From 0be9898adb6f58fee44f0fec0bbc0eac997ea9eb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 19 May 2008 12:31:28 +0200
Subject: [ALSA] ASoC: Clarify API for bias configuration

Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.

Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.

Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/soc/codecs/wm8753.c | 36 ++++++++++++++++++------------------
 1 file changed, 18 insertions(+), 18 deletions(-)

(limited to 'sound/soc/codecs/wm8753.c')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index fb41826c4c4..9032b0c07c8 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1274,29 +1274,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8753_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* set vmid to 50k and unmute dac */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		/* set vmid to 5k for quick power up */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* mute dac and set vmid to 500k, enable VREF */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		wm8753_write(codec, WM8753_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -1500,7 +1500,7 @@ static void wm8753_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec =
 		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8753_dapm_event(codec, codec->dapm_state);
+	wm8753_set_bias_level(codec, codec->bias_level);
 }
 
 static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
@@ -1512,7 +1512,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 	if (!codec->card)
 		return 0;
 
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -1537,12 +1537,12 @@ static int wm8753_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8753 caps */
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-		codec->dapm_state = SNDRV_CTL_POWER_D0;
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+		wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+		codec->bias_level = SND_SOC_BIAS_ON;
 		schedule_delayed_work(&codec->delayed_work,
 			msecs_to_jiffies(caps_charge));
 	}
@@ -1563,7 +1563,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8753_read_reg_cache;
 	codec->write = wm8753_write;
-	codec->dapm_event = wm8753_dapm_event;
+	codec->set_bias_level = wm8753_set_bias_level;
 	codec->dai = wm8753_dai;
 	codec->num_dai = 2;
 	codec->reg_cache_size = sizeof(wm8753_reg);
@@ -1584,8 +1584,8 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	}
 
 	/* charge output caps */
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work,
 		msecs_to_jiffies(caps_charge));
 
@@ -1792,7 +1792,7 @@ static int wm8753_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-- 
cgit v1.2.3-70-g09d2


From a65f0568f6cc8433877fb71dd7d36b551854b0bc Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 13 May 2008 14:54:43 +0200
Subject: [ALSA] soc - Convert Wolfson codec drivers to use bulk DAPM
 registration

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/soc/codecs/wm8731.c | 16 ++++------------
 sound/soc/codecs/wm8750.c | 16 ++++------------
 sound/soc/codecs/wm8753.c | 17 ++++-------------
 sound/soc/codecs/wm9712.c | 15 ++++-----------
 sound/soc/codecs/wm9713.c | 15 ++++-----------
 5 files changed, 20 insertions(+), 59 deletions(-)

(limited to 'sound/soc/codecs/wm8753.c')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0f28aa4bccc..5acf43ab104 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -193,7 +193,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"),
 SND_SOC_DAPM_INPUT("LLINEIN"),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
 	{"Output Mixer", "HiFi Playback Switch", "DAC"},
@@ -214,22 +214,14 @@ static const char *intercon[][3] = {
 	{"Line Input", NULL, "LLINEIN"},
 	{"Line Input", NULL, "RLINEIN"},
 	{"Mic Bias", NULL, "MICIN"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8731_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
-	/* set up audio path interconnects */
-	for (i = 0; intercon[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-			intercon[i][1], intercon[i][2]);
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 62423f4493b..1f11ad24551 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -378,7 +378,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("RINPUT3"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left mixer */
 	{"Left Mixer", "Playback Switch", "Left DAC"},
 	{"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
@@ -470,22 +470,14 @@ static const char *audio_map[][3] = {
 	/* ADC */
 	{"Left ADC", NULL, "Left ADC Mux"},
 	{"Right ADC", NULL, "Right ADC Mux"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8750_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+				  ARRAY_SIZE(wm8750_dapm_widgets));
 
-	/* set up audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 9032b0c07c8..c32e6326be6 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -523,7 +523,7 @@ SND_SOC_DAPM_INPUT("MIC2"),
 SND_SOC_DAPM_VMID("VREF"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left mixer */
 	{"Left Mixer", "Left Playback Switch", "Left DAC"},
 	{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
@@ -674,23 +674,14 @@ static const char *audio_map[][3] = {
 
 	/* ACOP */
 	{"ACOP", NULL, "ALC Mixer"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8753_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
 
-	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-
-	/* set up the WM8753 audio map */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index e26cfcf0b4f..d9789f1c890 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -348,7 +348,7 @@ SND_SOC_DAPM_INPUT("MIC1"),
 SND_SOC_DAPM_INPUT("MIC2"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* virtual mixer - mixes left & right channels for spk and mono */
 	{"AC97 Mixer", NULL, "Left DAC"},
 	{"AC97 Mixer", NULL, "Right DAC"},
@@ -443,21 +443,14 @@ static const char *audio_map[][3] = {
 	{"Speaker PGA", NULL, "Speaker Mux"},
 	{"LOUT2", NULL, "Speaker PGA"},
 	{"ROUT2", NULL, "Speaker PGA"},
-
-	{NULL, NULL, NULL},
 };
 
 static int wm9712_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
+				  ARRAY_SIZE(wm9712_dapm_widgets));
 
-	/* set up audio path connects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-					   audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 4863636e9d5..4f516a5a561 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -453,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"),
 SND_SOC_DAPM_VMID("VMID"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left HP mixer */
 	{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
 	{"Left HP Mixer", "Voice Playback Switch",   "Voice DAC"},
@@ -604,21 +604,14 @@ static const char *audio_map[][3] = {
 	{"Capture Mono Mux", "Stereo", "Capture Mixer"},
 	{"Capture Mono Mux", "Left", "Left Capture Source"},
 	{"Capture Mono Mux", "Right", "Right Capture Source"},
-
-	{NULL, NULL, NULL},
 };
 
 static int wm9713_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+				  ARRAY_SIZE(wm9713_dapm_widgets));
 
-	/* set up audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
-- 
cgit v1.2.3-70-g09d2


From d751b233bb8568f1de1ccbe3824ca69090326251 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 11 Jun 2008 13:47:06 +0100
Subject: ALSA: ASoC: Fix register cache sizes for Wolfson codecs

The register cache size is used by the codec_reg sysfs file which works in
terms of the register cache access functions rather than in terms of raw
access to the cache so the size specified needs to be in terms of the
number of elements.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/soc/codecs/wm8731.c | 2 +-
 sound/soc/codecs/wm8750.c | 2 +-
 sound/soc/codecs/wm8753.c | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/soc/codecs/wm8753.c')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5acf43ab104..77880537a3c 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -534,7 +534,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
 	codec->set_bias_level = wm8731_set_bias_level;
 	codec->dai = &wm8731_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = sizeof(wm8731_reg);
+	codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
 	codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
 	if (codec->reg_cache == NULL)
 		return -ENOMEM;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 1f11ad24551..1ae670a98c5 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -798,7 +798,7 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	codec->set_bias_level = wm8750_set_bias_level;
 	codec->dai = &wm8750_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = sizeof(wm8750_reg);
+	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;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index c32e6326be6..285c5eaefe0 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1557,7 +1557,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	codec->set_bias_level = wm8753_set_bias_level;
 	codec->dai = wm8753_dai;
 	codec->num_dai = 2;
-	codec->reg_cache_size = sizeof(wm8753_reg);
+	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
 	codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
 
 	if (codec->reg_cache == NULL)
-- 
cgit v1.2.3-70-g09d2


From 2cc8c609798957b90adf90b5bfb9859d1643fade Mon Sep 17 00:00:00 2001
From: Mike Montour <mail@mmontour.net>
Date: Wed, 11 Jun 2008 13:47:12 +0100
Subject: ALSA: ASoC: Add TLV information to remaining WM8753 controls

Signed-off-by: Mike Montour <mail@mmontour.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/soc/codecs/wm8753.c | 75 +++++++++++++++++++++++++++++++----------------
 1 file changed, 50 insertions(+), 25 deletions(-)

(limited to 'sound/soc/codecs/wm8753.c')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 285c5eaefe0..00b481183d4 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -260,28 +260,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const unsigned int out_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	/* 0000000 - 0101111 = "Analogue mute" */
+	0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0),
+	48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0),
+};
+static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
 
 static const struct snd_kcontrol_new wm8753_snd_controls[] = {
-SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0),
-
-SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0),
-
-SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0),
-SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0),
-
-SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0),
-
-SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1),
-SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1),
-SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1),
-
-SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0),
-SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0),
-
-SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
-SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
-SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1),
+SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0,
+		 adc_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V,
+		 0, 127, 0, out_tlv),
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0,
+		 127, 0, out_tlv),
+
+SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7,
+		 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4,
+		 7, 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7,
+		 1, voice_mix_tlv),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7,
+	     1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7,
+	     1, 0),
+
+SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv),
+SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1,
+	       voice_mix_tlv),
 SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
 
 SOC_ENUM("Bass Boost", wm8753_enum[0]),
@@ -291,10 +313,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1),
 SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1),
 SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
 
-SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv),
-SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv),
+SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1,
+	       rec_mix_tlv),
+SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1,
+	       rec_mix_tlv),
 
-SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0,
+		 pga_tlv),
 SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
 
@@ -326,8 +351,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]),
 SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
 SOC_ENUM("Playback Phase", wm8753_enum[10]),
 
-SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0),
-SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
+SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv),
+SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv),
 
 SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
 
-- 
cgit v1.2.3-70-g09d2


From a5c95e90c1baa9c1114875264bbd283526eb8377 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 23 Jun 2008 14:51:29 +0100
Subject: ALSA: ASoC: Replace custom debug macros with pr_ equivalents

Several ASoC codec drivers use custom macros equivalent to the standard
pr_ macros, most of which are not actually used. Replace these custom
macros with the standard ones.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/soc/codecs/wm8510.c | 25 +++----------------------
 sound/soc/codecs/wm8731.c | 25 +++----------------------
 sound/soc/codecs/wm8750.c | 25 +++----------------------
 sound/soc/codecs/wm8753.c | 25 +++----------------------
 sound/soc/codecs/wm8990.c | 25 +++----------------------
 5 files changed, 15 insertions(+), 110 deletions(-)

(limited to 'sound/soc/codecs/wm8753.c')

diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 152e6f21154..b549f6753ab 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -30,25 +30,6 @@
 #define AUDIO_NAME "wm8510"
 #define WM8510_VERSION "0.6"
 
-/*
- * Debug
- */
-
-#define WM8510_DEBUG 0
-
-#ifdef WM8510_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 struct snd_soc_codec_device soc_codec_dev_wm8510;
 
 /*
@@ -721,13 +702,13 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8510_init(socdev);
 	if (ret < 0) {
-		err("failed to initialise WM8510\n");
+		pr_err("failed to initialise WM8510\n");
 		goto err;
 	}
 	return ret;
@@ -777,7 +758,7 @@ static int wm8510_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	info("WM8510 Audio Codec %s", WM8510_VERSION);
+	pr_info("WM8510 Audio Codec %s", WM8510_VERSION);
 
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 77880537a3c..3ff42ad65ed 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -31,25 +31,6 @@
 #define AUDIO_NAME "wm8731"
 #define WM8731_VERSION "0.13"
 
-/*
- * Debug
- */
-
-#define WM8731_DEBUG 0
-
-#ifdef WM8731_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 /* codec private data */
@@ -624,13 +605,13 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8731_init(socdev);
 	if (ret < 0) {
-		err("failed to initialise WM8731\n");
+		pr_err("failed to initialise WM8731\n");
 		goto err;
 	}
 	return ret;
@@ -681,7 +662,7 @@ static int wm8731_probe(struct platform_device *pdev)
 	struct wm8731_priv *wm8731;
 	int ret = 0;
 
-	info("WM8731 Audio Codec %s", WM8731_VERSION);
+	pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
 
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 1ae670a98c5..eb460c9aa63 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -31,25 +31,6 @@
 #define AUDIO_NAME "WM8750"
 #define WM8750_VERSION "0.12"
 
-/*
- * Debug
- */
-
-#define WM8750_DEBUG 0
-
-#ifdef WM8750_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 /* codec private data */
 struct wm8750_priv {
 	unsigned int sysclk;
@@ -896,13 +877,13 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8750_init(socdev);
 	if (ret < 0) {
-	err("failed to initialise WM8750\n");
+		pr_err("failed to initialise WM8750\n");
 		goto err;
 	}
 	return ret;
@@ -953,7 +934,7 @@ static int wm8750_probe(struct platform_device *pdev)
 	struct wm8750_priv *wm8750;
 	int ret = 0;
 
-	info("WM8750 Audio Codec %s", WM8750_VERSION);
+	pr_info("WM8750 Audio Codec %s", WM8750_VERSION);
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 00b481183d4..be01a738f18 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -55,25 +55,6 @@
 #define AUDIO_NAME "wm8753"
 #define WM8753_VERSION "0.16"
 
-/*
- * Debug
- */
-
-#define WM8753_DEBUG 0
-
-#ifdef WM8753_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -1689,13 +1670,13 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8753_init(socdev);
 	if (ret < 0) {
-		err("failed to initialise WM8753\n");
+		pr_err("failed to initialise WM8753\n");
 		goto err;
 	}
 
@@ -1747,7 +1728,7 @@ static int wm8753_probe(struct platform_device *pdev)
 	struct wm8753_priv *wm8753;
 	int ret = 0;
 
-	info("WM8753 Audio Codec %s", WM8753_VERSION);
+	pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
 
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index a7d25e2f252..a1371b73ba7 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -33,25 +33,6 @@
 #define AUDIO_NAME "wm8990"
 #define WM8990_VERSION "0.2"
 
-/*
- * Debug
- */
-
-#define WM8990_DEBUG 0
-
-#ifdef WM8990_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 /* codec private data */
 struct wm8990_priv {
 	unsigned int sysclk;
@@ -1524,13 +1505,13 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8990_init(socdev);
 	if (ret < 0) {
-		err("failed to initialise WM8990\n");
+		pr_err("failed to initialise WM8990\n");
 		goto err;
 	}
 	return ret;
@@ -1579,7 +1560,7 @@ static int wm8990_probe(struct platform_device *pdev)
 	struct wm8990_priv *wm8990;
 	int ret = 0;
 
-	info("WM8990 Audio Codec %s\n", WM8990_VERSION);
+	pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION);
 
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-- 
cgit v1.2.3-70-g09d2


From e550e17ffeb8cf8db27724eaf2ad05f77388afb9 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Date: Mon, 7 Jul 2008 16:07:52 +0100
Subject: ALSA: asoc: codecs - merge structs snd_soc_codec_dai and
 snd_soc_cpu_dai.

This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers.

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/soc/codecs/ac97.c        |  2 +-
 sound/soc/codecs/ac97.h        |  2 +-
 sound/soc/codecs/ak4535.c      |  8 ++++----
 sound/soc/codecs/ak4535.h      |  2 +-
 sound/soc/codecs/cs4270.c      |  8 ++++----
 sound/soc/codecs/cs4270.h      |  2 +-
 sound/soc/codecs/tlv320aic3x.c |  8 ++++----
 sound/soc/codecs/tlv320aic3x.h |  2 +-
 sound/soc/codecs/uda1380.c     |  6 +++---
 sound/soc/codecs/uda1380.h     |  2 +-
 sound/soc/codecs/wm8510.c      | 10 +++++-----
 sound/soc/codecs/wm8510.h      |  2 +-
 sound/soc/codecs/wm8731.c      |  8 ++++----
 sound/soc/codecs/wm8731.h      |  2 +-
 sound/soc/codecs/wm8750.c      |  8 ++++----
 sound/soc/codecs/wm8750.h      |  2 +-
 sound/soc/codecs/wm8753.c      | 28 ++++++++++++++--------------
 sound/soc/codecs/wm8753.h      |  2 +-
 sound/soc/codecs/wm8990.c      | 12 ++++++------
 sound/soc/codecs/wm8990.h      |  2 +-
 sound/soc/codecs/wm9712.c      |  2 +-
 sound/soc/codecs/wm9712.h      |  2 +-
 sound/soc/codecs/wm9713.c      | 10 +++++-----
 sound/soc/codecs/wm9713.h      |  2 +-
 24 files changed, 67 insertions(+), 67 deletions(-)

(limited to 'sound/soc/codecs/wm8753.c')

diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index e4516f3ce64..61fd96ca7bc 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -41,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
-struct snd_soc_codec_dai ac97_dai = {
+struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
 	.type = SND_SOC_DAI_AC97,
 	.playback = {
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
index 2bf6d69fd06..281aa42e2bb 100644
--- a/sound/soc/codecs/ac97.h
+++ b/sound/soc/codecs/ac97.h
@@ -14,6 +14,6 @@
 #define __LINUX_SND_SOC_AC97_H
 
 extern struct snd_soc_codec_device soc_codec_dev_ac97;
-extern struct snd_soc_codec_dai ac97_dai;
+extern struct snd_soc_dai ac97_dai;
 
 #endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 469266e881d..b26003c4f3e 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -329,7 +329,7 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -369,7 +369,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -394,7 +394,7 @@ static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute)
+static int ak4535_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
@@ -436,7 +436,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
-struct snd_soc_codec_dai ak4535_dai = {
+struct snd_soc_dai ak4535_dai = {
 	.name = "AK4535",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h
index fc686ddf753..e9fe30e2c05 100644
--- a/sound/soc/codecs/ak4535.h
+++ b/sound/soc/codecs/ak4535.h
@@ -40,7 +40,7 @@ struct ak4535_setup_data {
 	unsigned short i2c_address;
 };
 
-extern struct snd_soc_codec_dai ak4535_dai;
+extern struct snd_soc_dai ak4535_dai;
 extern struct snd_soc_codec_device soc_codec_dev_ak4535;
 
 #endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index e73fcfd9f5c..9deb8c74fdf 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -201,7 +201,7 @@ static struct {
  * driver what the input settings can be.  This would need to be implemented
  * for stand-alone mode to work.
  */
-static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
  * data for playback only, but ASoC currently does not support different
  * formats for playback vs. record.
  */
-static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
  * board does not have the MUTEA or MUTEB pins connected to such circuitry,
  * then this function will do nothing.
  */
-static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute)
+static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	int reg6;
@@ -667,7 +667,7 @@ error:
 
 #endif /* USE_I2C*/
 
-struct snd_soc_codec_dai cs4270_dai = {
+struct snd_soc_dai cs4270_dai = {
 	.name = "CS4270",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h
index 0ced49b7804..adc6cd9667d 100644
--- a/sound/soc/codecs/cs4270.h
+++ b/sound/soc/codecs/cs4270.h
@@ -16,7 +16,7 @@
  * The ASoC codec DAI structure for the CS4270.  Assign this structure to
  * the .codec_dai field of your machine driver's snd_soc_dai_link structure.
  */
-extern struct snd_soc_codec_dai cs4270_dai;
+extern struct snd_soc_dai cs4270_dai;
 
 /*
  * The ASoC codec device structure for the CS4270.  Assign this structure
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 954d39b7c04..b1dce5f459d 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -814,7 +814,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
+static int aic3x_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON;
@@ -831,7 +831,7 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -841,7 +841,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -990,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-struct snd_soc_codec_dai aic3x_dai = {
+struct snd_soc_dai aic3x_dai = {
 	.name = "aic3x",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index e6009461063..d76c079b86e 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -228,7 +228,7 @@ struct aic3x_setup_data {
 	unsigned int gpio_func[2];
 };
 
-extern struct snd_soc_codec_dai aic3x_dai;
+extern struct snd_soc_dai aic3x_dai;
 extern struct snd_soc_codec_device soc_codec_dev_aic3x;
 
 #endif /* _AIC3X_H */
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 6d5335b14d5..a52d6d9e007 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -372,7 +372,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -499,7 +499,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
 	uda1380_write(codec, UDA1380_CLK, clk);
 }
 
-static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute)
+static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
@@ -542,7 +542,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 		       SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		       SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
-struct snd_soc_codec_dai uda1380_dai[] = {
+struct snd_soc_dai uda1380_dai[] = {
 {
 	.name = "UDA1380",
 	.playback = {
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
index f9d885c8bf0..50c603e2c9f 100644
--- a/sound/soc/codecs/uda1380.h
+++ b/sound/soc/codecs/uda1380.h
@@ -83,7 +83,7 @@ struct uda1380_setup_data {
 #define UDA1380_DAI_PLAYBACK	1 /* playback DAI */
 #define UDA1380_DAI_CAPTURE	2 /* capture DAI */
 
-extern struct snd_soc_codec_dai uda1380_dai[3];
+extern struct snd_soc_dai uda1380_dai[3];
 extern struct snd_soc_codec_device soc_codec_dev_uda1380;
 
 #endif /* _UDA1380_H */
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index b549f6753ab..67325fd9544 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -332,7 +332,7 @@ static void pll_factors(unsigned int target, unsigned int source)
 	pll_div.k = K;
 }
 
-static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
 		int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -368,7 +368,7 @@ static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
 /*
  * Configure WM8510 clock dividers.
  */
-static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -402,7 +402,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -510,7 +510,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8510_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
@@ -554,7 +554,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
-struct snd_soc_codec_dai wm8510_dai = {
+struct snd_soc_dai wm8510_dai = {
 	.name = "WM8510 HiFi",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
index c862e7b7d53..f5d2e42eb3f 100644
--- a/sound/soc/codecs/wm8510.h
+++ b/sound/soc/codecs/wm8510.h
@@ -97,7 +97,7 @@ struct wm8510_setup_data {
 	unsigned short i2c_address;
 };
 
-extern struct snd_soc_codec_dai wm8510_dai;
+extern struct snd_soc_dai wm8510_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8510;
 
 #endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 3ff42ad65ed..369d39c3f74 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -318,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream)
 	}
 }
 
-static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8731_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
@@ -330,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -349,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 }
 
 
-static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -443,7 +443,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-struct snd_soc_codec_dai wm8731_dai = {
+struct snd_soc_dai wm8731_dai = {
 	.name = "WM8731",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 5bcab6a7afb..99f2e3c60e3 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -38,7 +38,7 @@ struct wm8731_setup_data {
 	unsigned short i2c_address;
 };
 
-extern struct snd_soc_codec_dai wm8731_dai;
+extern struct snd_soc_dai wm8731_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 #endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index eb460c9aa63..e23cb09f0d1 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -536,7 +536,7 @@ static inline int get_coeff(int mclk, int rate)
 	return -EINVAL;
 }
 
-static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -554,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 	return -EINVAL;
 }
 
-static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -647,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8750_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
@@ -692,7 +692,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-struct snd_soc_codec_dai wm8750_dai = {
+struct snd_soc_dai wm8750_dai = {
 	.name = "WM8750",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
index a97a54a6348..8ef30e628b2 100644
--- a/sound/soc/codecs/wm8750.h
+++ b/sound/soc/codecs/wm8750.h
@@ -61,7 +61,7 @@ struct wm8750_setup_data {
 	unsigned short i2c_address;
 };
 
-extern struct snd_soc_codec_dai wm8750_dai;
+extern struct snd_soc_dai wm8750_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8750;
 
 #endif
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index be01a738f18..8604809f0c3 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -740,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 	pll_div->k = K;
 }
 
-static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
 		int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
 	u16 reg, enable;
@@ -863,7 +863,7 @@ static int get_coeff(int mclk, int rate)
 /*
  * Clock after PLL and dividers
  */
-static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -890,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's ADC and Voice DAC format.
  */
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -960,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
 /*
  * Set's PCM dai fmt and BCLK.
  */
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1026,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1054,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's HiFi DAC format.
  */
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1087,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's I2S DAI format.
  */
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1195,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1210,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
@@ -1218,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1233,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1250,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8753_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
@@ -1316,7 +1316,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
-static const struct snd_soc_codec_dai wm8753_all_dai[] = {
+static const struct snd_soc_dai wm8753_all_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "WM8753 HiFi",
 	.id = 1,
@@ -1456,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = {
 },
 };
 
-struct snd_soc_codec_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[2];
 EXPORT_SYMBOL_GPL(wm8753_dai);
 
 static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 95e2a1f5316..44f5f1ff0cc 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -120,7 +120,7 @@ struct wm8753_setup_data {
 #define WM8753_DAI_HIFI		0
 #define WM8753_DAI_VOICE		1
 
-extern struct snd_soc_codec_dai wm8753_dai[2];
+extern struct snd_soc_dai wm8753_dai[2];
 extern struct snd_soc_codec_device soc_codec_dev_wm8753;
 
 #endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index a1371b73ba7..3ecce5168e9 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1029,7 +1029,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 	pll_div->k = K;
 }
 
-static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
 		int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
 	u16 reg;
@@ -1065,7 +1065,7 @@ static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
 /*
  * Clock after PLL and dividers
  */
-static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+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;
@@ -1078,7 +1078,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's ADC and Voice DAC format.
  */
-static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1131,7 +1131,7 @@ static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1196,7 +1196,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8990_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 val;
@@ -1329,7 +1329,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
  * 1. ADC/DAC on Primary Interface
  * 2. ADC on Primary Interface/DAC on secondary
  */
-struct snd_soc_codec_dai wm8990_dai = {
+struct snd_soc_dai wm8990_dai = {
 /* ADC/DAC on primary */
 	.name = "WM8990 ADC/DAC Primary",
 	.id = 1,
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index bf9f8823dfc..6bea5748528 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -825,7 +825,7 @@ struct wm8990_setup_data {
 #define WM8990_ADCCLK_DIV 2
 #define WM8990_BCLK_DIV 3
 
-extern struct snd_soc_codec_dai wm8990_dai;
+extern struct snd_soc_dai wm8990_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8990;
 
 #endif	/* __WM8990REGISTERDEFS_H__ */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 47390113bd0..9fc8edd8222 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -532,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
-struct snd_soc_codec_dai wm9712_dai[] = {
+struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
 	.type = SND_SOC_DAI_AC97_BUS,
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
index 719105d61e6..d29e8a18ca6 100644
--- a/sound/soc/codecs/wm9712.h
+++ b/sound/soc/codecs/wm9712.h
@@ -8,7 +8,7 @@
 #define WM9712_DAI_AC97_HIFI	0
 #define WM9712_DAI_AC97_AUX		1
 
-extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_dai wm9712_dai[2];
 extern struct snd_soc_codec_device soc_codec_dev_wm9712;
 
 #endif
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a4806189044..38d1fe0971f 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
 		int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
  * Tristate the PCM DAI lines, tristate can be disabled by calling
  * wm9713_set_dai_fmt()
  */
-static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
 	int tristate)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
  * Configure WM9713 clock dividers.
  * Voice DAC needs 256 FS
  */
-static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1018,7 +1018,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
-struct snd_soc_codec_dai wm9713_dai[] = {
+struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
 	.type = SND_SOC_DAI_AC97_BUS,
diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h
index d357b6c8134..63b8d81756e 100644
--- a/sound/soc/codecs/wm9713.h
+++ b/sound/soc/codecs/wm9713.h
@@ -46,7 +46,7 @@
 #define WM9713_DAI_PCM_VOICE	2
 
 extern struct snd_soc_codec_device soc_codec_dev_wm9713;
-extern struct snd_soc_codec_dai wm9713_dai[3];
+extern struct snd_soc_dai wm9713_dai[3];
 
 int wm9713_reset(struct snd_soc_codec *codec,  int try_warm);
 
-- 
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