DAI links to be kept active over suspend

As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked.  This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link.  It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.

Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  3 +++
 sound/soc/soc-core.c | 38 +++++++++++++++++++++++++++++++++++++-
 2 files changed, 40 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9f306f0710df..8326fc3db1cf 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -505,6 +505,9 @@ struct snd_soc_dai_link  {
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_codec *codec);
 
+	/* Keep DAI active over suspend */
+	unsigned int ignore_suspend:1;
+
 	/* Symmetry requirements */
 	unsigned int symmetric_rates:1;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 8661e5b4adb1..34f71bf60140 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -893,19 +893,31 @@ static int soc_suspend(struct device *dev)
 	/* mute any active DAC's */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (dai->ops->digital_mute && dai->playback.active)
 			dai->ops->digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
-	for (i = 0; i < card->num_links; i++)
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		snd_pcm_suspend_all(card->dai_link[i].pcm);
+	}
 
 	if (card->suspend_pre)
 		card->suspend_pre(pdev, PMSG_SUSPEND);
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (cpu_dai->suspend && !cpu_dai->ac97_control)
 			cpu_dai->suspend(cpu_dai);
 		if (platform->suspend)
@@ -918,6 +930,10 @@ static int soc_suspend(struct device *dev)
 
 	for (i = 0; i < codec->num_dai; i++) {
 		char *stream = codec->dai[i].playback.stream_name;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (stream != NULL)
 			snd_soc_dapm_stream_event(codec, stream,
 				SND_SOC_DAPM_STREAM_SUSPEND);
@@ -943,6 +959,10 @@ static int soc_suspend(struct device *dev)
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (cpu_dai->suspend && cpu_dai->ac97_control)
 			cpu_dai->suspend(cpu_dai);
 	}
@@ -982,6 +1002,10 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (cpu_dai->resume && cpu_dai->ac97_control)
 			cpu_dai->resume(cpu_dai);
 	}
@@ -1004,6 +1028,10 @@ static void soc_resume_deferred(struct work_struct *work)
 
 	for (i = 0; i < codec->num_dai; i++) {
 		char *stream = codec->dai[i].playback.stream_name;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (stream != NULL)
 			snd_soc_dapm_stream_event(codec, stream,
 				SND_SOC_DAPM_STREAM_RESUME);
@@ -1016,12 +1044,20 @@ static void soc_resume_deferred(struct work_struct *work)
 	/* unmute any active DACs */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (dai->ops->digital_mute && dai->playback.active)
 			dai->ops->digital_mute(dai, 0);
 	}
 
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+
+		if (card->dai_link[i].ignore_suspend)
+			continue;
+
 		if (cpu_dai->resume && !cpu_dai->ac97_control)
 			cpu_dai->resume(cpu_dai);
 		if (platform->resume)
-- 
cgit v1.2.3


From 06ae99888e599ea2e82386d618405c473c379185 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 19:14:45 +0100
Subject: ASoC: Refactor WM8731 regulator management into bias management

This allows more flexible integration with subsystem features.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8731.c | 51 ++++++++++++++++++++++++++---------------------
 1 file changed, 28 insertions(+), 23 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c4b63538bb77..2650fac30ee4 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -368,6 +368,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
+	int i, ret;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
 	u16 reg;
 
 	switch (level) {
@@ -376,6 +380,24 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
+						    wm8731->supplies);
+			if (ret != 0)
+				return ret;
+
+			/* Sync reg_cache with the hardware */
+			for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
+				if (cache[i] == wm8731_reg[i])
+					continue;
+
+				data[0] = (i << 1) | ((cache[i] >> 8)
+						      & 0x0001);
+				data[1] = cache[i] & 0x00ff;
+				codec->hw_write(codec->control_data, data, 2);
+			}
+		}
+
 		/* Clear PWROFF, gate CLKOUT, everything else as-is */
 		reg = snd_soc_read(codec, WM8731_PWR) & 0xff7f;
 		snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
@@ -383,6 +405,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_OFF:
 		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 		snd_soc_write(codec, WM8731_PWR, 0xffff);
+		regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
+				       wm8731->supplies);
 		break;
 	}
 	codec->bias_level = level;
@@ -427,12 +451,9 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
-	snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
-			       wm8731->supplies);
+
 	return 0;
 }
 
@@ -440,25 +461,7 @@ static int wm8731_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
-	int i, ret;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
-				    wm8731->supplies);
-	if (ret != 0)
-		return ret;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
-		if (cache[i] == wm8731_reg[i])
-			continue;
 
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
@@ -603,6 +606,9 @@ static int wm8731_register(struct wm8731_priv *wm8731,
 		goto err_codec;
 	}
 
+	/* Regulators will have been enabled by bias management */
+	regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
+
 	return 0;
 
 err_codec:
@@ -621,7 +627,6 @@ static void wm8731_unregister(struct wm8731_priv *wm8731)
 	wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dai(&wm8731_dai);
 	snd_soc_unregister_codec(&wm8731->codec);
-	regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
 	regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
 	kfree(wm8731);
 	wm8731_codec = NULL;
-- 
cgit v1.2.3


From 896060c76bdfd8a45eb33b3dd1a8307fe37f6c04 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 7 May 2010 18:39:25 +0100
Subject: ASoC: Use more idiomatic driver name for WM8731

Make dev_() prints much prettier.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 drivers/mfd/wm831x-irq.c  | 30 ++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8731.c |  2 +-
 2 files changed, 31 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/drivers/mfd/wm831x-irq.c b/drivers/mfd/wm831x-irq.c
index 301327697117..4c1122ceb443 100644
--- a/drivers/mfd/wm831x-irq.c
+++ b/drivers/mfd/wm831x-irq.c
@@ -21,6 +21,7 @@
 
 #include <linux/mfd/wm831x/core.h>
 #include <linux/mfd/wm831x/pdata.h>
+#include <linux/mfd/wm831x/gpio.h>
 #include <linux/mfd/wm831x/irq.h>
 
 #include <linux/delay.h>
@@ -388,12 +389,41 @@ static void wm831x_irq_mask(unsigned int irq)
 	wm831x->irq_masks_cur[irq_data->reg - 1] |= irq_data->mask;
 }
 
+static int wm831x_irq_set_type(unsigned int irq, unsigned int type)
+{
+	struct wm831x *wm831x = get_irq_chip_data(irq);
+	int val;
+
+	irq = irq - wm831x->irq_base;
+
+	if (irq < WM831X_IRQ_GPIO_1 || irq > WM831X_IRQ_GPIO_11)
+		return -EINVAL;
+
+	switch (type) {
+	case IRQ_TYPE_EDGE_BOTH:
+		val = WM831X_GPN_INT_MODE;
+		break;
+	case IRQ_TYPE_EDGE_RISING:
+		val = WM831X_GPN_POL;
+		break;
+	case IRQ_TYPE_EDGE_FALLING:
+		val = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return wm831x_set_bits(wm831x, WM831X_GPIO1_CONTROL + irq,
+			       WM831X_GPN_INT_MODE | WM831X_GPN_POL, val);
+}
+
 static struct irq_chip wm831x_irq_chip = {
 	.name = "wm831x",
 	.bus_lock = wm831x_irq_lock,
 	.bus_sync_unlock = wm831x_irq_sync_unlock,
 	.mask = wm831x_irq_mask,
 	.unmask = wm831x_irq_unmask,
+	.set_type = wm831x_irq_set_type,
 };
 
 /* The processing of the primary interrupt occurs in a thread so that
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2650fac30ee4..e17c714196f2 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -707,7 +707,7 @@ MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
 
 static struct i2c_driver wm8731_i2c_driver = {
 	.driver = {
-		.name = "WM8731 I2C Codec",
+		.name = "wm8731",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8731_i2c_probe,
-- 
cgit v1.2.3


From d11bb4a925613fa814ed4ae350440eb24ebff336 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Mon, 10 May 2010 14:39:24 +0300
Subject: ASoC: core: Fix for the volume limiting when invert is in use

If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.

Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  | 23 ++++++++++++-----------
 sound/soc/soc-core.c | 30 +++++++++++++++++++++---------
 2 files changed, 33 insertions(+), 20 deletions(-)

(limited to 'sound/soc')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8326fc3db1cf..697e7ffe39d7 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -30,10 +30,10 @@
 #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \
-	.invert = xinvert})
+	.platform_max = xmax, .invert = xinvert})
 #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
-	{.reg = xreg, .max = xmax, .invert = xinvert})
+	{.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert})
 #define SOC_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
@@ -53,14 +53,14 @@
 	.put = snd_soc_put_volsw, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
-		 .max = xmax, .invert = xinvert} }
+		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.info = snd_soc_info_volsw_2r, \
 	.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		.max = xmax, .invert = xinvert} }
+		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -70,7 +70,7 @@
 	.put = snd_soc_put_volsw, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
-		 .max = xmax, .invert = xinvert} }
+		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -80,7 +80,7 @@
 	.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		.max = xmax, .invert = xinvert} }
+		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
 {	.iface  = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -89,7 +89,8 @@
 	.info   = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \
 	.put    = snd_soc_put_volsw_s8, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = xreg, .min = xmin, .max = xmax} }
+		{.reg = xreg, .min = xmin, .max = xmax, \
+		 .platform_max = xmax} }
 #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \
 {	.reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
 	.max = xmax, .texts = xtexts }
@@ -126,7 +127,7 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
-		 .max = xmax, .invert = xinvert} }
+		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -146,7 +147,7 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
-		.max = xmax, .invert = xinvert} }
+		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@@ -157,7 +158,7 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&(struct soc_mixer_control) \
 		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		.max = xmax, .invert = xinvert} }
+		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_bool_ext, \
@@ -572,7 +573,7 @@ struct snd_soc_pcm_runtime {
 
 /* mixer control */
 struct soc_mixer_control {
-	int min, max;
+	int min, max, platform_max;
 	unsigned int reg, rreg, shift, rshift, invert;
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 34f71bf60140..e1043f644730 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2017,18 +2017,22 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	int max = mc->max;
+	int platform_max;
 	unsigned int shift = mc->shift;
 	unsigned int rshift = mc->rshift;
 
-	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
+	if (!mc->platform_max)
+		mc->platform_max = mc->max;
+	platform_max = mc->platform_max;
+
+	if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
 	else
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 
 	uinfo->count = shift == rshift ? 1 : 2;
 	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = max;
+	uinfo->value.integer.max = platform_max;
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
@@ -2126,16 +2130,20 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	int max = mc->max;
+	int platform_max;
 
-	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
+	if (!mc->platform_max)
+		mc->platform_max = mc->max;
+	platform_max = mc->platform_max;
+
+	if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume"))
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
 	else
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 
 	uinfo->count = 2;
 	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = max;
+	uinfo->value.integer.max = platform_max;
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
@@ -2236,13 +2244,17 @@ int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
 {
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
-	int max = mc->max;
+	int platform_max;
 	int min = mc->min;
 
+	if (!mc->platform_max)
+		mc->platform_max = mc->max;
+	platform_max = mc->platform_max;
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = 2;
 	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = max-min;
+	uinfo->value.integer.max = platform_max - min;
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
@@ -2331,7 +2343,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec,
 	if (found) {
 		mc = (struct soc_mixer_control *)kctl->private_value;
 		if (max <= mc->max) {
-			mc->max = max;
+			mc->platform_max = max;
 			ret = 0;
 		}
 	}
-- 
cgit v1.2.3


From fd5722e5cdb233bbc47e782e10b3b5e3488a22fa Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 10 May 2010 14:55:04 +0100
Subject: ASoC: Add register write logging for WM8994

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8994.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index f04a116e678b..c6b3f1ebd62e 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1710,6 +1710,8 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (!wm8994_volatile(reg))
 		wm8994->reg_cache[reg] = value;
 
+	dev_dbg(codec->dev, "0x%x = 0x%x\n", reg, value);
+
 	return wm8994_reg_write(codec->control_data, reg, value);
 }
 
-- 
cgit v1.2.3


From 3254d2850075c88083d870b3ee5e1ce0922cd3d9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 10 May 2010 14:56:03 +0100
Subject: ASoC: Add additional WM hubs DC servo trace

Log the values we're getting back from the DC servo and the values we
write to it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm_hubs.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index defdbd08e135..8ad43bb32eb0 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -127,6 +127,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			break;
 		}
 
+		dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
+
 		/* HPOUT1L */
 		if (reg_l + hubs->dcs_codes > 0 &&
 		    reg_l + hubs->dcs_codes < 0xff)
@@ -139,6 +141,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			reg_r += hubs->dcs_codes;
 		dcs_cfg |= reg_r;
 
+		dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
+
 		/* Do it */
 		snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
 		wait_for_dc_servo(codec,
-- 
cgit v1.2.3


From 6adb26bd03f6056357298bc29d5c6acdea2b216a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 10 May 2010 16:13:11 +0100
Subject: ASoC: Reorder power down sequence for WM hubs devices

Disable the output stage prior to the delay stage rather than the
other way around. Fixes merge issue with previous headphone output
path corrections.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm_hubs.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 8ad43bb32eb0..16f1a57da08a 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -401,14 +401,14 @@ static int hp_event(struct snd_soc_dapm_widget *w,
 
 	case SND_SOC_DAPM_PRE_PMD:
 		snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
-				    WM8993_HPOUT1L_DLY |
-				    WM8993_HPOUT1R_DLY |
+				    WM8993_HPOUT1L_OUTP |
+				    WM8993_HPOUT1R_OUTP |
 				    WM8993_HPOUT1L_RMV_SHORT |
 				    WM8993_HPOUT1R_RMV_SHORT, 0);
 
 		snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
-				    WM8993_HPOUT1L_OUTP |
-				    WM8993_HPOUT1R_OUTP, 0);
+				    WM8993_HPOUT1L_DLY |
+				    WM8993_HPOUT1R_DLY, 0);
 
 		snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
 				    WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
-- 
cgit v1.2.3


From 6a2f1ee1f9bb5346644105c9355e9e06f6a847d3 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 10 May 2010 18:36:37 +0100
Subject: ASoC: Don't restart unconfigured WM8994 FLLs

If the FLL is not configured attempting to resume it will produce a
warning message so skip the resume.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/wm8994.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index c6b3f1ebd62e..dda356591fdb 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3548,6 +3548,9 @@ static int wm8994_resume(struct platform_device *pdev)
 	wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
+		if (!wm8994->fll_suspend[i].out)
+			continue;
+
 		ret = wm8994_set_fll(&codec->dai[0], i + 1,
 				     wm8994->fll_suspend[i].src,
 				     wm8994->fll_suspend[i].in,
-- 
cgit v1.2.3


From 83f2889643d8efbc7fe925afbaa114e775699d7c Mon Sep 17 00:00:00 2001
From: Eric Miao <eric.y.miao@gmail.com>
Date: Tue, 16 Mar 2010 17:03:20 +0800
Subject: [ARM] pxa: merge regs-ssp.h into ssp.h

No need to separate them as they should be together from the begining.

Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
---
 arch/arm/mach-pxa/include/mach/regs-ssp.h | 127 ------------------------------
 arch/arm/mach-pxa/include/mach/ssp.h      | 122 ++++++++++++++++++++++++++++
 arch/arm/mach-pxa/ssp.c                   |   1 -
 drivers/spi/pxa2xx_spi.c                  |   1 -
 sound/soc/pxa/pxa-ssp.c                   |   1 -
 5 files changed, 122 insertions(+), 130 deletions(-)
 delete mode 100644 arch/arm/mach-pxa/include/mach/regs-ssp.h

(limited to 'sound/soc')

diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h
deleted file mode 100644
index dd15dc35a240..000000000000
--- a/arch/arm/mach-pxa/include/mach/regs-ssp.h
+++ /dev/null
@@ -1,127 +0,0 @@
-#ifndef __ASM_ARCH_REGS_SSP_H
-#define __ASM_ARCH_REGS_SSP_H
-
-/*
- * SSP Serial Port Registers
- * PXA250, PXA255, PXA26x and PXA27x SSP controllers are all slightly different.
- * PXA255, PXA26x and PXA27x have extra ports, registers and bits.
- */
-
-#define SSCR0		(0x00)  /* SSP Control Register 0 */
-#define SSCR1		(0x04)  /* SSP Control Register 1 */
-#define SSSR		(0x08)  /* SSP Status Register */
-#define SSITR		(0x0C)  /* SSP Interrupt Test Register */
-#define SSDR		(0x10)  /* SSP Data Write/Data Read Register */
-
-#define SSTO		(0x28)  /* SSP Time Out Register */
-#define SSPSP		(0x2C)  /* SSP Programmable Serial Protocol */
-#define SSTSA		(0x30)  /* SSP Tx Timeslot Active */
-#define SSRSA		(0x34)  /* SSP Rx Timeslot Active */
-#define SSTSS		(0x38)  /* SSP Timeslot Status */
-#define SSACD		(0x3C)  /* SSP Audio Clock Divider */
-
-#if defined(CONFIG_PXA3xx)
-#define SSACDD		(0x40)	/* SSP Audio Clock Dither Divider */
-#endif
-
-/* Common PXA2xx bits first */
-#define SSCR0_DSS	(0x0000000f)	/* Data Size Select (mask) */
-#define SSCR0_DataSize(x)  ((x) - 1)	/* Data Size Select [4..16] */
-#define SSCR0_FRF	(0x00000030)	/* FRame Format (mask) */
-#define SSCR0_Motorola	(0x0 << 4)	/* Motorola's Serial Peripheral Interface (SPI) */
-#define SSCR0_TI	(0x1 << 4)	/* Texas Instruments' Synchronous Serial Protocol (SSP) */
-#define SSCR0_National	(0x2 << 4)	/* National Microwire */
-#define SSCR0_ECS	(1 << 6)	/* External clock select */
-#define SSCR0_SSE	(1 << 7)	/* Synchronous Serial Port Enable */
-#define SSCR0_SCR(x)	((x) << 8)	/* Serial Clock Rate (mask) */
-
-#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx)
-#define SSCR0_EDSS	(1 << 20)	/* Extended data size select */
-#define SSCR0_NCS	(1 << 21)	/* Network clock select */
-#define SSCR0_RIM	(1 << 22)	/* Receive FIFO overrrun interrupt mask */
-#define SSCR0_TUM	(1 << 23)	/* Transmit FIFO underrun interrupt mask */
-#define SSCR0_FRDC	(0x07000000)	/* Frame rate divider control (mask) */
-#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24)	/* Time slots per frame [1..8] */
-#define SSCR0_ACS	(1 << 30)	/* Audio clock select */
-#define SSCR0_MOD	(1 << 31)	/* Mode (normal or network) */
-#endif
-
-#if defined(CONFIG_PXA3xx)
-#define SSCR0_FPCKE	(1 << 29)	/* FIFO packing enable */
-#endif
-
-#define SSCR1_RIE	(1 << 0)	/* Receive FIFO Interrupt Enable */
-#define SSCR1_TIE	(1 << 1)	/* Transmit FIFO Interrupt Enable */
-#define SSCR1_LBM	(1 << 2)	/* Loop-Back Mode */
-#define SSCR1_SPO	(1 << 3)	/* Motorola SPI SSPSCLK polarity setting */
-#define SSCR1_SPH	(1 << 4)	/* Motorola SPI SSPSCLK phase setting */
-#define SSCR1_MWDS	(1 << 5)	/* Microwire Transmit Data Size */
-#define SSCR1_TFT	(0x000003c0)	/* Transmit FIFO Threshold (mask) */
-#define SSCR1_TxTresh(x) (((x) - 1) << 6) /* level [1..16] */
-#define SSCR1_RFT	(0x00003c00)	/* Receive FIFO Threshold (mask) */
-#define SSCR1_RxTresh(x) (((x) - 1) << 10) /* level [1..16] */
-
-#define SSSR_TNF	(1 << 2)	/* Transmit FIFO Not Full */
-#define SSSR_RNE	(1 << 3)	/* Receive FIFO Not Empty */
-#define SSSR_BSY	(1 << 4)	/* SSP Busy */
-#define SSSR_TFS	(1 << 5)	/* Transmit FIFO Service Request */
-#define SSSR_RFS	(1 << 6)	/* Receive FIFO Service Request */
-#define SSSR_ROR	(1 << 7)	/* Receive FIFO Overrun */
-
-#define SSCR0_TIM		(1 << 23)	/* Transmit FIFO Under Run Interrupt Mask */
-#define SSCR0_RIM		(1 << 22)	/* Receive FIFO Over Run interrupt Mask */
-#define SSCR0_NCS		(1 << 21)	/* Network Clock Select */
-#define SSCR0_EDSS		(1 << 20)	/* Extended Data Size Select */
-
-/* extra bits in PXA255, PXA26x and PXA27x SSP ports */
-#define SSCR0_TISSP		(1 << 4)	/* TI Sync Serial Protocol */
-#define SSCR0_PSP		(3 << 4)	/* PSP - Programmable Serial Protocol */
-#define SSCR1_TTELP		(1 << 31)	/* TXD Tristate Enable Last Phase */
-#define SSCR1_TTE		(1 << 30)	/* TXD Tristate Enable */
-#define SSCR1_EBCEI		(1 << 29)	/* Enable Bit Count Error interrupt */
-#define SSCR1_SCFR		(1 << 28)	/* Slave Clock free Running */
-#define SSCR1_ECRA		(1 << 27)	/* Enable Clock Request A */
-#define SSCR1_ECRB		(1 << 26)	/* Enable Clock request B */
-#define SSCR1_SCLKDIR		(1 << 25)	/* Serial Bit Rate Clock Direction */
-#define SSCR1_SFRMDIR		(1 << 24)	/* Frame Direction */
-#define SSCR1_RWOT		(1 << 23)	/* Receive Without Transmit */
-#define SSCR1_TRAIL		(1 << 22)	/* Trailing Byte */
-#define SSCR1_TSRE		(1 << 21)	/* Transmit Service Request Enable */
-#define SSCR1_RSRE		(1 << 20)	/* Receive Service Request Enable */
-#define SSCR1_TINTE		(1 << 19)	/* Receiver Time-out Interrupt enable */
-#define SSCR1_PINTE		(1 << 18)	/* Peripheral Trailing Byte Interupt Enable */
-#define SSCR1_IFS		(1 << 16)	/* Invert Frame Signal */
-#define SSCR1_STRF		(1 << 15)	/* Select FIFO or EFWR */
-#define SSCR1_EFWR		(1 << 14)	/* Enable FIFO Write/Read */
-
-#define SSSR_BCE		(1 << 23)	/* Bit Count Error */
-#define SSSR_CSS		(1 << 22)	/* Clock Synchronisation Status */
-#define SSSR_TUR		(1 << 21)	/* Transmit FIFO Under Run */
-#define SSSR_EOC		(1 << 20)	/* End Of Chain */
-#define SSSR_TINT		(1 << 19)	/* Receiver Time-out Interrupt */
-#define SSSR_PINT		(1 << 18)	/* Peripheral Trailing Byte Interrupt */
-
-#if defined(CONFIG_PXA3xx)
-#define SSPSP_EDMYSTOP(x)	((x) << 28)     /* Extended Dummy Stop */
-#define SSPSP_EDMYSTRT(x)	((x) << 26)     /* Extended Dummy Start */
-#endif
-
-#define SSPSP_FSRT		(1 << 25)	/* Frame Sync Relative Timing */
-#define SSPSP_DMYSTOP(x)	((x) << 23)	/* Dummy Stop */
-#define SSPSP_SFRMWDTH(x)	((x) << 16)	/* Serial Frame Width */
-#define SSPSP_SFRMDLY(x)	((x) << 9)	/* Serial Frame Delay */
-#define SSPSP_DMYSTRT(x)	((x) << 7)	/* Dummy Start */
-#define SSPSP_STRTDLY(x)	((x) << 4)	/* Start Delay */
-#define SSPSP_ETDS		(1 << 3)	/* End of Transfer data State */
-#define SSPSP_SFRMP		(1 << 2)	/* Serial Frame Polarity */
-#define SSPSP_SCMODE(x)		((x) << 0)	/* Serial Bit Rate Clock Mode */
-
-#define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass */
-#define SSACD_ACPS(x)		((x) << 4)	/* Audio clock PLL select */
-#define SSACD_ACDS(x)		((x) << 0)	/* Audio clock divider select */
-#if defined(CONFIG_PXA3xx)
-#define SSACD_SCDX8		(1 << 7)	/* SYSCLK division ratio select */
-#endif
-
-
-#endif /* __ASM_ARCH_REGS_SSP_H */
diff --git a/arch/arm/mach-pxa/include/mach/ssp.h b/arch/arm/mach-pxa/include/mach/ssp.h
index 8470a73d18b9..4c4ce62390d0 100644
--- a/arch/arm/mach-pxa/include/mach/ssp.h
+++ b/arch/arm/mach-pxa/include/mach/ssp.h
@@ -22,6 +22,128 @@
 #include <linux/list.h>
 #include <linux/io.h>
 
+/*
+ * SSP Serial Port Registers
+ * PXA250, PXA255, PXA26x and PXA27x SSP controllers are all slightly different.
+ * PXA255, PXA26x and PXA27x have extra ports, registers and bits.
+ */
+
+#define SSCR0		(0x00)  /* SSP Control Register 0 */
+#define SSCR1		(0x04)  /* SSP Control Register 1 */
+#define SSSR		(0x08)  /* SSP Status Register */
+#define SSITR		(0x0C)  /* SSP Interrupt Test Register */
+#define SSDR		(0x10)  /* SSP Data Write/Data Read Register */
+
+#define SSTO		(0x28)  /* SSP Time Out Register */
+#define SSPSP		(0x2C)  /* SSP Programmable Serial Protocol */
+#define SSTSA		(0x30)  /* SSP Tx Timeslot Active */
+#define SSRSA		(0x34)  /* SSP Rx Timeslot Active */
+#define SSTSS		(0x38)  /* SSP Timeslot Status */
+#define SSACD		(0x3C)  /* SSP Audio Clock Divider */
+
+#if defined(CONFIG_PXA3xx)
+#define SSACDD		(0x40)	/* SSP Audio Clock Dither Divider */
+#endif
+
+/* Common PXA2xx bits first */
+#define SSCR0_DSS	(0x0000000f)	/* Data Size Select (mask) */
+#define SSCR0_DataSize(x)  ((x) - 1)	/* Data Size Select [4..16] */
+#define SSCR0_FRF	(0x00000030)	/* FRame Format (mask) */
+#define SSCR0_Motorola	(0x0 << 4)	/* Motorola's Serial Peripheral Interface (SPI) */
+#define SSCR0_TI	(0x1 << 4)	/* Texas Instruments' Synchronous Serial Protocol (SSP) */
+#define SSCR0_National	(0x2 << 4)	/* National Microwire */
+#define SSCR0_ECS	(1 << 6)	/* External clock select */
+#define SSCR0_SSE	(1 << 7)	/* Synchronous Serial Port Enable */
+#define SSCR0_SCR(x)	((x) << 8)	/* Serial Clock Rate (mask) */
+
+#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx)
+#define SSCR0_EDSS	(1 << 20)	/* Extended data size select */
+#define SSCR0_NCS	(1 << 21)	/* Network clock select */
+#define SSCR0_RIM	(1 << 22)	/* Receive FIFO overrrun interrupt mask */
+#define SSCR0_TUM	(1 << 23)	/* Transmit FIFO underrun interrupt mask */
+#define SSCR0_FRDC	(0x07000000)	/* Frame rate divider control (mask) */
+#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24)	/* Time slots per frame [1..8] */
+#define SSCR0_ACS	(1 << 30)	/* Audio clock select */
+#define SSCR0_MOD	(1 << 31)	/* Mode (normal or network) */
+#endif
+
+#if defined(CONFIG_PXA3xx)
+#define SSCR0_FPCKE	(1 << 29)	/* FIFO packing enable */
+#endif
+
+#define SSCR1_RIE	(1 << 0)	/* Receive FIFO Interrupt Enable */
+#define SSCR1_TIE	(1 << 1)	/* Transmit FIFO Interrupt Enable */
+#define SSCR1_LBM	(1 << 2)	/* Loop-Back Mode */
+#define SSCR1_SPO	(1 << 3)	/* Motorola SPI SSPSCLK polarity setting */
+#define SSCR1_SPH	(1 << 4)	/* Motorola SPI SSPSCLK phase setting */
+#define SSCR1_MWDS	(1 << 5)	/* Microwire Transmit Data Size */
+#define SSCR1_TFT	(0x000003c0)	/* Transmit FIFO Threshold (mask) */
+#define SSCR1_TxTresh(x) (((x) - 1) << 6) /* level [1..16] */
+#define SSCR1_RFT	(0x00003c00)	/* Receive FIFO Threshold (mask) */
+#define SSCR1_RxTresh(x) (((x) - 1) << 10) /* level [1..16] */
+
+#define SSSR_TNF	(1 << 2)	/* Transmit FIFO Not Full */
+#define SSSR_RNE	(1 << 3)	/* Receive FIFO Not Empty */
+#define SSSR_BSY	(1 << 4)	/* SSP Busy */
+#define SSSR_TFS	(1 << 5)	/* Transmit FIFO Service Request */
+#define SSSR_RFS	(1 << 6)	/* Receive FIFO Service Request */
+#define SSSR_ROR	(1 << 7)	/* Receive FIFO Overrun */
+
+#define SSCR0_TIM		(1 << 23)	/* Transmit FIFO Under Run Interrupt Mask */
+#define SSCR0_RIM		(1 << 22)	/* Receive FIFO Over Run interrupt Mask */
+#define SSCR0_NCS		(1 << 21)	/* Network Clock Select */
+#define SSCR0_EDSS		(1 << 20)	/* Extended Data Size Select */
+
+/* extra bits in PXA255, PXA26x and PXA27x SSP ports */
+#define SSCR0_TISSP		(1 << 4)	/* TI Sync Serial Protocol */
+#define SSCR0_PSP		(3 << 4)	/* PSP - Programmable Serial Protocol */
+#define SSCR1_TTELP		(1 << 31)	/* TXD Tristate Enable Last Phase */
+#define SSCR1_TTE		(1 << 30)	/* TXD Tristate Enable */
+#define SSCR1_EBCEI		(1 << 29)	/* Enable Bit Count Error interrupt */
+#define SSCR1_SCFR		(1 << 28)	/* Slave Clock free Running */
+#define SSCR1_ECRA		(1 << 27)	/* Enable Clock Request A */
+#define SSCR1_ECRB		(1 << 26)	/* Enable Clock request B */
+#define SSCR1_SCLKDIR		(1 << 25)	/* Serial Bit Rate Clock Direction */
+#define SSCR1_SFRMDIR		(1 << 24)	/* Frame Direction */
+#define SSCR1_RWOT		(1 << 23)	/* Receive Without Transmit */
+#define SSCR1_TRAIL		(1 << 22)	/* Trailing Byte */
+#define SSCR1_TSRE		(1 << 21)	/* Transmit Service Request Enable */
+#define SSCR1_RSRE		(1 << 20)	/* Receive Service Request Enable */
+#define SSCR1_TINTE		(1 << 19)	/* Receiver Time-out Interrupt enable */
+#define SSCR1_PINTE		(1 << 18)	/* Peripheral Trailing Byte Interupt Enable */
+#define SSCR1_IFS		(1 << 16)	/* Invert Frame Signal */
+#define SSCR1_STRF		(1 << 15)	/* Select FIFO or EFWR */
+#define SSCR1_EFWR		(1 << 14)	/* Enable FIFO Write/Read */
+
+#define SSSR_BCE		(1 << 23)	/* Bit Count Error */
+#define SSSR_CSS		(1 << 22)	/* Clock Synchronisation Status */
+#define SSSR_TUR		(1 << 21)	/* Transmit FIFO Under Run */
+#define SSSR_EOC		(1 << 20)	/* End Of Chain */
+#define SSSR_TINT		(1 << 19)	/* Receiver Time-out Interrupt */
+#define SSSR_PINT		(1 << 18)	/* Peripheral Trailing Byte Interrupt */
+
+#if defined(CONFIG_PXA3xx)
+#define SSPSP_EDMYSTOP(x)	((x) << 28)     /* Extended Dummy Stop */
+#define SSPSP_EDMYSTRT(x)	((x) << 26)     /* Extended Dummy Start */
+#endif
+
+#define SSPSP_FSRT		(1 << 25)	/* Frame Sync Relative Timing */
+#define SSPSP_DMYSTOP(x)	((x) << 23)	/* Dummy Stop */
+#define SSPSP_SFRMWDTH(x)	((x) << 16)	/* Serial Frame Width */
+#define SSPSP_SFRMDLY(x)	((x) << 9)	/* Serial Frame Delay */
+#define SSPSP_DMYSTRT(x)	((x) << 7)	/* Dummy Start */
+#define SSPSP_STRTDLY(x)	((x) << 4)	/* Start Delay */
+#define SSPSP_ETDS		(1 << 3)	/* End of Transfer data State */
+#define SSPSP_SFRMP		(1 << 2)	/* Serial Frame Polarity */
+#define SSPSP_SCMODE(x)		((x) << 0)	/* Serial Bit Rate Clock Mode */
+
+#define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass */
+#define SSACD_ACPS(x)		((x) << 4)	/* Audio clock PLL select */
+#define SSACD_ACDS(x)		((x) << 0)	/* Audio clock divider select */
+#if defined(CONFIG_PXA3xx)
+#define SSACD_SCDX8		(1 << 7)	/* SYSCLK division ratio select */
+#endif
+
 enum pxa_ssp_type {
 	SSP_UNDEFINED = 0,
 	PXA25x_SSP,  /* pxa 210, 250, 255, 26x */
diff --git a/arch/arm/mach-pxa/ssp.c b/arch/arm/mach-pxa/ssp.c
index 6e7df7af61e1..a9b2d3eb8f32 100644
--- a/arch/arm/mach-pxa/ssp.c
+++ b/arch/arm/mach-pxa/ssp.c
@@ -33,7 +33,6 @@
 #include <asm/irq.h>
 #include <mach/hardware.h>
 #include <mach/ssp.h>
-#include <mach/regs-ssp.h>
 
 static DEFINE_MUTEX(ssp_lock);
 static LIST_HEAD(ssp_list);
diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c
index 35d4e5cdb631..959e78bb2f04 100644
--- a/drivers/spi/pxa2xx_spi.c
+++ b/drivers/spi/pxa2xx_spi.c
@@ -36,7 +36,6 @@
 #include <asm/delay.h>
 
 #include <mach/dma.h>
-#include <mach/regs-ssp.h>
 #include <mach/ssp.h>
 #include <mach/pxa2xx_spi.h>
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 544fd9566f4d..2e27962220ad 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -32,7 +32,6 @@
 
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <mach/regs-ssp.h>
 #include <mach/audio.h>
 #include <mach/ssp.h>
 
-- 
cgit v1.2.3


From 54c39b420f017c546fb130a05c14bdb0a7910f98 Mon Sep 17 00:00:00 2001
From: Haojian Zhuang <haojian.zhuang@marvell.com>
Date: Tue, 16 Mar 2010 17:12:37 +0800
Subject: [ARM] pxa: move ssp into common plat-pxa

Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
---
 arch/arm/mach-pxa/Kconfig            |   5 -
 arch/arm/mach-pxa/Makefile           |   1 -
 arch/arm/mach-pxa/include/mach/ssp.h | 196 ------------------------------
 arch/arm/mach-pxa/poodle.c           |   2 +-
 arch/arm/mach-pxa/ssp.c              | 223 -----------------------------------
 arch/arm/plat-pxa/Kconfig            |   5 +
 arch/arm/plat-pxa/Makefile           |   1 +
 arch/arm/plat-pxa/include/plat/ssp.h | 196 ++++++++++++++++++++++++++++++
 arch/arm/plat-pxa/ssp.c              | 223 +++++++++++++++++++++++++++++++++++
 drivers/spi/pxa2xx_spi.c             |   2 +-
 sound/soc/pxa/pxa-ssp.c              |   2 +-
 11 files changed, 428 insertions(+), 428 deletions(-)
 delete mode 100644 arch/arm/mach-pxa/include/mach/ssp.h
 delete mode 100644 arch/arm/mach-pxa/ssp.c
 create mode 100644 arch/arm/plat-pxa/include/plat/ssp.h
 create mode 100644 arch/arm/plat-pxa/ssp.c

(limited to 'sound/soc')

diff --git a/arch/arm/mach-pxa/Kconfig b/arch/arm/mach-pxa/Kconfig
index 8f7eb12776a9..6dd450a7dea6 100644
--- a/arch/arm/mach-pxa/Kconfig
+++ b/arch/arm/mach-pxa/Kconfig
@@ -648,11 +648,6 @@ config PXA_SHARP_Cxx00
 	help
 	  Enable common support for Sharp Cxx00 models
 
-config PXA_SSP
-	tristate
-	help
-	  Enable support for PXA2xx SSP ports
-
 config TOSA_BT
 	tristate "Control the state of built-in bluetooth chip on Sharp SL-6000"
 	depends on MACH_TOSA
diff --git a/arch/arm/mach-pxa/Makefile b/arch/arm/mach-pxa/Makefile
index a83aff386fba..cf57fb2f73d8 100644
--- a/arch/arm/mach-pxa/Makefile
+++ b/arch/arm/mach-pxa/Makefile
@@ -14,7 +14,6 @@ obj-$(CONFIG_PXA3xx)		+= cpufreq-pxa3xx.o
 endif
 
 # Generic drivers that other drivers may depend upon
-obj-$(CONFIG_PXA_SSP)		+= ssp.o
 
 # SoC-specific code
 obj-$(CONFIG_PXA25x)		+= mfp-pxa2xx.o pxa2xx.o pxa25x.o
diff --git a/arch/arm/mach-pxa/include/mach/ssp.h b/arch/arm/mach-pxa/include/mach/ssp.h
deleted file mode 100644
index 4c4ce62390d0..000000000000
--- a/arch/arm/mach-pxa/include/mach/ssp.h
+++ /dev/null
@@ -1,196 +0,0 @@
-/*
- *  ssp.h
- *
- *  Copyright (C) 2003 Russell King, All Rights Reserved.
- *
- * 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 driver supports the following PXA CPU/SSP ports:-
- *
- *       PXA250     SSP
- *       PXA255     SSP, NSSP
- *       PXA26x     SSP, NSSP, ASSP
- *       PXA27x     SSP1, SSP2, SSP3
- *       PXA3xx     SSP1, SSP2, SSP3, SSP4
- */
-
-#ifndef __ASM_ARCH_SSP_H
-#define __ASM_ARCH_SSP_H
-
-#include <linux/list.h>
-#include <linux/io.h>
-
-/*
- * SSP Serial Port Registers
- * PXA250, PXA255, PXA26x and PXA27x SSP controllers are all slightly different.
- * PXA255, PXA26x and PXA27x have extra ports, registers and bits.
- */
-
-#define SSCR0		(0x00)  /* SSP Control Register 0 */
-#define SSCR1		(0x04)  /* SSP Control Register 1 */
-#define SSSR		(0x08)  /* SSP Status Register */
-#define SSITR		(0x0C)  /* SSP Interrupt Test Register */
-#define SSDR		(0x10)  /* SSP Data Write/Data Read Register */
-
-#define SSTO		(0x28)  /* SSP Time Out Register */
-#define SSPSP		(0x2C)  /* SSP Programmable Serial Protocol */
-#define SSTSA		(0x30)  /* SSP Tx Timeslot Active */
-#define SSRSA		(0x34)  /* SSP Rx Timeslot Active */
-#define SSTSS		(0x38)  /* SSP Timeslot Status */
-#define SSACD		(0x3C)  /* SSP Audio Clock Divider */
-
-#if defined(CONFIG_PXA3xx)
-#define SSACDD		(0x40)	/* SSP Audio Clock Dither Divider */
-#endif
-
-/* Common PXA2xx bits first */
-#define SSCR0_DSS	(0x0000000f)	/* Data Size Select (mask) */
-#define SSCR0_DataSize(x)  ((x) - 1)	/* Data Size Select [4..16] */
-#define SSCR0_FRF	(0x00000030)	/* FRame Format (mask) */
-#define SSCR0_Motorola	(0x0 << 4)	/* Motorola's Serial Peripheral Interface (SPI) */
-#define SSCR0_TI	(0x1 << 4)	/* Texas Instruments' Synchronous Serial Protocol (SSP) */
-#define SSCR0_National	(0x2 << 4)	/* National Microwire */
-#define SSCR0_ECS	(1 << 6)	/* External clock select */
-#define SSCR0_SSE	(1 << 7)	/* Synchronous Serial Port Enable */
-#define SSCR0_SCR(x)	((x) << 8)	/* Serial Clock Rate (mask) */
-
-#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx)
-#define SSCR0_EDSS	(1 << 20)	/* Extended data size select */
-#define SSCR0_NCS	(1 << 21)	/* Network clock select */
-#define SSCR0_RIM	(1 << 22)	/* Receive FIFO overrrun interrupt mask */
-#define SSCR0_TUM	(1 << 23)	/* Transmit FIFO underrun interrupt mask */
-#define SSCR0_FRDC	(0x07000000)	/* Frame rate divider control (mask) */
-#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24)	/* Time slots per frame [1..8] */
-#define SSCR0_ACS	(1 << 30)	/* Audio clock select */
-#define SSCR0_MOD	(1 << 31)	/* Mode (normal or network) */
-#endif
-
-#if defined(CONFIG_PXA3xx)
-#define SSCR0_FPCKE	(1 << 29)	/* FIFO packing enable */
-#endif
-
-#define SSCR1_RIE	(1 << 0)	/* Receive FIFO Interrupt Enable */
-#define SSCR1_TIE	(1 << 1)	/* Transmit FIFO Interrupt Enable */
-#define SSCR1_LBM	(1 << 2)	/* Loop-Back Mode */
-#define SSCR1_SPO	(1 << 3)	/* Motorola SPI SSPSCLK polarity setting */
-#define SSCR1_SPH	(1 << 4)	/* Motorola SPI SSPSCLK phase setting */
-#define SSCR1_MWDS	(1 << 5)	/* Microwire Transmit Data Size */
-#define SSCR1_TFT	(0x000003c0)	/* Transmit FIFO Threshold (mask) */
-#define SSCR1_TxTresh(x) (((x) - 1) << 6) /* level [1..16] */
-#define SSCR1_RFT	(0x00003c00)	/* Receive FIFO Threshold (mask) */
-#define SSCR1_RxTresh(x) (((x) - 1) << 10) /* level [1..16] */
-
-#define SSSR_TNF	(1 << 2)	/* Transmit FIFO Not Full */
-#define SSSR_RNE	(1 << 3)	/* Receive FIFO Not Empty */
-#define SSSR_BSY	(1 << 4)	/* SSP Busy */
-#define SSSR_TFS	(1 << 5)	/* Transmit FIFO Service Request */
-#define SSSR_RFS	(1 << 6)	/* Receive FIFO Service Request */
-#define SSSR_ROR	(1 << 7)	/* Receive FIFO Overrun */
-
-#define SSCR0_TIM		(1 << 23)	/* Transmit FIFO Under Run Interrupt Mask */
-#define SSCR0_RIM		(1 << 22)	/* Receive FIFO Over Run interrupt Mask */
-#define SSCR0_NCS		(1 << 21)	/* Network Clock Select */
-#define SSCR0_EDSS		(1 << 20)	/* Extended Data Size Select */
-
-/* extra bits in PXA255, PXA26x and PXA27x SSP ports */
-#define SSCR0_TISSP		(1 << 4)	/* TI Sync Serial Protocol */
-#define SSCR0_PSP		(3 << 4)	/* PSP - Programmable Serial Protocol */
-#define SSCR1_TTELP		(1 << 31)	/* TXD Tristate Enable Last Phase */
-#define SSCR1_TTE		(1 << 30)	/* TXD Tristate Enable */
-#define SSCR1_EBCEI		(1 << 29)	/* Enable Bit Count Error interrupt */
-#define SSCR1_SCFR		(1 << 28)	/* Slave Clock free Running */
-#define SSCR1_ECRA		(1 << 27)	/* Enable Clock Request A */
-#define SSCR1_ECRB		(1 << 26)	/* Enable Clock request B */
-#define SSCR1_SCLKDIR		(1 << 25)	/* Serial Bit Rate Clock Direction */
-#define SSCR1_SFRMDIR		(1 << 24)	/* Frame Direction */
-#define SSCR1_RWOT		(1 << 23)	/* Receive Without Transmit */
-#define SSCR1_TRAIL		(1 << 22)	/* Trailing Byte */
-#define SSCR1_TSRE		(1 << 21)	/* Transmit Service Request Enable */
-#define SSCR1_RSRE		(1 << 20)	/* Receive Service Request Enable */
-#define SSCR1_TINTE		(1 << 19)	/* Receiver Time-out Interrupt enable */
-#define SSCR1_PINTE		(1 << 18)	/* Peripheral Trailing Byte Interupt Enable */
-#define SSCR1_IFS		(1 << 16)	/* Invert Frame Signal */
-#define SSCR1_STRF		(1 << 15)	/* Select FIFO or EFWR */
-#define SSCR1_EFWR		(1 << 14)	/* Enable FIFO Write/Read */
-
-#define SSSR_BCE		(1 << 23)	/* Bit Count Error */
-#define SSSR_CSS		(1 << 22)	/* Clock Synchronisation Status */
-#define SSSR_TUR		(1 << 21)	/* Transmit FIFO Under Run */
-#define SSSR_EOC		(1 << 20)	/* End Of Chain */
-#define SSSR_TINT		(1 << 19)	/* Receiver Time-out Interrupt */
-#define SSSR_PINT		(1 << 18)	/* Peripheral Trailing Byte Interrupt */
-
-#if defined(CONFIG_PXA3xx)
-#define SSPSP_EDMYSTOP(x)	((x) << 28)     /* Extended Dummy Stop */
-#define SSPSP_EDMYSTRT(x)	((x) << 26)     /* Extended Dummy Start */
-#endif
-
-#define SSPSP_FSRT		(1 << 25)	/* Frame Sync Relative Timing */
-#define SSPSP_DMYSTOP(x)	((x) << 23)	/* Dummy Stop */
-#define SSPSP_SFRMWDTH(x)	((x) << 16)	/* Serial Frame Width */
-#define SSPSP_SFRMDLY(x)	((x) << 9)	/* Serial Frame Delay */
-#define SSPSP_DMYSTRT(x)	((x) << 7)	/* Dummy Start */
-#define SSPSP_STRTDLY(x)	((x) << 4)	/* Start Delay */
-#define SSPSP_ETDS		(1 << 3)	/* End of Transfer data State */
-#define SSPSP_SFRMP		(1 << 2)	/* Serial Frame Polarity */
-#define SSPSP_SCMODE(x)		((x) << 0)	/* Serial Bit Rate Clock Mode */
-
-#define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass */
-#define SSACD_ACPS(x)		((x) << 4)	/* Audio clock PLL select */
-#define SSACD_ACDS(x)		((x) << 0)	/* Audio clock divider select */
-#if defined(CONFIG_PXA3xx)
-#define SSACD_SCDX8		(1 << 7)	/* SYSCLK division ratio select */
-#endif
-
-enum pxa_ssp_type {
-	SSP_UNDEFINED = 0,
-	PXA25x_SSP,  /* pxa 210, 250, 255, 26x */
-	PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */
-	PXA27x_SSP,
-};
-
-struct ssp_device {
-	struct platform_device *pdev;
-	struct list_head	node;
-
-	struct clk	*clk;
-	void __iomem	*mmio_base;
-	unsigned long	phys_base;
-
-	const char	*label;
-	int		port_id;
-	int		type;
-	int		use_count;
-	int		irq;
-	int		drcmr_rx;
-	int		drcmr_tx;
-};
-
-/**
- * ssp_write_reg - Write to a SSP register
- *
- * @dev: SSP device to access
- * @reg: Register to write to
- * @val: Value to be written.
- */
-static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val)
-{
-	__raw_writel(val, dev->mmio_base + reg);
-}
-
-/**
- * ssp_read_reg - Read from a SSP register
- *
- * @dev: SSP device to access
- * @reg: Register to read from
- */
-static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg)
-{
-	return __raw_readl(dev->mmio_base + reg);
-}
-
-struct ssp_device *ssp_request(int port, const char *label);
-void ssp_free(struct ssp_device *);
-#endif /* __ASM_ARCH_SSP_H */
diff --git a/arch/arm/mach-pxa/poodle.c b/arch/arm/mach-pxa/poodle.c
index 48b10c783a28..dcb178807046 100644
--- a/arch/arm/mach-pxa/poodle.c
+++ b/arch/arm/mach-pxa/poodle.c
@@ -40,12 +40,12 @@
 #include <mach/pxa25x.h>
 #include <mach/mmc.h>
 #include <mach/udc.h>
-#include <plat/i2c.h>
 #include <mach/irda.h>
 #include <mach/poodle.h>
 #include <mach/pxafb.h>
 #include <mach/sharpsl.h>
 #include <mach/pxa2xx_spi.h>
+#include <plat/i2c.h>
 
 #include <asm/hardware/scoop.h>
 #include <asm/hardware/locomo.h>
diff --git a/arch/arm/mach-pxa/ssp.c b/arch/arm/mach-pxa/ssp.c
deleted file mode 100644
index a9b2d3eb8f32..000000000000
--- a/arch/arm/mach-pxa/ssp.c
+++ /dev/null
@@ -1,223 +0,0 @@
-/*
- *  linux/arch/arm/mach-pxa/ssp.c
- *
- *  based on linux/arch/arm/mach-sa1100/ssp.c by Russell King
- *
- *  Copyright (C) 2003 Russell King.
- *  Copyright (C) 2003 Wolfson Microelectronics PLC
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- *  PXA2xx SSP driver.  This provides the generic core for simple
- *  IO-based SSP applications and allows easy port setup for DMA access.
- *
- *  Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
- */
-
-#include <linux/module.h>
-#include <linux/kernel.h>
-#include <linux/sched.h>
-#include <linux/slab.h>
-#include <linux/errno.h>
-#include <linux/interrupt.h>
-#include <linux/ioport.h>
-#include <linux/init.h>
-#include <linux/mutex.h>
-#include <linux/clk.h>
-#include <linux/err.h>
-#include <linux/platform_device.h>
-#include <linux/io.h>
-
-#include <asm/irq.h>
-#include <mach/hardware.h>
-#include <mach/ssp.h>
-
-static DEFINE_MUTEX(ssp_lock);
-static LIST_HEAD(ssp_list);
-
-struct ssp_device *ssp_request(int port, const char *label)
-{
-	struct ssp_device *ssp = NULL;
-
-	mutex_lock(&ssp_lock);
-
-	list_for_each_entry(ssp, &ssp_list, node) {
-		if (ssp->port_id == port && ssp->use_count == 0) {
-			ssp->use_count++;
-			ssp->label = label;
-			break;
-		}
-	}
-
-	mutex_unlock(&ssp_lock);
-
-	if (&ssp->node == &ssp_list)
-		return NULL;
-
-	return ssp;
-}
-EXPORT_SYMBOL(ssp_request);
-
-void ssp_free(struct ssp_device *ssp)
-{
-	mutex_lock(&ssp_lock);
-	if (ssp->use_count) {
-		ssp->use_count--;
-		ssp->label = NULL;
-	} else
-		dev_err(&ssp->pdev->dev, "device already free\n");
-	mutex_unlock(&ssp_lock);
-}
-EXPORT_SYMBOL(ssp_free);
-
-static int __devinit ssp_probe(struct platform_device *pdev)
-{
-	const struct platform_device_id *id = platform_get_device_id(pdev);
-	struct resource *res;
-	struct ssp_device *ssp;
-	int ret = 0;
-
-	ssp = kzalloc(sizeof(struct ssp_device), GFP_KERNEL);
-	if (ssp == NULL) {
-		dev_err(&pdev->dev, "failed to allocate memory");
-		return -ENOMEM;
-	}
-	ssp->pdev = pdev;
-
-	ssp->clk = clk_get(&pdev->dev, NULL);
-	if (IS_ERR(ssp->clk)) {
-		ret = PTR_ERR(ssp->clk);
-		goto err_free;
-	}
-
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (res == NULL) {
-		dev_err(&pdev->dev, "no memory resource defined\n");
-		ret = -ENODEV;
-		goto err_free_clk;
-	}
-
-	res = request_mem_region(res->start, res->end - res->start + 1,
-			pdev->name);
-	if (res == NULL) {
-		dev_err(&pdev->dev, "failed to request memory resource\n");
-		ret = -EBUSY;
-		goto err_free_clk;
-	}
-
-	ssp->phys_base = res->start;
-
-	ssp->mmio_base = ioremap(res->start, res->end - res->start + 1);
-	if (ssp->mmio_base == NULL) {
-		dev_err(&pdev->dev, "failed to ioremap() registers\n");
-		ret = -ENODEV;
-		goto err_free_mem;
-	}
-
-	ssp->irq = platform_get_irq(pdev, 0);
-	if (ssp->irq < 0) {
-		dev_err(&pdev->dev, "no IRQ resource defined\n");
-		ret = -ENODEV;
-		goto err_free_io;
-	}
-
-	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (res == NULL) {
-		dev_err(&pdev->dev, "no SSP RX DRCMR defined\n");
-		ret = -ENODEV;
-		goto err_free_io;
-	}
-	ssp->drcmr_rx = res->start;
-
-	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (res == NULL) {
-		dev_err(&pdev->dev, "no SSP TX DRCMR defined\n");
-		ret = -ENODEV;
-		goto err_free_io;
-	}
-	ssp->drcmr_tx = res->start;
-
-	/* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id
-	 * starts from 0, do a translation here
-	 */
-	ssp->port_id = pdev->id + 1;
-	ssp->use_count = 0;
-	ssp->type = (int)id->driver_data;
-
-	mutex_lock(&ssp_lock);
-	list_add(&ssp->node, &ssp_list);
-	mutex_unlock(&ssp_lock);
-
-	platform_set_drvdata(pdev, ssp);
-	return 0;
-
-err_free_io:
-	iounmap(ssp->mmio_base);
-err_free_mem:
-	release_mem_region(res->start, res->end - res->start + 1);
-err_free_clk:
-	clk_put(ssp->clk);
-err_free:
-	kfree(ssp);
-	return ret;
-}
-
-static int __devexit ssp_remove(struct platform_device *pdev)
-{
-	struct resource *res;
-	struct ssp_device *ssp;
-
-	ssp = platform_get_drvdata(pdev);
-	if (ssp == NULL)
-		return -ENODEV;
-
-	iounmap(ssp->mmio_base);
-
-	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(res->start, res->end - res->start + 1);
-
-	clk_put(ssp->clk);
-
-	mutex_lock(&ssp_lock);
-	list_del(&ssp->node);
-	mutex_unlock(&ssp_lock);
-
-	kfree(ssp);
-	return 0;
-}
-
-static const struct platform_device_id ssp_id_table[] = {
-	{ "pxa25x-ssp",		PXA25x_SSP },
-	{ "pxa25x-nssp",	PXA25x_NSSP },
-	{ "pxa27x-ssp",		PXA27x_SSP },
-	{ },
-};
-
-static struct platform_driver ssp_driver = {
-	.probe		= ssp_probe,
-	.remove		= __devexit_p(ssp_remove),
-	.driver		= {
-		.owner	= THIS_MODULE,
-		.name	= "pxa2xx-ssp",
-	},
-	.id_table	= ssp_id_table,
-};
-
-static int __init pxa_ssp_init(void)
-{
-	return platform_driver_register(&ssp_driver);
-}
-
-static void __exit pxa_ssp_exit(void)
-{
-	platform_driver_unregister(&ssp_driver);
-}
-
-arch_initcall(pxa_ssp_init);
-module_exit(pxa_ssp_exit);
-
-MODULE_DESCRIPTION("PXA SSP driver");
-MODULE_AUTHOR("Liam Girdwood");
-MODULE_LICENSE("GPL");
diff --git a/arch/arm/plat-pxa/Kconfig b/arch/arm/plat-pxa/Kconfig
index b158e98038ed..da53395a17c6 100644
--- a/arch/arm/plat-pxa/Kconfig
+++ b/arch/arm/plat-pxa/Kconfig
@@ -1,3 +1,8 @@
 if PLAT_PXA
 
+config PXA_SSP
+	tristate
+	help
+	  Enable support for PXA2xx SSP ports
+
 endif
diff --git a/arch/arm/plat-pxa/Makefile b/arch/arm/plat-pxa/Makefile
index 0264bfb0ca4f..4aacdd12c9cc 100644
--- a/arch/arm/plat-pxa/Makefile
+++ b/arch/arm/plat-pxa/Makefile
@@ -9,3 +9,4 @@ obj-$(CONFIG_PXA3xx)		+= mfp.o
 obj-$(CONFIG_ARCH_MMP)		+= mfp.o
 
 obj-$(CONFIG_HAVE_PWM)		+= pwm.o
+obj-$(CONFIG_PXA_SSP)		+= ssp.o
diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h
new file mode 100644
index 000000000000..4c4ce62390d0
--- /dev/null
+++ b/arch/arm/plat-pxa/include/plat/ssp.h
@@ -0,0 +1,196 @@
+/*
+ *  ssp.h
+ *
+ *  Copyright (C) 2003 Russell King, All Rights Reserved.
+ *
+ * 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 driver supports the following PXA CPU/SSP ports:-
+ *
+ *       PXA250     SSP
+ *       PXA255     SSP, NSSP
+ *       PXA26x     SSP, NSSP, ASSP
+ *       PXA27x     SSP1, SSP2, SSP3
+ *       PXA3xx     SSP1, SSP2, SSP3, SSP4
+ */
+
+#ifndef __ASM_ARCH_SSP_H
+#define __ASM_ARCH_SSP_H
+
+#include <linux/list.h>
+#include <linux/io.h>
+
+/*
+ * SSP Serial Port Registers
+ * PXA250, PXA255, PXA26x and PXA27x SSP controllers are all slightly different.
+ * PXA255, PXA26x and PXA27x have extra ports, registers and bits.
+ */
+
+#define SSCR0		(0x00)  /* SSP Control Register 0 */
+#define SSCR1		(0x04)  /* SSP Control Register 1 */
+#define SSSR		(0x08)  /* SSP Status Register */
+#define SSITR		(0x0C)  /* SSP Interrupt Test Register */
+#define SSDR		(0x10)  /* SSP Data Write/Data Read Register */
+
+#define SSTO		(0x28)  /* SSP Time Out Register */
+#define SSPSP		(0x2C)  /* SSP Programmable Serial Protocol */
+#define SSTSA		(0x30)  /* SSP Tx Timeslot Active */
+#define SSRSA		(0x34)  /* SSP Rx Timeslot Active */
+#define SSTSS		(0x38)  /* SSP Timeslot Status */
+#define SSACD		(0x3C)  /* SSP Audio Clock Divider */
+
+#if defined(CONFIG_PXA3xx)
+#define SSACDD		(0x40)	/* SSP Audio Clock Dither Divider */
+#endif
+
+/* Common PXA2xx bits first */
+#define SSCR0_DSS	(0x0000000f)	/* Data Size Select (mask) */
+#define SSCR0_DataSize(x)  ((x) - 1)	/* Data Size Select [4..16] */
+#define SSCR0_FRF	(0x00000030)	/* FRame Format (mask) */
+#define SSCR0_Motorola	(0x0 << 4)	/* Motorola's Serial Peripheral Interface (SPI) */
+#define SSCR0_TI	(0x1 << 4)	/* Texas Instruments' Synchronous Serial Protocol (SSP) */
+#define SSCR0_National	(0x2 << 4)	/* National Microwire */
+#define SSCR0_ECS	(1 << 6)	/* External clock select */
+#define SSCR0_SSE	(1 << 7)	/* Synchronous Serial Port Enable */
+#define SSCR0_SCR(x)	((x) << 8)	/* Serial Clock Rate (mask) */
+
+#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx)
+#define SSCR0_EDSS	(1 << 20)	/* Extended data size select */
+#define SSCR0_NCS	(1 << 21)	/* Network clock select */
+#define SSCR0_RIM	(1 << 22)	/* Receive FIFO overrrun interrupt mask */
+#define SSCR0_TUM	(1 << 23)	/* Transmit FIFO underrun interrupt mask */
+#define SSCR0_FRDC	(0x07000000)	/* Frame rate divider control (mask) */
+#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24)	/* Time slots per frame [1..8] */
+#define SSCR0_ACS	(1 << 30)	/* Audio clock select */
+#define SSCR0_MOD	(1 << 31)	/* Mode (normal or network) */
+#endif
+
+#if defined(CONFIG_PXA3xx)
+#define SSCR0_FPCKE	(1 << 29)	/* FIFO packing enable */
+#endif
+
+#define SSCR1_RIE	(1 << 0)	/* Receive FIFO Interrupt Enable */
+#define SSCR1_TIE	(1 << 1)	/* Transmit FIFO Interrupt Enable */
+#define SSCR1_LBM	(1 << 2)	/* Loop-Back Mode */
+#define SSCR1_SPO	(1 << 3)	/* Motorola SPI SSPSCLK polarity setting */
+#define SSCR1_SPH	(1 << 4)	/* Motorola SPI SSPSCLK phase setting */
+#define SSCR1_MWDS	(1 << 5)	/* Microwire Transmit Data Size */
+#define SSCR1_TFT	(0x000003c0)	/* Transmit FIFO Threshold (mask) */
+#define SSCR1_TxTresh(x) (((x) - 1) << 6) /* level [1..16] */
+#define SSCR1_RFT	(0x00003c00)	/* Receive FIFO Threshold (mask) */
+#define SSCR1_RxTresh(x) (((x) - 1) << 10) /* level [1..16] */
+
+#define SSSR_TNF	(1 << 2)	/* Transmit FIFO Not Full */
+#define SSSR_RNE	(1 << 3)	/* Receive FIFO Not Empty */
+#define SSSR_BSY	(1 << 4)	/* SSP Busy */
+#define SSSR_TFS	(1 << 5)	/* Transmit FIFO Service Request */
+#define SSSR_RFS	(1 << 6)	/* Receive FIFO Service Request */
+#define SSSR_ROR	(1 << 7)	/* Receive FIFO Overrun */
+
+#define SSCR0_TIM		(1 << 23)	/* Transmit FIFO Under Run Interrupt Mask */
+#define SSCR0_RIM		(1 << 22)	/* Receive FIFO Over Run interrupt Mask */
+#define SSCR0_NCS		(1 << 21)	/* Network Clock Select */
+#define SSCR0_EDSS		(1 << 20)	/* Extended Data Size Select */
+
+/* extra bits in PXA255, PXA26x and PXA27x SSP ports */
+#define SSCR0_TISSP		(1 << 4)	/* TI Sync Serial Protocol */
+#define SSCR0_PSP		(3 << 4)	/* PSP - Programmable Serial Protocol */
+#define SSCR1_TTELP		(1 << 31)	/* TXD Tristate Enable Last Phase */
+#define SSCR1_TTE		(1 << 30)	/* TXD Tristate Enable */
+#define SSCR1_EBCEI		(1 << 29)	/* Enable Bit Count Error interrupt */
+#define SSCR1_SCFR		(1 << 28)	/* Slave Clock free Running */
+#define SSCR1_ECRA		(1 << 27)	/* Enable Clock Request A */
+#define SSCR1_ECRB		(1 << 26)	/* Enable Clock request B */
+#define SSCR1_SCLKDIR		(1 << 25)	/* Serial Bit Rate Clock Direction */
+#define SSCR1_SFRMDIR		(1 << 24)	/* Frame Direction */
+#define SSCR1_RWOT		(1 << 23)	/* Receive Without Transmit */
+#define SSCR1_TRAIL		(1 << 22)	/* Trailing Byte */
+#define SSCR1_TSRE		(1 << 21)	/* Transmit Service Request Enable */
+#define SSCR1_RSRE		(1 << 20)	/* Receive Service Request Enable */
+#define SSCR1_TINTE		(1 << 19)	/* Receiver Time-out Interrupt enable */
+#define SSCR1_PINTE		(1 << 18)	/* Peripheral Trailing Byte Interupt Enable */
+#define SSCR1_IFS		(1 << 16)	/* Invert Frame Signal */
+#define SSCR1_STRF		(1 << 15)	/* Select FIFO or EFWR */
+#define SSCR1_EFWR		(1 << 14)	/* Enable FIFO Write/Read */
+
+#define SSSR_BCE		(1 << 23)	/* Bit Count Error */
+#define SSSR_CSS		(1 << 22)	/* Clock Synchronisation Status */
+#define SSSR_TUR		(1 << 21)	/* Transmit FIFO Under Run */
+#define SSSR_EOC		(1 << 20)	/* End Of Chain */
+#define SSSR_TINT		(1 << 19)	/* Receiver Time-out Interrupt */
+#define SSSR_PINT		(1 << 18)	/* Peripheral Trailing Byte Interrupt */
+
+#if defined(CONFIG_PXA3xx)
+#define SSPSP_EDMYSTOP(x)	((x) << 28)     /* Extended Dummy Stop */
+#define SSPSP_EDMYSTRT(x)	((x) << 26)     /* Extended Dummy Start */
+#endif
+
+#define SSPSP_FSRT		(1 << 25)	/* Frame Sync Relative Timing */
+#define SSPSP_DMYSTOP(x)	((x) << 23)	/* Dummy Stop */
+#define SSPSP_SFRMWDTH(x)	((x) << 16)	/* Serial Frame Width */
+#define SSPSP_SFRMDLY(x)	((x) << 9)	/* Serial Frame Delay */
+#define SSPSP_DMYSTRT(x)	((x) << 7)	/* Dummy Start */
+#define SSPSP_STRTDLY(x)	((x) << 4)	/* Start Delay */
+#define SSPSP_ETDS		(1 << 3)	/* End of Transfer data State */
+#define SSPSP_SFRMP		(1 << 2)	/* Serial Frame Polarity */
+#define SSPSP_SCMODE(x)		((x) << 0)	/* Serial Bit Rate Clock Mode */
+
+#define SSACD_SCDB		(1 << 3)	/* SSPSYSCLK Divider Bypass */
+#define SSACD_ACPS(x)		((x) << 4)	/* Audio clock PLL select */
+#define SSACD_ACDS(x)		((x) << 0)	/* Audio clock divider select */
+#if defined(CONFIG_PXA3xx)
+#define SSACD_SCDX8		(1 << 7)	/* SYSCLK division ratio select */
+#endif
+
+enum pxa_ssp_type {
+	SSP_UNDEFINED = 0,
+	PXA25x_SSP,  /* pxa 210, 250, 255, 26x */
+	PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */
+	PXA27x_SSP,
+};
+
+struct ssp_device {
+	struct platform_device *pdev;
+	struct list_head	node;
+
+	struct clk	*clk;
+	void __iomem	*mmio_base;
+	unsigned long	phys_base;
+
+	const char	*label;
+	int		port_id;
+	int		type;
+	int		use_count;
+	int		irq;
+	int		drcmr_rx;
+	int		drcmr_tx;
+};
+
+/**
+ * ssp_write_reg - Write to a SSP register
+ *
+ * @dev: SSP device to access
+ * @reg: Register to write to
+ * @val: Value to be written.
+ */
+static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val)
+{
+	__raw_writel(val, dev->mmio_base + reg);
+}
+
+/**
+ * ssp_read_reg - Read from a SSP register
+ *
+ * @dev: SSP device to access
+ * @reg: Register to read from
+ */
+static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg)
+{
+	return __raw_readl(dev->mmio_base + reg);
+}
+
+struct ssp_device *ssp_request(int port, const char *label);
+void ssp_free(struct ssp_device *);
+#endif /* __ASM_ARCH_SSP_H */
diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c
new file mode 100644
index 000000000000..5fe230178f0a
--- /dev/null
+++ b/arch/arm/plat-pxa/ssp.c
@@ -0,0 +1,223 @@
+/*
+ *  linux/arch/arm/mach-pxa/ssp.c
+ *
+ *  based on linux/arch/arm/mach-sa1100/ssp.c by Russell King
+ *
+ *  Copyright (C) 2003 Russell King.
+ *  Copyright (C) 2003 Wolfson Microelectronics PLC
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ *  PXA2xx SSP driver.  This provides the generic core for simple
+ *  IO-based SSP applications and allows easy port setup for DMA access.
+ *
+ *  Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/interrupt.h>
+#include <linux/ioport.h>
+#include <linux/init.h>
+#include <linux/mutex.h>
+#include <linux/clk.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <asm/irq.h>
+#include <mach/hardware.h>
+#include <plat/ssp.h>
+
+static DEFINE_MUTEX(ssp_lock);
+static LIST_HEAD(ssp_list);
+
+struct ssp_device *ssp_request(int port, const char *label)
+{
+	struct ssp_device *ssp = NULL;
+
+	mutex_lock(&ssp_lock);
+
+	list_for_each_entry(ssp, &ssp_list, node) {
+		if (ssp->port_id == port && ssp->use_count == 0) {
+			ssp->use_count++;
+			ssp->label = label;
+			break;
+		}
+	}
+
+	mutex_unlock(&ssp_lock);
+
+	if (&ssp->node == &ssp_list)
+		return NULL;
+
+	return ssp;
+}
+EXPORT_SYMBOL(ssp_request);
+
+void ssp_free(struct ssp_device *ssp)
+{
+	mutex_lock(&ssp_lock);
+	if (ssp->use_count) {
+		ssp->use_count--;
+		ssp->label = NULL;
+	} else
+		dev_err(&ssp->pdev->dev, "device already free\n");
+	mutex_unlock(&ssp_lock);
+}
+EXPORT_SYMBOL(ssp_free);
+
+static int __devinit ssp_probe(struct platform_device *pdev)
+{
+	const struct platform_device_id *id = platform_get_device_id(pdev);
+	struct resource *res;
+	struct ssp_device *ssp;
+	int ret = 0;
+
+	ssp = kzalloc(sizeof(struct ssp_device), GFP_KERNEL);
+	if (ssp == NULL) {
+		dev_err(&pdev->dev, "failed to allocate memory");
+		return -ENOMEM;
+	}
+	ssp->pdev = pdev;
+
+	ssp->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(ssp->clk)) {
+		ret = PTR_ERR(ssp->clk);
+		goto err_free;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "no memory resource defined\n");
+		ret = -ENODEV;
+		goto err_free_clk;
+	}
+
+	res = request_mem_region(res->start, res->end - res->start + 1,
+			pdev->name);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "failed to request memory resource\n");
+		ret = -EBUSY;
+		goto err_free_clk;
+	}
+
+	ssp->phys_base = res->start;
+
+	ssp->mmio_base = ioremap(res->start, res->end - res->start + 1);
+	if (ssp->mmio_base == NULL) {
+		dev_err(&pdev->dev, "failed to ioremap() registers\n");
+		ret = -ENODEV;
+		goto err_free_mem;
+	}
+
+	ssp->irq = platform_get_irq(pdev, 0);
+	if (ssp->irq < 0) {
+		dev_err(&pdev->dev, "no IRQ resource defined\n");
+		ret = -ENODEV;
+		goto err_free_io;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "no SSP RX DRCMR defined\n");
+		ret = -ENODEV;
+		goto err_free_io;
+	}
+	ssp->drcmr_rx = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (res == NULL) {
+		dev_err(&pdev->dev, "no SSP TX DRCMR defined\n");
+		ret = -ENODEV;
+		goto err_free_io;
+	}
+	ssp->drcmr_tx = res->start;
+
+	/* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id
+	 * starts from 0, do a translation here
+	 */
+	ssp->port_id = pdev->id + 1;
+	ssp->use_count = 0;
+	ssp->type = (int)id->driver_data;
+
+	mutex_lock(&ssp_lock);
+	list_add(&ssp->node, &ssp_list);
+	mutex_unlock(&ssp_lock);
+
+	platform_set_drvdata(pdev, ssp);
+	return 0;
+
+err_free_io:
+	iounmap(ssp->mmio_base);
+err_free_mem:
+	release_mem_region(res->start, res->end - res->start + 1);
+err_free_clk:
+	clk_put(ssp->clk);
+err_free:
+	kfree(ssp);
+	return ret;
+}
+
+static int __devexit ssp_remove(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct ssp_device *ssp;
+
+	ssp = platform_get_drvdata(pdev);
+	if (ssp == NULL)
+		return -ENODEV;
+
+	iounmap(ssp->mmio_base);
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	release_mem_region(res->start, res->end - res->start + 1);
+
+	clk_put(ssp->clk);
+
+	mutex_lock(&ssp_lock);
+	list_del(&ssp->node);
+	mutex_unlock(&ssp_lock);
+
+	kfree(ssp);
+	return 0;
+}
+
+static const struct platform_device_id ssp_id_table[] = {
+	{ "pxa25x-ssp",		PXA25x_SSP },
+	{ "pxa25x-nssp",	PXA25x_NSSP },
+	{ "pxa27x-ssp",		PXA27x_SSP },
+	{ },
+};
+
+static struct platform_driver ssp_driver = {
+	.probe		= ssp_probe,
+	.remove		= __devexit_p(ssp_remove),
+	.driver		= {
+		.owner	= THIS_MODULE,
+		.name	= "pxa2xx-ssp",
+	},
+	.id_table	= ssp_id_table,
+};
+
+static int __init pxa_ssp_init(void)
+{
+	return platform_driver_register(&ssp_driver);
+}
+
+static void __exit pxa_ssp_exit(void)
+{
+	platform_driver_unregister(&ssp_driver);
+}
+
+arch_initcall(pxa_ssp_init);
+module_exit(pxa_ssp_exit);
+
+MODULE_DESCRIPTION("PXA SSP driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c
index 959e78bb2f04..50d6b780bc67 100644
--- a/drivers/spi/pxa2xx_spi.c
+++ b/drivers/spi/pxa2xx_spi.c
@@ -36,7 +36,7 @@
 #include <asm/delay.h>
 
 #include <mach/dma.h>
-#include <mach/ssp.h>
+#include <plat/ssp.h>
 #include <mach/pxa2xx_spi.h>
 
 MODULE_AUTHOR("Stephen Street");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 2e27962220ad..6271a100f355 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -33,7 +33,7 @@
 #include <mach/hardware.h>
 #include <mach/dma.h>
 #include <mach/audio.h>
-#include <mach/ssp.h>
+#include <plat/ssp.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa-ssp.h"
-- 
cgit v1.2.3


From 866d091dcb043594626d4facf7c6fcffcc7c8d35 Mon Sep 17 00:00:00 2001
From: Eric Miao <eric.y.miao@gmail.com>
Date: Tue, 16 Mar 2010 17:30:15 +0800
Subject: [ARM] pxa: remove incorrect select PXA_SSP in Kconfig

PXA_SSP is actually used by drivers like drivers/spi/pxa2xx_spi.c and
sound/soc/pxa/pxa-ssp.c, not by boards. Remove those incorrect 'select'
from Kconfig and make SOC_PXA_SSP to select.

Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
---
 arch/arm/mach-pxa/Kconfig | 13 -------------
 sound/soc/pxa/Kconfig     |  1 +
 2 files changed, 1 insertion(+), 13 deletions(-)

(limited to 'sound/soc')

diff --git a/arch/arm/mach-pxa/Kconfig b/arch/arm/mach-pxa/Kconfig
index 6dd450a7dea6..352dc4f3820c 100644
--- a/arch/arm/mach-pxa/Kconfig
+++ b/arch/arm/mach-pxa/Kconfig
@@ -19,7 +19,6 @@ config MACH_MAINSTONE
 config MACH_ZYLONITE
 	bool
 	select PXA3xx
-	select PXA_SSP
 	select HAVE_PWM
 	select PXA_HAVE_BOARD_IRQS
 
@@ -39,7 +38,6 @@ config MACH_LITTLETON
 	select PXA3xx
 	select CPU_PXA300
 	select CPU_PXA310
-	select PXA_SSP
 
 config MACH_TAVOREVB
 	bool "PXA930 Evaluation Board (aka TavorEVB)"
@@ -98,7 +96,6 @@ config MACH_ARMCORE
 	select PXA27x
 	select IWMMXT
 	select PXA25x
-	select PXA_SSP
 
 config MACH_EM_X270
 	bool "CompuLab EM-x270 platform"
@@ -161,7 +158,6 @@ config MACH_XCEP
 	select MTD_CFI
 	select MTD_CHAR
 	select SMC91X
-	select PXA_SSP
 	help
 	  PXA255 based Single Board Computer with SMC 91C111 ethernet chip and 64 MB of flash.
 	  Tuned for usage in Libera instruments for particle accelerators.
@@ -180,7 +176,6 @@ config MACH_TRIZEPS4WL
 	depends on TRIZEPS_PXA
 	select TRIZEPS_PCMCIA
 	select PXA27x
-	select PXA_SSP
 
 choice
 	prompt "Select base board for Trizeps module"
@@ -217,7 +212,6 @@ config MACH_PCM027
 	bool "Phytec phyCORE-PXA270 CPU module (PCM-027)"
 	select PXA27x
 	select IWMMXT
-	select PXA_SSP
 	select PXA_HAVE_BOARD_IRQS
 
 config MACH_PCM990_BASEBOARD
@@ -261,7 +255,6 @@ config MACH_H4700
 	bool "HP iPAQ hx4700"
 	select PXA27x
 	select IWMMXT
-	select PXA_SSP
 	select HAVE_PWM
 	select PXA_HAVE_BOARD_IRQS
 
@@ -277,7 +270,6 @@ config MACH_MAGICIAN
 	bool "Enable HTC Magician Support"
 	select PXA27x
 	select IWMMXT
-	select PXA_SSP
 	select HAVE_PWM
 	select PXA_HAVE_BOARD_IRQS
 
@@ -431,13 +423,11 @@ config MACH_RAUMFELD_CONNECTOR
 	bool "Raumfeld Connector"
 	select PXA3xx
 	select CPU_PXA300
-	select PXA_SSP
 
 config MACH_RAUMFELD_SPEAKER
 	bool "Raumfeld Speaker"
 	select PXA3xx
 	select CPU_PXA300
-	select PXA_SSP
 
 config PXA_SHARPSL
 	bool "SHARP Zaurus SL-5600, SL-C7xx and SL-Cxx00 Models"
@@ -466,7 +456,6 @@ config MACH_POODLE
 	depends on PXA_SHARPSL
 	select PXA25x
 	select SHARP_LOCOMO
-	select PXA_SSP
 	select PXA_HAVE_BOARD_IRQS
 
 config MACH_CORGI
@@ -636,14 +625,12 @@ config CPU_PXA950
 
 config PXA_SHARP_C7xx
 	bool
-	select PXA_SSP
 	select SHARPSL_PM
 	help
 	  Enable support for all Sharp C7xx models
 
 config PXA_SHARP_Cxx00
 	bool
-	select PXA_SSP
 	select SHARPSL_PM
 	help
 	  Enable common support for Sharp Cxx00 models
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 376e14a9c273..78e612100782 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -23,6 +23,7 @@ config SND_PXA2XX_SOC_I2S
 
 config SND_PXA_SOC_SSP
 	tristate
+	select PXA_SSP
 
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
-- 
cgit v1.2.3


From baffe1699c68da5040de6b274054adc995795782 Mon Sep 17 00:00:00 2001
From: Haojian Zhuang <haojian.zhuang@marvell.com>
Date: Wed, 5 May 2010 10:11:15 -0400
Subject: [ARM] pxa: add namespace on ssp

In order to prevent code ambiguous, add namespace on functions in ssp driver.

Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
---
 arch/arm/plat-pxa/include/plat/ssp.h |  12 ++--
 arch/arm/plat-pxa/ssp.c              |  22 +++---
 drivers/spi/pxa2xx_spi.c             |   8 +--
 sound/soc/pxa/pxa-ssp.c              | 132 +++++++++++++++++------------------
 4 files changed, 87 insertions(+), 87 deletions(-)

(limited to 'sound/soc')

diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h
index d16d79a3a48a..fe43150690ed 100644
--- a/arch/arm/plat-pxa/include/plat/ssp.h
+++ b/arch/arm/plat-pxa/include/plat/ssp.h
@@ -159,28 +159,28 @@ struct ssp_device {
 };
 
 /**
- * ssp_write_reg - Write to a SSP register
+ * pxa_ssp_write_reg - Write to a SSP register
  *
  * @dev: SSP device to access
  * @reg: Register to write to
  * @val: Value to be written.
  */
-static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val)
+static inline void pxa_ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val)
 {
 	__raw_writel(val, dev->mmio_base + reg);
 }
 
 /**
- * ssp_read_reg - Read from a SSP register
+ * pxa_ssp_read_reg - Read from a SSP register
  *
  * @dev: SSP device to access
  * @reg: Register to read from
  */
-static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg)
+static inline u32 pxa_ssp_read_reg(struct ssp_device *dev, u32 reg)
 {
 	return __raw_readl(dev->mmio_base + reg);
 }
 
-struct ssp_device *ssp_request(int port, const char *label);
-void ssp_free(struct ssp_device *);
+struct ssp_device *pxa_ssp_request(int port, const char *label);
+void pxa_ssp_free(struct ssp_device *);
 #endif /* __ASM_ARCH_SSP_H */
diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c
index 52c07cc756d7..c6357e554aba 100644
--- a/arch/arm/plat-pxa/ssp.c
+++ b/arch/arm/plat-pxa/ssp.c
@@ -37,7 +37,7 @@
 static DEFINE_MUTEX(ssp_lock);
 static LIST_HEAD(ssp_list);
 
-struct ssp_device *ssp_request(int port, const char *label)
+struct ssp_device *pxa_ssp_request(int port, const char *label)
 {
 	struct ssp_device *ssp = NULL;
 
@@ -58,9 +58,9 @@ struct ssp_device *ssp_request(int port, const char *label)
 
 	return ssp;
 }
-EXPORT_SYMBOL(ssp_request);
+EXPORT_SYMBOL(pxa_ssp_request);
 
-void ssp_free(struct ssp_device *ssp)
+void pxa_ssp_free(struct ssp_device *ssp)
 {
 	mutex_lock(&ssp_lock);
 	if (ssp->use_count) {
@@ -70,9 +70,9 @@ void ssp_free(struct ssp_device *ssp)
 		dev_err(&ssp->pdev->dev, "device already free\n");
 	mutex_unlock(&ssp_lock);
 }
-EXPORT_SYMBOL(ssp_free);
+EXPORT_SYMBOL(pxa_ssp_free);
 
-static int __devinit ssp_probe(struct platform_device *pdev)
+static int __devinit pxa_ssp_probe(struct platform_device *pdev)
 {
 	const struct platform_device_id *id = platform_get_device_id(pdev);
 	struct resource *res;
@@ -164,7 +164,7 @@ err_free:
 	return ret;
 }
 
-static int __devexit ssp_remove(struct platform_device *pdev)
+static int __devexit pxa_ssp_remove(struct platform_device *pdev)
 {
 	struct resource *res;
 	struct ssp_device *ssp;
@@ -196,9 +196,9 @@ static const struct platform_device_id ssp_id_table[] = {
 	{ },
 };
 
-static struct platform_driver ssp_driver = {
-	.probe		= ssp_probe,
-	.remove		= __devexit_p(ssp_remove),
+static struct platform_driver pxa_ssp_driver = {
+	.probe		= pxa_ssp_probe,
+	.remove		= __devexit_p(pxa_ssp_remove),
 	.driver		= {
 		.owner	= THIS_MODULE,
 		.name	= "pxa2xx-ssp",
@@ -208,12 +208,12 @@ static struct platform_driver ssp_driver = {
 
 static int __init pxa_ssp_init(void)
 {
-	return platform_driver_register(&ssp_driver);
+	return platform_driver_register(&pxa_ssp_driver);
 }
 
 static void __exit pxa_ssp_exit(void)
 {
-	platform_driver_unregister(&ssp_driver);
+	platform_driver_unregister(&pxa_ssp_driver);
 }
 
 arch_initcall(pxa_ssp_init);
diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c
index 50d6b780bc67..e76b1afafe07 100644
--- a/drivers/spi/pxa2xx_spi.c
+++ b/drivers/spi/pxa2xx_spi.c
@@ -1465,7 +1465,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev)
 
 	platform_info = dev->platform_data;
 
-	ssp = ssp_request(pdev->id, pdev->name);
+	ssp = pxa_ssp_request(pdev->id, pdev->name);
 	if (ssp == NULL) {
 		dev_err(&pdev->dev, "failed to request SSP%d\n", pdev->id);
 		return -ENODEV;
@@ -1475,7 +1475,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev)
 	master = spi_alloc_master(dev, sizeof(struct driver_data) + 16);
 	if (!master) {
 		dev_err(&pdev->dev, "cannot alloc spi_master\n");
-		ssp_free(ssp);
+		pxa_ssp_free(ssp);
 		return -ENOMEM;
 	}
 	drv_data = spi_master_get_devdata(master);
@@ -1604,7 +1604,7 @@ out_error_irq_alloc:
 
 out_error_master_alloc:
 	spi_master_put(master);
-	ssp_free(ssp);
+	pxa_ssp_free(ssp);
 	return status;
 }
 
@@ -1648,7 +1648,7 @@ static int pxa2xx_spi_remove(struct platform_device *pdev)
 	free_irq(ssp->irq, drv_data);
 
 	/* Release SSP */
-	ssp_free(ssp);
+	pxa_ssp_free(ssp);
 
 	/* Disconnect from the SPI framework */
 	spi_unregister_master(drv_data->master);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 6271a100f355..a1fd23e0e3d0 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -56,15 +56,15 @@ struct ssp_priv {
 static void dump_registers(struct ssp_device *ssp)
 {
 	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
-		 ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1),
-		 ssp_read_reg(ssp, SSTO));
+		 pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1),
+		 pxa_ssp_read_reg(ssp, SSTO));
 
 	dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
-		 ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR),
-		 ssp_read_reg(ssp, SSACD));
+		 pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR),
+		 pxa_ssp_read_reg(ssp, SSACD));
 }
 
-static void ssp_enable(struct ssp_device *ssp)
+static void pxa_ssp_enable(struct ssp_device *ssp)
 {
 	uint32_t sscr0;
 
@@ -72,7 +72,7 @@ static void ssp_enable(struct ssp_device *ssp)
 	__raw_writel(sscr0, ssp->mmio_base + SSCR0);
 }
 
-static void ssp_disable(struct ssp_device *ssp)
+static void pxa_ssp_disable(struct ssp_device *ssp)
 {
 	uint32_t sscr0;
 
@@ -86,7 +86,7 @@ struct pxa2xx_pcm_dma_data {
 };
 
 static struct pxa2xx_pcm_dma_params *
-ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
+pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
 {
 	struct pxa2xx_pcm_dma_data *dma;
 
@@ -118,7 +118,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 
 	if (!cpu_dai->active) {
 		clk_enable(ssp->clk);
-		ssp_disable(ssp);
+		pxa_ssp_disable(ssp);
 	}
 
 	kfree(snd_soc_dai_get_dma_data(cpu_dai, substream));
@@ -136,7 +136,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 	struct ssp_device *ssp = priv->ssp;
 
 	if (!cpu_dai->active) {
-		ssp_disable(ssp);
+		pxa_ssp_disable(ssp);
 		clk_disable(ssp->clk);
 	}
 
@@ -159,7 +159,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
 	priv->to  = __raw_readl(ssp->mmio_base + SSTO);
 	priv->psp = __raw_readl(ssp->mmio_base + SSPSP);
 
-	ssp_disable(ssp);
+	pxa_ssp_disable(ssp);
 	clk_disable(ssp->clk);
 	return 0;
 }
@@ -179,7 +179,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
 	__raw_writel(priv->psp, ssp->mmio_base + SSPSP);
 
 	if (cpu_dai->active)
-		ssp_enable(ssp);
+		pxa_ssp_enable(ssp);
 	else
 		clk_disable(ssp->clk);
 
@@ -195,9 +195,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
  * ssp_set_clkdiv - set SSP clock divider
  * @div: serial clock rate divider
  */
-static void ssp_set_scr(struct ssp_device *ssp, u32 div)
+static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
 {
-	u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 
 	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
 		sscr0 &= ~0x0000ff00;
@@ -206,15 +206,15 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div)
 		sscr0 &= ~0x000fff00;
 		sscr0 |= (div - 1) << 8;     /* 1..4096 */
 	}
-	ssp_write_reg(ssp, SSCR0, sscr0);
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 }
 
 /**
- * ssp_get_clkdiv - get SSP clock divider
+ * pxa_ssp_get_clkdiv - get SSP clock divider
  */
-static u32 ssp_get_scr(struct ssp_device *ssp)
+static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
 {
-	u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 	u32 div;
 
 	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
@@ -234,7 +234,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	struct ssp_device *ssp = priv->ssp;
 	int val;
 
-	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
+	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
 		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
 	dev_dbg(&ssp->pdev->dev,
@@ -262,7 +262,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		break;
 	case PXA_SSP_CLK_AUDIO:
 		priv->sysclk = 0;
-		ssp_set_scr(ssp, 1);
+		pxa_ssp_set_scr(ssp, 1);
 		sscr0 |= SSCR0_ACS;
 		break;
 	default:
@@ -273,8 +273,8 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
 	if (!cpu_is_pxa3xx())
 		clk_disable(ssp->clk);
-	val = ssp_read_reg(ssp, SSCR0) | sscr0;
-	ssp_write_reg(ssp, SSCR0, val);
+	val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
+	pxa_ssp_write_reg(ssp, SSCR0, val);
 	if (!cpu_is_pxa3xx())
 		clk_enable(ssp->clk);
 
@@ -293,11 +293,11 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 
 	switch (div_id) {
 	case PXA_SSP_AUDIO_DIV_ACDS:
-		val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
-		ssp_write_reg(ssp, SSACD, val);
+		val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+		pxa_ssp_write_reg(ssp, SSACD, val);
 		break;
 	case PXA_SSP_AUDIO_DIV_SCDB:
-		val = ssp_read_reg(ssp, SSACD);
+		val = pxa_ssp_read_reg(ssp, SSACD);
 		val &= ~SSACD_SCDB;
 #if defined(CONFIG_PXA3xx)
 		if (cpu_is_pxa3xx())
@@ -320,10 +320,10 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 		default:
 			return -EINVAL;
 		}
-		ssp_write_reg(ssp, SSACD, val);
+		pxa_ssp_write_reg(ssp, SSACD, val);
 		break;
 	case PXA_SSP_DIV_SCR:
-		ssp_set_scr(ssp, div);
+		pxa_ssp_set_scr(ssp, div);
 		break;
 	default:
 		return -ENODEV;
@@ -340,11 +340,11 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 {
 	struct ssp_priv *priv = cpu_dai->private_data;
 	struct ssp_device *ssp = priv->ssp;
-	u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70;
+	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
 
 #if defined(CONFIG_PXA3xx)
 	if (cpu_is_pxa3xx())
-		ssp_write_reg(ssp, SSACDD, 0);
+		pxa_ssp_write_reg(ssp, SSACDD, 0);
 #endif
 
 	switch (freq_out) {
@@ -382,7 +382,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 			val = tmp;
 
 			val = (val << 16) | 64;
-			ssp_write_reg(ssp, SSACDD, val);
+			pxa_ssp_write_reg(ssp, SSACDD, val);
 
 			ssacd |= (0x6 << 4);
 
@@ -396,7 +396,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 		return -EINVAL;
 	}
 
-	ssp_write_reg(ssp, SSACD, ssacd);
+	pxa_ssp_write_reg(ssp, SSACD, ssacd);
 
 	return 0;
 }
@@ -411,7 +411,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 	struct ssp_device *ssp = priv->ssp;
 	u32 sscr0;
 
-	sscr0 = ssp_read_reg(ssp, SSCR0);
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 	sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS);
 
 	/* set slot width */
@@ -428,10 +428,10 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 		sscr0 |= SSCR0_SlotsPerFrm(slots);
 
 		/* set active slot mask */
-		ssp_write_reg(ssp, SSTSA, tx_mask);
-		ssp_write_reg(ssp, SSRSA, rx_mask);
+		pxa_ssp_write_reg(ssp, SSTSA, tx_mask);
+		pxa_ssp_write_reg(ssp, SSRSA, rx_mask);
 	}
-	ssp_write_reg(ssp, SSCR0, sscr0);
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 
 	return 0;
 }
@@ -446,12 +446,12 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
 	struct ssp_device *ssp = priv->ssp;
 	u32 sscr1;
 
-	sscr1 = ssp_read_reg(ssp, SSCR1);
+	sscr1 = pxa_ssp_read_reg(ssp, SSCR1);
 	if (tristate)
 		sscr1 &= ~SSCR1_TTE;
 	else
 		sscr1 |= SSCR1_TTE;
-	ssp_write_reg(ssp, SSCR1, sscr1);
+	pxa_ssp_write_reg(ssp, SSCR1, sscr1);
 
 	return 0;
 }
@@ -475,14 +475,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		return 0;
 
 	/* we can only change the settings if the port is not in use */
-	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
+	if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
 		dev_err(&ssp->pdev->dev,
 			"can't change hardware dai format: stream is in use");
 		return -EINVAL;
 	}
 
 	/* reset port settings */
-	sscr0 = ssp_read_reg(ssp, SSCR0) &
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
 		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
 	sspsp = 0;
@@ -534,9 +534,9 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		return -EINVAL;
 	}
 
-	ssp_write_reg(ssp, SSCR0, sscr0);
-	ssp_write_reg(ssp, SSCR1, sscr1);
-	ssp_write_reg(ssp, SSPSP, sspsp);
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
+	pxa_ssp_write_reg(ssp, SSCR1, sscr1);
+	pxa_ssp_write_reg(ssp, SSPSP, sspsp);
 
 	dump_registers(ssp);
 
@@ -565,7 +565,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	u32 sscr0;
 	u32 sspsp;
 	int width = snd_pcm_format_physical_width(params_format(params));
-	int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
+	int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
 	struct pxa2xx_pcm_dma_params *dma_data;
 
 	dma_data = snd_soc_dai_get_dma_data(dai, substream);
@@ -577,22 +577,22 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	 * to force 16-bit frame width on the wire (for S16_LE), even
 	 * with two channels. Use 16-bit DMA transfers for this case.
 	 */
-	dma_data = ssp_get_dma_params(ssp,
+	dma_data = pxa_ssp_get_dma_params(ssp,
 			((chn == 2) && (ttsa != 1)) || (width == 32),
 			substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	snd_soc_dai_set_dma_data(dai, substream, dma_data);
 
 	/* we can only change the settings if the port is not in use */
-	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+	if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
 		return 0;
 
 	/* clear selected SSP bits */
-	sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
-	ssp_write_reg(ssp, SSCR0, sscr0);
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 
 	/* bit size */
-	sscr0 = ssp_read_reg(ssp, SSCR0);
+	sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 #ifdef CONFIG_PXA3xx
@@ -608,13 +608,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
 		break;
 	}
-	ssp_write_reg(ssp, SSCR0, sscr0);
+	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 
 	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-	       sspsp = ssp_read_reg(ssp, SSPSP);
+	       sspsp = pxa_ssp_read_reg(ssp, SSPSP);
 
-		if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
+		if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) {
 			/* This is a special case where the bitclk is 64fs
 			* and we're not dealing with 2*32 bits of audio
 			* samples.
@@ -648,7 +648,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			sspsp |= SSPSP_DMYSTRT(1);
 		}
 
-		ssp_write_reg(ssp, SSPSP, sspsp);
+		pxa_ssp_write_reg(ssp, SSPSP, sspsp);
 		break;
 	default:
 		break;
@@ -679,45 +679,45 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_RESUME:
-		ssp_enable(ssp);
+		pxa_ssp_enable(ssp);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		val = ssp_read_reg(ssp, SSCR1);
+		val = pxa_ssp_read_reg(ssp, SSCR1);
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			val |= SSCR1_TSRE;
 		else
 			val |= SSCR1_RSRE;
-		ssp_write_reg(ssp, SSCR1, val);
-		val = ssp_read_reg(ssp, SSSR);
-		ssp_write_reg(ssp, SSSR, val);
+		pxa_ssp_write_reg(ssp, SSCR1, val);
+		val = pxa_ssp_read_reg(ssp, SSSR);
+		pxa_ssp_write_reg(ssp, SSSR, val);
 		break;
 	case SNDRV_PCM_TRIGGER_START:
-		val = ssp_read_reg(ssp, SSCR1);
+		val = pxa_ssp_read_reg(ssp, SSCR1);
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			val |= SSCR1_TSRE;
 		else
 			val |= SSCR1_RSRE;
-		ssp_write_reg(ssp, SSCR1, val);
-		ssp_enable(ssp);
+		pxa_ssp_write_reg(ssp, SSCR1, val);
+		pxa_ssp_enable(ssp);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
-		val = ssp_read_reg(ssp, SSCR1);
+		val = pxa_ssp_read_reg(ssp, SSCR1);
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			val &= ~SSCR1_TSRE;
 		else
 			val &= ~SSCR1_RSRE;
-		ssp_write_reg(ssp, SSCR1, val);
+		pxa_ssp_write_reg(ssp, SSCR1, val);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		ssp_disable(ssp);
+		pxa_ssp_disable(ssp);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		val = ssp_read_reg(ssp, SSCR1);
+		val = pxa_ssp_read_reg(ssp, SSCR1);
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			val &= ~SSCR1_TSRE;
 		else
 			val &= ~SSCR1_RSRE;
-		ssp_write_reg(ssp, SSCR1, val);
+		pxa_ssp_write_reg(ssp, SSCR1, val);
 		break;
 
 	default:
@@ -739,7 +739,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
 	if (!priv)
 		return -ENOMEM;
 
-	priv->ssp = ssp_request(dai->id + 1, "SoC audio");
+	priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
 	if (priv->ssp == NULL) {
 		ret = -ENODEV;
 		goto err_priv;
@@ -759,7 +759,7 @@ static void pxa_ssp_remove(struct platform_device *pdev,
 			      struct snd_soc_dai *dai)
 {
 	struct ssp_priv *priv = dai->private_data;
-	ssp_free(priv->ssp);
+	pxa_ssp_free(priv->ssp);
 }
 
 #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-- 
cgit v1.2.3


From 36aeff6146925025033e2bcd45fa1e9725bc4599 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Wed, 12 May 2010 10:35:36 +0300
Subject: ASoC: TWL4030: Add control for digimic Left Right swap

The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/twl4030.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 0fe74d1e2c5f..6a34f562b563 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1102,6 +1102,16 @@ static const struct soc_enum twl4030_vibradir_enum =
 			ARRAY_SIZE(twl4030_vibradir_texts),
 			twl4030_vibradir_texts);
 
+/* Digimic Left and right swapping */
+static const char *twl4030_digimicswap_texts[] = {
+	"Not swapped", "Swapped",
+};
+
+static const struct soc_enum twl4030_digimicswap_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0,
+			ARRAY_SIZE(twl4030_digimicswap_texts),
+			twl4030_digimicswap_texts);
+
 static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 	/* Codec operation mode control */
 	SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum,
@@ -1178,6 +1188,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 
 	SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
 	SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
+
+	SOC_ENUM("Digimic LR Swap", twl4030_digimicswap_enum),
 };
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
-- 
cgit v1.2.3


From d98508a121e8f4b1ccf876fea463fa0afddc4e19 Mon Sep 17 00:00:00 2001
From: Sergey Lapin <slapin@ossfans.org>
Date: Thu, 13 May 2010 19:48:16 +0400
Subject: OMAP: McBSP: Add 32-bit mode support

This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec,
or others.

Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/omap-mcbsp.c | 28 ++++++++++++++++++++++++----
 1 file changed, 24 insertions(+), 4 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 2d33a89f147a..6f44cb4d30b8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -320,8 +320,18 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
 	omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
 	omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
-	omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
-							OMAP_DMA_DATA_TYPE_S16;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
+			 OMAP_DMA_DATA_TYPE_S16;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
+			 OMAP_DMA_DATA_TYPE_S32;
+		break;
+	default:
+		return -EINVAL;
+	}
 
 	snd_soc_dai_set_dma_data(cpu_dai, substream,
 		&omap_mcbsp_dai_dma_params[id][substream->stream]);
@@ -356,6 +366,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_16);
 		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_16);
 		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		/* Set word lengths */
+		wlen = 32;
+		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_32);
+		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_32);
+		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_32);
+		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_32);
+		break;
 	default:
 		/* Unsupported PCM format */
 		return -EINVAL;
@@ -659,13 +677,15 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
 		.channels_min = 1,				\
 		.channels_max = 16,				\
 		.rates = OMAP_MCBSP_RATES,			\
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |		\
+			   SNDRV_PCM_FMTBIT_S32_LE,		\
 	},							\
 	.capture = {						\
 		.channels_min = 1,				\
 		.channels_max = 16,				\
 		.rates = OMAP_MCBSP_RATES,			\
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
+		.formats = SNDRV_PCM_FMTBIT_S16_LE |		\
+			   SNDRV_PCM_FMTBIT_S32_LE,		\
 	},							\
 	.ops = &omap_mcbsp_dai_ops,				\
 	.private_data = &mcbsp_data[(link_id)].bus_id,		\
-- 
cgit v1.2.3


From 7fd1d74bfc0ecf3dfa139b47daa7941841724886 Mon Sep 17 00:00:00 2001
From: Felipe Balbi <felipe.balbi@nokia.com>
Date: Mon, 17 May 2010 14:21:45 +0300
Subject: ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function

Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.

Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 007fe830be46..ad1795a83acb 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -352,7 +352,7 @@ static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 
 	/* Safety check */
 	if (unlikely(power == dac33->chip_power)) {
-		dev_warn(codec->dev, "Trying to set the same power state: %s\n",
+		dev_dbg(codec->dev, "Trying to set the same power state: %s\n",
 			power ? "ON" : "OFF");
 		goto exit;
 	}
-- 
cgit v1.2.3


From 2d4cdd6fc974716555fcbaf7ec1d4dda22784c1b Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Mon, 17 May 2010 14:21:46 +0300
Subject: ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF

Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tlv320dac33.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index ad1795a83acb..bcf6d934499a 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -589,6 +589,9 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
 		}
 		break;
 	case SND_SOC_BIAS_OFF:
+		/* Do not power off, when the codec is already off */
+		if (codec->bias_level == SND_SOC_BIAS_OFF)
+			return 0;
 		ret = dac33_hard_power(codec, 0);
 		if (ret != 0)
 			return ret;
-- 
cgit v1.2.3


From 058276303dbc4ed089c1f7dad0871810b1f5ddf1 Mon Sep 17 00:00:00 2001
From: Linus Walleij <linus.walleij@stericsson.com>
Date: Mon, 17 May 2010 16:30:42 -0700
Subject: DMAENGINE: extend the control command to include an arg

This adds an argument to the DMAengine control function, so that
we can later provide control commands that need some external data
passed in through an argument akin to the ioctl() operation
prototype.

[dan.j.williams@intel.com: fix up some missed conversions]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
---
 drivers/dma/at_hdmac.c       |  3 ++-
 drivers/dma/coh901318.c      |  5 +++--
 drivers/dma/dw_dmac.c        |  3 ++-
 drivers/dma/fsldma.c         |  2 +-
 drivers/dma/ipu/ipu_idmac.c  | 12 +++++++-----
 drivers/dma/shdma.c          |  3 ++-
 drivers/dma/ste_dma40.c      |  3 ++-
 drivers/dma/timb_dma.c       |  3 ++-
 drivers/dma/txx9dmac.c       |  3 ++-
 drivers/mmc/host/atmel-mci.c |  2 +-
 drivers/serial/sh-sci.c      |  2 +-
 drivers/video/mx3fb.c        |  2 +-
 include/linux/dmaengine.h    |  3 ++-
 sound/soc/txx9/txx9aclc.c    |  7 ++++---
 14 files changed, 32 insertions(+), 21 deletions(-)

(limited to 'sound/soc')

diff --git a/drivers/dma/at_hdmac.c b/drivers/dma/at_hdmac.c
index 93ed99c84cf1..ee805a43f879 100644
--- a/drivers/dma/at_hdmac.c
+++ b/drivers/dma/at_hdmac.c
@@ -759,7 +759,8 @@ err_desc_get:
 	return NULL;
 }
 
-static int atc_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int atc_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+		       unsigned long arg)
 {
 	struct at_dma_chan	*atchan = to_at_dma_chan(chan);
 	struct at_dma		*atdma = to_at_dma(chan->device);
diff --git a/drivers/dma/coh901318.c b/drivers/dma/coh901318.c
index 4233440741a2..a724e6be1b4d 100644
--- a/drivers/dma/coh901318.c
+++ b/drivers/dma/coh901318.c
@@ -942,7 +942,7 @@ coh901318_free_chan_resources(struct dma_chan *chan)
 
 	spin_unlock_irqrestore(&cohc->lock, flags);
 
-	chan->device->device_control(chan, DMA_TERMINATE_ALL);
+	chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 }
 
 
@@ -1176,7 +1176,8 @@ coh901318_issue_pending(struct dma_chan *chan)
 }
 
 static int
-coh901318_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+coh901318_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+		  unsigned long arg)
 {
 	unsigned long flags;
 	struct coh901318_chan *cohc = to_coh901318_chan(chan);
diff --git a/drivers/dma/dw_dmac.c b/drivers/dma/dw_dmac.c
index 18fb5b41cedf..a3991ab0d67e 100644
--- a/drivers/dma/dw_dmac.c
+++ b/drivers/dma/dw_dmac.c
@@ -781,7 +781,8 @@ err_desc_get:
 	return NULL;
 }
 
-static int dwc_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int dwc_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+		       unsigned long arg)
 {
 	struct dw_dma_chan	*dwc = to_dw_dma_chan(chan);
 	struct dw_dma		*dw = to_dw_dma(chan->device);
diff --git a/drivers/dma/fsldma.c b/drivers/dma/fsldma.c
index cb1924f46c9e..005329d496bd 100644
--- a/drivers/dma/fsldma.c
+++ b/drivers/dma/fsldma.c
@@ -775,7 +775,7 @@ fail:
 }
 
 static int fsl_dma_device_control(struct dma_chan *dchan,
-				  enum dma_ctrl_cmd cmd)
+				  enum dma_ctrl_cmd cmd, unsigned long arg)
 {
 	struct fsldma_chan *chan;
 	unsigned long flags;
diff --git a/drivers/dma/ipu/ipu_idmac.c b/drivers/dma/ipu/ipu_idmac.c
index 246a6143e4a7..cb26ee9773d6 100644
--- a/drivers/dma/ipu/ipu_idmac.c
+++ b/drivers/dma/ipu/ipu_idmac.c
@@ -1472,7 +1472,8 @@ static void idmac_issue_pending(struct dma_chan *chan)
 	 */
 }
 
-static int __idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int __idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+			   unsigned long arg)
 {
 	struct idmac_channel *ichan = to_idmac_chan(chan);
 	struct idmac *idmac = to_idmac(chan->device);
@@ -1513,14 +1514,15 @@ static int __idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
 	return 0;
 }
 
-static int idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int idmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+			 unsigned long arg)
 {
 	struct idmac_channel *ichan = to_idmac_chan(chan);
 	int ret;
 
 	mutex_lock(&ichan->chan_mutex);
 
-	ret = __idmac_control(chan, cmd);
+	ret = __idmac_control(chan, cmd, arg);
 
 	mutex_unlock(&ichan->chan_mutex);
 
@@ -1616,7 +1618,7 @@ static void idmac_free_chan_resources(struct dma_chan *chan)
 
 	mutex_lock(&ichan->chan_mutex);
 
-	__idmac_control(chan, DMA_TERMINATE_ALL);
+	__idmac_control(chan, DMA_TERMINATE_ALL, 0);
 
 	if (ichan->status > IPU_CHANNEL_FREE) {
 #ifdef DEBUG
@@ -1709,7 +1711,7 @@ static void __exit ipu_idmac_exit(struct ipu *ipu)
 	for (i = 0; i < IPU_CHANNELS_NUM; i++) {
 		struct idmac_channel *ichan = ipu->channel + i;
 
-		idmac_control(&ichan->dma_chan, DMA_TERMINATE_ALL);
+		idmac_control(&ichan->dma_chan, DMA_TERMINATE_ALL, 0);
 		idmac_prep_slave_sg(&ichan->dma_chan, NULL, 0, DMA_NONE, 0);
 	}
 
diff --git a/drivers/dma/shdma.c b/drivers/dma/shdma.c
index 98f82cdb404c..e9de1d35c20d 100644
--- a/drivers/dma/shdma.c
+++ b/drivers/dma/shdma.c
@@ -580,7 +580,8 @@ static struct dma_async_tx_descriptor *sh_dmae_prep_slave_sg(
 			       direction, flags);
 }
 
-static int sh_dmae_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int sh_dmae_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+			   unsigned long arg)
 {
 	struct sh_dmae_chan *sh_chan = to_sh_chan(chan);
 
diff --git a/drivers/dma/ste_dma40.c b/drivers/dma/ste_dma40.c
index 81fec95312b6..c426829f6ab8 100644
--- a/drivers/dma/ste_dma40.c
+++ b/drivers/dma/ste_dma40.c
@@ -2065,7 +2065,8 @@ static void d40_issue_pending(struct dma_chan *chan)
 	spin_unlock_irqrestore(&d40c->lock, flags);
 }
 
-static int d40_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int d40_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+		       unsigned long arg)
 {
 	unsigned long flags;
 	struct d40_chan *d40c = container_of(chan, struct d40_chan, chan);
diff --git a/drivers/dma/timb_dma.c b/drivers/dma/timb_dma.c
index 44b346d8d319..0172fa3c7a2b 100644
--- a/drivers/dma/timb_dma.c
+++ b/drivers/dma/timb_dma.c
@@ -611,7 +611,8 @@ static struct dma_async_tx_descriptor *td_prep_slave_sg(struct dma_chan *chan,
 	return &td_desc->txd;
 }
 
-static int td_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int td_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+		      unsigned long arg)
 {
 	struct timb_dma_chan *td_chan =
 		container_of(chan, struct timb_dma_chan, chan);
diff --git a/drivers/dma/txx9dmac.c b/drivers/dma/txx9dmac.c
index e523737639aa..d02edb05910e 100644
--- a/drivers/dma/txx9dmac.c
+++ b/drivers/dma/txx9dmac.c
@@ -938,7 +938,8 @@ txx9dmac_prep_slave_sg(struct dma_chan *chan, struct scatterlist *sgl,
 	return &first->txd;
 }
 
-static int txx9dmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd)
+static int txx9dmac_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+			    unsigned long arg)
 {
 	struct txx9dmac_chan *dc = to_txx9dmac_chan(chan);
 	struct txx9dmac_desc *desc, *_desc;
diff --git a/drivers/mmc/host/atmel-mci.c b/drivers/mmc/host/atmel-mci.c
index ae6d24ba4f08..fd2d24f81478 100644
--- a/drivers/mmc/host/atmel-mci.c
+++ b/drivers/mmc/host/atmel-mci.c
@@ -578,7 +578,7 @@ static void atmci_stop_dma(struct atmel_mci *host)
 	struct dma_chan *chan = host->data_chan;
 
 	if (chan) {
-		chan->device->device_control(chan, DMA_TERMINATE_ALL);
+	  chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 		atmci_dma_cleanup(host);
 	} else {
 		/* Data transfer was stopped by the interrupt handler */
diff --git a/drivers/serial/sh-sci.c b/drivers/serial/sh-sci.c
index 690988237971..882f3d5ed028 100644
--- a/drivers/serial/sh-sci.c
+++ b/drivers/serial/sh-sci.c
@@ -1087,7 +1087,7 @@ static void work_fn_rx(struct work_struct *work)
 		unsigned long flags;
 		int count;
 
-		chan->device->device_control(chan, DMA_TERMINATE_ALL);
+		chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 		dev_dbg(port->dev, "Read %u bytes with cookie %d\n",
 			sh_desc->partial, sh_desc->cookie);
 
diff --git a/drivers/video/mx3fb.c b/drivers/video/mx3fb.c
index 3aa50bc276eb..7cfc170bce19 100644
--- a/drivers/video/mx3fb.c
+++ b/drivers/video/mx3fb.c
@@ -388,7 +388,7 @@ static void sdc_disable_channel(struct mx3fb_info *mx3_fbi)
 	spin_unlock_irqrestore(&mx3fb->lock, flags);
 
 	mx3_fbi->txd->chan->device->device_control(mx3_fbi->txd->chan,
-						   DMA_TERMINATE_ALL);
+						   DMA_TERMINATE_ALL, 0);
 	mx3_fbi->txd = NULL;
 	mx3_fbi->cookie = -EINVAL;
 }
diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h
index 50b7b3e0d572..17456571ff7a 100644
--- a/include/linux/dmaengine.h
+++ b/include/linux/dmaengine.h
@@ -347,7 +347,8 @@ struct dma_device {
 		struct dma_chan *chan, struct scatterlist *sgl,
 		unsigned int sg_len, enum dma_data_direction direction,
 		unsigned long flags);
-	int (*device_control)(struct dma_chan *chan, enum dma_ctrl_cmd cmd);
+	int (*device_control)(struct dma_chan *chan, enum dma_ctrl_cmd cmd,
+		unsigned long arg);
 
 	enum dma_status (*device_tx_status)(struct dma_chan *chan,
 					    dma_cookie_t cookie,
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index b35d00706c0e..9398f507f77f 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -159,7 +159,7 @@ static void txx9aclc_dma_tasklet(unsigned long data)
 		void __iomem *base = drvdata->base;
 
 		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
-		chan->device->device_control(chan, DMA_TERMINATE_ALL);
+		chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 		/* first time */
 		for (i = 0; i < NR_DMA_CHAIN; i++) {
 			desc = txx9aclc_dma_submit(dmadata,
@@ -267,7 +267,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
 	struct dma_chan *chan = dmadata->dma_chan;
 
 	dmadata->frag_count = -1;
-	chan->device->device_control(chan, DMA_TERMINATE_ALL);
+	chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 	return 0;
 }
 
@@ -396,7 +396,8 @@ static int txx9aclc_pcm_remove(struct platform_device *pdev)
 		struct dma_chan *chan = dmadata->dma_chan;
 		if (chan) {
 			dmadata->frag_count = -1;
-			chan->device->device_control(chan, DMA_TERMINATE_ALL);
+			chan->device->device_control(chan,
+						     DMA_TERMINATE_ALL, 0);
 			dma_release_channel(chan);
 		}
 		dev->dmadata[i].dma_chan = NULL;
-- 
cgit v1.2.3


From 5e64d6aadd2b8d5d480c3ff402f5dbbd009f58c5 Mon Sep 17 00:00:00 2001
From: Misael Lopez Cruz <x0052729@ti.com>
Date: Mon, 17 May 2010 19:53:10 -0500
Subject: ASoC: SDP4430: Add sdp4430 machine driver

Add ASoC support for TI SDP4430.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/Kconfig   |   9 ++
 sound/soc/omap/Makefile  |   2 +
 sound/soc/omap/sdp4430.c | 228 +++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 239 insertions(+)
 create mode 100644 sound/soc/omap/sdp4430.c

(limited to 'sound/soc')

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 83be4a76d2bb..d542ea2ff6be 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -98,6 +98,15 @@ config SND_OMAP_SOC_SDP3430
 	  Say Y if you want to add support for SoC audio on Texas Instruments
 	  SDP3430.
 
+config SND_OMAP_SOC_SDP4430
+	tristate "SoC Audio support for Texas Instruments SDP4430"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP
+	select SND_OMAP_SOC_MCPDM
+	select SND_SOC_TWL6040
+	help
+	  Say Y if you want to add support for SoC audio on Texas Instruments
+	  SDP4430.
+
 config SND_OMAP_SOC_OMAP3_PANDORA
 	tristate "SoC Audio support for OMAP3 Pandora"
 	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3a75755f25e4..ba9fc650db28 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -17,6 +17,7 @@ snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-am3517evm-objs := am3517evm.o
 snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-sdp4430-objs := sdp4430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
 snd-soc-zoom2-objs := zoom2.o
@@ -31,6 +32,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
 obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
 obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
new file mode 100644
index 000000000000..83831ee35d07
--- /dev/null
+++ b/sound/soc/omap/sdp4430.c
@@ -0,0 +1,228 @@
+/*
+ * sdp4430.c  --  SoC audio for TI OMAP4430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+
+#include "mcpdm.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+
+static int twl6040_power_mode;
+
+static int sdp4430_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;
+	int clk_id, freq;
+	int ret;
+
+	if (twl6040_power_mode) {
+		clk_id = TWL6040_SYSCLK_SEL_HPPLL;
+		freq = 38400000;
+	} else {
+		clk_id = TWL6040_SYSCLK_SEL_LPPLL;
+		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");
+		return ret;
+	}
+}
+
+static struct snd_soc_ops sdp4430_ops = {
+	.hw_params = sdp4430_hw_params,
+};
+
+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];
+
+	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),
+};
+
+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 Stereophone (Headphone): HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+};
+
+static int sdp4430_twl6040_init(struct snd_soc_codec *codec)
+{
+	int 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(codec, 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(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* SDP4430 connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_enable_pin(codec, "Headset Mic");
+	snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+
+	/* TWL6040 not connected pins */
+	snd_soc_dapm_nc_pin(codec, "AFML");
+	snd_soc_dapm_nc_pin(codec, "AFMR");
+
+	ret = snd_soc_dapm_sync(codec);
+
+	return ret;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp4430_dai = {
+	.name = "TWL6040",
+	.stream_name = "TWL6040",
+	.cpu_dai = &omap_mcpdm_dai,
+	.codec_dai = &twl6040_dai,
+	.init = sdp4430_twl6040_init,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sdp4430 = {
+	.name = "SDP4430",
+	.platform = &omap_soc_platform,
+	.dai_link = &sdp4430_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device sdp4430_snd_devdata = {
+	.card = &snd_soc_sdp4430,
+	.codec_dev = &soc_codec_dev_twl6040,
+};
+
+static struct platform_device *sdp4430_snd_device;
+
+static int __init sdp4430_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_4430sdp()) {
+		pr_debug("Not SDP4430!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "SDP4430 SoC init\n");
+
+	sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!sdp4430_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sdp4430_snd_device, &sdp4430_snd_devdata);
+	sdp4430_snd_devdata.dev = &sdp4430_snd_device->dev;
+
+	ret = platform_device_add(sdp4430_snd_device);
+	if (ret)
+		goto err;
+
+	/* Codec starts in HP mode */
+	twl6040_power_mode = 1;
+
+	return 0;
+
+err:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(sdp4430_snd_device);
+	return ret;
+}
+module_init(sdp4430_soc_init);
+
+static void __exit sdp4430_soc_exit(void)
+{
+	platform_device_unregister(sdp4430_snd_device);
+}
+module_exit(sdp4430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP4430");
+MODULE_LICENSE("GPL");
+
-- 
cgit v1.2.3


From 7254e2bddc96c47cb5edd30cfd3e1f12a2df9149 Mon Sep 17 00:00:00 2001
From: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Date: Tue, 18 May 2010 12:44:17 -0500
Subject: ASoC: SDP4430: Add support for Earphone speaker

Enable earphone speaker in sdp4430 machine driver.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/sdp4430.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
index 83831ee35d07..3a0c19866121 100644
--- a/sound/soc/omap/sdp4430.c
+++ b/sound/soc/omap/sdp4430.c
@@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = {
 	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),
 };
 
 static const struct snd_soc_dapm_route audio_map[] = {
@@ -120,6 +121,9 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	/* Headset Stereophone (Headphone): HSOL, HSOR */
 	{"Headset Stereophone", NULL, "HSOL"},
 	{"Headset Stereophone", NULL, "HSOR"},
+
+	/* Earphone speaker */
+	{"Earphone Spk", NULL, "EP"},
 };
 
 static int sdp4430_twl6040_init(struct snd_soc_codec *codec)
-- 
cgit v1.2.3


From 871a05a78b4879d768bc2fde4b75439f517e2839 Mon Sep 17 00:00:00 2001
From: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Date: Tue, 18 May 2010 12:44:18 -0500
Subject: ASoC: TWL6040: Enable earphone path in codec

Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/twl6040.c | 21 ++++++++++++++++++++-
 1 file changed, 20 insertions(+), 1 deletion(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 2ae442edeb9a..af36346ff336 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -432,6 +432,12 @@ static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0);
  */
 static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0);
 
+/*
+ * EPGAIN volume control:
+ * from -24 to 6 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(ep_tlv, -2400, 200, 0);
+
 /* Left analog microphone selection */
 static const char *twl6040_amicl_texts[] =
 	{"Headset Mic", "Main Mic", "Aux/FM Left", "Off"};
@@ -479,6 +485,9 @@ static const struct snd_kcontrol_new hfl_driver_switch_controls =
 static const struct snd_kcontrol_new hfr_driver_switch_controls =
 	SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0);
 
+static const struct snd_kcontrol_new ep_driver_switch_controls =
+	SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0);
+
 static const struct snd_kcontrol_new twl6040_snd_controls[] = {
 	/* Capture gains */
 	SOC_DOUBLE_TLV("Capture Preamplifier Volume",
@@ -491,7 +500,8 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
 		TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv),
 	SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
 		TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv),
-
+	SOC_SINGLE_TLV("Earphone Playback Volume",
+		TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv),
 };
 
 static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
@@ -507,6 +517,7 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HSOR"),
 	SND_SOC_DAPM_OUTPUT("HFL"),
 	SND_SOC_DAPM_OUTPUT("HFR"),
+	SND_SOC_DAPM_OUTPUT("EP"),
 
 	/* Analog input muxes for the capture amplifiers */
 	SND_SOC_DAPM_MUX("Analog Left Capture Route",
@@ -572,6 +583,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls,
 			twl6040_power_mode_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH_E("Earphone Driver",
+			SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls,
+			twl6040_power_mode_event,
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 	/* Analog playback PGAs */
 	SND_SOC_DAPM_PGA("HFDAC Left PGA",
@@ -607,6 +622,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"HSOL", NULL, "Headset Left Driver"},
 	{"HSOR", NULL, "Headset Right Driver"},
 
+	/* Earphone playback path */
+	{"Earphone Driver", "Switch", "HSDAC Left"},
+	{"EP", NULL, "Earphone Driver"},
+
 	/* Handsfree playback path */
 	{"HFDAC Left Playback", "Switch", "HFDAC Left"},
 	{"HFDAC Right Playback", "Switch", "HFDAC Right"},
-- 
cgit v1.2.3


From d8b55d2cd00df4d599985440fd75b38d153bffcb Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@slimlogic.co.uk>
Date: Wed, 19 May 2010 14:14:51 +0100
Subject: ASoC: sdp4430 - add sdp4430 pcm ops to DAI.

Fix build warning about unused ops and add ops
to the sdp4430 DAI link.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/omap/sdp4430.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/soc')

diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
index 3a0c19866121..4ebbde6b565f 100644
--- a/sound/soc/omap/sdp4430.c
+++ b/sound/soc/omap/sdp4430.c
@@ -167,6 +167,7 @@ static struct snd_soc_dai_link sdp4430_dai = {
 	.cpu_dai = &omap_mcpdm_dai,
 	.codec_dai = &twl6040_dai,
 	.init = sdp4430_twl6040_init,
+	.ops = &sdp4430_ops,
 };
 
 /* Audio machine driver */
-- 
cgit v1.2.3


From 266d38c8e3d7f62152b1448fd9a7265f32f32d87 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Wed, 19 May 2010 13:55:26 +0300
Subject: ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT

Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise
external widgets doesn't alter the output state.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tpa6130a2.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index fa4fa33a51aa..89788921280e 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -371,8 +371,8 @@ static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = {
 			0, 0, tpa6130a2_supply_event,
 			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
 	/* Outputs */
-	SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL),
-	SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL),
+	SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Left"),
+	SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Right"),
 };
 
 static const struct snd_soc_dapm_route audio_map[] = {
-- 
cgit v1.2.3


From ad8332c1302bcb4f80d593fd3eb477be9d7f5604 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Wed, 19 May 2010 10:52:28 +0300
Subject: ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies

These pins are for decoupling capacitors for the internal charge pumps
in TPA6130A2 and TPA6140A2 and not for connecting external supply.

Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the
issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with
TPA6140A2.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
---
 sound/soc/codecs/tpa6130a2.c | 52 +++++++++++++++-----------------------------
 1 file changed, 18 insertions(+), 34 deletions(-)

(limited to 'sound/soc')

diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 89788921280e..20ac67700395 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -35,22 +35,11 @@
 
 static struct i2c_client *tpa6130a2_client;
 
-#define TPA6130A2_NUM_SUPPLIES 2
-static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
-	"CPVSS",
-	"Vdd",
-};
-
-static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
-	"HPVdd",
-	"AVdd",
-};
-
 /* This struct is used to save the context */
 struct tpa6130a2_data {
 	struct mutex mutex;
 	unsigned char regs[TPA6130A2_CACHEREGNUM];
-	struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES];
+	struct regulator *supply;
 	int power_gpio;
 	unsigned char power_state;
 	enum tpa_model id;
@@ -135,11 +124,10 @@ static int tpa6130a2_power(int power)
 		if (data->power_gpio >= 0)
 			gpio_set_value(data->power_gpio, 1);
 
-		ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies),
-					    data->supplies);
+		ret = regulator_enable(data->supply);
 		if (ret != 0) {
 			dev_err(&tpa6130a2_client->dev,
-				"Failed to enable supplies: %d\n", ret);
+				"Failed to enable supply: %d\n", ret);
 			goto exit;
 		}
 
@@ -160,11 +148,10 @@ static int tpa6130a2_power(int power)
 		if (data->power_gpio >= 0)
 			gpio_set_value(data->power_gpio, 0);
 
-		ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies),
-					     data->supplies);
+		ret = regulator_disable(data->supply);
 		if (ret != 0) {
 			dev_err(&tpa6130a2_client->dev,
-				"Failed to disable supplies: %d\n", ret);
+				"Failed to disable supply: %d\n", ret);
 			goto exit;
 		}
 
@@ -411,7 +398,8 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 	struct device *dev;
 	struct tpa6130a2_data *data;
 	struct tpa6130a2_platform_data *pdata;
-	int i, ret;
+	const char *regulator;
+	int ret;
 
 	dev = &client->dev;
 
@@ -453,25 +441,21 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 	}
 
 	switch (data->id) {
+	default:
+		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
+			 pdata->id);
 	case TPA6130A2:
-		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
-			data->supplies[i].supply = tpa6130a2_supply_names[i];
+		regulator = "Vdd";
 		break;
 	case TPA6140A2:
-		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
-			data->supplies[i].supply = tpa6140a2_supply_names[i];;
+		regulator = "AVdd";
 		break;
-	default:
-		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
-			 pdata->id);
-		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
-			data->supplies[i].supply = tpa6130a2_supply_names[i];
 	}
 
-	ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
-				 data->supplies);
-	if (ret != 0) {
-		dev_err(dev, "Failed to request supplies: %d\n", ret);
+	data->supply = regulator_get(dev, regulator);
+	if (IS_ERR(data->supply)) {
+		ret = PTR_ERR(data->supply);
+		dev_err(dev, "Failed to request supply: %d\n", ret);
 		goto err_regulator;
 	}
 
@@ -494,7 +478,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 	return 0;
 
 err_power:
-	regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies);
+	regulator_put(data->supply);
 err_regulator:
 	if (data->power_gpio >= 0)
 		gpio_free(data->power_gpio);
@@ -515,7 +499,7 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client)
 	if (data->power_gpio >= 0)
 		gpio_free(data->power_gpio);
 
-	regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies);
+	regulator_put(data->supply);
 
 	kfree(data);
 	tpa6130a2_client = NULL;
-- 
cgit v1.2.3