From a6ba2b2dabb583e7820e567fb309d771b50cb9ff Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 8 Jan 2009 15:16:16 +0000
Subject: ASoC: Implement WM8350 headphone jack detection

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8350.c | 116 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8350.h |   8 ++++
 2 files changed, 124 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e3989d406f54..47a9dabb5235 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -51,10 +51,17 @@ struct wm8350_output {
 	u16 mute;
 };
 
+struct wm8350_jack_data {
+	struct snd_soc_jack *jack;
+	int report;
+};
+
 struct wm8350_data {
 	struct snd_soc_codec codec;
 	struct wm8350_output out1;
 	struct wm8350_output out2;
+	struct wm8350_jack_data hpl;
+	struct wm8350_jack_data hpr;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 };
 
@@ -1328,6 +1335,95 @@ static int wm8350_resume(struct platform_device *pdev)
 	return 0;
 }
 
+static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+{
+	struct wm8350_data *priv = data;
+	u16 reg;
+	int report;
+	int mask;
+	struct wm8350_jack_data *jack = NULL;
+
+	switch (irq) {
+	case WM8350_IRQ_CODEC_JCK_DET_L:
+		jack = &priv->hpl;
+		mask = WM8350_JACK_L_LVL;
+		break;
+
+	case WM8350_IRQ_CODEC_JCK_DET_R:
+		jack = &priv->hpr;
+		mask = WM8350_JACK_R_LVL;
+		break;
+
+	default:
+		BUG();
+	}
+
+	if (!jack->jack) {
+		dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
+		return;
+	}
+
+	/* Debounce */
+	msleep(200);
+
+	reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
+	if (reg & mask)
+		report = jack->report;
+	else
+		report = 0;
+
+	snd_soc_jack_report(jack->jack, report, jack->report);
+}
+
+/**
+ * wm8350_hp_jack_detect - Enable headphone jack detection.
+ *
+ * @codec:  WM8350 codec
+ * @which:  left or right jack detect signal
+ * @jack:   jack to report detection events on
+ * @report: value to report
+ *
+ * Enables the headphone jack detection of the WM8350.
+ */
+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 *wm8350 = codec->control_data;
+	int irq;
+	int ena;
+
+	switch (which) {
+	case WM8350_JDL:
+		priv->hpl.jack = jack;
+		priv->hpl.report = report;
+		irq = WM8350_IRQ_CODEC_JCK_DET_L;
+		ena = WM8350_JDL_ENA;
+		break;
+
+	case WM8350_JDR:
+		priv->hpr.jack = jack;
+		priv->hpr.report = report;
+		irq = WM8350_IRQ_CODEC_JCK_DET_R;
+		ena = WM8350_JDR_ENA;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+	wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
+
+	/* Sync status */
+	wm8350_hp_jack_handler(wm8350, irq, priv);
+
+	wm8350_unmask_irq(wm8350, irq);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
+
 static struct snd_soc_codec *wm8350_codec;
 
 static int wm8350_probe(struct platform_device *pdev)
@@ -1381,6 +1477,13 @@ static int wm8350_probe(struct platform_device *pdev)
 	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
 			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
 
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
+			    wm8350_hp_jack_handler, priv);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
+			    wm8350_hp_jack_handler, priv);
+
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "failed to create pcms\n");
@@ -1411,8 +1514,21 @@ static int wm8350_remove(struct platform_device *pdev)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
 	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *priv = codec->private_data;
 	int ret;
 
+	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+			  WM8350_JDL_ENA | WM8350_JDR_ENA);
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+
+	priv->hpl.jack = NULL;
+	priv->hpr.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 cc2887aa6c38..d11bd9288cf9 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -17,4 +17,12 @@
 extern struct snd_soc_dai wm8350_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8350;
 
+enum wm8350_jack {
+	WM8350_JDL = 1,
+	WM8350_JDR = 2,
+};
+
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+			  struct snd_soc_jack *jack, int report);
+
 #endif
-- 
cgit v1.2.3


From 3e8e1952e3a3dd59b11233a532ca68e6471742d9 Mon Sep 17 00:00:00 2001
From: Ian Molton <ian@mnementh.co.uk>
Date: Fri, 9 Jan 2009 00:23:21 +0000
Subject: ASoC: cleanup duplicated code.

Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.

This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.

[Edited to raise priority of error log message and document parameters.
 -- broonie]

Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h            |  2 ++
 sound/soc/codecs/ad1980.c      | 17 ++------------
 sound/soc/codecs/ak4535.c      | 18 ++-------------
 sound/soc/codecs/ssm2602.c     | 18 ++-------------
 sound/soc/codecs/tlv320aic23.c | 21 ++----------------
 sound/soc/codecs/tlv320aic3x.c | 19 ++--------------
 sound/soc/codecs/twl4030.c     | 19 ++--------------
 sound/soc/codecs/uda134x.c     | 50 ++++++++++++++----------------------------
 sound/soc/codecs/uda1380.c     | 18 ++-------------
 sound/soc/codecs/wm8350.c      | 18 ++-------------
 sound/soc/codecs/wm8510.c      | 19 ++--------------
 sound/soc/codecs/wm8580.c      | 17 ++------------
 sound/soc/codecs/wm8728.c      | 18 ++-------------
 sound/soc/codecs/wm8731.c      | 19 ++--------------
 sound/soc/codecs/wm8750.c      | 18 ++-------------
 sound/soc/codecs/wm8753.c      | 18 ++-------------
 sound/soc/codecs/wm8900.c      | 19 ++--------------
 sound/soc/codecs/wm8903.c      | 18 ++-------------
 sound/soc/codecs/wm8971.c      | 18 ++-------------
 sound/soc/codecs/wm8990.c      | 18 ++-------------
 sound/soc/codecs/wm9712.c      | 18 ++-------------
 sound/soc/codecs/wm9713.c      | 18 ++-------------
 sound/soc/soc-core.c           | 31 ++++++++++++++++++++++++++
 23 files changed, 89 insertions(+), 360 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9c3ef6a3e9fb..9d5a0362a055 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -214,6 +214,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
  */
 struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 	void *data, char *long_name);
+int snd_soc_add_controls(struct snd_soc_codec *codec,
+	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 73fdbb4d4a3d..c3c5d0eee37a 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src),
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
 };
 
-/* add non dapm controls */
-static int ad1980_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card, snd_soc_cnew(
-				&ad1980_snd_ac97_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 static unsigned int ac97_read(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
@@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 	ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
 	ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
 
-	ad1980_add_controls(codec);
+	snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
+				ARRAY_SIZE(ad1980_snd_ac97_controls));
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register card\n");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 81300d8d42ca..f17c363cb1db 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = {
 	SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
 };
 
-/* add non dapm controls */
-static int ak4535_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Mono 1 Mixer */
 static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
@@ -510,7 +495,8 @@ static int ak4535_init(struct snd_soc_device *socdev)
 	/* power on device */
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	ak4535_add_controls(codec);
+	snd_soc_add_controls(codec, ak4535_snd_controls,
+				ARRAY_SIZE(ak4535_snd_controls));
 	ak4535_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index cac373616768..ec7fe3b7b0cb 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]),
 SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
 };
 
-/* add non dapm controls */
-static int ssm2602_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Output Mixer */
 static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
@@ -622,7 +607,8 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 			APANA_ENABLE_MIC_BOOST);
 	ssm2602_write(codec, SSM2602_PWR, 0);
 
-	ssm2602_add_controls(codec);
+	snd_soc_add_controls(codec, ssm2602_snd_controls,
+				ARRAY_SIZE(ssm2602_snd_controls));
 	ssm2602_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index cfdea007c4cb..a0e47c1dcd64 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
 	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
 };
 
-/* add non dapm controls */
-static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
-{
-
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-
-}
-
 /* PGA Mixer controls for Line and Mic switch */
 static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
@@ -718,7 +700,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
 
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
 
-	tlv320aic23_add_controls(codec);
+	snd_soc_add_controls(codec, tlv320aic23_snd_controls,
+				ARRAY_SIZE(tlv320aic23_snd_controls));
 	tlv320aic23_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b47a749c5ea2..36ab0198ca3f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -311,22 +311,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
-/* add non dapm controls */
-static int aic3x_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&aic3x_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
 SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@@ -1224,7 +1208,8 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
 	aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
 
-	aic3x_add_controls(codec);
+	snd_soc_add_controls(codec, aic3x_snd_controls,
+				ARRAY_SIZE(aic3x_snd_controls));
 	aic3x_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index fd0f338374a7..253063fd319a 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -670,22 +670,6 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		0, 3, 5, 0, input_gain_tlv),
 };
 
-/* add non dapm controls */
-static int twl4030_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&twl4030_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	/* Left channel inputs */
 	SND_SOC_DAPM_INPUT("MAINMIC"),
@@ -1233,7 +1217,8 @@ static int twl4030_init(struct snd_soc_device *socdev)
 	/* power on device */
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	twl4030_add_controls(codec);
+	snd_soc_add_controls(codec, twl4030_snd_controls,
+				ARRAY_SIZE(twl4030_snd_controls));
 	twl4030_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index a2c5064a774b..277825d155a6 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -431,39 +431,6 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
 };
 
-static int uda134x_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i, n;
-	const struct snd_kcontrol_new *ctrls;
-	struct uda134x_platform_data *pd = codec->control_data;
-
-	switch (pd->model) {
-	case UDA134X_UDA1340:
-	case UDA134X_UDA1344:
-		n = ARRAY_SIZE(uda1340_snd_controls);
-		ctrls = uda1340_snd_controls;
-		break;
-	case UDA134X_UDA1341:
-		n = ARRAY_SIZE(uda1341_snd_controls);
-		ctrls = uda1341_snd_controls;
-		break;
-	default:
-		printk(KERN_ERR "%s unkown codec type: %d",
-		       __func__, pd->model);
-		return -EINVAL;
-	}
-
-	for (i = 0; i < n; i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&ctrls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 struct snd_soc_dai uda134x_dai = {
 	.name = "UDA134X",
 	/* playback capabilities */
@@ -572,7 +539,22 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = uda134x_add_controls(codec);
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1344:
+		ret = snd_soc_add_controls(codec, uda1340_snd_controls,
+					ARRAY_SIZE(uda1340_snd_controls));
+	break;
+	case UDA134X_UDA1341:
+		ret = snd_soc_add_controls(codec, uda1341_snd_controls,
+					ARRAY_SIZE(uda1341_snd_controls));
+	break;
+	default:
+		printk(KERN_ERR "%s unkown codec type: %d",
+			__func__, pd->model);
+	return -EINVAL;
+	}
+
 	if (ret < 0) {
 		printk(KERN_ERR "UDA134X: failed to register controls\n");
 		goto pcm_err;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index e6bf0844fbf3..a957b4365b9d 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -271,21 +271,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
 	SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
 };
 
-/* add non dapm controls */
-static int uda1380_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Input mux */
 static const struct snd_kcontrol_new uda1380_input_mux_control =
 	SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
@@ -675,7 +660,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	}
 
 	/* uda1380 init */
-	uda1380_add_controls(codec);
+	snd_soc_add_controls(codec, uda1380_snd_controls,
+				ARRAY_SIZE(uda1380_snd_controls));
 	uda1380_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 47a9dabb5235..2e0db29b4998 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -782,21 +782,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Beep", NULL, "IN3R PGA"},
 };
 
-static int wm8350_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8350_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static int wm8350_add_widgets(struct snd_soc_codec *codec)
 {
 	int ret;
@@ -1490,7 +1475,8 @@ static int wm8350_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	wm8350_add_controls(codec);
+	snd_soc_add_controls(codec, wm8350_snd_controls,
+				ARRAY_SIZE(wm8350_snd_controls));
 	wm8350_add_widgets(codec);
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 40f8238df717..abe7cce87714 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
 SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
 };
 
-/* add non dapm controls */
-static int wm8510_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8510_snd_controls[i], codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Speaker Output Mixer */
 static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
@@ -656,7 +640,8 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	/* power on device */
 	codec->bias_level = SND_SOC_BIAS_OFF;
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8510_add_controls(codec);
+	snd_soc_add_controls(codec, wm8510_snd_controls,
+				ARRAY_SIZE(wm8510_snd_controls));
 	wm8510_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d004e5845298..9b75a377453e 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -330,20 +330,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0),
 SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0),
 };
 
-/* Add non-DAPM controls */
-static int wm8580_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8580_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
 static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = {
 SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1),
 SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1),
@@ -866,7 +852,8 @@ static int wm8580_init(struct snd_soc_device *socdev)
 		goto pcm_err;
 	}
 
-	wm8580_add_controls(codec);
+	snd_soc_add_controls(codec, wm8580_snd_controls,
+				ARRAY_SIZE(wm8580_snd_controls));
 	wm8580_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 80b11983e137..defa310bc7d9 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
 SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
 };
 
-static int wm8728_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8728_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /*
  * DAPM controls.
  */
@@ -330,7 +315,8 @@ static int wm8728_init(struct snd_soc_device *socdev)
 	/* power on device */
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	wm8728_add_controls(codec);
+	snd_soc_add_controls(codec, wm8728_snd_controls,
+				ARRAY_SIZE(wm8728_snd_controls));
 	wm8728_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c444b9f2701e..96d6e1aeaf43 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -129,22 +129,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
 SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
 };
 
-/* add non dapm controls */
-static int wm8731_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8731_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Output Mixer */
 static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
@@ -543,7 +527,8 @@ static int wm8731_init(struct snd_soc_device *socdev)
 	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
 	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
 
-	wm8731_add_controls(codec);
+	snd_soc_add_controls(codec, wm8731_snd_controls,
+				ARRAY_SIZE(wm8731_snd_controls));
 	wm8731_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 5997fa68e0d5..1578569793a2 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
 
 };
 
-/* add non dapm controls */
-static int wm8750_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8750_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * DAPM Controls
  */
@@ -816,7 +801,8 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
 	wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
 
-	wm8750_add_controls(codec);
+	snd_soc_add_controls(codec, wm8750_snd_controls,
+				ARRAY_SIZE(wm8750_snd_controls));
 	wm8750_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6c21b50c9375..7283178e0eb5 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]),
 SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
 };
 
-/* add non dapm controls */
-static int wm8753_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * _DAPM_ Controls
  */
@@ -1603,7 +1588,8 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
 	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
 
-	wm8753_add_controls(codec);
+	snd_soc_add_controls(codec, wm8753_snd_controls,
+				ARRAY_SIZE(wm8753_snd_controls));
 	wm8753_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 6767de10ded0..1e08d4f065f2 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1,
 
 };
 
-/* add non dapm controls */
-static int wm8900_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8900_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new wm8900_dapm_loutput2_control =
 SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0);
 
@@ -1439,7 +1423,8 @@ static int wm8900_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	wm8900_add_controls(codec);
+	snd_soc_add_controls(codec, wm8900_snd_controls,
+				ARRAY_SIZE(wm8900_snd_controls));
 	wm8900_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index bde74546db4a..6ff34b957dce 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume",
 		 0, 63, 0, out_tlv),
 };
 
-static int wm8903_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8903_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new linput_mode_mux =
 	SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum);
 
@@ -1737,7 +1722,8 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	wm8903_add_controls(socdev->codec);
+	snd_soc_add_controls(socdev->codec, wm8903_snd_controls,
+				ARRAY_SIZE(wm8903_snd_controls));
 	wm8903_add_widgets(socdev->codec);
 
 	ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 88ead7f8dd98..c8bd9b06f330 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = {
 	SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
 };
 
-/* add non-DAPM controls */
-static int wm8971_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8971_snd_controls[i],
-					     codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * DAPM Controls
  */
@@ -745,7 +730,8 @@ static int wm8971_init(struct snd_soc_device *socdev)
 	reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
 	wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
 
-	wm8971_add_controls(codec);
+	snd_soc_add_controls(codec, wm8971_snd_controls,
+				ARRAY_SIZE(wm8971_snd_controls));
 	wm8971_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5b5afc144478..6b2778632d5e 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -417,21 +417,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
 
 };
 
-/* add non dapm controls */
-static int wm8990_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8990_snd_controls[i], codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * _DAPM_ Controls
  */
@@ -1460,7 +1445,8 @@ static int wm8990_init(struct snd_soc_device *socdev)
 	wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
 	wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
 
-	wm8990_add_controls(codec);
+	snd_soc_add_controls(codec, wm8990_snd_controls,
+				ARRAY_SIZE(wm8990_snd_controls));
 	wm8990_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index af83d629078a..1b0ace0f4dca 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
 SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
 };
 
-/* add non dapm controls */
-static int wm9712_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm9712_snd_ac97_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path.
@@ -698,7 +683,8 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
 
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm9712_add_controls(codec);
+	snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
+				ARRAY_SIZE(wm9712_snd_ac97_controls));
 	wm9712_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index f3ca8aaf0139..a45622620db7 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -190,21 +190,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
 };
 
-/* add non dapm controls */
-static int wm9713_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm9713_snd_ac97_controls[i],
-					codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path using the current
@@ -1245,7 +1230,8 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	reg = ac97_read(codec, AC97_CD) & 0x7fff;
 	ac97_write(codec, AC97_CD, reg);
 
-	wm9713_add_controls(codec);
+	snd_soc_add_controls(codec, wm9713_snd_ac97_controls,
+				ARRAY_SIZE(wm9713_snd_ac97_controls));
 	wm9713_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6cbe7e82f238..d3b97a7542e0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1494,6 +1494,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 }
 EXPORT_SYMBOL_GPL(snd_soc_cnew);
 
+/**
+ * snd_soc_add_controls - add an array of controls to a codec.
+ * Convienience function to add a list of controls. Many codecs were
+ * duplicating this code.
+ *
+ * @codec: codec to add controls to
+ * @controls: array of controls to add
+ * @num_controls: number of elements in the array
+ *
+ * Return 0 for success, else error.
+ */
+int snd_soc_add_controls(struct snd_soc_codec *codec,
+	const struct snd_kcontrol_new *controls, int num_controls)
+{
+	struct snd_card *card = codec->card;
+	int err, i;
+
+	for (i = 0; i < num_controls; i++) {
+		const struct snd_kcontrol_new *control = &controls[i];
+		err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
+		if (err < 0) {
+			dev_err(codec->dev, "%s: Failed to add %s\n",
+				codec->name, control->name);
+			return err;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_controls);
+
 /**
  * snd_soc_info_enum_double - enumerated double mixer info callback
  * @kcontrol: mixer control
-- 
cgit v1.2.3


From 2aceefefc891e85d336c1d95d9d89fd785f5d44c Mon Sep 17 00:00:00 2001
From: Ian Molton <ian@mnementh.co.uk>
Date: Fri, 16 Jan 2009 11:04:18 +0000
Subject: ASoC: Driver for the WM9705 AC97 codec.

This driver adds support for the wm9705 ac97 codec. The driver supports
audio input and output.

Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   4 +
 sound/soc/codecs/Makefile |   3 +
 sound/soc/codecs/wm9705.c | 410 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm9705.h |  14 ++
 4 files changed, 431 insertions(+)
 create mode 100644 sound/soc/codecs/wm9705.c
 create mode 100644 sound/soc/codecs/wm9705.h

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d0e0d691ae51..cb5fcd605acc 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -34,6 +34,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8903 if I2C
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8990 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
         help
@@ -144,6 +145,9 @@ config SND_SOC_WM8971
 config SND_SOC_WM8990
 	tristate
 
+config SND_SOC_WM9705
+	tristate
+
 config SND_SOC_WM9712
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index c4ddc9aa2bbd..3664cdc300b2 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,6 +23,7 @@ snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 
@@ -51,5 +52,7 @@ obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
+obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
new file mode 100644
index 000000000000..cb26b6a77ffb
--- /dev/null
+++ b/sound/soc/codecs/wm9705.c
@@ -0,0 +1,410 @@
+/*
+ * wm9705.c  --  ALSA Soc WM9705 codec support
+ *
+ * Copyright 2008 Ian Molton <spyro@f2s.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; Version 2 of the  License only.
+ *
+ */
+
+#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/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+/*
+ * WM9705 register cache
+ */
+static const u16 wm9705_reg[] = {
+	0x6150, 0x8000, 0x8000, 0x8000, /* 0x0  */
+	0x0000, 0x8000, 0x8008, 0x8008, /* 0x8  */
+	0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */
+	0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */
+	0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */
+	0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */
+	0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */
+};
+
+static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = {
+	SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+	SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
+	SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+	SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+	SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+	SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
+	SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+	SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+	SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1),
+	SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1),
+	SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1),
+	SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1),
+	SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1),
+	SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0),
+	SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
+	SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+};
+
+static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
+static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
+	"Line", "Stereo Mix", "Mono Mix", "Phone"};
+
+static const struct soc_enum wm9705_enum_mic =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
+static const struct soc_enum wm9705_enum_rec_l =
+	SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
+static const struct soc_enum wm9705_enum_rec_r =
+	SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
+
+/* Headphone Mixer */
+static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+	SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1),
+	SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1),
+	SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1),
+	SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1),
+};
+
+/* Mic source */
+static const struct snd_kcontrol_new wm9705_mic_src_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_mic);
+
+/* Capture source */
+static const struct snd_kcontrol_new wm9705_capture_selectl_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_rec_l);
+static const struct snd_kcontrol_new wm9705_capture_selectr_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_rec_r);
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_mic_src_controls),
+	SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_capture_selectl_controls),
+	SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_capture_selectr_controls),
+	SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0,
+		&wm9705_hp_mixer_controls[0],
+		ARRAY_SIZE(wm9705_hp_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+	SND_SOC_DAPM_OUTPUT("MONOOUT"),
+	SND_SOC_DAPM_INPUT("PHONE"),
+	SND_SOC_DAPM_INPUT("LINEINL"),
+	SND_SOC_DAPM_INPUT("LINEINR"),
+	SND_SOC_DAPM_INPUT("CDINL"),
+	SND_SOC_DAPM_INPUT("CDINR"),
+	SND_SOC_DAPM_INPUT("PCBEEP"),
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+/* Audio map
+ * WM9705 has no switches to disable the route from the inputs to the HP mixer
+ * so in order to prevent active inputs from forcing the audio outputs to be
+ * constantly enabled, we use the mutes on those inputs to simulate such
+ * controls.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* HP mixer */
+	{"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"},
+	{"HP Mixer", "CD Playback Switch", "CD PGA"},
+	{"HP Mixer", "Mic Playback Switch", "Mic PGA"},
+	{"HP Mixer", "Phone Playback Switch", "Phone PGA"},
+	{"HP Mixer", "Line Playback Switch", "Line PGA"},
+	{"HP Mixer", NULL, "Left DAC"},
+	{"HP Mixer", NULL, "Right DAC"},
+
+	/* mono mixer */
+	{"Mono Mixer", NULL, "HP Mixer"},
+
+	/* outputs */
+	{"Headphone PGA", NULL, "HP Mixer"},
+	{"HPOUTL", NULL, "Headphone PGA"},
+	{"HPOUTR", NULL, "Headphone PGA"},
+	{"Line out PGA", NULL, "HP Mixer"},
+	{"LOUT", NULL, "Line out PGA"},
+	{"ROUT", NULL, "Line out PGA"},
+	{"Mono PGA", NULL, "Mono Mixer"},
+	{"MONOOUT", NULL, "Mono PGA"},
+
+	/* inputs */
+	{"CD PGA", NULL, "CDINL"},
+	{"CD PGA", NULL, "CDINR"},
+	{"Line PGA", NULL, "LINEINL"},
+	{"Line PGA", NULL, "LINEINR"},
+	{"Phone PGA", NULL, "PHONE"},
+	{"Mic Source", "Mic 1", "MIC1"},
+	{"Mic Source", "Mic 2", "MIC2"},
+	{"Mic PGA", NULL, "Mic Source"},
+	{"PCBEEP PGA", NULL, "PCBEEP"},
+
+	/* Left capture selector */
+	{"Left Capture Source", "Mic", "Mic Source"},
+	{"Left Capture Source", "CD", "CDINL"},
+	{"Left Capture Source", "Line", "LINEINL"},
+	{"Left Capture Source", "Stereo Mix", "HP Mixer"},
+	{"Left Capture Source", "Mono Mix", "HP Mixer"},
+	{"Left Capture Source", "Phone", "PHONE"},
+
+	/* Right capture source */
+	{"Right Capture Source", "Mic", "Mic Source"},
+	{"Right Capture Source", "CD", "CDINR"},
+	{"Right Capture Source", "Line", "LINEINR"},
+	{"Right Capture Source", "Stereo Mix", "HP Mixer"},
+	{"Right Capture Source", "Mono Mix", "HP Mixer"},
+	{"Right Capture Source", "Phone", "PHONE"},
+
+	{"ADC PGA", NULL, "Left Capture Source"},
+	{"ADC PGA", NULL, "Right Capture Source"},
+
+	/* ADC's */
+	{"Left ADC",  NULL, "ADC PGA"},
+	{"Right ADC", NULL, "ADC PGA"},
+};
+
+static int wm9705_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
+					ARRAY_SIZE(wm9705_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+/* We use a register cache to enhance read performance. */
+static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	switch (reg) {
+	case AC97_RESET:
+	case AC97_VENDOR_ID1:
+	case AC97_VENDOR_ID2:
+		return soc_ac97_ops.read(codec->ac97, reg);
+	default:
+		reg = reg >> 1;
+
+		if (reg >= (ARRAY_SIZE(wm9705_reg)))
+			return -EIO;
+
+		return cache[reg];
+	}
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	reg = reg >> 1;
+	if (reg < (ARRAY_SIZE(wm9705_reg)))
+		cache[reg] = val;
+
+	return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int reg;
+	u16 vra;
+
+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return ac97_write(codec, reg, runtime->rate);
+}
+
+#define WM9705_AC97_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)
+
+struct snd_soc_dai wm9705_dai[] = {
+	{
+		.name = "AC97 HiFi",
+		.ac97_control = 1,
+		.playback = {
+			.stream_name = "HiFi Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.stream_name = "HiFi Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.ops = {
+			.prepare = ac97_prepare,
+		},
+	},
+	{
+		.name = "AC97 Aux",
+		.playback = {
+			.stream_name = "Aux Playback",
+			.channels_min = 1,
+			.channels_max = 1,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+	}
+};
+EXPORT_SYMBOL_GPL(wm9705_dai);
+
+static int wm9705_reset(struct snd_soc_codec *codec)
+{
+	if (soc_ac97_ops.reset) {
+		soc_ac97_ops.reset(codec->ac97);
+		if (ac97_read(codec, 0) == wm9705_reg[0])
+			return 0; /* Success */
+	}
+
+	return -EIO;
+}
+
+static int wm9705_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "WM9705 SoC Audio Codec\n");
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+	codec->reg_cache_size = sizeof(wm9705_reg);
+	codec->reg_cache_step = 2;
+
+	codec->name = "WM9705";
+	codec->owner = THIS_MODULE;
+	codec->dai = wm9705_dai;
+	codec->num_dai = ARRAY_SIZE(wm9705_dai);
+	codec->write = ac97_write;
+	codec->read = ac97_read;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9705: failed to register AC97 codec\n");
+		goto codec_err;
+	}
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+	ret = wm9705_reset(codec);
+	if (ret)
+		goto reset_err;
+
+	snd_soc_add_controls(codec, wm9705_snd_ac97_controls,
+				ARRAY_SIZE(wm9705_snd_ac97_controls));
+	wm9705_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9705: failed to register card\n");
+		goto pcm_err;
+	}
+
+	return 0;
+
+reset_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	snd_soc_free_ac97_codec(codec);
+codec_err:
+	kfree(codec->reg_cache);
+cache_err:
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+	return ret;
+}
+
+static int wm9705_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec == NULL)
+		return 0;
+
+	snd_soc_dapm_free(socdev);
+	snd_soc_free_pcms(socdev);
+	snd_soc_free_ac97_codec(codec);
+	kfree(codec->reg_cache);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9705 = {
+	.probe = 	wm9705_soc_probe,
+	.remove = 	wm9705_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
+
+MODULE_DESCRIPTION("ASoC WM9705 driver");
+MODULE_AUTHOR("Ian Molton");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h
new file mode 100644
index 000000000000..d380f110f9e2
--- /dev/null
+++ b/sound/soc/codecs/wm9705.h
@@ -0,0 +1,14 @@
+/*
+ * wm9705.h  --  WM9705 Soc Audio driver
+ */
+
+#ifndef _WM9705_H
+#define _WM9705_H
+
+#define WM9705_DAI_AC97_HIFI	0
+#define WM9705_DAI_AC97_AUX	1
+
+extern struct snd_soc_dai wm9705_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9705;
+
+#endif
-- 
cgit v1.2.3


From c42f69bb064333624dcc1452ed109441c3c9e7b4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 16 Jan 2009 16:31:03 +0000
Subject: ASoC: Ignore output frequency for WM9713 PLL

The WM9713 driver does not support configuring the PLL output frequency
so the output frequency parameter is irrelevant. Allow users to set it
to zero by ignoring it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm9713.c | 10 ++++------
 1 file changed, 4 insertions(+), 6 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a45622620db7..e636d8a18ed6 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -32,7 +32,6 @@
 
 struct wm9713_priv {
 	u32 pll_in; /* PLL input frequency */
-	u32 pll_out; /* PLL output frequency */
 };
 
 static unsigned int ac97_read(struct snd_soc_codec *codec,
@@ -723,13 +722,13 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	struct _pll_div pll_div;
 
 	/* turn PLL off ? */
-	if (freq_in == 0 || freq_out == 0) {
+	if (freq_in == 0) {
 		/* disable PLL power and select ext source */
 		reg = ac97_read(codec, AC97_HANDSET_RATE);
 		ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
 		reg = ac97_read(codec, AC97_EXTENDED_MID);
 		ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
-		wm9713->pll_out = 0;
+		wm9713->pll_in = 0;
 		return 0;
 	}
 
@@ -773,7 +772,6 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
 	reg = ac97_read(codec, AC97_HANDSET_RATE);
 	ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
-	wm9713->pll_out = freq_out;
 	wm9713->pll_in = freq_in;
 
 	/* wait 10ms AC97 link frames for the link to stabilise */
@@ -1149,8 +1147,8 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* do we need to re-start the PLL ? */
-	if (wm9713->pll_out)
-		wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+	if (wm9713->pll_in)
+		wm9713_set_pll(codec, 0, wm9713->pll_in, 0);
 
 	/* only synchronise the codec if warm reset failed */
 	if (ret == 0) {
-- 
cgit v1.2.3


From 91432e976ff1323e5dd6f52498969602953c6ee9 Mon Sep 17 00:00:00 2001
From: Ian Molton <ian@mnementh.co.uk>
Date: Sat, 17 Jan 2009 17:44:23 +0000
Subject: ASoC: fixes to caching implementations

This patch takes fixes a number of bugs in the caching code used by
several ASoC codec drivers. Mostly off-by-one fixes.

Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ac97.c    | 2 --
 sound/soc/codecs/ad1980.c  | 4 ++--
 sound/soc/codecs/twl4030.c | 3 +++
 sound/soc/codecs/wm8580.c  | 4 ++--
 sound/soc/codecs/wm8728.c  | 4 ++--
 sound/soc/codecs/wm8753.c  | 4 ++--
 sound/soc/codecs/wm8990.c  | 4 ++--
 sound/soc/codecs/wm9712.c  | 4 ++--
 sound/soc/codecs/wm9713.c  | 4 ++--
 9 files changed, 17 insertions(+), 16 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fb53e6511af2..89d41277616d 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -123,7 +123,6 @@ bus_err:
 	snd_soc_free_pcms(socdev);
 
 err:
-	kfree(socdev->codec->reg_cache);
 	kfree(socdev->codec);
 	socdev->codec = NULL;
 	return ret;
@@ -138,7 +137,6 @@ static int ac97_soc_remove(struct platform_device *pdev)
 		return 0;
 
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec->reg_cache);
 	kfree(socdev->codec);
 
 	return 0;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index c3c5d0eee37a..faf358758e13 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -109,7 +109,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	default:
 		reg = reg >> 1;
 
-		if (reg >= (ARRAY_SIZE(ad1980_reg)))
+		if (reg >= ARRAY_SIZE(ad1980_reg))
 			return -EINVAL;
 
 		return cache[reg];
@@ -123,7 +123,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg < (ARRAY_SIZE(ad1980_reg)))
+	if (reg < ARRAY_SIZE(ad1980_reg))
 		cache[reg] = val;
 
 	return 0;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index ddc9f37d863f..f530c1e6d9e8 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -125,6 +125,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
 {
 	u8 *cache = codec->reg_cache;
 
+	if (reg >= TWL4030_CACHEREGNUM)
+		return -EIO;
+
 	return cache[reg];
 }
 
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 9b75a377453e..3faf0e70ce10 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -200,7 +200,7 @@ static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+	BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
 	return cache[reg];
 }
 
@@ -223,7 +223,7 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg,
 {
 	u8 data[2];
 
-	BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+	BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
 
 	/* Registers are 9 bits wide */
 	value &= 0x1ff;
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index defa310bc7d9..f90dc52e975d 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -47,7 +47,7 @@ static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
 	return cache[reg];
 }
 
@@ -55,7 +55,7 @@ static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
 	u16 reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
 	cache[reg] = value;
 }
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 7283178e0eb5..5a1c1fca120f 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
+	if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
 		return -1;
 	return cache[reg - 1];
 }
@@ -109,7 +109,7 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	if (reg < 1 || reg > 0x3f)
+	if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
 		return;
 	cache[reg - 1] = value;
 }
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 6b2778632d5e..f93c0955ed9d 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -116,7 +116,7 @@ static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+	BUG_ON(reg >= ARRAY_SIZE(wm8990_reg));
 	return cache[reg];
 }
 
@@ -129,7 +129,7 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
 	u16 *cache = codec->reg_cache;
 
 	/* Reset register and reserved registers are uncached */
-	if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
+	if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg))
 		return;
 
 	cache[reg] = value;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 1b0ace0f4dca..4dc90d67530e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -452,7 +452,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	else {
 		reg = reg >> 1;
 
-		if (reg > (ARRAY_SIZE(wm9712_reg)))
+		if (reg >= (ARRAY_SIZE(wm9712_reg)))
 			return -EIO;
 
 		return cache[reg];
@@ -466,7 +466,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg <= (ARRAY_SIZE(wm9712_reg)))
+	if (reg < (ARRAY_SIZE(wm9712_reg)))
 		cache[reg] = val;
 
 	return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index e636d8a18ed6..0e60e16973d4 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -620,7 +620,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	else {
 		reg = reg >> 1;
 
-		if (reg > (ARRAY_SIZE(wm9713_reg)))
+		if (reg >= (ARRAY_SIZE(wm9713_reg)))
 			return -EIO;
 
 		return cache[reg];
@@ -634,7 +634,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (reg < 0x7c)
 		soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg <= (ARRAY_SIZE(wm9713_reg)))
+	if (reg < (ARRAY_SIZE(wm9713_reg)))
 		cache[reg] = val;
 
 	return 0;
-- 
cgit v1.2.3


From 927b0aea93bb324d743e575659e10d6d76818e4b Mon Sep 17 00:00:00 2001
From: Ian Molton <ian@mnementh.co.uk>
Date: Mon, 19 Jan 2009 17:23:11 +0000
Subject: ASoC: Fix WM9705 capture switch name

This patch fixes the acpture switch name so that it better reflects its
purpose.

Signed-off-by: Ian Molton <iann@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm9705.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index cb26b6a77ffb..5e1937ac0b5e 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -57,8 +57,8 @@ static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = {
 	SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1),
 	SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1),
 	SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0),
-	SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
-	SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+	SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
+	SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
 };
 
 static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
-- 
cgit v1.2.3


From ff637d38ea6b9c54f708a2b9edabc1b0c73c6d0a Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Thu, 22 Jan 2009 18:23:39 -0600
Subject: ASoC: remove stand-alone mode support from CS4270 codec driver

The CS4270 supports stand-alone mode, where the codec is not connect to the
I2C or SPI buses.  Instead, input voltages configure the codec at power-on.
The CS4270 ASoC device driver has partial support for this mode, but the
code was never tested, and partial support doesn't help anyone.  It also made
the rest of the code more complicated than necessary.

[Removed redundant CS4270 dependency on I2C -- broonie]

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 92 ++++++++++++++---------------------------------
 sound/soc/fsl/Kconfig     |  3 +-
 2 files changed, 29 insertions(+), 66 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1aa0c34421c..2e4ce04925e2 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -12,11 +12,7 @@
  *
  * Current features/limitations:
  *
- * 1) Software mode is supported.  Stand-alone mode is automatically
- *    selected if I2C is disabled or if a CS4270 is not found on the I2C
- *    bus.  However, stand-alone mode is only partially implemented because
- *    there is no mechanism yet for this driver and the machine driver to
- *    communicate the values of the M0, M1, MCLK1, and MCLK2 pins.
+ * 1) Software mode is supported.  Stand-alone mode is not supported.
  * 2) Only I2C is supported, not SPI
  * 3) Only Master mode is supported, not Slave.
  * 4) The machine driver's 'startup' function must call
@@ -33,14 +29,6 @@
 #include <sound/initval.h>
 #include <linux/i2c.h>
 
-#include "cs4270.h"
-
-/* If I2C is defined, then we support software mode.  However, if we're
-   not compiled as module but I2C is, then we can't use I2C calls. */
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-#define USE_I2C
-#endif
-
 /* Private data for the CS4270 */
 struct cs4270_private {
 	unsigned int mclk; /* Input frequency of the MCLK pin */
@@ -60,8 +48,6 @@ struct cs4270_private {
 			SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
 			SNDRV_PCM_FMTBIT_S24_LE  | SNDRV_PCM_FMTBIT_S24_BE)
 
-#ifdef USE_I2C
-
 /* CS4270 registers addresses */
 #define CS4270_CHIPID	0x01	/* Chip ID */
 #define CS4270_PWRCTL	0x02	/* Power Control */
@@ -271,17 +257,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return ret;
 }
 
-/*
- * A list of addresses on which this CS4270 could use.  I2C addresses are
- * 7 bits.  For the CS4270, the upper four bits are always 1001, and the
- * lower three bits are determined via the AD2, AD1, and AD0 pins
- * (respectively).
- */
-static const unsigned short normal_i2c[] = {
-	0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END
-};
-I2C_CLIENT_INSMOD;
-
 /*
  * Pre-fill the CS4270 register cache.
  *
@@ -476,7 +451,6 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 }
 
 #ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-
 /*
  * Set the CS4270 external mute
  *
@@ -501,32 +475,16 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
-
+#else
+#define cs4270_mute NULL
 #endif
 
-static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *);
-
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
 		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1)
 };
 
-static const struct i2c_device_id cs4270_id[] = {
-	{"cs4270", 0},
-	{}
-};
-MODULE_DEVICE_TABLE(i2c, cs4270_id);
-
-static struct i2c_driver cs4270_i2c_driver = {
-	.driver = {
-		.name = "CS4270 I2C",
-		.owner = THIS_MODULE,
-	},
-	.id_table = cs4270_id,
-	.probe = cs4270_i2c_probe,
-};
-
 /*
  * Global variable to store socdev for i2c probe function.
  *
@@ -633,7 +591,20 @@ error:
 	return ret;
 }
 
-#endif /* USE_I2C*/
+static const struct i2c_device_id cs4270_id[] = {
+	{"cs4270", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs4270_id);
+
+static struct i2c_driver cs4270_i2c_driver = {
+	.driver = {
+		.name = "cs4270",
+		.owner = THIS_MODULE,
+	},
+	.id_table = cs4270_id,
+	.probe = cs4270_i2c_probe,
+};
 
 struct snd_soc_dai cs4270_dai = {
 	.name = "CS4270",
@@ -698,7 +669,6 @@ static int cs4270_probe(struct platform_device *pdev)
 		goto error_free_codec;
 	}
 
-#ifdef USE_I2C
 	cs4270_socdev = socdev;
 
 	ret = i2c_add_driver(&cs4270_i2c_driver);
@@ -708,20 +678,16 @@ static int cs4270_probe(struct platform_device *pdev)
 	}
 
 	/* Did we find a CS4270 on the I2C bus? */
-	if (codec->control_data) {
-		/* Initialize codec ops */
-		cs4270_dai.ops.hw_params = cs4270_hw_params;
-		cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
-		cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
-#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-		cs4270_dai.ops.digital_mute = cs4270_mute;
-#endif
-	} else
-		printk(KERN_INFO "cs4270: no I2C device found, "
-			"using stand-alone mode\n");
-#else
-	printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
-#endif
+	if (!codec->control_data) {
+		printk(KERN_ERR "cs4270: failed to attach driver");
+		goto error_del_driver;
+	}
+
+	/* Initialize codec ops */
+	cs4270_dai.ops.hw_params = cs4270_hw_params;
+	cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
+	cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
+	cs4270_dai.ops.digital_mute = cs4270_mute;
 
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -732,11 +698,9 @@ static int cs4270_probe(struct platform_device *pdev)
 	return 0;
 
 error_del_driver:
-#ifdef USE_I2C
 	i2c_del_driver(&cs4270_i2c_driver);
 
 error_free_pcms:
-#endif
 	snd_soc_free_pcms(socdev);
 
 error_free_codec:
@@ -752,9 +716,7 @@ static int cs4270_remove(struct platform_device *pdev)
 
 	snd_soc_free_pcms(socdev);
 
-#ifdef USE_I2C
 	i2c_del_driver(&cs4270_i2c_driver);
-#endif
 
 	kfree(socdev->codec);
 	socdev->codec = NULL;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index c7c78c39cfed..9fc908283371 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -10,7 +10,8 @@ config SND_SOC_MPC8610
 
 config SND_SOC_MPC8610_HPCD
 	tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
-	depends on MPC8610_HPCD
+	# I2C is necessary for the CS4270 driver
+	depends on MPC8610_HPCD && I2C
 	select SND_SOC_MPC8610
 	select SND_SOC_CS4270
 	select SND_SOC_CS4270_VD33_ERRATA
-- 
cgit v1.2.3


From ef963dcf6879e500e6559b4327f6cbdc4439198e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 23 Jan 2009 14:53:58 +0000
Subject: ASoC: Fix spurious codec driver dependencies

Kbuild ignores dependency from things that are themselves selected so
ASoC machine drivers need to ensure that the control bus is being built.
This also avoids issues where multiple buses are supported by a given
codec.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig | 4 ----
 sound/soc/omap/Kconfig   | 4 ++--
 2 files changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index cb5fcd605acc..656f180b2c1f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -91,7 +91,6 @@ config SND_SOC_SSM2602
 
 config SND_SOC_TLV320AIC23
 	tristate
-	depends on I2C
 
 config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
@@ -99,15 +98,12 @@ config SND_SOC_TLV320AIC26
 
 config SND_SOC_TLV320AIC3X
 	tristate
-	depends on I2C
 
 config SND_SOC_TWL4030
 	tristate
-	depends on TWL4030_CORE
 
 config SND_SOC_UDA134X
        tristate
-       select SND_SOC_L3
 
 config SND_SOC_UDA1380
         tristate
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index ccd8973683db..675732e724d5 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -8,7 +8,7 @@ config SND_OMAP_SOC_MCBSP
 
 config SND_OMAP_SOC_N810
 	tristate "SoC Audio support for Nokia N810"
-	depends on SND_OMAP_SOC && MACH_NOKIA_N810
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
 	select SND_OMAP_SOC_MCBSP
 	select OMAP_MUX
 	select SND_SOC_TLV320AIC3X
@@ -17,7 +17,7 @@ config SND_OMAP_SOC_N810
 
 config SND_OMAP_SOC_OSK5912
 	tristate "SoC Audio support for omap osk5912"
-	depends on SND_OMAP_SOC && MACH_OMAP_OSK
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC23
 	help
-- 
cgit v1.2.3


From 01e097d6c409a6eb64758dce9fcde0c70073fe36 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 23 Jan 2009 15:07:45 +0000
Subject: ASoC: Include header file in cs4270 and wm9705

Ensures that the DAI and socdev exported by the codec match up with
their exported prototype.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 2 ++
 sound/soc/codecs/wm9705.c | 2 ++
 2 files changed, 4 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2e4ce04925e2..e2130d7b1e41 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -29,6 +29,8 @@
 #include <sound/initval.h>
 #include <linux/i2c.h>
 
+#include "cs4270.h"
+
 /* Private data for the CS4270 */
 struct cs4270_private {
 	unsigned int mclk; /* Input frequency of the MCLK pin */
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 5e1937ac0b5e..d5c81bb3decb 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -20,6 +20,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
+#include "wm9705.h"
+
 /*
  * WM9705 register cache
  */
-- 
cgit v1.2.3


From 070504ade7a95a0f4395673717f3bb7d41793ca8 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 23 Jan 2009 15:34:54 +0000
Subject: ASoC: Fix L3 bus handling in Kconfig

It has no external dependencies but needs to be selected for L3 based
codecs to work.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  | 1 +
 sound/soc/s3c24xx/Kconfig | 1 +
 2 files changed, 2 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 656f180b2c1f..a195303603e0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311 if I2C
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index fcd03acf10f6..e05a71084c32 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -48,4 +48,5 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X
 	tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
        	depends on SND_S3C24XX_SOC
        	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_L3
        	select SND_SOC_UDA134X
-- 
cgit v1.2.3


From 0db4d0705260dd4bddf1e5a5441c58bdf08bdc9f Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Fri, 23 Jan 2009 16:31:19 -0600
Subject: ASoC: improve I2C initialization code in CS4270 driver

Further improvements in the I2C initialization sequence of the CS4270 driver.
All ASoC initialization is now done in the I2C probe function.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 252 +++++++++++++++++++++-------------------------
 1 file changed, 113 insertions(+), 139 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index e2130d7b1e41..2aa12fdbd2ca 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -31,12 +31,6 @@
 
 #include "cs4270.h"
 
-/* Private data for the CS4270 */
-struct cs4270_private {
-	unsigned int mclk; /* Input frequency of the MCLK pin */
-	unsigned int mode; /* The mode (I2S or left-justified) */
-};
-
 /*
  * The codec isn't really big-endian or little-endian, since the I2S
  * interface requires data to be sent serially with the MSbit first.
@@ -109,6 +103,14 @@ struct cs4270_private {
 #define CS4270_MUTE_DAC_A	0x01
 #define CS4270_MUTE_DAC_B	0x02
 
+/* Private data for the CS4270 */
+struct cs4270_private {
+	struct snd_soc_codec codec;
+	u8 reg_cache[CS4270_NUMREGS];
+	unsigned int mclk; /* Input frequency of the MCLK pin */
+	unsigned int mode; /* The mode (I2S or left-justified) */
+};
+
 /*
  * Clock Ratio Selection for Master Mode with I2C enabled
  *
@@ -504,6 +506,31 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
  */
 static struct snd_soc_device *cs4270_socdev;
 
+struct snd_soc_dai cs4270_dai = {
+	.name = "cs4270",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = 0,
+		.formats = CS4270_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = 0,
+		.formats = CS4270_FORMATS,
+	},
+	.ops = {
+		.hw_params = cs4270_hw_params,
+		.set_sysclk = cs4270_set_dai_sysclk,
+		.set_fmt = cs4270_set_dai_fmt,
+		.digital_mute = cs4270_mute,
+	},
+};
+EXPORT_SYMBOL_GPL(cs4270_dai);
+
 /*
  * Initialize the I2C interface of the CS4270
  *
@@ -517,47 +544,52 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = cs4270_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec;
+	struct cs4270_private *cs4270;
 	int i;
 	int ret = 0;
 
-	/* Probing all possible addresses has one drawback: if there are
-	   multiple CS4270s on the bus, then you cannot specify which
-	   socdev is matched with which CS4270.  For now, we just reject
-	   this I2C device if the socdev already has one attached. */
-	if (codec->control_data)
-		return -ENODEV;
-
-	/* Note: codec_dai->codec is NULL here */
-
-	codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL);
-	if (!codec->reg_cache) {
-		printk(KERN_ERR "cs4270: could not allocate register cache\n");
-		ret = -ENOMEM;
-		goto error;
-	}
-
 	/* Verify that we have a CS4270 */
 
 	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
 	if (ret < 0) {
 		printk(KERN_ERR "cs4270: failed to read I2C\n");
-		goto error;
+		return ret;
 	}
 	/* The top four bits of the chip ID should be 1100. */
 	if ((ret & 0xF0) != 0xC0) {
-		/* The device at this address is not a CS4270 codec */
-		ret = -ENODEV;
-		goto error;
+		printk(KERN_ERR "cs4270: device at addr %X is not a CS4270\n",
+		       i2c_client->addr);
+		return -ENODEV;
 	}
 
 	printk(KERN_INFO "cs4270: found device at I2C address %X\n",
 		i2c_client->addr);
 	printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF);
 
+	/* Allocate enough space for the snd_soc_codec structure
+	   and our private data together. */
+	cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL);
+	if (!cs4270) {
+		printk(KERN_ERR "cs4270: Could not allocate codec structure\n");
+		return -ENOMEM;
+	}
+	codec = &cs4270->codec;
+	socdev->codec = codec;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->name = "CS4270";
+	codec->owner = THIS_MODULE;
+	codec->dai = &cs4270_dai;
+	codec->num_dai = 1;
+	codec->private_data = cs4270;
 	codec->control_data = i2c_client;
 	codec->read = cs4270_read_reg_cache;
 	codec->write = cs4270_i2c_write;
+	codec->reg_cache = cs4270->reg_cache;
 	codec->reg_cache_size = CS4270_NUMREGS;
 
 	/* The I2C interface is set up, so pre-fill our register cache */
@@ -565,35 +597,72 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	ret = cs4270_fill_cache(codec);
 	if (ret < 0) {
 		printk(KERN_ERR "cs4270: failed to fill register cache\n");
-		goto error;
+		goto error_free_codec;
+	}
+
+	/* Register PCMs */
+
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "cs4270: failed to create PCMs\n");
+		goto error_free_codec;
 	}
 
 	/* Add the non-DAPM controls */
 
 	for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
-		struct snd_kcontrol *kctrl =
-		snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
+		struct snd_kcontrol *kctrl;
+
+		kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
+		if (!kctrl) {
+			printk(KERN_ERR "cs4270: error creating control '%s'\n",
+			       cs4270_snd_controls[i].name);
+			ret = -ENOMEM;
+			goto error_free_pcms;
+		}
 
 		ret = snd_ctl_add(codec->card, kctrl);
-		if (ret < 0)
-			goto error;
+		if (ret < 0) {
+			printk(KERN_ERR "cs4270: error adding control '%s'\n",
+			       cs4270_snd_controls[i].name);
+			goto error_free_pcms;
+		}
 	}
 
-	i2c_set_clientdata(i2c_client, codec);
+	/* Initialize the SOC device */
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "cs4270: failed to register card\n");
+		goto error_free_pcms;;
+	}
+
+	i2c_set_clientdata(i2c_client, socdev);
 
 	return 0;
 
-error:
-	codec->control_data = NULL;
+error_free_pcms:
+	snd_soc_free_pcms(socdev);
 
-	kfree(codec->reg_cache);
-	codec->reg_cache = NULL;
-	codec->reg_cache_size = 0;
+error_free_codec:
+	kfree(cs4270);
 
 	return ret;
 }
 
-static const struct i2c_device_id cs4270_id[] = {
+static int cs4270_i2c_remove(struct i2c_client *i2c_client)
+{
+	struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct cs4270_private *cs4270 = codec->private_data;
+
+	snd_soc_free_pcms(socdev);
+	kfree(cs4270);
+
+	return 0;
+}
+
+static struct i2c_device_id cs4270_id[] = {
 	{"cs4270", 0},
 	{}
 };
@@ -606,27 +675,9 @@ static struct i2c_driver cs4270_i2c_driver = {
 	},
 	.id_table = cs4270_id,
 	.probe = cs4270_i2c_probe,
+	.remove = cs4270_i2c_remove,
 };
 
-struct snd_soc_dai cs4270_dai = {
-	.name = "CS4270",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = 0,
-		.formats = CS4270_FORMATS,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = 0,
-		.formats = CS4270_FORMATS,
-	},
-};
-EXPORT_SYMBOL_GPL(cs4270_dai);
-
 /*
  * ASoC probe function
  *
@@ -635,94 +686,15 @@ EXPORT_SYMBOL_GPL(cs4270_dai);
  */
 static int cs4270_probe(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec;
-	int ret = 0;
-
-	printk(KERN_INFO "CS4270 ALSA SoC Codec\n");
-
-	/* Allocate enough space for the snd_soc_codec structure
-	   and our private data together. */
-	codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) +
-			sizeof(struct cs4270_private), GFP_KERNEL);
-	if (!codec) {
-		printk(KERN_ERR "cs4270: Could not allocate codec structure\n");
-		return -ENOMEM;
-	}
-
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	codec->name = "CS4270";
-	codec->owner = THIS_MODULE;
-	codec->dai = &cs4270_dai;
-	codec->num_dai = 1;
-	codec->private_data = (void *) codec +
-		ALIGN(sizeof(struct snd_soc_codec), 4);
-
-	socdev->codec = codec;
-
-	/* Register PCMs */
-
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to create PCMs\n");
-		goto error_free_codec;
-	}
-
-	cs4270_socdev = socdev;
-
-	ret = i2c_add_driver(&cs4270_i2c_driver);
-	if (ret) {
-		printk(KERN_ERR "cs4270: failed to attach driver");
-		goto error_free_pcms;
-	}
-
-	/* Did we find a CS4270 on the I2C bus? */
-	if (!codec->control_data) {
-		printk(KERN_ERR "cs4270: failed to attach driver");
-		goto error_del_driver;
-	}
+	cs4270_socdev = platform_get_drvdata(pdev);;
 
-	/* Initialize codec ops */
-	cs4270_dai.ops.hw_params = cs4270_hw_params;
-	cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
-	cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
-	cs4270_dai.ops.digital_mute = cs4270_mute;
-
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to register card\n");
-		goto error_del_driver;
-	}
-
-	return 0;
-
-error_del_driver:
-	i2c_del_driver(&cs4270_i2c_driver);
-
-error_free_pcms:
-	snd_soc_free_pcms(socdev);
-
-error_free_codec:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
-
-	return ret;
+	return i2c_add_driver(&cs4270_i2c_driver);
 }
 
 static int cs4270_remove(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-
-	snd_soc_free_pcms(socdev);
-
 	i2c_del_driver(&cs4270_i2c_driver);
 
-	kfree(socdev->codec);
-	socdev->codec = NULL;
-
 	return 0;
 }
 
@@ -740,6 +712,8 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
 static int __init cs4270_init(void)
 {
+	printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
+
 	return snd_soc_register_dai(&cs4270_dai);
 }
 module_init(cs4270_init);
-- 
cgit v1.2.3


From 6627a653bceb3a54e55e5cdc478ec5b8d5c9cc44 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 23 Jan 2009 22:55:23 +0000
Subject: ASoC: Push the codec runtime storage into the card structure

This is a further stage on the road to refactoring away the ASoC
platform device.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h            |  3 ++-
 sound/soc/codecs/ac97.c        | 20 ++++++++++----------
 sound/soc/codecs/ad1980.c      | 12 ++++++------
 sound/soc/codecs/ad73311.c     |  8 ++++----
 sound/soc/codecs/ak4535.c      | 14 +++++++-------
 sound/soc/codecs/cs4270.c      |  6 +++---
 sound/soc/codecs/pcm3008.c     | 12 ++++++------
 sound/soc/codecs/ssm2602.c     | 20 ++++++++++----------
 sound/soc/codecs/tlv320aic23.c | 18 +++++++++---------
 sound/soc/codecs/tlv320aic26.c |  4 ++--
 sound/soc/codecs/tlv320aic3x.c | 14 +++++++-------
 sound/soc/codecs/twl4030.c     | 12 ++++++------
 sound/soc/codecs/uda134x.c     | 18 +++++++++---------
 sound/soc/codecs/uda1380.c     | 18 +++++++++---------
 sound/soc/codecs/wm8350.c      | 10 +++++-----
 sound/soc/codecs/wm8510.c      | 16 ++++++++--------
 sound/soc/codecs/wm8580.c      | 10 +++++-----
 sound/soc/codecs/wm8728.c      | 16 ++++++++--------
 sound/soc/codecs/wm8731.c      | 20 ++++++++++----------
 sound/soc/codecs/wm8750.c      | 16 ++++++++--------
 sound/soc/codecs/wm8753.c      | 18 +++++++++---------
 sound/soc/codecs/wm8900.c      |  8 ++++----
 sound/soc/codecs/wm8903.c      | 18 +++++++++---------
 sound/soc/codecs/wm8971.c      | 14 +++++++-------
 sound/soc/codecs/wm8990.c      | 14 +++++++-------
 sound/soc/codecs/wm9705.c      | 15 ++++++++-------
 sound/soc/codecs/wm9712.c      | 21 +++++++++++----------
 sound/soc/codecs/wm9713.c      | 17 +++++++++--------
 sound/soc/soc-core.c           | 37 +++++++++++++++++--------------------
 sound/soc/soc-dapm.c           |  6 +++---
 sound/soc/soc-jack.c           |  4 ++--
 31 files changed, 220 insertions(+), 219 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 7039343e8a78..0e7735264169 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -418,6 +418,8 @@ struct snd_soc_card {
 
 	struct snd_soc_device *socdev;
 
+	struct snd_soc_codec *codec;
+
 	struct snd_soc_platform *platform;
 	struct delayed_work delayed_work;
 	struct work_struct deferred_resume_work;
@@ -427,7 +429,6 @@ struct snd_soc_card {
 struct snd_soc_device {
 	struct device *dev;
 	struct snd_soc_card *card;
-	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
 	void *codec_data;
 };
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 89d41277616d..11f84b6e5cb8 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
@@ -84,10 +84,10 @@ static int ac97_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (!socdev->codec)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->card->codec)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->name = "AC97";
@@ -123,21 +123,21 @@ bus_err:
 	snd_soc_free_pcms(socdev);
 
 err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ac97_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (!codec)
 		return 0;
 
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return 0;
 }
@@ -147,7 +147,7 @@ static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	snd_ac97_suspend(socdev->codec->ac97);
+	snd_ac97_suspend(socdev->card->codec->ac97);
 
 	return 0;
 }
@@ -156,7 +156,7 @@ static int ac97_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	snd_ac97_resume(socdev->codec->ac97);
+	snd_ac97_resume(socdev->card->codec->ac97);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index faf358758e13..ddb3b08ac23c 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -186,10 +186,10 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache =
@@ -275,15 +275,15 @@ codec_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ad1980_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index b09289a1e55a..e61dac5e7b8f 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -53,7 +53,7 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 	codec->owner = THIS_MODULE;
 	codec->dai = &ad73311_dai;
 	codec->num_dai = 1;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -75,15 +75,15 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 register_err:
 	snd_soc_free_pcms(socdev);
 pcm_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ad73311_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index f17c363cb1db..d56e6bb1fedb 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -329,7 +329,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ak4535_priv *ak4535 = codec->private_data;
 	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
 	int rate = params_rate(params), fs = 256;
@@ -447,7 +447,7 @@ EXPORT_SYMBOL_GPL(ak4535_dai);
 static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -456,7 +456,7 @@ static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
 static int ak4535_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	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);
@@ -469,7 +469,7 @@ static int ak4535_resume(struct platform_device *pdev)
  */
 static int ak4535_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "AK4535";
@@ -523,7 +523,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = ak4535_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -622,7 +622,7 @@ static int ak4535_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = ak4535;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -649,7 +649,7 @@ static int ak4535_probe(struct platform_device *pdev)
 static int ak4535_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2aa12fdbd2ca..21253b48289f 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -350,7 +350,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct cs4270_private *cs4270 = codec->private_data;
 	int ret;
 	unsigned int i;
@@ -575,7 +575,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 		return -ENOMEM;
 	}
 	codec = &cs4270->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -653,7 +653,7 @@ error_free_codec:
 static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 {
 	struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct cs4270_private *cs4270 = codec->private_data;
 
 	snd_soc_free_pcms(socdev);
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 9a3e67e5319c..5cda9e6b5a74 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -67,11 +67,11 @@ static int pcm3008_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (!socdev->codec)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->card->codec)
 		return -ENOMEM;
 
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->name = "PCM3008";
@@ -139,7 +139,7 @@ gpio_err:
 card_err:
 	snd_soc_free_pcms(socdev);
 pcm_err:
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return ret;
 }
@@ -147,7 +147,7 @@ pcm_err:
 static int pcm3008_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct pcm3008_setup_data *setup = socdev->codec_data;
 
 	if (!codec)
@@ -155,7 +155,7 @@ static int pcm3008_soc_remove(struct platform_device *pdev)
 
 	pcm3008_gpio_free(setup);
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index ec7fe3b7b0cb..58e225dadc7e 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -276,7 +276,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	u16 srate;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
@@ -321,7 +321,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
@@ -358,7 +358,7 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	/* set active */
 	ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
 
@@ -370,7 +370,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	/* deactivate */
 	if (!codec->active)
@@ -535,7 +535,7 @@ EXPORT_SYMBOL_GPL(ssm2602_dai);
 static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -544,7 +544,7 @@ static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
 static int ssm2602_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -566,7 +566,7 @@ static int ssm2602_resume(struct platform_device *pdev)
  */
 static int ssm2602_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "SSM2602";
@@ -639,7 +639,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = ssm2602_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -733,7 +733,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = ssm2602;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -754,7 +754,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 static int ssm2602_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a0e47c1dcd64..8b20c360adf5 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -405,7 +405,7 @@ static int tlv320aic23_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 iface_reg;
 	int ret;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
@@ -453,7 +453,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* set active */
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
@@ -466,7 +466,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	/* deactivate */
@@ -609,7 +609,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 			       pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -620,7 +620,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 static int tlv320aic23_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u16 reg;
 
@@ -642,7 +642,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
  */
 static int tlv320aic23_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 	u16 reg;
 
@@ -729,7 +729,7 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c,
 				   const struct i2c_device_id *i2c_id)
 {
 	struct snd_soc_device *socdev = tlv320aic23_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
@@ -787,7 +787,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 	if (aic23 == NULL)
 		return -ENOMEM;
 	codec = &aic23->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -806,7 +806,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 static int tlv320aic23_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	if (codec->control_data)
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 29f2f1a017fd..229e464cf713 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -130,7 +130,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic26 *aic26 = codec->private_data;
 	int fsref, divisor, wlen, pval, jval, dval, qval;
 	u16 reg;
@@ -338,7 +338,7 @@ static int aic26_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 	codec = &aic26->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
 		&pdev->dev, socdev->dev);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 36ab0198ca3f..ba64b0c617e6 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -727,7 +727,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
@@ -1079,7 +1079,7 @@ EXPORT_SYMBOL_GPL(aic3x_dai);
 static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1089,7 +1089,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
 static int aic3x_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u8 *cache = codec->reg_cache;
@@ -1112,7 +1112,7 @@ static int aic3x_resume(struct platform_device *pdev)
  */
 static int aic3x_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
@@ -1243,7 +1243,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 			   const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = aic3x_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1348,7 +1348,7 @@ static int aic3x_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = aic3x;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1374,7 +1374,7 @@ static int aic3x_probe(struct platform_device *pdev)
 static int aic3x_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* power down chip */
 	if (codec->control_data)
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index f530c1e6d9e8..796f34cac85d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -981,7 +981,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u8 mode, old_mode, format, old_format;
 
 
@@ -1166,7 +1166,7 @@ EXPORT_SYMBOL_GPL(twl4030_dai);
 static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1176,7 +1176,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 static int twl4030_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	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);
@@ -1190,7 +1190,7 @@ static int twl4030_resume(struct platform_device *pdev)
 
 static int twl4030_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	printk(KERN_INFO "TWL4030 Audio Codec init \n");
@@ -1251,7 +1251,7 @@ static int twl4030_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1265,7 +1265,7 @@ static int twl4030_probe(struct platform_device *pdev)
 static int twl4030_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
 	snd_soc_free_pcms(socdev);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 277825d155a6..661599295ca7 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -173,7 +173,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 	struct snd_pcm_runtime *master_runtime;
 
@@ -206,7 +206,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 
 	if (uda134x->master_substream == substream)
@@ -221,7 +221,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 	u8 hw_params;
 
@@ -492,11 +492,11 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return ret;
 
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 
 	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
 	if (uda134x == NULL)
@@ -584,7 +584,7 @@ priv_err:
 static int uda134x_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -604,7 +604,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev,
 						pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -614,7 +614,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev,
 static int uda134x_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a957b4365b9d..98e4a6560f06 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -397,7 +397,7 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, reg_start, reg_end, clk;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -430,7 +430,7 @@ static int uda1380_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* set WSPLL power and divider if running from this clock */
@@ -469,7 +469,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* shut down WSPLL power if running from this clock */
@@ -591,7 +591,7 @@ EXPORT_SYMBOL_GPL(uda1380_dai);
 static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -600,7 +600,7 @@ static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
 static int uda1380_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -622,7 +622,7 @@ static int uda1380_resume(struct platform_device *pdev)
  */
 static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "UDA1380";
@@ -688,7 +688,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c,
 {
 	struct snd_soc_device *socdev = uda1380_socdev;
 	struct uda1380_setup_data *setup = socdev->codec_data;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -779,7 +779,7 @@ static int uda1380_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -803,7 +803,7 @@ static int uda1380_probe(struct platform_device *pdev)
 static int uda1380_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 2e0db29b4998..75d3438ccb87 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1301,7 +1301,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -1310,7 +1310,7 @@ static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8350_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1423,8 +1423,8 @@ static int wm8350_probe(struct platform_device *pdev)
 
 	BUG_ON(!wm8350_codec);
 
-	socdev->codec = wm8350_codec;
-	codec = socdev->codec;
+	socdev->card->codec = wm8350_codec;
+	codec = socdev->card->codec;
 	wm8350 = codec->control_data;
 	priv = codec->private_data;
 
@@ -1498,7 +1498,7 @@ card_err:
 static int wm8350_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8350 *wm8350 = codec->control_data;
 	struct wm8350_data *priv = codec->private_data;
 	int ret;
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index abe7cce87714..f01078cfbd72 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -452,7 +452,7 @@ static int wm8510_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
 	u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
 
@@ -581,7 +581,7 @@ EXPORT_SYMBOL_GPL(wm8510_dai);
 static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -590,7 +590,7 @@ static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8510_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -612,7 +612,7 @@ static int wm8510_resume(struct platform_device *pdev)
  */
 static int wm8510_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8510";
@@ -670,7 +670,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -751,7 +751,7 @@ err_driver:
 static int __devinit wm8510_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -817,7 +817,7 @@ static int wm8510_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -847,7 +847,7 @@ static int wm8510_probe(struct platform_device *pdev)
 static int wm8510_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 3faf0e70ce10..d3c51ba5e6f9 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -539,7 +539,7 @@ static int wm8580_paif_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
 
 	paifb &= ~WM8580_AIF_LENGTH_MASK;
@@ -816,7 +816,7 @@ EXPORT_SYMBOL_GPL(wm8580_dai);
  */
 static int wm8580_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8580";
@@ -888,7 +888,7 @@ static int wm8580_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8580_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -986,7 +986,7 @@ static int wm8580_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8580;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1007,7 +1007,7 @@ static int wm8580_probe(struct platform_device *pdev)
 static int wm8580_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index f90dc52e975d..f8363b308895 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -137,7 +137,7 @@ static int wm8728_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
 
 	dac &= ~0x18;
@@ -264,7 +264,7 @@ EXPORT_SYMBOL_GPL(wm8728_dai);
 static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -274,7 +274,7 @@ static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8728_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8728_set_bias_level(codec, codec->suspend_bias_level);
 
@@ -287,7 +287,7 @@ static int wm8728_resume(struct platform_device *pdev)
  */
 static int wm8728_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8728";
@@ -349,7 +349,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8728_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -430,7 +430,7 @@ err_driver:
 static int __devinit wm8728_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8728_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -494,7 +494,7 @@ static int wm8728_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -527,7 +527,7 @@ static int wm8728_probe(struct platform_device *pdev)
 static int wm8728_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 96d6e1aeaf43..0150fe53a65d 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -253,7 +253,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8731_priv *wm8731 = codec->private_data;
 	u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
 	int i = get_coeff(wm8731->sysclk, params_rate(params));
@@ -283,7 +283,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* set active */
 	wm8731_write(codec, WM8731_ACTIVE, 0x0001);
@@ -296,7 +296,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* deactivate */
 	if (!codec->active) {
@@ -458,7 +458,7 @@ EXPORT_SYMBOL_GPL(wm8731_dai);
 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8731_write(codec, WM8731_ACTIVE, 0x0);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -468,7 +468,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8731_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -490,7 +490,7 @@ static int wm8731_resume(struct platform_device *pdev)
  */
 static int wm8731_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8731";
@@ -561,7 +561,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -642,7 +642,7 @@ err_driver:
 static int __devinit wm8731_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -716,7 +716,7 @@ static int wm8731_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8731;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -750,7 +750,7 @@ static int wm8731_probe(struct platform_device *pdev)
 static int wm8731_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 1578569793a2..96afb86addc6 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -604,7 +604,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8750_priv *wm8750 = codec->private_data;
 	u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
 	u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
@@ -712,7 +712,7 @@ static void wm8750_work(struct work_struct *work)
 static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -721,7 +721,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8750_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -754,7 +754,7 @@ static int wm8750_resume(struct platform_device *pdev)
  */
 static int wm8750_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8750";
@@ -836,7 +836,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -917,7 +917,7 @@ err_driver:
 static int __devinit wm8750_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -989,7 +989,7 @@ static int wm8750_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8750;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1043,7 +1043,7 @@ static int run_delayed_work(struct delayed_work *dwork)
 static int wm8750_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 5a1c1fca120f..502766dce861 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -912,7 +912,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8753_priv *wm8753 = codec->private_data;
 	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
@@ -1146,7 +1146,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8753_priv *wm8753 = codec->private_data;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
 	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
@@ -1483,7 +1483,7 @@ static void wm8753_work(struct work_struct *work)
 static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* we only need to suspend if we are a valid card */
 	if (!codec->card)
@@ -1496,7 +1496,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8753_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -1533,7 +1533,7 @@ static int wm8753_resume(struct platform_device *pdev)
  */
 static int wm8753_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8753";
@@ -1624,7 +1624,7 @@ static int wm8753_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1705,7 +1705,7 @@ err_driver:
 static int __devinit wm8753_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -1780,7 +1780,7 @@ static int wm8753_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8753;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1832,7 +1832,7 @@ static int run_delayed_work(struct delayed_work *dwork)
 static int wm8753_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 1e08d4f065f2..85c0f1bc6766 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -720,7 +720,7 @@ static int wm8900_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 
 	reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60;
@@ -1210,7 +1210,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8900_priv *wm8900 = codec->private_data;
 	int fll_out = wm8900->fll_out;
 	int fll_in  = wm8900->fll_in;
@@ -1234,7 +1234,7 @@ static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8900_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8900_priv *wm8900 = codec->private_data;
 	u16 *cache;
 	int i, ret;
@@ -1414,7 +1414,7 @@ static int wm8900_probe(struct platform_device *pdev)
 	}
 
 	codec = wm8900_codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	/* Register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 6ff34b957dce..d36b2b1edf19 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1261,7 +1261,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
@@ -1303,7 +1303,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1323,7 +1323,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	int fs = params_rate(params);
@@ -1527,7 +1527,7 @@ EXPORT_SYMBOL_GPL(wm8903_dai);
 static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1537,7 +1537,7 @@ static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8903_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
@@ -1713,7 +1713,7 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	socdev->codec = wm8903_codec;
+	socdev->card->codec = wm8903_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -1722,9 +1722,9 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	snd_soc_add_controls(socdev->codec, wm8903_snd_controls,
+	snd_soc_add_controls(socdev->card->codec, wm8903_snd_controls,
 				ARRAY_SIZE(wm8903_snd_controls));
-	wm8903_add_widgets(socdev->codec);
+	wm8903_add_widgets(socdev->card->codec);
 
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1745,7 +1745,7 @@ err:
 static int wm8903_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index c8bd9b06f330..24d4c905a011 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -531,7 +531,7 @@ 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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8971_priv *wm8971 = codec->private_data;
 	u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
 	u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
@@ -637,7 +637,7 @@ static void wm8971_work(struct work_struct *work)
 static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -646,7 +646,7 @@ static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8971_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -677,7 +677,7 @@ static int wm8971_resume(struct platform_device *pdev)
 
 static int wm8971_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8971";
@@ -758,7 +758,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8971_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -859,7 +859,7 @@ static int wm8971_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8971;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -894,7 +894,7 @@ static int wm8971_probe(struct platform_device *pdev)
 static int wm8971_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index f93c0955ed9d..6af1d399b316 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1162,7 +1162,7 @@ static int wm8990_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->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
 
 	audio1 &= ~WM8990_AIF_WL_MASK;
@@ -1361,7 +1361,7 @@ EXPORT_SYMBOL_GPL(wm8990_dai);
 static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* we only need to suspend if we are a valid card */
 	if (!codec->card)
@@ -1374,7 +1374,7 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8990_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -1402,7 +1402,7 @@ static int wm8990_resume(struct platform_device *pdev)
  */
 static int wm8990_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 	int ret = 0;
 
@@ -1480,7 +1480,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8990_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1579,7 +1579,7 @@ static int wm8990_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8990;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1605,7 +1605,7 @@ static int wm8990_probe(struct platform_device *pdev)
 static int wm8990_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index d5c81bb3decb..2e9e06b2daaf 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -249,7 +249,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg;
 	u16 vra;
 
@@ -323,10 +323,11 @@ static int wm9705_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9705 SoC Audio Codec\n");
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL);
@@ -380,15 +381,15 @@ pcm_err:
 codec_err:
 	kfree(codec->reg_cache);
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9705_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 4dc90d67530e..b3a8be77676e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -478,7 +478,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg;
 	u16 vra;
 
@@ -499,7 +499,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 vra, xsle;
 
 	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
@@ -592,7 +592,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 	pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -601,7 +601,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 static int wm9712_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i, ret;
 	u16 *cache = codec->reg_cache;
 
@@ -637,10 +637,11 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL);
@@ -704,15 +705,15 @@ codec_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9712_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 0e60e16973d4..54db9c524988 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1115,7 +1115,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev,
 	pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 
 	/* Disable everything except touchpanel - that will be handled
@@ -1133,7 +1133,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev,
 static int wm9713_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm9713_priv *wm9713 = codec->private_data;
 	int i, ret;
 	u16 *cache = codec->reg_cache;
@@ -1174,10 +1174,11 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL);
@@ -1249,15 +1250,15 @@ priv_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9713_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 8313d52a6e8c..f18c7a3e36d1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -234,7 +234,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		cpu_dai->capture.active = codec_dai->capture.active = 1;
 	cpu_dai->active = codec_dai->active = 1;
 	cpu_dai->runtime = runtime;
-	socdev->codec->active++;
+	card->codec->active++;
 	mutex_unlock(&pcm_mutex);
 	return 0;
 
@@ -264,7 +264,7 @@ static void close_delayed_work(struct work_struct *work)
 	struct snd_soc_card *card = container_of(work, struct snd_soc_card,
 						 delayed_work.work);
 	struct snd_soc_device *socdev = card->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_dai *codec_dai;
 	int i;
 
@@ -319,7 +319,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 
 	mutex_lock(&pcm_mutex);
 
@@ -387,7 +387,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0;
 
 	mutex_lock(&pcm_mutex);
@@ -553,7 +553,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 
 	mutex_lock(&pcm_mutex);
 
@@ -629,7 +629,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	int i;
 
 	/* Due to the resume being scheduled into a workqueue we could
@@ -705,7 +705,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	struct snd_soc_device *socdev = card->socdev;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	struct platform_device *pdev = to_platform_device(socdev->dev);
 	int i;
 
@@ -982,8 +982,8 @@ static struct platform_driver soc_driver = {
 static int soc_new_pcm(struct snd_soc_device *socdev,
 	struct snd_soc_dai_link *dai_link, int num)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
 	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
@@ -998,7 +998,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 
 	rtd->dai = dai_link;
 	rtd->socdev = socdev;
-	codec_dai->codec = socdev->codec;
+	codec_dai->codec = card->codec;
 
 	/* check client and interface hw capabilities */
 	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
@@ -1048,9 +1048,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 }
 
 /* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 {
-	struct snd_soc_codec *codec = devdata->codec;
 	int i, step = 1, count = 0;
 
 	if (!codec->reg_cache_size)
@@ -1090,7 +1089,7 @@ static ssize_t codec_reg_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	return soc_codec_reg_show(devdata, buf);
+	return soc_codec_reg_show(devdata->card->codec, buf);
 }
 
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
@@ -1107,12 +1106,10 @@ static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
 {
 	ssize_t ret;
 	struct snd_soc_codec *codec = file->private_data;
-	struct device *card_dev = codec->card->dev;
-	struct snd_soc_device *devdata = card_dev->driver_data;
 	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
-	ret = soc_codec_reg_show(devdata, buf);
+	ret = soc_codec_reg_show(codec, buf);
 	if (ret >= 0)
 		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 	kfree(buf);
@@ -1309,8 +1306,8 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
  */
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0, i;
 
 	mutex_lock(&codec->mutex);
@@ -1355,8 +1352,8 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
  */
 int snd_soc_init_card(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0, i, ac97 = 0, err = 0;
 
 	for (i = 0; i < card->num_links; i++) {
@@ -1404,7 +1401,7 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 	if (err < 0)
 		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
-	soc_init_codec_debugfs(socdev->codec);
+	soc_init_codec_debugfs(codec);
 	mutex_unlock(&codec->mutex);
 
 out:
@@ -1421,14 +1418,14 @@ EXPORT_SYMBOL_GPL(snd_soc_init_card);
  */
 void snd_soc_free_pcms(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	struct snd_soc_dai *codec_dai;
 	int i;
 #endif
 
 	mutex_lock(&codec->mutex);
-	soc_cleanup_codec_debugfs(socdev->codec);
+	soc_cleanup_codec_debugfs(codec);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 54b4564b82b4..f4a8753c84c0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -817,7 +817,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	struct snd_soc_codec *codec = devdata->codec;
+	struct snd_soc_codec *codec = devdata->card->codec;
 	struct snd_soc_dapm_widget *w;
 	int count = 0;
 	char *state = "not set";
@@ -1552,8 +1552,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 				enum snd_soc_bias_level level)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	if (card->set_bias_level)
@@ -1645,7 +1645,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
  */
 void snd_soc_dapm_free(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	snd_soc_dapm_sys_remove(socdev->dev);
 	dapm_free_widgets(codec);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 8cc00c3cdf34..ab64a30bedde 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -34,7 +34,7 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
 	jack->card = card;
 	INIT_LIST_HEAD(&jack->pins);
 
-	return snd_jack_new(card->socdev->codec->card, id, type, &jack->jack);
+	return snd_jack_new(card->codec->card, id, type, &jack->jack);
 }
 EXPORT_SYMBOL_GPL(snd_soc_jack_new);
 
@@ -54,7 +54,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new);
  */
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 {
-	struct snd_soc_codec *codec = jack->card->socdev->codec;
+	struct snd_soc_codec *codec = jack->card->codec;
 	struct snd_soc_jack_pin *pin;
 	int enable;
 	int oldstatus;
-- 
cgit v1.2.3


From 3fc93030e5a792fdd0da3321487f5cbfd1143c2b Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Tue, 27 Jan 2009 11:29:39 +0200
Subject: ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset
 cancelation

The offset cancelation bit in ANAMICL register is self cleanig.
Make sure that the reg_cache holds the same value as the HW
register.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 796f34cac85d..24419afd319e 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -913,6 +913,9 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 		 ((byte & TWL4030_CNCL_OFFSET_START) ==
 		  TWL4030_CNCL_OFFSET_START));
 
+	/* Make sure that the reg_cache has the same value as the HW */
+	twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
+
 	/* anti-pop when changing analog gain */
 	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
 	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
-- 
cgit v1.2.3


From db04e2c58a65364218b89f1372b4b3b78d206423 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Tue, 27 Jan 2009 11:29:40 +0200
Subject: ASoC: TWL4030: Code clean up for codec power up and down

Merge the codec up and down functions to a simple one.
Codec is powered down by default (reg_cache change).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 43 +++++++++++++++++--------------------------
 1 file changed, 17 insertions(+), 26 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 24419afd319e..af7b433d4f53 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -42,7 +42,7 @@
  */
 static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* this register not used		*/
-	0x93, /* REG_CODEC_MODE		(0x1)	*/
+	0x91, /* REG_CODEC_MODE		(0x1)	*/
 	0xc3, /* REG_OPTION		(0x2)	*/
 	0x00, /* REG_UNKNOWN		(0x3)	*/
 	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
@@ -154,26 +154,17 @@ static int twl4030_write(struct snd_soc_codec *codec,
 	return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 }
 
-static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 {
 	u8 mode;
 
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
-		mode & ~TWL4030_CODECPDZ);
+	if (enable)
+		mode |= TWL4030_CODECPDZ;
+	else
+		mode &= ~TWL4030_CODECPDZ;
 
-	/* REVISIT: this delay is present in TI sample drivers */
-	/* but there seems to be no TRM requirement for it     */
-	udelay(10);
-}
-
-static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
-{
-	u8 mode;
-
-	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
-		mode | TWL4030_CODECPDZ);
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
 
 	/* REVISIT: this delay is present in TI sample drivers */
 	/* but there seems to be no TRM requirement for it     */
@@ -185,7 +176,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	int i;
 
 	/* clear CODECPDZ prior to setting register defaults */
-	twl4030_clear_codecpdz(codec);
+	twl4030_codec_enable(codec, 0);
 
 	/* set all audio section registers to reasonable defaults */
 	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
@@ -895,7 +886,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	int i = 0;
 
 	/* set CODECPDZ to turn on codec */
-	twl4030_set_codecpdz(codec);
+	twl4030_codec_enable(codec, 1);
 
 	/* initiate offset cancellation */
 	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
@@ -922,8 +913,8 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
 
 	/* toggle CODECPDZ as per TRM */
-	twl4030_clear_codecpdz(codec);
-	twl4030_set_codecpdz(codec);
+	twl4030_codec_enable(codec, 0);
+	twl4030_codec_enable(codec, 1);
 
 	/* program anti-pop with bias ramp delay */
 	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
@@ -952,7 +943,7 @@ static void twl4030_power_down(struct snd_soc_codec *codec)
 	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
 
 	/* power down */
-	twl4030_clear_codecpdz(codec);
+	twl4030_codec_enable(codec, 0);
 }
 
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
@@ -1030,7 +1021,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	if (mode != old_mode) {
 		/* change rate and set CODECPDZ */
 		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 
 	/* sample size */
@@ -1053,13 +1044,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	if (format != old_format) {
 
 		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_clear_codecpdz(codec);
+		twl4030_codec_enable(codec, 0);
 
 		/* change format */
 		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
 
 		/* set CODECPDZ afterwards */
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 	return 0;
 }
@@ -1129,13 +1120,13 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if (format != old_format) {
 
 		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_clear_codecpdz(codec);
+		twl4030_codec_enable(codec, 0);
 
 		/* change format */
 		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
 
 		/* set CODECPDZ afterwards */
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 
 	return 0;
-- 
cgit v1.2.3


From aad749e51a66d473f5cef4a050e3e36795261be3 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Tue, 27 Jan 2009 11:29:41 +0200
Subject: ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the
 Headset Left is in use

The Headset Left anti-pop and bias ramp does not need to be
enabled, if the headset is not in use.
Move the code to DAPM event handler for HeadsetL.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 71 +++++++++++++++++++++++++++++-----------------
 1 file changed, 45 insertions(+), 26 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index af7b433d4f53..900486ef633d 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -414,6 +414,47 @@ static int handsfree_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int headsetl_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	unsigned char hs_gain, hs_pop;
+
+	/* Save the current volume */
+	hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* Do the anti-pop/bias ramp enable according to the TRM */
+		hs_pop = TWL4030_RAMP_DELAY_645MS;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		hs_pop |= TWL4030_VMID_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Is this needed? Can we just use whatever gain here? */
+		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
+				(hs_gain & (~0x0f)) | 0x0a);
+		hs_pop |= TWL4030_RAMP_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+
+		/* Restore the original volume */
+		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Do the anti-pop/bias ramp disable according to the TRM */
+		hs_pop = twl4030_read_reg_cache(w->codec,
+						TWL4030_REG_HS_POPN_SET);
+		hs_pop &= ~TWL4030_RAMP_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Bypass the reg_cache to mute the headset */
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					hs_gain & (~0x0f),
+					TWL4030_REG_HS_GAIN_SET);
+		hs_pop &= ~TWL4030_VMID_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		break;
+	}
+	return 0;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -720,8 +761,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_predriver_control),
 	/* HeadsetL/R */
-	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_hsol_control),
+	SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsol_control, headsetl_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_hsor_control),
 	/* CarkitL/R */
@@ -882,7 +924,7 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 
 static void twl4030_power_up(struct snd_soc_codec *codec)
 {
-	u8 anamicl, regmisc1, byte, popn;
+	u8 anamicl, regmisc1, byte;
 	int i = 0;
 
 	/* set CODECPDZ to turn on codec */
@@ -915,33 +957,10 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	/* toggle CODECPDZ as per TRM */
 	twl4030_codec_enable(codec, 0);
 	twl4030_codec_enable(codec, 1);
-
-	/* program anti-pop with bias ramp delay */
-	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-	popn &= TWL4030_RAMP_DELAY;
-	popn |=	TWL4030_RAMP_DELAY_645MS;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-	popn |=	TWL4030_VMID_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* enable anti-pop ramp */
-	popn |= TWL4030_RAMP_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
 }
 
 static void twl4030_power_down(struct snd_soc_codec *codec)
 {
-	u8 popn;
-
-	/* disable anti-pop ramp */
-	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-	popn &= ~TWL4030_RAMP_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* disable bias out */
-	popn &= ~TWL4030_VMID_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
 	/* power down */
 	twl4030_codec_enable(codec, 0);
 }
-- 
cgit v1.2.3


From fb2a2f84908460c18c3894963990518b224dd9ff Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Tue, 27 Jan 2009 11:29:42 +0200
Subject: ASoC: TWL4030: Physical ADC and amplifier power switch change

Change the power switches for the physical ADC and for the
amplifiers for the analog capture path to map more closely
the actual path inside of the codec.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 900486ef633d..8fe059e3e9ab 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -802,16 +802,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
 		SND_SOC_DAPM_POST_REG),
 
-	/* Analog input muxes with power switch for the physical ADCL/R */
+	/* Analog input muxes with switch for the capture amplifiers */
 	SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
-		TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control),
+		TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
 	SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
-		TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control),
+		TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
 
-	SND_SOC_DAPM_PGA("Analog Left Amplifier",
-		TWL4030_REG_ANAMICL, 4, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("Analog Right Amplifier",
-		TWL4030_REG_ANAMICR, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC Physical Left",
+		TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC Physical Right",
+		TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0),
 
 	SND_SOC_DAPM_PGA("Digimic0 Enable",
 		TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
@@ -885,23 +885,23 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Analog Right Capture Route", "Sub mic", "SUBMIC"},
 	{"Analog Right Capture Route", "AUXR", "AUXR"},
 
-	{"Analog Left Amplifier", NULL, "Analog Left Capture Route"},
-	{"Analog Right Amplifier", NULL, "Analog Right Capture Route"},
+	{"ADC Physical Left", NULL, "Analog Left Capture Route"},
+	{"ADC Physical Right", NULL, "Analog Right Capture Route"},
 
 	{"Digimic0 Enable", NULL, "DIGIMIC0"},
 	{"Digimic1 Enable", NULL, "DIGIMIC1"},
 
 	/* TX1 Left capture path */
-	{"TX1 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX1 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
 	/* TX1 Right capture path */
-	{"TX1 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX1 Capture Route", "Analog", "ADC Physical Right"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
 	/* TX2 Left capture path */
-	{"TX2 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX2 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
 	/* TX2 Right capture path */
-	{"TX2 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX2 Capture Route", "Analog", "ADC Physical Right"},
 	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
 
 	{"ADC Virtual Left1", NULL, "TX1 Capture Route"},
-- 
cgit v1.2.3


From 006f367e38fb45e2f161c0f500c74449ae63e866 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Tue, 27 Jan 2009 11:29:43 +0200
Subject: ASoC: TWL4030: Move the twl4030_power_up and _power_down function

Move the twl4030_power_up and twl4030_power_down function
earlier to facilitate the analog bypass implementation.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 85 +++++++++++++++++++++++-----------------------
 1 file changed, 42 insertions(+), 43 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 8fe059e3e9ab..f985bef40a39 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -184,6 +184,48 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 }
 
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+	u8 anamicl, regmisc1, byte;
+	int i = 0;
+
+	/* set CODECPDZ to turn on codec */
+	twl4030_codec_enable(codec, 1);
+
+	/* initiate offset cancellation */
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	twl4030_write(codec, TWL4030_REG_ANAMICL,
+		anamicl | TWL4030_CNCL_OFFSET_START);
+
+	/* wait for offset cancellation to complete */
+	do {
+		/* this takes a little while, so don't slam i2c */
+		udelay(2000);
+		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+				    TWL4030_REG_ANAMICL);
+	} while ((i++ < 100) &&
+		 ((byte & TWL4030_CNCL_OFFSET_START) ==
+		  TWL4030_CNCL_OFFSET_START));
+
+	/* Make sure that the reg_cache has the same value as the HW */
+	twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
+
+	/* anti-pop when changing analog gain */
+	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
+
+	/* toggle CODECPDZ as per TRM */
+	twl4030_codec_enable(codec, 0);
+	twl4030_codec_enable(codec, 1);
+}
+
+static void twl4030_power_down(struct snd_soc_codec *codec)
+{
+	/* power down */
+	twl4030_codec_enable(codec, 0);
+}
+
 /* Earpiece */
 static const char *twl4030_earpiece_texts[] =
 		{"Off", "DACL1", "DACL2", "DACR1"};
@@ -922,49 +964,6 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static void twl4030_power_up(struct snd_soc_codec *codec)
-{
-	u8 anamicl, regmisc1, byte;
-	int i = 0;
-
-	/* set CODECPDZ to turn on codec */
-	twl4030_codec_enable(codec, 1);
-
-	/* initiate offset cancellation */
-	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
-	twl4030_write(codec, TWL4030_REG_ANAMICL,
-		anamicl | TWL4030_CNCL_OFFSET_START);
-
-
-	/* wait for offset cancellation to complete */
-	do {
-		/* this takes a little while, so don't slam i2c */
-		udelay(2000);
-		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
-				    TWL4030_REG_ANAMICL);
-	} while ((i++ < 100) &&
-		 ((byte & TWL4030_CNCL_OFFSET_START) ==
-		  TWL4030_CNCL_OFFSET_START));
-
-	/* Make sure that the reg_cache has the same value as the HW */
-	twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
-
-	/* anti-pop when changing analog gain */
-	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
-	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
-		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
-
-	/* toggle CODECPDZ as per TRM */
-	twl4030_codec_enable(codec, 0);
-	twl4030_codec_enable(codec, 1);
-}
-
-static void twl4030_power_down(struct snd_soc_codec *codec)
-{
-	/* power down */
-	twl4030_codec_enable(codec, 0);
-}
-
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
-- 
cgit v1.2.3


From 7393958f630ac91e591e62058f2bdb61523ec60c Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 29 Jan 2009 14:57:50 +0200
Subject: ASoC: TWL4030: Add analog loopback support

This patch adds the analog loopback/bypass support for twl4030 codec.

Details for the implementation:
It seams that the analog loopback needs the DAC powered on on the channel,
where the loopback is selected. The switch for the DACs has been moved from
the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be
powered while the audio playback is used or/and the loopback is enabled for
the channel.

The twl4030 codec powering has been reworked. Now the codec will be powered as
long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are
when the given change in the registers needs the codec power down/up cycle in
order to take effect. Otherwise the codec is on.

When the codec enters to STANDBY state and none of the loopback paths are
enabled, than the amplifiers, which are no in the DAPM path are forced to turn
off and the PLL is disabled. When playback/capture starts the disabled gains
are restored and the PLL is enabled.

When one of the loopback enabled in STANDBY mode, the disabled gains are
restored and the PLL is enabled also.

In short: the codec always goes to the lowest power state based on the
bias_level and the bypass_state.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 212 ++++++++++++++++++++++++++++++++++++++++++---
 sound/soc/codecs/twl4030.h |  15 ++++
 2 files changed, 214 insertions(+), 13 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index f985bef40a39..c26854b398d3 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -117,6 +117,13 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_MISC_SET_2		(0x49)	*/
 };
 
+/* codec private data */
+struct twl4030_priv {
+	unsigned int bypass_state;
+	unsigned int codec_powered;
+	unsigned int codec_muted;
+};
+
 /*
  * read twl4030 register cache
  */
@@ -156,8 +163,12 @@ 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;
 	u8 mode;
 
+	if (enable == twl4030->codec_powered)
+		return;
+
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
 	if (enable)
 		mode |= TWL4030_CODECPDZ;
@@ -165,6 +176,7 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 		mode &= ~TWL4030_CODECPDZ;
 
 	twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+	twl4030->codec_powered = enable;
 
 	/* REVISIT: this delay is present in TI sample drivers */
 	/* but there seems to be no TRM requirement for it     */
@@ -184,11 +196,82 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 }
 
+static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
+{
+	struct twl4030_priv *twl4030 = codec->private_data;
+	u8 reg_val;
+
+	if (mute == twl4030->codec_muted)
+		return;
+
+	if (mute) {
+		/* Bypass the reg_cache and mute the volumes
+		 * Headset mute is done in it's own event handler
+		 * Things to mute:  Earpiece, PreDrivL/R, CarkitL/R
+		 */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_EAR_GAIN),
+					TWL4030_REG_EAR_CTL);
+
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PREDL_GAIN),
+					TWL4030_REG_PREDL_CTL);
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PREDR_GAIN),
+					TWL4030_REG_PREDL_CTL);
+
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PRECKL_GAIN),
+					TWL4030_REG_PRECKL_CTL);
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PRECKL_GAIN),
+					TWL4030_REG_PRECKR_CTL);
+
+		/* Disable PLL */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
+		reg_val &= ~TWL4030_APLL_EN;
+		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
+	} else {
+		/* Restore the volumes
+		 * Headset mute is done in it's own event handler
+		 * Things to restore:  Earpiece, PreDrivL/R, CarkitL/R
+		 */
+		twl4030_write(codec, TWL4030_REG_EAR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL));
+
+		twl4030_write(codec, TWL4030_REG_PREDL_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL));
+		twl4030_write(codec, TWL4030_REG_PREDR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL));
+
+		twl4030_write(codec, TWL4030_REG_PRECKL_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL));
+		twl4030_write(codec, TWL4030_REG_PRECKR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL));
+
+		/* Enable PLL */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
+		reg_val |= TWL4030_APLL_EN;
+		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
+	}
+
+	twl4030->codec_muted = mute;
+}
+
 static void twl4030_power_up(struct snd_soc_codec *codec)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
 	u8 anamicl, regmisc1, byte;
 	int i = 0;
 
+	if (twl4030->codec_powered)
+		return;
+
 	/* set CODECPDZ to turn on codec */
 	twl4030_codec_enable(codec, 1);
 
@@ -220,6 +303,9 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
 	twl4030_codec_enable(codec, 1);
 }
 
+/*
+ * Unconditional power down
+ */
 static void twl4030_power_down(struct snd_soc_codec *codec)
 {
 	/* power down */
@@ -402,6 +488,22 @@ static const struct soc_enum twl4030_micpathtx2_enum =
 static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
 SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
 
+/* Analog bypass for AudioR1 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassr1_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR1_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioL1 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassl1_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL1_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioR2 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR2_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioL2 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
+
 static int micpath_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -497,6 +599,31 @@ static int headsetl_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int bypass_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_mixer_control *m =
+		(struct soc_mixer_control *)w->kcontrols->private_value;
+	struct twl4030_priv *twl4030 = w->codec->private_data;
+	unsigned char reg;
+
+	reg = twl4030_read_reg_cache(w->codec, m->reg);
+	if (reg & (1 << m->shift))
+		twl4030->bypass_state |=
+			(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+	else
+		twl4030->bypass_state &=
+			~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+
+	if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(w->codec, 0);
+		else
+			twl4030_codec_mute(w->codec, 1);
+	}
+	return 0;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -775,13 +902,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
-			TWL4030_REG_AVDAC_CTL, 0, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
-			TWL4030_REG_AVDAC_CTL, 1, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
-			TWL4030_REG_AVDAC_CTL, 2, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
-			TWL4030_REG_AVDAC_CTL, 3, 0),
+			SND_SOC_NOPM, 0, 0),
 
 	/* Analog PGAs */
 	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
@@ -793,6 +920,29 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
 			0, 0, NULL, 0),
 
+	/* Analog bypasses */
+	SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr1_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl1_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr2_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl2_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+
+	SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			3, 0, NULL, 0),
+
 	/* Output MUX controls */
 	/* Earpiece */
 	SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
@@ -863,13 +1013,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
 	SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
 	SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0),
+
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
-	{"ARXL1_APGA", NULL, "DAC Left1"},
-	{"ARXR1_APGA", NULL, "DAC Right1"},
-	{"ARXL2_APGA", NULL, "DAC Left2"},
-	{"ARXR2_APGA", NULL, "DAC Right2"},
+	{"Analog L1 Playback Mixer", NULL, "DAC Left1"},
+	{"Analog R1 Playback Mixer", NULL, "DAC Right1"},
+	{"Analog L2 Playback Mixer", NULL, "DAC Left2"},
+	{"Analog R2 Playback Mixer", NULL, "DAC Right2"},
+
+	{"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
+	{"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
+	{"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
+	{"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
 
 	/* Internal playback routings */
 	/* Earpiece */
@@ -951,6 +1107,17 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
 	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
 
+	/* Analog bypass routes */
+	{"Right1 Analog Loopback", "Switch", "Analog Right Capture Route"},
+	{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
+	{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
+	{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+
+	{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
+	{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
+	{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
+	{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+
 };
 
 static int twl4030_add_widgets(struct snd_soc_codec *codec)
@@ -967,16 +1134,25 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		twl4030_power_up(codec);
+		twl4030_codec_mute(codec, 0);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		/* TODO: develop a twl4030_prepare function */
+		twl4030_power_up(codec);
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(codec, 0);
+		else
+			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* TODO: develop a twl4030_standby function */
-		twl4030_power_down(codec);
+		twl4030_power_up(codec);
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(codec, 0);
+		else
+			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_OFF:
 		twl4030_power_down(codec);
@@ -996,7 +1172,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = socdev->card->codec;
 	u8 mode, old_mode, format, old_format;
 
-
 	/* bit rate */
 	old_mode = twl4030_read_reg_cache(codec,
 			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1038,6 +1213,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 
 	if (mode != old_mode) {
 		/* change rate and set CODECPDZ */
+		twl4030_codec_enable(codec, 0);
 		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
 		twl4030_codec_enable(codec, 1);
 	}
@@ -1258,11 +1434,19 @@ static int twl4030_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
+	struct twl4030_priv *twl4030;
 
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
 
+	twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL);
+	if (twl4030 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = twl4030;
 	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -1280,8 +1464,10 @@ static int twl4030_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->card->codec;
 
 	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
+	kfree(codec->private_data);
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 442e5a828617..33dbb144dad1 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -170,6 +170,9 @@
 #define TWL4030_CLK256FS_EN		0x02
 #define TWL4030_AIF_EN			0x01
 
+/* EAR_CTL (0x21) */
+#define TWL4030_EAR_GAIN		0x30
+
 /* HS_GAIN_SET (0x23) Fields */
 
 #define TWL4030_HSR_GAIN		0x0C
@@ -198,6 +201,18 @@
 #define TWL4030_RAMP_DELAY_2581MS	0x1C
 #define TWL4030_RAMP_EN			0x02
 
+/* PREDL_CTL (0x25) */
+#define TWL4030_PREDL_GAIN		0x30
+
+/* PREDR_CTL (0x26) */
+#define TWL4030_PREDR_GAIN		0x30
+
+/* PRECKL_CTL (0x27) */
+#define TWL4030_PRECKL_GAIN		0x30
+
+/* PRECKR_CTL (0x28) */
+#define TWL4030_PRECKR_GAIN		0x30
+
 /* HFL_CTL (0x29, 0x2A) Fields */
 #define TWL4030_HF_CTL_HB_EN		0x04
 #define TWL4030_HF_CTL_LOOP_EN		0x08
-- 
cgit v1.2.3


From 04eb093c7c81d118efeb96228f69bc0179f71897 Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Thu, 29 Jan 2009 14:28:37 -0600
Subject: ASoC: fix initialization order of the CS4270 codec driver

ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself.
When a codec driver registers with ASoC, a probe function is called.  Most
codec drivers call ASoC first, and then register with the I2C bus in the ASoC
probe function.

However, in order to support multiple codecs on one board, it's easier if the
codec driver is probed via the I2C bus first.  This is because the call to
i2c_add_driver() can result in the I2C probe function being called multiple
times - once for each codec.  In the current design, the driver registers
once with ASoC, and in the ASoC probe function, it calls i2c_add_driver().
The results in the I2C probe function being called multiple times before the
driver can register with ASoC again.

The new design has the driver call i2c_add_driver() first.  In the I2C probe
function, the driver registers with ASoC.  This allows the ASoC probe function
to be called once per I2C device.

Also add code to check if the I2C probe function is called more than once,
since that is not supported with the current ASoC design.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 177 +++++++++++++++++++++++++---------------------
 1 file changed, 97 insertions(+), 80 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 21253b48289f..adc1150ddb00 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -490,21 +490,17 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 };
 
 /*
- * Global variable to store socdev for i2c probe function.
+ * Global variable to store codec for the ASoC probe function.
  *
  * If struct i2c_driver had a private_data field, we wouldn't need to use
- * cs4270_socdec.  This is the only way to pass the socdev structure to
- * cs4270_i2c_probe().
- *
- * The real solution to cs4270_socdev is to create a mechanism
- * that maps I2C addresses to snd_soc_device structures.  Perhaps the
- * creation of the snd_soc_device object should be moved out of
- * cs4270_probe() and into cs4270_i2c_probe(), but that would make this
- * driver dependent on I2C.  The CS4270 supports "stand-alone" mode, whereby
- * the chip is *not* connected to the I2C bus, but is instead configured via
- * input pins.
+ * cs4270_codec.  This is the only way to pass the codec structure from
+ * cs4270_i2c_probe() to cs4270_probe().  Unfortunately, there is no good
+ * way to synchronize these two functions.  cs4270_i2c_probe() can be called
+ * multiple times before cs4270_probe() is called even once.  So for now, we
+ * also only allow cs4270_i2c_probe() to be run once.  That means that we do
+ * not support more than one cs4270 device in the system, at least for now.
  */
-static struct snd_soc_device *cs4270_socdev;
+static struct snd_soc_codec *cs4270_codec;
 
 struct snd_soc_dai cs4270_dai = {
 	.name = "cs4270",
@@ -531,6 +527,70 @@ struct snd_soc_dai cs4270_dai = {
 };
 EXPORT_SYMBOL_GPL(cs4270_dai);
 
+/*
+ * ASoC probe function
+ */
+static int cs4270_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = cs4270_codec;
+	unsigned int i;
+	int ret;
+
+	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
+	socdev->card->codec = codec;
+
+	/* Register PCMs */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "cs4270: failed to create PCMs\n");
+		return ret;
+	}
+
+	/* Add the non-DAPM controls */
+	for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
+		struct snd_kcontrol *kctrl;
+
+		kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
+		if (!kctrl) {
+			printk(KERN_ERR "cs4270: error creating control '%s'\n",
+			       cs4270_snd_controls[i].name);
+			ret = -ENOMEM;
+			goto error_free_pcms;
+		}
+
+		ret = snd_ctl_add(codec->card, kctrl);
+		if (ret < 0) {
+			printk(KERN_ERR "cs4270: error adding control '%s'\n",
+			       cs4270_snd_controls[i].name);
+			goto error_free_pcms;
+		}
+	}
+
+	/* And finally, register the socdev */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "cs4270: failed to register card\n");
+		goto error_free_pcms;
+	}
+
+	return 0;
+
+error_free_pcms:
+	snd_soc_free_pcms(socdev);
+
+	return ret;
+}
+
+static int cs4270_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+
+	return 0;
+};
+
 /*
  * Initialize the I2C interface of the CS4270
  *
@@ -543,17 +603,27 @@ EXPORT_SYMBOL_GPL(cs4270_dai);
 static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = cs4270_socdev;
 	struct snd_soc_codec *codec;
 	struct cs4270_private *cs4270;
-	int i;
-	int ret = 0;
+	int ret;
+
+	/* For now, we only support one cs4270 device in the system.  See the
+	 * comment for cs4270_codec.
+	 */
+	if (cs4270_codec) {
+		printk(KERN_ERR "cs4270: ignoring CS4270 at addr %X\n",
+		       i2c_client->addr);
+		printk(KERN_ERR "cs4270: only one CS4270 per board allowed\n");
+		/* Should we return something other than ENODEV here? */
+		return -ENODEV;
+	}
 
 	/* Verify that we have a CS4270 */
 
 	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to read I2C\n");
+		printk(KERN_ERR "cs4270: failed to read I2C at addr %X\n",
+		       i2c_client->addr);
 		return ret;
 	}
 	/* The top four bits of the chip ID should be 1100. */
@@ -575,7 +645,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 		return -ENOMEM;
 	}
 	codec = &cs4270->codec;
-	socdev->card->codec = codec;
+	cs4270_codec = codec;
 
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -600,50 +670,20 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 		goto error_free_codec;
 	}
 
-	/* Register PCMs */
-
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	/* Register the DAI.  If all the other ASoC driver have already
+	 * registered, then this will call our probe function, so
+	 * cs4270_codec needs to be ready.
+	 */
+	ret = snd_soc_register_dai(&cs4270_dai);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to create PCMs\n");
+		printk(KERN_ERR "cs4270: failed to register DAIe\n");
 		goto error_free_codec;
 	}
 
-	/* Add the non-DAPM controls */
-
-	for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
-		struct snd_kcontrol *kctrl;
-
-		kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
-		if (!kctrl) {
-			printk(KERN_ERR "cs4270: error creating control '%s'\n",
-			       cs4270_snd_controls[i].name);
-			ret = -ENOMEM;
-			goto error_free_pcms;
-		}
-
-		ret = snd_ctl_add(codec->card, kctrl);
-		if (ret < 0) {
-			printk(KERN_ERR "cs4270: error adding control '%s'\n",
-			       cs4270_snd_controls[i].name);
-			goto error_free_pcms;
-		}
-	}
-
-	/* Initialize the SOC device */
-
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to register card\n");
-		goto error_free_pcms;;
-	}
-
-	i2c_set_clientdata(i2c_client, socdev);
+	i2c_set_clientdata(i2c_client, cs4270);
 
 	return 0;
 
-error_free_pcms:
-	snd_soc_free_pcms(socdev);
-
 error_free_codec:
 	kfree(cs4270);
 
@@ -652,11 +692,8 @@ error_free_codec:
 
 static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 {
-	struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client);
-	struct snd_soc_codec *codec = socdev->card->codec;
-	struct cs4270_private *cs4270 = codec->private_data;
+	struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client);
 
-	snd_soc_free_pcms(socdev);
 	kfree(cs4270);
 
 	return 0;
@@ -678,26 +715,6 @@ static struct i2c_driver cs4270_i2c_driver = {
 	.remove = cs4270_i2c_remove,
 };
 
-/*
- * ASoC probe function
- *
- * This function is called when the machine driver calls
- * platform_device_add().
- */
-static int cs4270_probe(struct platform_device *pdev)
-{
-	cs4270_socdev = platform_get_drvdata(pdev);;
-
-	return i2c_add_driver(&cs4270_i2c_driver);
-}
-
-static int cs4270_remove(struct platform_device *pdev)
-{
-	i2c_del_driver(&cs4270_i2c_driver);
-
-	return 0;
-}
-
 /*
  * ASoC codec device structure
  *
@@ -714,13 +731,13 @@ static int __init cs4270_init(void)
 {
 	printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
 
-	return snd_soc_register_dai(&cs4270_dai);
+	return i2c_add_driver(&cs4270_i2c_driver);
 }
 module_init(cs4270_init);
 
 static void __exit cs4270_exit(void)
 {
-	snd_soc_unregister_dai(&cs4270_dai);
+	i2c_del_driver(&cs4270_i2c_driver);
 }
 module_exit(cs4270_exit);
 
-- 
cgit v1.2.3


From ff7bf02f630ae93cad4feda0f6a5a19b25a5019a Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Fri, 30 Jan 2009 11:14:49 -0600
Subject: ASoC: fix documentation in CS4270 codec driver

Spruce up the documentation in the CS4270 codec.  Use kerneldoc where
appropriate.  Fix incorrect comments.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 163 +++++++++++++++++++++++++++++-----------------
 1 file changed, 105 insertions(+), 58 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index adc1150ddb00..e5f5afdd3427 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -3,10 +3,10 @@
  *
  * Author: Timur Tabi <timur@freescale.com>
  *
- * Copyright 2007 Freescale Semiconductor, Inc.  This file is licensed under
- * the terms of the GNU General Public License version 2.  This program
- * is licensed "as is" without any warranty of any kind, whether express
- * or implied.
+ * Copyright 2007-2009 Freescale Semiconductor, Inc.  This file is licensed
+ * under the terms of the GNU General Public License version 2.  This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
  *
  * This is an ASoC device driver for the Cirrus Logic CS4270 codec.
  *
@@ -111,8 +111,13 @@ struct cs4270_private {
 	unsigned int mode; /* The mode (I2S or left-justified) */
 };
 
-/*
- * Clock Ratio Selection for Master Mode with I2C enabled
+/**
+ * struct cs4270_mode_ratios - clock ratio tables
+ * @ratio: the ratio of MCLK to the sample rate
+ * @speed_mode: the Speed Mode bits to set in the Mode Control register for
+ *              this ratio
+ * @mclk: the Ratio Select bits to set in the Mode Control register for this
+ *        ratio
  *
  * The data for this chart is taken from Table 5 of the CS4270 reference
  * manual.
@@ -121,31 +126,30 @@ struct cs4270_private {
  * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling
  * rates the CS4270 currently supports.
  *
- * Each element in this array corresponds to the ratios in mclk_ratios[].
- * These two arrays need to be in sync.
- *
- * 'speed_mode' is the corresponding bit pattern to be written to the
+ * @speed_mode is the corresponding bit pattern to be written to the
  * MODE bits of the Mode Control Register
  *
- * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of
+ * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of
  * the Mode Control Register.
  *
  * In situations where a single ratio is represented by multiple speed
  * modes, we favor the slowest speed.  E.g, for a ratio of 128, we pick
  * double-speed instead of quad-speed.  However, the CS4270 errata states
- * that Divide-By-1.5 can cause failures, so we avoid that mode where
+ * that divide-By-1.5 can cause failures, so we avoid that mode where
  * possible.
  *
- * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not
- * work if VD = 3.3V.  If this effects you, select the
+ * Errata: There is an errata for the CS4270 where divide-by-1.5 does not
+ * work if Vd is 3.3V.  If this effects you, select the
  * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will
  * never select any sample rates that require divide-by-1.5.
  */
-static struct {
+struct cs4270_mode_ratios {
 	unsigned int ratio;
 	u8 speed_mode;
 	u8 mclk;
-} cs4270_mode_ratios[] = {
+};
+
+static struct cs4270_mode_ratios[] = {
 	{64, CS4270_MODE_4X, CS4270_MODE_DIV1},
 #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA
 	{96, CS4270_MODE_4X, CS4270_MODE_DIV15},
@@ -162,34 +166,27 @@ static struct {
 /* The number of MCLK/LRCK ratios supported by the CS4270 */
 #define NUM_MCLK_RATIOS		ARRAY_SIZE(cs4270_mode_ratios)
 
-/*
- * Determine the CS4270 samples rates.
+/**
+ * cs4270_set_dai_sysclk - determine the CS4270 samples rates.
+ * @codec_dai: the codec DAI
+ * @clk_id: the clock ID (ignored)
+ * @freq: the MCLK input frequency
+ * @dir: the clock direction (ignored)
  *
- * 'freq' is the input frequency to MCLK.  The other parameters are ignored.
+ * This function is used to tell the codec driver what the input MCLK
+ * frequency is.
  *
  * The value of MCLK is used to determine which sample rates are supported
  * by the CS4270.  The ratio of MCLK / Fs must be equal to one of nine
- * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
+ * supported values - 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
  *
  * This function calculates the nine ratios and determines which ones match
  * a standard sample rate.  If there's a match, then it is added to the list
- * of support sample rates.
+ * of supported sample rates.
  *
  * This function must be called by the machine driver's 'startup' function,
  * otherwise the list of supported sample rates will not be available in
  * time for ALSA.
- *
- * Note that in stand-alone mode, the sample rate is determined by input
- * pins M0, M1, MDIV1, and MDIV2.  Also in stand-alone mode, divide-by-3
- * is not a programmable option.  However, divide-by-3 is not an available
- * option in stand-alone mode.  This cases two problems: a ratio of 768 is
- * not available (it requires divide-by-3) and B) ratios 192 and 384 can
- * only be selected with divide-by-1.5, but there is an errate that make
- * this selection difficult.
- *
- * In addition, there is no mechanism for communicating with the machine
- * 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_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
@@ -230,8 +227,10 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-/*
- * Configure the codec for the selected audio format
+/**
+ * cs4270_set_dai_fmt - configure the codec for the selected audio format
+ * @codec_dai: the codec DAI
+ * @format: a SND_SOC_DAIFMT_x value indicating the data format
  *
  * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the
  * codec accordingly.
@@ -261,8 +260,16 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return ret;
 }
 
-/*
- * Pre-fill the CS4270 register cache.
+/**
+ * cs4270_fill_cache - pre-fill the CS4270 register cache.
+ * @codec: the codec for this CS4270
+ *
+ * This function fills in the CS4270 register cache by reading the register
+ * values from the hardware.
+ *
+ * This CS4270 registers are cached to avoid excessive I2C I/O operations.
+ * After the initial read to pre-fill the cache, the CS4270 never updates
+ * the register values, so we won't have a cache coherency problem.
  *
  * We use the auto-increment feature of the CS4270 to read all registers in
  * one shot.
@@ -285,12 +292,17 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 	return 0;
 }
 
-/*
- * Read from the CS4270 register cache.
+/**
+ * cs4270_read_reg_cache - read from the CS4270 register cache.
+ * @codec: the codec for this CS4270
+ * @reg: the register to read
+ *
+ * This function returns the value for a given register.  It reads only from
+ * the register cache, not the hardware itself.
  *
  * This CS4270 registers are cached to avoid excessive I2C I/O operations.
  * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherncy problem.
+ * the register values, so we won't have a cache coherency problem.
  */
 static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
@@ -303,8 +315,11 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
 	return cache[reg - CS4270_FIRSTREG];
 }
 
-/*
- * Write to a CS4270 register via the I2C bus.
+/**
+ * cs4270_i2c_write - write to a CS4270 register via the I2C bus.
+ * @codec: the codec for this CS4270
+ * @reg: the register to write
+ * @value: the value to write to the register
  *
  * This function writes the given value to the given CS4270 register, and
  * also updates the register cache.
@@ -336,11 +351,17 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
-/*
- * Program the CS4270 with the given hardware parameters.
+/**
+ * cs4270_hw_params - program the CS4270 with the given hardware parameters.
+ * @substream: the audio stream
+ * @params: the hardware parameters to set
+ * @dai: the SOC DAI (ignored)
  *
- * The .ops functions are used to provide board-specific data, like
- * input frequencies, to this driver.  This function takes that information,
+ * This function programs the hardware with the values provided.
+ * Specifically, the sample rate and the data format.
+ *
+ * The .ops functions are used to provide board-specific data, like input
+ * frequencies, to this driver.  This function takes that information,
  * combines it with the hardware parameters provided, and programs the
  * hardware accordingly.
  */
@@ -455,8 +476,10 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 }
 
 #ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-/*
- * Set the CS4270 external mute
+/**
+ * cs4270_mute - enable/disable the CS4270 external mute
+ * @dai: the SOC DAI
+ * @mute: 0 = disable mute, 1 = enable mute
  *
  * This function toggles the mute bits in the MUTE register.  The CS4270's
  * mute capability is intended for external muting circuitry, so if the
@@ -490,7 +513,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 };
 
 /*
- * Global variable to store codec for the ASoC probe function.
+ * cs4270_codec - global variable to store codec for the ASoC probe function
  *
  * If struct i2c_driver had a private_data field, we wouldn't need to use
  * cs4270_codec.  This is the only way to pass the codec structure from
@@ -527,8 +550,12 @@ struct snd_soc_dai cs4270_dai = {
 };
 EXPORT_SYMBOL_GPL(cs4270_dai);
 
-/*
- * ASoC probe function
+/**
+ * cs4270_probe - ASoC probe function
+ * @pdev: platform device
+ *
+ * This function is called when ASoC has all the pieces it needs to
+ * instantiate a sound driver.
  */
 static int cs4270_probe(struct platform_device *pdev)
 {
@@ -582,6 +609,12 @@ error_free_pcms:
 	return ret;
 }
 
+/**
+ * cs4270_remove - ASoC remove function
+ * @pdev: platform device
+ *
+ * This function is the counterpart to cs4270_probe().
+ */
 static int cs4270_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -591,14 +624,13 @@ static int cs4270_remove(struct platform_device *pdev)
 	return 0;
 };
 
-/*
- * Initialize the I2C interface of the CS4270
- *
- * This function is called for whenever the I2C subsystem finds a device
- * at a particular address.
+/**
+ * cs4270_i2c_probe - initialize the I2C interface of the CS4270
+ * @i2c_client: the I2C client object
+ * @id: the I2C device ID (ignored)
  *
- * Note: snd_soc_new_pcms() must be called before this function can be called,
- * because of snd_ctl_add().
+ * This function is called whenever the I2C subsystem finds a device that
+ * matches the device ID given via a prior call to i2c_add_driver().
  */
 static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	const struct i2c_device_id *id)
@@ -690,6 +722,12 @@ error_free_codec:
 	return ret;
 }
 
+/**
+ * cs4270_i2c_remove - remove an I2C device
+ * @i2c_client: the I2C client object
+ *
+ * This function is the counterpart to cs4270_i2c_probe().
+ */
 static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 {
 	struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client);
@@ -699,12 +737,21 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 	return 0;
 }
 
+/*
+ * cs4270_id - I2C device IDs supported by this driver
+ */
 static struct i2c_device_id cs4270_id[] = {
 	{"cs4270", 0},
 	{}
 };
 MODULE_DEVICE_TABLE(i2c, cs4270_id);
 
+/*
+ * cs4270_i2c_driver - I2C device identification
+ *
+ * This structure tells the I2C subsystem how to identify and support a
+ * given I2C device type.
+ */
 static struct i2c_driver cs4270_i2c_driver = {
 	.driver = {
 		.name = "cs4270",
-- 
cgit v1.2.3


From d9fb7fbddc9a14569aad517984c1a5b0b07002ea Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Mon, 2 Feb 2009 14:50:45 -0600
Subject: ASoC: fix build break in CS4270 codec driver

Fix a oversight in the CS4270 codec driver that caused a build break.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index e5f5afdd3427..7962874258fb 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -149,7 +149,7 @@ struct cs4270_mode_ratios {
 	u8 mclk;
 };
 
-static struct cs4270_mode_ratios[] = {
+static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
 	{64, CS4270_MODE_4X, CS4270_MODE_DIV1},
 #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA
 	{96, CS4270_MODE_4X, CS4270_MODE_DIV15},
-- 
cgit v1.2.3


From a6c255e0945160b76eabe983f59e2129e0c66246 Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Mon, 2 Feb 2009 15:08:29 -0600
Subject: ASoC: fix message display in CS4270 codec driver

Replace printk calls with dev_xxx calls.  Set the 'dev' field of the codec
and codec_dai structures so that these calls work.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 66 ++++++++++++++++++++++++++++-------------------
 1 file changed, 39 insertions(+), 27 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7962874258fb..2c79a24186fd 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -212,7 +212,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	rates &= ~SNDRV_PCM_RATE_KNOT;
 
 	if (!rates) {
-		printk(KERN_ERR "cs4270: could not find a valid sample rate\n");
+		dev_err(codec->dev, "could not find a valid sample rate\n");
 		return -EINVAL;
 	}
 
@@ -253,7 +253,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
 		break;
 	default:
-		printk(KERN_ERR "cs4270: invalid DAI format\n");
+		dev_err(codec->dev, "invalid dai format\n");
 		ret = -EINVAL;
 	}
 
@@ -284,7 +284,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 		CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
 
 	if (length != CS4270_NUMREGS) {
-		printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n",
+		dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
 		       i2c_client->addr);
 		return -EIO;
 	}
@@ -340,7 +340,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (cache[reg - CS4270_FIRSTREG] != value) {
 		struct i2c_client *client = codec->control_data;
 		if (i2c_smbus_write_byte_data(client, reg, value)) {
-			printk(KERN_ERR "cs4270: I2C write failed\n");
+			dev_err(codec->dev, "i2c write failed\n");
 			return -EIO;
 		}
 
@@ -391,7 +391,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 
 	if (i == NUM_MCLK_RATIOS) {
 		/* We did not find a matching ratio */
-		printk(KERN_ERR "cs4270: could not find matching ratio\n");
+		dev_err(codec->dev, "could not find matching ratio\n");
 		return -EINVAL;
 	}
 
@@ -401,7 +401,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 			    CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC |
 			    CS4270_PWRCTL_PDN);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
@@ -413,7 +413,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 
 	ret = snd_soc_write(codec, CS4270_MODE, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
@@ -430,13 +430,13 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ;
 		break;
 	default:
-		printk(KERN_ERR "cs4270: unknown format\n");
+		dev_err(codec->dev, "unknown dai format\n");
 		return -EINVAL;
 	}
 
 	ret = snd_soc_write(codec, CS4270_FORMAT, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
@@ -447,7 +447,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 	reg &= ~CS4270_MUTE_AUTO;
 	ret = snd_soc_write(codec, CS4270_MUTE, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
@@ -460,7 +460,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
 	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
@@ -468,7 +468,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 
 	ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
@@ -570,7 +570,7 @@ static int cs4270_probe(struct platform_device *pdev)
 	/* Register PCMs */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to create PCMs\n");
+		dev_err(codec->dev, "failed to create pcms\n");
 		return ret;
 	}
 
@@ -580,7 +580,7 @@ static int cs4270_probe(struct platform_device *pdev)
 
 		kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
 		if (!kctrl) {
-			printk(KERN_ERR "cs4270: error creating control '%s'\n",
+			dev_err(codec->dev, "error creating control '%s'\n",
 			       cs4270_snd_controls[i].name);
 			ret = -ENOMEM;
 			goto error_free_pcms;
@@ -588,7 +588,7 @@ static int cs4270_probe(struct platform_device *pdev)
 
 		ret = snd_ctl_add(codec->card, kctrl);
 		if (ret < 0) {
-			printk(KERN_ERR "cs4270: error adding control '%s'\n",
+			dev_err(codec->dev, "error adding control '%s'\n",
 			       cs4270_snd_controls[i].name);
 			goto error_free_pcms;
 		}
@@ -597,7 +597,7 @@ static int cs4270_probe(struct platform_device *pdev)
 	/* And finally, register the socdev */
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to register card\n");
+		dev_err(codec->dev, "failed to register card\n");
 		goto error_free_pcms;
 	}
 
@@ -643,9 +643,9 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	 * comment for cs4270_codec.
 	 */
 	if (cs4270_codec) {
-		printk(KERN_ERR "cs4270: ignoring CS4270 at addr %X\n",
+		dev_err(&i2c_client->dev, "ignoring CS4270 at addr %X\n",
 		       i2c_client->addr);
-		printk(KERN_ERR "cs4270: only one CS4270 per board allowed\n");
+		dev_err(&i2c_client->dev, "only one per board allowed\n");
 		/* Should we return something other than ENODEV here? */
 		return -ENODEV;
 	}
@@ -654,35 +654,35 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 
 	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to read I2C at addr %X\n",
+		dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n",
 		       i2c_client->addr);
 		return ret;
 	}
 	/* The top four bits of the chip ID should be 1100. */
 	if ((ret & 0xF0) != 0xC0) {
-		printk(KERN_ERR "cs4270: device at addr %X is not a CS4270\n",
+		dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n",
 		       i2c_client->addr);
 		return -ENODEV;
 	}
 
-	printk(KERN_INFO "cs4270: found device at I2C address %X\n",
+	dev_info(&i2c_client->dev, "found device at i2c address %X\n",
 		i2c_client->addr);
-	printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF);
+	dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF);
 
 	/* Allocate enough space for the snd_soc_codec structure
 	   and our private data together. */
 	cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL);
 	if (!cs4270) {
-		printk(KERN_ERR "cs4270: Could not allocate codec structure\n");
+		dev_err(&i2c_client->dev, "could not allocate codec\n");
 		return -ENOMEM;
 	}
 	codec = &cs4270->codec;
-	cs4270_codec = codec;
 
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
+	codec->dev = &i2c_client->dev;
 	codec->name = "CS4270";
 	codec->owner = THIS_MODULE;
 	codec->dai = &cs4270_dai;
@@ -698,17 +698,25 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 
 	ret = cs4270_fill_cache(codec);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to fill register cache\n");
+		dev_err(&i2c_client->dev, "failed to fill register cache\n");
 		goto error_free_codec;
 	}
 
+	/* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI
+	 * structure for each CS4270 device, but the machine driver needs to
+	 * have a pointer to the DAI structure, so for now it must be a global
+	 * variable.
+	 */
+	cs4270_dai.dev = &i2c_client->dev;
+
 	/* Register the DAI.  If all the other ASoC driver have already
 	 * registered, then this will call our probe function, so
 	 * cs4270_codec needs to be ready.
 	 */
+	cs4270_codec = codec;
 	ret = snd_soc_register_dai(&cs4270_dai);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to register DAIe\n");
+		dev_err(&i2c_client->dev, "failed to register DAIe\n");
 		goto error_free_codec;
 	}
 
@@ -718,6 +726,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 
 error_free_codec:
 	kfree(cs4270);
+	cs4270_codec = NULL;
+	cs4270_dai.dev = NULL;
 
 	return ret;
 }
@@ -733,6 +743,8 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 	struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client);
 
 	kfree(cs4270);
+	cs4270_codec = NULL;
+	cs4270_dai.dev = NULL;
 
 	return 0;
 }
@@ -776,7 +788,7 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
 static int __init cs4270_init(void)
 {
-	printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
+	pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
 
 	return i2c_add_driver(&cs4270_i2c_driver);
 }
-- 
cgit v1.2.3


From 111f6fbeb73fc350fe3a08c4ecd0ccdf3e13bef0 Mon Sep 17 00:00:00 2001
From: Vasily Khoruzhick <anarsoul@gmail.com>
Date: Tue, 3 Feb 2009 15:52:56 +0200
Subject: ASoC: Don't unconditionally use the PLL in UDA1380

Without this fix driver switches to WSPLL in uda1380_pcm_prepare
even if SYSCLK was chosen (uda1380_pcm_prepare modifies UDA1380_CLK
register to disable R00_DAC_CLK before flushing reg cache)

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/uda1380.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 98e4a6560f06..6e4a1770ce8d 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -418,8 +418,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
 		uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
 	}
 
-	/* FIXME enable DAC_CLK */
-	uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+	/* FIXME restore DAC_CLK */
+	uda1380_write(codec, UDA1380_CLK, clk);
 
 	return 0;
 }
-- 
cgit v1.2.3


From 5b2474425ed2a625b75dcd8d648701e473b7d764 Mon Sep 17 00:00:00 2001
From: Philipp Zabel <philipp.zabel@gmail.com>
Date: Tue, 3 Feb 2009 17:19:40 +0100
Subject: ASoC: uda1380: split set_dai_fmt into _both, _playback and _capture
 variants

This patch splits set_dai_fmt into three variants (single interface,
dual interface playback only, dual interface capture only) so that
data input and output formats can be configured separately for dual
interface setups.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/uda1380.c | 68 +++++++++++++++++++++++++++++++++++++++++-----
 1 file changed, 61 insertions(+), 7 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 6e4a1770ce8d..5242b8156b38 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -356,7 +356,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -366,16 +366,70 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
 	iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
 
-	/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		iface |= R01_SFORI_I2S | R01_SFORO_I2S;
 		break;
 	case SND_SOC_DAIFMT_LSB:
-		iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+		iface |= R01_SFORI_LSB16 | R01_SFORO_LSB16;
 		break;
 	case SND_SOC_DAIFMT_MSB:
-		iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+		iface |= R01_SFORI_MSB | R01_SFORO_MSB;
+	}
+
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+		iface |= R01_SIM;
+
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~R01_SFORI_MASK;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORI_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORI_LSB16;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORI_MSB;
+	}
+
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~(R01_SIM | R01_SFORO_MASK);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORO_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORO_LSB16;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORO_MSB;
 	}
 
 	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
@@ -549,7 +603,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
 		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt,
+		.set_fmt = uda1380_set_dai_fmt_both,
 	},
 },
 { /* playback only - dual interface */
@@ -566,7 +620,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
 		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt,
+		.set_fmt = uda1380_set_dai_fmt_playback,
 	},
 },
 { /* capture only - dual interface*/
@@ -582,7 +636,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-		.set_fmt = uda1380_set_dai_fmt,
+		.set_fmt = uda1380_set_dai_fmt_capture,
 	},
 },
 };
-- 
cgit v1.2.3


From b93f74f604c53b546fced33d11aee722560de249 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jarkko.nikula@nokia.com>
Date: Mon, 9 Feb 2009 14:27:06 +0200
Subject: ASoC: TLV320AIC3X: Fix volume ranges

This is a minor fix but helps to define dB ranges for volume controls.

Only DAC digital volume has full register value range from 0 to 127 but
ADC PGA gain and output stage volume controls don't.

For ADC PGA, maximum value is 119 and then it saturates to the same
gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB
and is muted for values 118 and above.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 38 +++++++++++++++++++-------------------
 1 file changed, 19 insertions(+), 19 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index ac73e692a99b..cdb6dec14e1c 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -255,51 +255,51 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1),
 
 	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
+		     DACR1_2_RLOPM_VOL, 0, 118, 1),
 	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
 	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
 	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
+		     DACR1_2_LLOPM_VOL, 0, 118, 1),
 	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     0, 0x7f, 1),
+		     0, 118, 1),
 	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
-		     0, 0x7f, 1),
+		     0, 118, 1),
 	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
-		     LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
+		     LINE2R_2_LLOPM_VOL, 0, 118, 1),
 	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
-		     LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
+		     LINE2R_2_RLOPM_VOL, 0, 118, 1),
 
 	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
-		     DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1),
+		     DACR1_2_MONOLOPM_VOL, 0, 118, 1),
 	SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
 	SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL,
-		     PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1),
+		     PGAR_2_MONOLOPM_VOL, 0, 118, 1),
 	SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL,
-		     LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1),
+		     LINE2R_2_MONOLOPM_VOL, 0, 118, 1),
 
 	SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL,
-		     DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
+		     DACR1_2_HPROUT_VOL, 0, 118, 1),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
 	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
-		     PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
+		     PGAR_2_HPROUT_VOL, 0, 118, 1),
 	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
-		     0, 0x7f, 1),
+		     0, 118, 1),
 	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
-		     0, 0x7f, 1),
+		     0, 118, 1),
 	SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
-		     LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
+		     LINE2R_2_HPROUT_VOL, 0, 118, 1),
 
 	SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL,
-		     DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
+		     DACR1_2_HPRCOM_VOL, 0, 118, 1),
 	SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
 		     0x01, 0),
 	SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
-		     0, 0x7f, 1),
+		     0, 118, 1),
 	SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
-		     0, 0x7f, 1),
+		     0, 118, 1),
 	SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
-		     LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
+		     LINE2R_2_HPRCOM_VOL, 0, 118, 1),
 
 	/*
 	 * Note: enable Automatic input Gain Controller with care. It can
@@ -308,7 +308,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
 
 	/* Input */
-	SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0),
+	SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0),
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
 
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
-- 
cgit v1.2.3


From 7565fc38cc8c3a2742d63e957199d70a82d2faf1 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jarkko.nikula@nokia.com>
Date: Mon, 9 Feb 2009 14:27:07 +0200
Subject: ASoC: TLV320AIC3X: Add TLV information for volume controls

TLV320AIC3X volume controls are logarithmic. Export their dB ranges.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 114 ++++++++++++++++++++++++++---------------
 1 file changed, 73 insertions(+), 41 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cdb6dec14e1c..d638e3f0728b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -45,6 +45,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
+#include <sound/tlv.h>
 
 #include "tlv320aic3x.h"
 
@@ -250,56 +251,86 @@ static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
 };
 
+/*
+ * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0);
+/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */
+static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0);
+/*
+ * Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB.
+ * Step size is approximately 0.5 dB over most of the scale but increasing
+ * near the very low levels.
+ * Define dB scale so that it is mostly correct for range about -55 to 0 dB
+ * but having increasing dB difference below that (and where it doesn't count
+ * so much). This setting shows -50 dB (actual is -50.3 dB) for register
+ * value 100 and -58.5 dB (actual is -78.3 dB) for register value 117.
+ */
+static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1);
+
 static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	/* Output */
-	SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("PCM Playback Volume",
+			 LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv),
 
-	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_RLOPM_VOL, 0, 118, 1),
+	SOC_DOUBLE_R_TLV("Line DAC Playback Volume",
+			 DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
 	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
-	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_LLOPM_VOL, 0, 118, 1),
-	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     0, 118, 1),
-	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
-		     0, 118, 1),
-	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
-		     LINE2R_2_LLOPM_VOL, 0, 118, 1),
-	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
-		     LINE2R_2_RLOPM_VOL, 0, 118, 1),
-
-	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
-		     DACR1_2_MONOLOPM_VOL, 0, 118, 1),
+	SOC_DOUBLE_R_TLV("LineL DAC Playback Volume",
+			 DACL1_2_LLOPM_VOL, DACR1_2_LLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("LineL Left PGA Bypass Playback Volume",
+		       PGAL_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("LineR Right PGA Bypass Playback Volume",
+		       PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("LineL Line2 Bypass Playback Volume",
+			 LINE2L_2_LLOPM_VOL, LINE2R_2_LLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("LineR Line2 Bypass Playback Volume",
+			 LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("Mono DAC Playback Volume",
+			 DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
-	SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL,
-		     PGAR_2_MONOLOPM_VOL, 0, 118, 1),
-	SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL,
-		     LINE2R_2_MONOLOPM_VOL, 0, 118, 1),
-
-	SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL,
-		     DACR1_2_HPROUT_VOL, 0, 118, 1),
+	SOC_DOUBLE_R_TLV("Mono PGA Bypass Playback Volume",
+			 PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("Mono Line2 Bypass Playback Volume",
+			 LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("HP DAC Playback Volume",
+			 DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
-		     PGAR_2_HPROUT_VOL, 0, 118, 1),
-	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
-		     0, 118, 1),
-	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
-		     0, 118, 1),
-	SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
-		     LINE2R_2_HPROUT_VOL, 0, 118, 1),
-
-	SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL,
-		     DACR1_2_HPRCOM_VOL, 0, 118, 1),
+	SOC_DOUBLE_R_TLV("HP Right PGA Bypass Playback Volume",
+			 PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPL PGA Bypass Playback Volume",
+		       PGAL_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPR PGA Bypass Playback Volume",
+		       PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("HP Line2 Bypass Playback Volume",
+			 LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume",
+			 DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
 		     0x01, 0),
-	SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
-		     0, 118, 1),
-	SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
-		     0, 118, 1),
-	SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
-		     LINE2R_2_HPRCOM_VOL, 0, 118, 1),
+	SOC_SINGLE_TLV("HPLCOM PGA Bypass Playback Volume",
+		       PGAL_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPRCOM PGA Bypass Playback Volume",
+		       PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Playback Volume",
+			 LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL,
+			 0, 118, 1, output_stage_tlv),
 
 	/*
 	 * Note: enable Automatic input Gain Controller with care. It can
@@ -308,7 +339,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
 
 	/* Input */
-	SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0),
+	SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
+			 0, 119, 0, adc_tlv),
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
 
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
-- 
cgit v1.2.3


From d5e9ba1d58b6da1c58a91113fc350ece97ec5a0b Mon Sep 17 00:00:00 2001
From: Timur Tabi <timur@freescale.com>
Date: Tue, 3 Feb 2009 11:09:32 -0600
Subject: ASoC: add additional controls to the CS4270 codec driver

Update the CS4270 codec driver to allow applications to use the mixer to
control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute.
Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first
initializes the hardware, but these features either don't work or interfere
with normal ALSA behavior.  However, they can now be re-enabled by an
application if desired.

Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute
bits.  The driver previously and erroneously assumed that these bits
control only external muting circuitry, but they also control internal
muting circuitry, so they should always be used.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |  6 ---
 sound/soc/codecs/cs4270.c | 93 ++++++++++++++++++++---------------------------
 2 files changed, 40 insertions(+), 59 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a195303603e0..628a591c728f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -67,12 +67,6 @@ config SND_SOC_AK4535
 config SND_SOC_CS4270
 	tristate
 
-# Cirrus Logic CS4270 Codec Hardware Mute Support
-# Select if you have external muting circuitry attached to your CS4270.
-config SND_SOC_CS4270_HWMUTE
-	bool
-	depends on SND_SOC_CS4270
-
 # Cirrus Logic CS4270 Codec VD = 3.3V Errata
 # Select if you are affected by the errata where the part will not function
 # if MCLK divide-by-1.5 is selected and VD is set to 3.3V.  The driver will
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2c79a24186fd..cd4a9ee38e46 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -395,17 +395,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/* Freeze and power-down the codec */
-
-	ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE |
-			    CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC |
-			    CS4270_PWRCTL_PDN);
-	if (ret < 0) {
-		dev_err(codec->dev, "i2c write failed\n");
-		return ret;
-	}
-
-	/* Program the mode control register */
+	/* Set the sample rate */
 
 	reg = snd_soc_read(codec, CS4270_MODE);
 	reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK);
@@ -417,7 +407,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	/* Program the format register */
+	/* Set the DAI format */
 
 	reg = snd_soc_read(codec, CS4270_FORMAT);
 	reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK);
@@ -440,42 +430,9 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	/* Disable auto-mute.  This feature appears to be buggy, because in
-	   some situations, auto-mute will not deactivate when it should. */
-
-	reg = snd_soc_read(codec, CS4270_MUTE);
-	reg &= ~CS4270_MUTE_AUTO;
-	ret = snd_soc_write(codec, CS4270_MUTE, reg);
-	if (ret < 0) {
-		dev_err(codec->dev, "i2c write failed\n");
-		return ret;
-	}
-
-	/* Disable automatic volume control.  It's enabled by default, and
-	 * it causes volume change commands to be delayed, sometimes until
-	 * after playback has started.
-	 */
-
-	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
-	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
-	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
-	if (ret < 0) {
-		dev_err(codec->dev, "i2c write failed\n");
-		return ret;
-	}
-
-	/* Thaw and power-up the codec */
-
-	ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
-	if (ret < 0) {
-		dev_err(codec->dev, "i2c write failed\n");
-		return ret;
-	}
-
 	return ret;
 }
 
-#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
 /**
  * cs4270_mute - enable/disable the CS4270 external mute
  * @dai: the SOC DAI
@@ -494,22 +451,23 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
 
 	if (mute)
-		reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
-			CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
+		reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
 	else
-		reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
-			  CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+		reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
-#else
-#define cs4270_mute NULL
-#endif
 
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
-		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1)
+		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1),
+	SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0),
+	SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0),
+	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
+	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
+	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
 };
 
 /*
@@ -637,6 +595,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 {
 	struct snd_soc_codec *codec;
 	struct cs4270_private *cs4270;
+	unsigned int reg;
 	int ret;
 
 	/* For now, we only support one cs4270 device in the system.  See the
@@ -702,6 +661,34 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 		goto error_free_codec;
 	}
 
+	/* Disable auto-mute.  This feature appears to be buggy.  In some
+	 * situations, auto-mute will not deactivate when it should, so we want
+	 * this feature disabled by default.  An application (e.g. alsactl) can
+	 * re-enabled it by using the controls.
+	 */
+
+	reg = cs4270_read_reg_cache(codec, CS4270_MUTE);
+	reg &= ~CS4270_MUTE_AUTO;
+	ret = cs4270_i2c_write(codec, CS4270_MUTE, reg);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "i2c write failed\n");
+		return ret;
+	}
+
+	/* Disable automatic volume control.  The hardware enables, and it
+	 * causes volume change commands to be delayed, sometimes until after
+	 * playback has started.  An application (e.g. alsactl) can
+	 * re-enabled it by using the controls.
+	 */
+
+	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "i2c write failed\n");
+		return ret;
+	}
+
 	/* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI
 	 * structure for each CS4270 device, but the machine driver needs to
 	 * have a pointer to the DAI structure, so for now it must be a global
-- 
cgit v1.2.3


From 31b59cf9cebb5bb675f49fe44814bbb7270374cc Mon Sep 17 00:00:00 2001
From: Paul Fertser <fercerpav@gmail.com>
Date: Mon, 16 Feb 2009 02:49:41 +0300
Subject: ASoC: Fix WM8753 DAIs unregistering

WM8753 uses a tricky way to switch DAIs "on the fly", for that it
registers 2 dummy DAIs and substitutes them depending on mixer control.

List element of registered dummy DAIs should be preserved to allow
unregistering of DAIs on module unload.

Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6f9e6beabb14..dc6042c6424f 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1451,30 +1451,35 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
 	if (mode < 4) {
 		int playback_active, capture_active, codec_active, pop_wait;
 		void *private_data;
+		struct list_head list;
 
 		playback_active = wm8753_dai[0].playback.active;
 		capture_active = wm8753_dai[0].capture.active;
 		codec_active = wm8753_dai[0].active;
 		private_data = wm8753_dai[0].private_data;
 		pop_wait = wm8753_dai[0].pop_wait;
+		list = wm8753_dai[0].list;
 		wm8753_dai[0] = wm8753_all_dai[mode << 1];
 		wm8753_dai[0].playback.active = playback_active;
 		wm8753_dai[0].capture.active = capture_active;
 		wm8753_dai[0].active = codec_active;
 		wm8753_dai[0].private_data = private_data;
 		wm8753_dai[0].pop_wait = pop_wait;
+		wm8753_dai[0].list = list;
 
 		playback_active = wm8753_dai[1].playback.active;
 		capture_active = wm8753_dai[1].capture.active;
 		codec_active = wm8753_dai[1].active;
 		private_data = wm8753_dai[1].private_data;
 		pop_wait = wm8753_dai[1].pop_wait;
+		list = wm8753_dai[1].list;
 		wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
 		wm8753_dai[1].playback.active = playback_active;
 		wm8753_dai[1].capture.active = capture_active;
 		wm8753_dai[1].active = codec_active;
 		wm8753_dai[1].private_data = private_data;
 		wm8753_dai[1].pop_wait = pop_wait;
+		wm8753_dai[1].list = list;
 	}
 	wm8753_dai[0].codec = codec;
 	wm8753_dai[1].codec = codec;
-- 
cgit v1.2.3


From 7b317b692a03a870d7acda0a0bd4d211f1c064fe Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 16 Feb 2009 14:08:22 +0000
Subject: ASoC: Remove version display from the WM8731 driver

It makes boot a bit more noisy and I never remember to update it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0150fe53a65d..816e5bf9edc1 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,8 +29,6 @@
 
 #include "wm8731.h"
 
-#define WM8731_VERSION "0.13"
-
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 /* codec private data */
@@ -702,8 +700,6 @@ static int wm8731_probe(struct platform_device *pdev)
 	struct wm8731_priv *wm8731;
 	int ret = 0;
 
-	pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
-- 
cgit v1.2.3


From 22d22ee5146ae823b1e93fe2887a7cba56015091 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 16 Feb 2009 19:20:15 +0000
Subject: ASoC: Clean up WM8731 bias level configuration

The WM8731 bias level configuration function was written slightly
obscurely - streamline the code a little and refresh the comments.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 8 +++-----
 1 file changed, 3 insertions(+), 5 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 816e5bf9edc1..c6db67793777 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -396,21 +396,19 @@ 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)
 {
-	u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
+	u16 reg;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		/* vref/mid, osc on, dac unmute */
-		wm8731_write(codec, WM8731_PWR, reg);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* everything off except vref/vmid, */
+		/* Clear PWROFF, gate CLKOUT, everything else as-is */
+		reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
 		wm8731_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
 		wm8731_write(codec, WM8731_ACTIVE, 0x0);
 		wm8731_write(codec, WM8731_PWR, 0xffff);
 		break;
-- 
cgit v1.2.3


From a8035c8f04477895207b92915b908344749be336 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 16 Feb 2009 19:35:43 +0000
Subject: ASoC: Shuffle WM8731 SPI and I2C device registration

This is a pure code motion patch intended to improve reviewability of a
following patch moving WM8731 to use more standard device registration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 215 ++++++++++++++++++++++++----------------------
 1 file changed, 112 insertions(+), 103 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c6db67793777..3ff971aeba21 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -36,6 +36,11 @@ struct wm8731_priv {
 	unsigned int sysclk;
 };
 
+#ifdef CONFIG_SPI_MASTER
+static int wm8731_spi_write(struct spi_device *spi, const char *data, int len);
+static struct spi_driver wm8731_spi_driver;
+#endif
+
 /*
  * wm8731 register cache
  * We can't read the WM8731 register space when we are
@@ -544,54 +549,9 @@ pcm_err:
 
 static struct snd_soc_device *wm8731_socdev;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-/*
- * WM8731 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
-static int wm8731_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
-{
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8731\n");
-
-	return ret;
-}
 
-static int wm8731_i2c_remove(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
-	return 0;
-}
-
-static const struct i2c_device_id wm8731_i2c_id[] = {
-	{ "wm8731", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
-
-static struct i2c_driver wm8731_i2c_driver = {
-	.driver = {
-		.name = "WM8731 I2C Codec",
-		.owner = THIS_MODULE,
-	},
-	.probe =    wm8731_i2c_probe,
-	.remove =   wm8731_i2c_remove,
-	.id_table = wm8731_i2c_id,
-};
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static struct i2c_driver wm8731_i2c_driver;
 
 static int wm8731_add_i2c_device(struct platform_device *pdev,
 				 const struct wm8731_setup_data *setup)
@@ -634,62 +594,6 @@ err_driver:
 }
 #endif
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8731_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
-
-	codec->control_data = spi;
-
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8731\n");
-
-	return ret;
-}
-
-static int __devexit wm8731_spi_remove(struct spi_device *spi)
-{
-	return 0;
-}
-
-static struct spi_driver wm8731_spi_driver = {
-	.driver = {
-		.name	= "wm8731",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8731_spi_probe,
-	.remove		= __devexit_p(wm8731_spi_remove),
-};
-
-static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
-{
-	struct spi_transfer t;
-	struct spi_message m;
-	u8 msg[2];
-
-	if (len <= 0)
-		return 0;
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-
-	spi_message_init(&m);
-	memset(&t, 0, (sizeof t));
-
-	t.tx_buf = &msg[0];
-	t.len = len;
-
-	spi_message_add_tail(&t, &m);
-	spi_sync(spi, &m);
-
-	return len;
-}
-#endif /* CONFIG_SPI_MASTER */
-
 static int wm8731_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -772,6 +676,111 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8731_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8731_socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8731_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8731\n");
+
+	return ret;
+}
+
+static int __devexit wm8731_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8731_spi_driver = {
+	.driver = {
+		.name	= "wm8731",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8731_spi_probe,
+	.remove		= __devexit_p(wm8731_spi_remove),
+};
+
+static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * WM8731 2 wire address is determined by GPIO5
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+
+static int wm8731_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct snd_soc_device *socdev = wm8731_socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int ret;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = wm8731_init(socdev);
+	if (ret < 0)
+		pr_err("failed to initialise WM8731\n");
+
+	return ret;
+}
+
+static int wm8731_i2c_remove(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	kfree(codec->reg_cache);
+	return 0;
+}
+
+static const struct i2c_device_id wm8731_i2c_id[] = {
+	{ "wm8731", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
+
+static struct i2c_driver wm8731_i2c_driver = {
+	.driver = {
+		.name = "WM8731 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8731_i2c_probe,
+	.remove =   wm8731_i2c_remove,
+	.id_table = wm8731_i2c_id,
+};
+#endif
+
 static int __init wm8731_modinit(void)
 {
 	return snd_soc_register_dai(&wm8731_dai);
-- 
cgit v1.2.3


From 5998102b9095fdb7c67755812038612afea315c5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 16 Feb 2009 20:49:16 +0000
Subject: ASoC: Refactor WM8731 device registration

Move the WM8731 driver to use a more standard device registration
scheme where the device can be registered independantly of the ASoC
probe.

As a transition measure push the current manual code for registering
the WM8731 into the individual machine driver probes. This allows
separate patches to update the relevant architecture files with less
risk of merge issues.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/sam9g20_wm8731.c |  43 ++++-
 sound/soc/codecs/wm8731.c        | 334 +++++++++++++++++----------------------
 sound/soc/codecs/wm8731.h        |   6 -
 sound/soc/pxa/corgi.c            |  43 ++++-
 sound/soc/pxa/poodle.c           |  41 ++++-
 5 files changed, 256 insertions(+), 211 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index aa524235fd98..173a239a541c 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -36,6 +36,7 @@
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
+#include <linux/i2c.h>
 
 #include <linux/atmel-ssc.h>
 
@@ -280,15 +281,41 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = {
 	.set_bias_level = at91sam9g20ek_set_bias_level,
 };
 
-static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_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, "wm8731", 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 snd_soc_device at91sam9g20ek_snd_devdata = {
 	.card = &snd_soc_at91sam9g20ek,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &at91sam9g20ek_wm8731_setup,
 };
 
 static struct platform_device *at91sam9g20ek_snd_device;
@@ -340,6 +367,10 @@ static int __init at91sam9g20ek_init(void)
 	}
 	ssc_p->ssc = ssc;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		goto err_ssc;
+
 	at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!at91sam9g20ek_snd_device) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
@@ -359,6 +390,8 @@ static int __init at91sam9g20ek_init(void)
 	return ret;
 
 err_ssc:
+	ssc_free(ssc);
+	ssc_p->ssc = NULL;
 err_mclk:
 	clk_put(mclk);
 	mclk = NULL;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 3ff971aeba21..a2c478e53d54 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,16 +29,18 @@
 
 #include "wm8731.h"
 
+static struct snd_soc_codec *wm8731_codec;
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 /* codec private data */
 struct wm8731_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8731_CACHEREGNUM];
 	unsigned int sysclk;
 };
 
 #ifdef CONFIG_SPI_MASTER
 static int wm8731_spi_write(struct spi_device *spi, const char *data, int len);
-static struct spi_driver wm8731_spi_driver;
 #endif
 
 /*
@@ -485,55 +487,33 @@ static int wm8731_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8731 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8731_init(struct snd_soc_device *socdev)
+static int wm8731_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int reg, ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
 
-	codec->name = "WM8731";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8731_read_reg_cache;
-	codec->write = wm8731_write;
-	codec->set_bias_level = wm8731_set_bias_level;
-	codec->dai = &wm8731_dai;
-	codec->num_dai = 1;
-	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;
+	if (wm8731_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
 
-	wm8731_reset(codec);
+	socdev->card->codec = wm8731_codec;
+	codec = wm8731_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8731: failed to create pcms\n");
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
 		goto pcm_err;
 	}
 
-	/* power on device */
-	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	/* set the update bits */
-	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
-	wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
-	wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
-	wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
-	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
-
 	snd_soc_add_controls(codec, wm8731_snd_controls,
-				ARRAY_SIZE(wm8731_snd_controls));
+			     ARRAY_SIZE(wm8731_snd_controls));
 	wm8731_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8731: failed to register card\n");
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
 		goto card_err;
 	}
 
@@ -543,104 +523,6 @@ card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
-	return ret;
-}
-
-static struct snd_soc_device *wm8731_socdev;
-
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static struct i2c_driver wm8731_i2c_driver;
-
-static int wm8731_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8731_setup_data *setup)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&wm8731_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, "wm8731", 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(&wm8731_i2c_driver);
-	return -ENODEV;
-}
-#endif
-
-static int wm8731_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8731_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8731_priv *wm8731;
-	int ret = 0;
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
-	if (wm8731 == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
-
-	codec->private_data = wm8731;
-	socdev->card->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	wm8731_socdev = socdev;
-	ret = -ENODEV;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8731_add_i2c_device(pdev, setup);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8731_spi_write;
-		ret = spi_register_driver(&wm8731_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
-#endif
-
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
 	return ret;
 }
 
@@ -648,22 +530,9 @@ static int wm8731_probe(struct platform_device *pdev)
 static int wm8731_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	if (codec->control_data)
-		wm8731_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(&wm8731_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8731_spi_driver);
-#endif
-	kfree(codec->private_data);
-	kfree(codec);
 
 	return 0;
 }
@@ -676,37 +545,78 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8731_spi_probe(struct spi_device *spi)
+static int wm8731_register(struct wm8731_priv *wm8731)
 {
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
+	struct snd_soc_codec *codec = &wm8731->codec;
+	u16 reg;
 
-	codec->control_data = spi;
+	if (wm8731_codec) {
+		dev_err(codec->dev, "Another WM8731 is registered\n");
+		return -EINVAL;
+	}
 
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8731\n");
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	return ret;
-}
+	codec->private_data = wm8731;
+	codec->name = "WM8731";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8731_read_reg_cache;
+	codec->write = wm8731_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8731_set_bias_level;
+	codec->dai = &wm8731_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8731_CACHEREGNUM;
+	codec->reg_cache = &wm8731->reg_cache;
+
+	memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg));
+
+	wm8731_dai.dev = codec->dev;
+
+	wm8731_reset(codec);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* Latch the update bits */
+	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
+	wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
+	wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
+	wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
+	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
+
+	wm8731_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8731_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
 
-static int __devexit wm8731_spi_remove(struct spi_device *spi)
-{
 	return 0;
 }
 
-static struct spi_driver wm8731_spi_driver = {
-	.driver = {
-		.name	= "wm8731",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8731_spi_probe,
-	.remove		= __devexit_p(wm8731_spi_remove),
-};
+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);
+	kfree(wm8731);
+	wm8731_codec = NULL;
+}
 
+#if defined(CONFIG_SPI_MASTER)
 static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
 {
 	struct spi_transfer t;
@@ -730,37 +640,67 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
 
 	return len;
 }
+
+static int __devinit wm8731_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_codec *codec;
+	struct wm8731_priv *wm8731;
+
+	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	if (wm8731 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8731->codec;
+	codec->control_data = spi;
+	codec->hw_write = (hw_write_t)wm8731_spi_write;
+	codec->dev = &spi->dev;
+
+	return wm8731_register(wm8731);
+}
+
+static int __devexit wm8731_spi_remove(struct spi_device *spi)
+{
+	/* FIXME: This isn't actually implemented... */
+	return 0;
+}
+
+static struct spi_driver wm8731_spi_driver = {
+	.driver = {
+		.name	= "wm8731",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8731_spi_probe,
+	.remove		= __devexit_p(wm8731_spi_remove),
+};
 #endif /* CONFIG_SPI_MASTER */
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-/*
- * WM8731 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
 static int wm8731_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
+	struct wm8731_priv *wm8731;
+	struct snd_soc_codec *codec;
 
-	i2c_set_clientdata(i2c, codec);
+	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	if (wm8731 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8731->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8731);
 	codec->control_data = i2c;
 
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8731\n");
+	codec->dev = &i2c->dev;
 
-	return ret;
+	return wm8731_register(wm8731);
 }
 
 static int wm8731_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+	struct wm8731_priv *wm8731 = i2c_get_clientdata(client);
+	wm8731_unregister(wm8731);
 	return 0;
 }
 
@@ -783,13 +723,33 @@ static struct i2c_driver wm8731_i2c_driver = {
 
 static int __init wm8731_modinit(void)
 {
-	return snd_soc_register_dai(&wm8731_dai);
+	int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8731_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n",
+		       ret);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8731_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n",
+		       ret);
+	}
+#endif
+	return 0;
 }
 module_init(wm8731_modinit);
 
 static void __exit wm8731_exit(void)
 {
-	snd_soc_unregister_dai(&wm8731_dai);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8731_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8731_spi_driver);
+#endif
 }
 module_exit(wm8731_exit);
 
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 95190e9c0c14..cd7b806e8ad0 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -34,12 +34,6 @@
 #define WM8731_SYSCLK	0
 #define WM8731_DAI		0
 
-struct wm8731_setup_data {
-	int            spi;
-	int            i2c_bus;
-	unsigned short i2c_address;
-};
-
 extern struct snd_soc_dai wm8731_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8731;
 
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 0d41be33d572..eaa66915a324 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -317,19 +317,44 @@ static struct snd_soc_card snd_soc_corgi = {
 	.num_links = 1,
 };
 
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
-
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
 	.card = &snd_soc_corgi,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &corgi_wm8731_setup,
 };
 
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_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, "wm8731", 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 *corgi_snd_device;
 
 static int __init corgi_init(void)
@@ -340,6 +365,10 @@ static int __init corgi_init(void)
 	      machine_is_husky()))
 		return -ENODEV;
 
+	ret = wm8731_i2c_setup();
+	if (ret != 0)
+		return ret;
+
 	corgi_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!corgi_snd_device)
 		return -ENOMEM;
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 3a62d4354ef6..fd683a0b742d 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -283,17 +283,42 @@ static struct snd_soc_card snd_soc_poodle = {
 	.num_links = 1,
 };
 
-/* poodle audio private data */
-static struct wm8731_setup_data poodle_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_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, "wm8731", 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;
+}
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
 	.card = &snd_soc_poodle,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &poodle_wm8731_setup,
 };
 
 static struct platform_device *poodle_snd_device;
@@ -305,6 +330,10 @@ static int __init poodle_init(void)
 	if (!machine_is_poodle())
 		return -ENODEV;
 
+	ret = wm8731_i2c_setup();
+	if (ret != 0)
+		return ret;
+
 	locomo_gpio_set_dir(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_AMP_ON, 0);
 	/* should we mute HP at startup - burning power ?*/
-- 
cgit v1.2.3


From 59544d33ff3118f22a484d8be06cdf5cfc2fdca5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Feb 2009 11:36:44 +0000
Subject: ASoC: Remove version display from the WM8753 driver

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index dc6042c6424f..31ff337f8225 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -51,8 +51,6 @@
 
 #include "wm8753.h"
 
-#define WM8753_VERSION "0.16"
-
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -1778,8 +1776,6 @@ static int wm8753_probe(struct platform_device *pdev)
 	struct wm8753_priv *wm8753;
 	int ret = 0;
 
-	pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
-- 
cgit v1.2.3


From 93b760b7072ca6972c15c798e97af3f830d8bbba Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Feb 2009 12:44:40 +0000
Subject: ASoC: Implement SPI device unregistration for WM8731

Completely untested.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index a2c478e53d54..4191bdb803bf 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -655,12 +655,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
 	codec->hw_write = (hw_write_t)wm8731_spi_write;
 	codec->dev = &spi->dev;
 
+	spi->dev.driver_data = wm8731;
+
 	return wm8731_register(wm8731);
 }
 
 static int __devexit wm8731_spi_remove(struct spi_device *spi)
 {
-	/* FIXME: This isn't actually implemented... */
+	struct wm8731_priv *wm8731 = spi->dev.driver_data;
+
+	wm8731_unregister(wm8731);
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From 6bab83fd886564e96abcff62862732159535f600 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Wed, 18 Feb 2009 14:39:05 +0200
Subject: ASoC: TWL4030: Add digital loopback support

This patch adds the digital loopback/bypass support for twl4030 codec.

The digital loopback will let the digimic0 (routed in the TX1 capture path
inside of TWL4030) data to be routed back to the RX2 playback path
(I2S stereo). It can also route the analog capture date routed through the
TX1 back to RX2.

Effectively the digital loopback is routing the audio from the TX1 capture path
to the RX2 playback path.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 56 +++++++++++++++++++++++++++++++++++++++++-----
 1 file changed, 50 insertions(+), 6 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index c26854b398d3..535d8ce2c328 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -504,6 +504,25 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
 static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
 	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
 
+/* Digital bypass gain, 0 mutes the bypass */
+static const unsigned int twl4030_dapm_dbypass_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1),
+	4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
+};
+
+/* Digital bypass left (TX1L -> RX2L) */
+static const struct snd_kcontrol_new twl4030_dapm_dbypassl_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_ATX2ARXPGA, 3, 7, 0,
+			twl4030_dapm_dbypass_tlv);
+
+/* Digital bypass right (TX1R -> RX2R) */
+static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
+			twl4030_dapm_dbypass_tlv);
+
 static int micpath_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -608,12 +627,22 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
 	unsigned char reg;
 
 	reg = twl4030_read_reg_cache(w->codec, m->reg);
-	if (reg & (1 << m->shift))
-		twl4030->bypass_state |=
-			(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
-	else
-		twl4030->bypass_state &=
-			~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+
+	if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
+		/* Analog bypass */
+		if (reg & (1 << m->shift))
+			twl4030->bypass_state |=
+				(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+		else
+			twl4030->bypass_state &=
+				~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+	} else {
+		/* Digital bypass */
+		if (reg & (0x7 << m->shift))
+			twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+		else
+			twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+	}
 
 	if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
 		if (twl4030->bypass_state)
@@ -934,6 +963,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 			&twl4030_dapm_abypassl2_control,
 			bypass_event, SND_SOC_DAPM_POST_REG),
 
+	/* Digital bypasses */
+	SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassl_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassr_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+
 	SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
 			0, 0, NULL, 0),
 	SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
@@ -1118,6 +1155,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
 	{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
 
+	/* Digital bypass routes */
+	{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
+	{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+
+	{"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
+	{"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+
 };
 
 static int twl4030_add_widgets(struct snd_soc_codec *codec)
-- 
cgit v1.2.3


From 519cf2df5fb50c6d24412b2421ce2d1ff0346163 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Feb 2009 21:06:01 +0000
Subject: ASoC: Check for errors when writing WM8731 reset register

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 4191bdb803bf..9c9fc3b5a6c8 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -574,9 +574,14 @@ static int wm8731_register(struct wm8731_priv *wm8731)
 
 	memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg));
 
+	ret = wm8731_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
+	}
+
 	wm8731_dai.dev = codec->dev;
 
-	wm8731_reset(codec);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch the update bits */
-- 
cgit v1.2.3


From c6f2981170272cce2c192087a16dd74dbde25ed2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Feb 2009 21:25:40 +0000
Subject: ASoC: Add device init/exit annotations to new-style Wolfson CODEC
 drivers

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8350.c | 2 +-
 sound/soc/codecs/wm8731.c | 8 ++++----
 sound/soc/codecs/wm8900.c | 8 ++++----
 sound/soc/codecs/wm8903.c | 8 ++++----
 4 files changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index d3562788d42b..359e5cc86f34 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1574,7 +1574,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8350 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
 
-static int wm8350_codec_probe(struct platform_device *pdev)
+static __devinit int wm8350_codec_probe(struct platform_device *pdev)
 {
 	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
 	struct wm8350_data *priv;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9c9fc3b5a6c8..4cac3195bfa3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -686,8 +686,8 @@ static struct spi_driver wm8731_spi_driver = {
 #endif /* CONFIG_SPI_MASTER */
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static int wm8731_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8731_priv *wm8731;
 	struct snd_soc_codec *codec;
@@ -707,7 +707,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c,
 	return wm8731_register(wm8731);
 }
 
-static int wm8731_i2c_remove(struct i2c_client *client)
+static __devexit int wm8731_i2c_remove(struct i2c_client *client)
 {
 	struct wm8731_priv *wm8731 = i2c_get_clientdata(client);
 	wm8731_unregister(wm8731);
@@ -726,7 +726,7 @@ static struct i2c_driver wm8731_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8731_i2c_probe,
-	.remove =   wm8731_i2c_remove,
+	.remove =   __devexit_p(wm8731_i2c_remove),
 	.id_table = wm8731_i2c_id,
 };
 #endif
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 85c0f1bc6766..da5ca64f89bb 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1272,8 +1272,8 @@ static int wm8900_resume(struct platform_device *pdev)
 
 static struct snd_soc_codec *wm8900_codec;
 
-static int wm8900_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8900_priv *wm8900;
 	struct snd_soc_codec *codec;
@@ -1372,7 +1372,7 @@ err:
 	return ret;
 }
 
-static int wm8900_i2c_remove(struct i2c_client *client)
+static __devexit int wm8900_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_dai(&wm8900_dai);
 	snd_soc_unregister_codec(wm8900_codec);
@@ -1398,7 +1398,7 @@ static struct i2c_driver wm8900_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = wm8900_i2c_probe,
-	.remove = wm8900_i2c_remove,
+	.remove = __devexit_p(wm8900_i2c_remove),
 	.id_table = wm8900_i2c_id,
 };
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index d36b2b1edf19..c6fa8a71b4dd 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1562,8 +1562,8 @@ static int wm8903_resume(struct platform_device *pdev)
 
 static struct snd_soc_codec *wm8903_codec;
 
-static int wm8903_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8903_priv *wm8903;
 	struct snd_soc_codec *codec;
@@ -1669,7 +1669,7 @@ err:
 	return ret;
 }
 
-static int wm8903_i2c_remove(struct i2c_client *client)
+static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
 
@@ -1699,7 +1699,7 @@ static struct i2c_driver wm8903_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe    = wm8903_i2c_probe,
-	.remove   = wm8903_i2c_remove,
+	.remove   = __devexit_p(wm8903_i2c_remove),
 	.id_table = wm8903_i2c_id,
 };
 
-- 
cgit v1.2.3


From ce3bdaa8710c10eec5a6dae67aaf73088d0ced4f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 19 Feb 2009 14:29:49 +0000
Subject: ASoC: Disable WM8731 line bypass by default
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This avoids temporarily enabling the ouput stages during startup which
can cause audible effets in the output stages.

Reported-by: Fredrik Redgård <rik@svep.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8731.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 4cac3195bfa3..9e7ebcc2c491 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -594,6 +594,10 @@ static int wm8731_register(struct wm8731_priv *wm8731)
 	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
 	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
 
+	/* Disable bypass path by default */
+	reg = wm8731_read_reg_cache(codec, WM8731_APANA);
+	wm8731_write(codec, WM8731_APANA, reg & ~0x4);
+
 	wm8731_codec = codec;
 
 	ret = snd_soc_register_codec(codec);
-- 
cgit v1.2.3


From eeb1080b29a0fa00e426ba77eb96f3a157b335ab Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 22 Feb 2009 14:19:23 +0000
Subject: ASoC: Report I/O errors from WM8753 reset

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 31ff337f8225..180ec94ad8ae 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1561,7 +1561,11 @@ static int wm8753_init(struct snd_soc_device *socdev)
 
 	wm8753_set_dai_mode(codec, 0);
 
-	wm8753_reset(codec);
+	ret = wm8753_reset(codec);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8753: failed to reset device\n");
+		return ret;
+	}
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-- 
cgit v1.2.3


From 93e051d2771e6bf70e86b8265bfbf296a457d044 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 22 Feb 2009 14:24:00 +0000
Subject: ASoC: Only unregister drivers we registered for WM8753

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 180ec94ad8ae..93c22c4f0826 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1845,6 +1845,7 @@ static int wm8753_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
+	struct wm8753_setup_data *setup = socdev->codec_data;
 
 	if (codec->control_data)
 		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -1852,11 +1853,14 @@ static int wm8753_remove(struct platform_device *pdev)
 	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(&wm8753_i2c_driver);
+	if (setup->i2c_address) {
+		i2c_unregister_device(codec->control_data);
+		i2c_del_driver(&wm8753_i2c_driver);
+	}
 #endif
 #if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8753_spi_driver);
+	if (setup->spi)
+		spi_unregister_driver(&wm8753_spi_driver);
 #endif
 	kfree(codec->private_data);
 	kfree(codec);
-- 
cgit v1.2.3


From 8056d9bbb57207854462b6b0a3a75d172300cce5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 23 Jan 2009 14:44:54 +0000
Subject: ASoC: Improve WM9713 voice DAC shutdown procedure

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm9713.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 54db9c524988..a93aea5c1878 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -940,13 +940,14 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u16 status;
+	u16 status, rate;
 
 	/* Gracefully shut down the voice interface. */
 	status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-	ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
 	schedule_timeout_interruptible(msecs_to_jiffies(1));
-	ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
 	ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
-- 
cgit v1.2.3


From 69e169da5a69cc991d54bb4d54f236523145756c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 22 Feb 2009 14:39:03 +0000
Subject: ASoC: Shuffle WM8753 device registration code

This patch should be pure code motion, separating that out from the
functional changes to move to new style device registration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 209 +++++++++++++++++++++++-----------------------
 1 file changed, 105 insertions(+), 104 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 93c22c4f0826..4b426888f98d 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -51,6 +51,11 @@
 
 #include "wm8753.h"
 
+#ifdef CONFIG_SPI_MASTER
+static struct spi_driver wm8753_spi_driver;
+static int wm8753_spi_write(struct spi_device *spi, const char *data, int len);
+#endif
+
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -1626,53 +1631,7 @@ pcm_err:
 static struct snd_soc_device *wm8753_socdev;
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-/*
- * WM8753 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
-static int wm8753_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
-{
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8753\n");
-
-	return ret;
-}
-
-static int wm8753_i2c_remove(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
-	return 0;
-}
-
-static const struct i2c_device_id wm8753_i2c_id[] = {
-	{ "wm8753", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
-
-static struct i2c_driver wm8753_i2c_driver = {
-	.driver = {
-		.name = "WM8753 I2C Codec",
-		.owner = THIS_MODULE,
-	},
-	.probe =    wm8753_i2c_probe,
-	.remove =   wm8753_i2c_remove,
-	.id_table = wm8753_i2c_id,
-};
+static struct i2c_driver wm8753_i2c_driver;
 
 static int wm8753_add_i2c_device(struct platform_device *pdev,
 				 const struct wm8753_setup_data *setup)
@@ -1715,63 +1674,6 @@ err_driver:
 }
 #endif
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8753_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
-
-	codec->control_data = spi;
-
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8753\n");
-
-	return ret;
-}
-
-static int __devexit wm8753_spi_remove(struct spi_device *spi)
-{
-	return 0;
-}
-
-static struct spi_driver wm8753_spi_driver = {
-	.driver = {
-		.name	= "wm8753",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8753_spi_probe,
-	.remove		= __devexit_p(wm8753_spi_remove),
-};
-
-static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
-{
-	struct spi_transfer t;
-	struct spi_message m;
-	u8 msg[2];
-
-	if (len <= 0)
-		return 0;
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-
-	spi_message_init(&m);
-	memset(&t, 0, (sizeof t));
-
-	t.tx_buf = &msg[0];
-	t.len = len;
-
-	spi_message_add_tail(&t, &m);
-	spi_sync(spi, &m);
-
-	return len;
-}
-#endif
-
-
 static int wm8753_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1876,6 +1778,105 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static int wm8753_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct snd_soc_device *socdev = wm8753_socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int ret;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = wm8753_init(socdev);
+	if (ret < 0)
+		pr_err("failed to initialise WM8753\n");
+
+	return ret;
+}
+
+static int wm8753_i2c_remove(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	kfree(codec->reg_cache);
+	return 0;
+}
+
+static const struct i2c_device_id wm8753_i2c_id[] = {
+	{ "wm8753", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
+
+static struct i2c_driver wm8753_i2c_driver = {
+	.driver = {
+		.name = "WM8753 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8753_i2c_probe,
+	.remove =   wm8753_i2c_remove,
+	.id_table = wm8753_i2c_id,
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8753_socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8753_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8753\n");
+
+	return ret;
+}
+
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8753_spi_driver = {
+	.driver = {
+		.name	= "wm8753",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8753_spi_probe,
+	.remove		= __devexit_p(wm8753_spi_remove),
+};
+#endif
+
 static int __init wm8753_modinit(void)
 {
 	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
-- 
cgit v1.2.3


From c2bac1606a937d4700f8fdd8e051a4c49593c41b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 24 Feb 2009 23:33:12 +0000
Subject: ASoC: Convert WM8753 to register via normal device probe

The base support for the only in-tree user, the GTA01, is out of tree
and will be updated separately.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c          | 369 +++++++++++++++++--------------------
 sound/soc/codecs/wm8753.h          |   6 -
 sound/soc/s3c24xx/neo1973_wm8753.c |   6 -
 3 files changed, 171 insertions(+), 210 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 4b426888f98d..bc29558149e9 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -63,12 +63,6 @@ MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
 static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
 	unsigned int mode);
 
-/* codec private data */
-struct wm8753_priv {
-	unsigned int sysclk;
-	unsigned int pcmclk;
-};
-
 /*
  * wm8753 register cache
  * We can't read the WM8753 register space when we
@@ -93,6 +87,14 @@ static const u16 wm8753_reg[] = {
 	0x0000, 0x0000
 };
 
+/* codec private data */
+struct wm8753_priv {
+	unsigned int sysclk;
+	unsigned int pcmclk;
+	struct snd_soc_codec codec;
+	u16 reg_cache[ARRAY_SIZE(wm8753_reg)];
+};
+
 /*
  * read wm8753 register cache
  */
@@ -1542,36 +1544,24 @@ static int wm8753_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8753 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8753_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8753_codec;
+
+static int wm8753_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int reg, ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
 
-	codec->name = "WM8753";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8753_read_reg_cache;
-	codec->write = wm8753_write;
-	codec->set_bias_level = wm8753_set_bias_level;
-	codec->dai = wm8753_dai;
-	codec->num_dai = 2;
-	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
-	codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
+	if (!wm8753_codec) {
+		dev_err(&pdev->dev, "WM8753 codec not yet registered\n");
+		return -EINVAL;
+	}
 
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
+	socdev->card->codec = wm8753_codec;
+	codec = wm8753_codec;
 
 	wm8753_set_dai_mode(codec, 0);
 
-	ret = wm8753_reset(codec);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8753: failed to reset device\n");
-		return ret;
-	}
-
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -1579,36 +1569,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
 		goto pcm_err;
 	}
 
-	/* charge output caps */
-	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));
-
-	/* set the update bits */
-	reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
-	wm8753_write(codec, WM8753_LDAC, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_RDAC);
-	wm8753_write(codec, WM8753_RDAC, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_LADC);
-	wm8753_write(codec, WM8753_LADC, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_RADC);
-	wm8753_write(codec, WM8753_RADC, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V);
-	wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V);
-	wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V);
-	wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V);
-	wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_LINVOL);
-	wm8753_write(codec, WM8753_LINVOL, reg | 0x0100);
-	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
-	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
-
-	snd_soc_add_controls(codec, wm8753_snd_controls,
-				ARRAY_SIZE(wm8753_snd_controls));
+	wm8753_add_controls(codec);
 	wm8753_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1616,110 +1577,13 @@ static int wm8753_init(struct snd_soc_device *socdev)
 		goto card_err;
 	}
 
-	return ret;
+	return 0;
 
 card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
-	return ret;
-}
-
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8753_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static struct i2c_driver wm8753_i2c_driver;
-
-static int wm8753_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8753_setup_data *setup)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&wm8753_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, "wm8753", 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(&wm8753_i2c_driver);
-	return -ENODEV;
-}
-#endif
-
-static int wm8753_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8753_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8753_priv *wm8753;
-	int ret = 0;
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
-	if (wm8753 == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
-
-	codec->private_data = wm8753;
-	socdev->card->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8753_socdev = socdev;
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8753_add_i2c_device(pdev, setup);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8753_spi_write;
-		ret = spi_register_driver(&wm8753_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
-#endif
 
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
+pcm_err:
 	return ret;
 }
 
@@ -1746,26 +1610,9 @@ static int run_delayed_work(struct delayed_work *dwork)
 static int wm8753_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-	struct wm8753_setup_data *setup = socdev->codec_data;
 
-	if (codec->control_data)
-		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);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		i2c_unregister_device(codec->control_data);
-		i2c_del_driver(&wm8753_i2c_driver);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi)
-		spi_unregister_driver(&wm8753_spi_driver);
-#endif
-	kfree(codec->private_data);
-	kfree(codec);
 
 	return 0;
 }
@@ -1778,30 +1625,134 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
+static int wm8753_register(struct wm8753_priv *wm8753)
+{
+	int ret, i;
+	struct snd_soc_codec *codec = &wm8753->codec;
+	u16 reg;
+
+	if (wm8753_codec) {
+		dev_err(codec->dev, "Multiple WM8753 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 = "WM8753";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8753_read_reg_cache;
+	codec->write = wm8753_write;
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
+	codec->set_bias_level = wm8753_set_bias_level;
+	codec->dai = wm8753_dai;
+	codec->num_dai = 2;
+	codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache);
+	codec->reg_cache = &wm8753->reg_cache;
+	codec->private_data = wm8753;
+
+	memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache));
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+
+	ret = wm8753_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		goto err;
+	}
+
+	/* charge output caps */
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	schedule_delayed_work(&codec->delayed_work,
+			      msecs_to_jiffies(caps_charge));
+
+	/* set the update bits */
+	reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
+	wm8753_write(codec, WM8753_LDAC, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_RDAC);
+	wm8753_write(codec, WM8753_RDAC, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_LADC);
+	wm8753_write(codec, WM8753_LADC, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_RADC);
+	wm8753_write(codec, WM8753_RADC, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V);
+	wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V);
+	wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V);
+	wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V);
+	wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_LINVOL);
+	wm8753_write(codec, WM8753_LINVOL, reg | 0x0100);
+	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
+	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
+
+	wm8753_codec = codec;
+
+	for (i = 0; i < ARRAY_SIZE(wm8753_dai); i++)
+		wm8753_dai[i].dev = codec->dev;
+
+	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(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	run_delayed_work(&codec->delayed_work);
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8753);
+	return ret;
+}
+
+static void wm8753_unregister(struct wm8753_priv *wm8753)
+{
+	wm8753_set_bias_level(&wm8753->codec, SND_SOC_BIAS_OFF);
+	run_delayed_work(&wm8753->codec.delayed_work);
+	snd_soc_unregister_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai));
+	snd_soc_unregister_codec(&wm8753->codec);
+	kfree(wm8753);
+	wm8753_codec = NULL;
+}
+
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 
 static int wm8753_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
+	struct snd_soc_codec *codec;
+	struct wm8753_priv *wm8753;
 
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
+	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
+	if (wm8753 == NULL)
+		return -ENOMEM;
 
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8753\n");
+        codec = &wm8753->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+        codec->control_data = i2c;
+        i2c_set_clientdata(i2c, wm8753);
 
-	return ret;
+        codec->dev = &i2c->dev;
+
+	return wm8753_register(wm8753);
 }
 
 static int wm8753_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
-	return 0;
+        struct wm8753_priv *wm8753 = i2c_get_clientdata(client);
+        wm8753_unregister(wm8753);
+        return 0;
 }
 
 static const struct i2c_device_id wm8753_i2c_id[] = {
@@ -1812,7 +1763,7 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
 
 static struct i2c_driver wm8753_i2c_driver = {
 	.driver = {
-		.name = "WM8753 I2C Codec",
+		.name = "wm8753",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8753_i2c_probe,
@@ -1848,21 +1799,27 @@ static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
 
 static int __devinit wm8753_spi_probe(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
+	struct snd_soc_codec *codec;
+	struct wm8753_priv *wm8753;
+
+	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
+	if (wm8753 == NULL)
+		return -ENOMEM;
 
+	codec = &wm8753->codec;
 	codec->control_data = spi;
+	codec->hw_write = (hw_write_t)wm8753_spi_write;
+	codec->dev = &spi->dev;
 
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8753\n");
+	spi->dev.driver_data = wm8753;
 
-	return ret;
+	return wm8753_register(wm8753);
 }
 
 static int __devexit wm8753_spi_remove(struct spi_device *spi)
 {
+	struct wm8753_priv *wm8753 = spi->dev.driver_data;
+	wm8753_unregister(wm8753);
 	return 0;
 }
 
@@ -1879,13 +1836,29 @@ static struct spi_driver wm8753_spi_driver = {
 
 static int __init wm8753_modinit(void)
 {
-	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+	int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8753_i2c_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8753 I2C driver: %d\n", ret);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8753_spi_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8753 SPI driver: %d\n", ret);
+#endif
+	return 0;
 }
 module_init(wm8753_modinit);
 
 static void __exit wm8753_exit(void)
 {
-	snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8753_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8753_spi_driver);
+#endif
 }
 module_exit(wm8753_exit);
 
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index f55704ce931b..57b2ba244040 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -77,12 +77,6 @@
 #define WM8753_BIASCTL		0x3d
 #define WM8753_ADCTL2		0x3f
 
-struct wm8753_setup_data {
-	int spi;
-	int i2c_bus;
-	unsigned short i2c_address;
-};
-
 #define WM8753_PLL1			0
 #define WM8753_PLL2			1
 
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 45bb12e8ea44..286e11ad50ea 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -585,15 +585,9 @@ static struct snd_soc_card neo1973 = {
 	.num_links = ARRAY_SIZE(neo1973_dai),
 };
 
-static struct wm8753_setup_data neo1973_wm8753_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1a,
-};
-
 static struct snd_soc_device neo1973_snd_devdata = {
 	.card = &neo1973,
 	.codec_dev = &soc_codec_dev_wm8753,
-	.codec_data = &neo1973_wm8753_setup,
 };
 
 static int lm4857_i2c_probe(struct i2c_client *client,
-- 
cgit v1.2.3


From e611bd82441130991d7f4600dfd4632cebd417c5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 22 Feb 2009 20:04:41 +0000
Subject: ASoC: Only write back non-default registers when resuming WM8753

This will reduce the number of writes done on resume, allowing that to
complete faster (especially on systems with very slow I2C like the
current Samsung driver).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index bc29558149e9..2241204b5151 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1526,6 +1526,11 @@ static int wm8753_resume(struct platform_device *pdev)
 	for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
 		if (i + 1 == WM8753_RESET)
 			continue;
+
+		/* No point in writing hardware default values back */
+		if (cache[i] == wm8753_reg[i])
+			continue;
+
 		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
 		data[1] = cache[i] & 0x00ff;
 		codec->hw_write(codec->control_data, data, 2);
-- 
cgit v1.2.3


From 6d5643455ced9ee45a4aa7403fe8056d826bde85 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 26 Feb 2009 11:29:58 +0100
Subject: ASoC: wm8753 - Fix build error

sound/soc/codecs/wm8753.c: In function 'wm8753_probe':
sound/soc/codecs/wm8753.c:1577: error: implicit declaration of function 'wm8753_add_controls'

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/soc/codecs/wm8753.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 2241204b5151..7f353e935d71 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1574,7 +1574,8 @@ static int wm8753_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	wm8753_add_controls(codec);
+	snd_soc_add_controls(codec, wm8753_snd_controls,
+			     ARRAY_SIZE(wm8753_snd_controls));
 	wm8753_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-- 
cgit v1.2.3


From 4eae080dda3a563160be2f642cfbda27ffc42178 Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Wed, 25 Feb 2009 14:37:21 +0100
Subject: ASoC: Add cs4270 support for slave mode configurations

Added support for scenarios where the Cirrus CS4270 audio codec is slave
to the bitclk and lrclk. Mixed setups are unsupported.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 24 +++++++++++++++++++++++-
 1 file changed, 23 insertions(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index cd4a9ee38e46..339e0f6b0fe2 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -109,6 +109,7 @@ struct cs4270_private {
 	u8 reg_cache[CS4270_NUMREGS];
 	unsigned int mclk; /* Input frequency of the MCLK pin */
 	unsigned int mode; /* The mode (I2S or left-justified) */
+	unsigned int slave_mode;
 };
 
 /**
@@ -247,6 +248,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct cs4270_private *cs4270 = codec->private_data;
 	int ret = 0;
 
+	/* set DAI format */
 	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 	case SND_SOC_DAIFMT_LEFT_J:
@@ -257,6 +259,21 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		ret = -EINVAL;
 	}
 
+	/* set master/slave audio interface */
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs4270->slave_mode = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs4270->slave_mode = 0;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		/* unsupported - cs4270 can eigther be slave or master to
+		 * both the bitclk and the lrclk. */
+	default:
+		ret = -EINVAL;
+	}
+
 	return ret;
 }
 
@@ -399,7 +416,12 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 
 	reg = snd_soc_read(codec, CS4270_MODE);
 	reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK);
-	reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk;
+	reg |= cs4270_mode_ratios[i].mclk;
+
+	if (cs4270->slave_mode)
+		reg |= CS4270_MODE_SLAVE;
+	else
+		reg |= cs4270_mode_ratios[i].speed_mode;
 
 	ret = snd_soc_write(codec, CS4270_MODE, reg);
 	if (ret < 0) {
-- 
cgit v1.2.3


From ff09d49ad0176a5f52a398c137a7ff5f669d6be4 Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Sat, 28 Feb 2009 13:21:03 +0100
Subject: ASoC: fix typo and removed unneeded switch case for cs4270

This removes a misspelled comment and got rid of superfluous switch
case.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 339e0f6b0fe2..f86f33cc1798 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -267,10 +267,8 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case SND_SOC_DAIFMT_CBM_CFM:
 		cs4270->slave_mode = 0;
 		break;
-	case SND_SOC_DAIFMT_CBM_CFS:
-		/* unsupported - cs4270 can eigther be slave or master to
-		 * both the bitclk and the lrclk. */
 	default:
+		/* all other modes are unsupported by the hardware */
 		ret = -EINVAL;
 	}
 
-- 
cgit v1.2.3


From a3c7729e6c5d41bbeb3e13befbcf8e4ef76e55dc Mon Sep 17 00:00:00 2001
From: Philipp Zabel <philipp.zabel@gmail.com>
Date: Tue, 3 Mar 2009 16:10:53 +0100
Subject: ASoC: Remove version display from the UDA1380 driver

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/uda1380.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 5242b8156b38..8686a554536e 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -35,8 +35,6 @@
 
 #include "uda1380.h"
 
-#define UDA1380_VERSION "0.6"
-
 /*
  * uda1380 register cache
  */
@@ -826,8 +824,6 @@ static int uda1380_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret;
 
-	pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
-- 
cgit v1.2.3


From ef9e5e5c31cb2c6254760611289ac13e4e41b964 Mon Sep 17 00:00:00 2001
From: Philipp Zabel <philipp.zabel@gmail.com>
Date: Tue, 3 Mar 2009 16:10:54 +0100
Subject: ASoC: UDA1380: change decimator/interpolator register handling

If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).

* Queue work in the alsa PCM_START .trigger to flush registers
  as soon as the link is running. This replaces the .prepare
  and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
  its alsa control to avoid confusion.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/uda1380.c | 102 +++++++++++++++++++++------------------------
 1 file changed, 48 insertions(+), 54 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 8686a554536e..1c9d2a75addb 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -25,6 +25,7 @@
 #include <linux/ioctl.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
+#include <linux/workqueue.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/initval.h>
@@ -35,6 +36,9 @@
 
 #include "uda1380.h"
 
+static struct work_struct uda1380_work;
+static struct snd_soc_codec *uda1380_codec;
+
 /*
  * uda1380 register cache
  */
@@ -50,6 +54,8 @@ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
 	0x0000, 0x8000, 0x0002, 0x0000,
 };
 
+static unsigned long uda1380_cache_dirty;
+
 /*
  * read uda1380 register cache
  */
@@ -71,8 +77,11 @@ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
 	u16 reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
+
 	if (reg >= UDA1380_CACHEREGNUM)
 		return;
+	if ((reg >= 0x10) && (cache[reg] != value))
+		set_bit(reg - 0x10, &uda1380_cache_dirty);
 	cache[reg] = value;
 }
 
@@ -111,6 +120,8 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
 					(data[0]<<8) | data[1]);
 			return -EIO;
 		}
+		if (reg >= 0x10)
+			clear_bit(reg - 0x10, &uda1380_cache_dirty);
 		return 0;
 	} else
 		return -EIO;
@@ -118,6 +129,20 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
 
 #define uda1380_reset(c)	uda1380_write(c, UDA1380_RESET, 0)
 
+static void uda1380_flush_work(struct work_struct *work)
+{
+	int bit, reg;
+
+	for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+		reg = 0x10 + bit;
+		pr_debug("uda1380: flush reg %x val %x:\n", reg,
+				uda1380_read_reg_cache(uda1380_codec, reg));
+		uda1380_write(uda1380_codec, reg,
+				uda1380_read_reg_cache(uda1380_codec, reg));
+		clear_bit(bit, &uda1380_cache_dirty);
+	}
+}
+
 /* declarations of ALSA reg_elem_REAL controls */
 static const char *uda1380_deemp[] = {
 	"None",
@@ -252,7 +277,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
 	SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0),	/* DA_POL_INV */
 	SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum),				/* SEL_NS */
 	SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum),		/* MIX_POS, MIX */
-	SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0),			/* SILENCE, force DAC output to silence */
 	SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0),		/* SDET_ON */
 	SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum),		/* SD_VALUE */
 	SOC_ENUM("Oversampling Input", uda1380_os_enum),			/* OS */
@@ -438,41 +462,28 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-/*
- * Flush reg cache
- * We can only write the interpolator and decimator registers
- * when the DAI is being clocked by the CPU DAI. It's up to the
- * machine and cpu DAI driver to do this before we are called.
- */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
+static int uda1380_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;
-	int reg, reg_start, reg_end, clk;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		reg_start = UDA1380_MVOL;
-		reg_end = UDA1380_MIXER;
-	} else {
-		reg_start = UDA1380_DEC;
-		reg_end = UDA1380_AGC;
-	}
-
-	/* FIXME disable DAC_CLK */
-	clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-	uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
-
-	for (reg = reg_start; reg <= reg_end; reg++) {
-		pr_debug("uda1380: flush reg %x val %x:", reg,
-				uda1380_read_reg_cache(codec, reg));
-		uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+	int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		uda1380_write_reg_cache(codec, UDA1380_MIXER,
+					mixer & ~R14_SILENCE);
+		schedule_work(&uda1380_work);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		uda1380_write_reg_cache(codec, UDA1380_MIXER,
+					mixer | R14_SILENCE);
+		schedule_work(&uda1380_work);
+		break;
 	}
-
-	/* FIXME restore DAC_CLK */
-	uda1380_write(codec, UDA1380_CLK, clk);
-
 	return 0;
 }
 
@@ -538,24 +549,6 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
 	uda1380_write(codec, UDA1380_CLK, clk);
 }
 
-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;
-
-	/* FIXME: mute(codec,0) is called when the magician clock is already
-	 * set to WSPLL, but for some unknown reason writing to interpolator
-	 * registers works only when clocked by SYSCLK */
-	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-	uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
-	if (mute)
-		uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
-	else
-		uda1380_write(codec, UDA1380_DEEMP, mute_reg);
-	uda1380_write(codec, UDA1380_CLK, clk);
-	return 0;
-}
-
 static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
@@ -597,10 +590,9 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
+		.trigger = uda1380_trigger,
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt_both,
 	},
 },
@@ -614,10 +606,9 @@ struct snd_soc_dai uda1380_dai[] = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.ops = {
+		.trigger = uda1380_trigger,
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt_playback,
 	},
 },
@@ -631,9 +622,9 @@ struct snd_soc_dai uda1380_dai[] = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.ops = {
+		.trigger = uda1380_trigger,
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
 		.set_fmt = uda1380_set_dai_fmt_capture,
 	},
 },
@@ -692,6 +683,9 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	codec->reg_cache_step = 1;
 	uda1380_reset(codec);
 
+	uda1380_codec = codec;
+	INIT_WORK(&uda1380_work, uda1380_flush_work);
+
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-- 
cgit v1.2.3


From 5f2a9384a9291d898b4bf85c4fbf497eef582977 Mon Sep 17 00:00:00 2001
From: Philipp Zabel <philipp.zabel@gmail.com>
Date: Tue, 3 Mar 2009 16:10:52 +0100
Subject: ASoC: UDA1380: DATAI is slave only

Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/uda1380.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 1c9d2a75addb..1b10f488328c 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -399,8 +399,9 @@ static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai,
 		iface |= R01_SFORI_MSB | R01_SFORO_MSB;
 	}
 
-	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
-		iface |= R01_SIM;
+	/* DATAI is slave only, so in single-link mode, this has to be slave */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
 
 	uda1380_write(codec, UDA1380_IFACE, iface);
 
@@ -428,6 +429,10 @@ static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai,
 		iface |= R01_SFORI_MSB;
 	}
 
+	/* DATAI is slave only, so this has to be slave */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
 	uda1380_write(codec, UDA1380_IFACE, iface);
 
 	return 0;
-- 
cgit v1.2.3


From 86027ae78c9294bb450b76eec28cfb431a8fb3ee Mon Sep 17 00:00:00 2001
From: Jonas Andersson <jonas@microbit.se>
Date: Wed, 4 Mar 2009 08:24:26 +0100
Subject: ASoC: wm8510 pll settings

When setting WM8510_MCLKDIV the pll was turned off.

When setting pll frequency you got twice the expected freq, because
the  code calculated  with postscaler of 8,  but  the hardware divide by 4.

Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/atmel/playpaq_wm8510.c | 24 ++++++++++++------------
 sound/soc/codecs/wm8510.c        |  4 ++--
 2 files changed, 14 insertions(+), 14 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 43dd8cee83c6..70657534e6b1 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -164,38 +164,38 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
 	 */
 	switch (params_rate(params)) {
 	case 48000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_1;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_2;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 44100:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_1;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_2;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 22050:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_2;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_4;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 16000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_3;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_6;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 11025:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_4;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_8;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 8000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_6;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_12;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index f01078cfbd72..6d4ef71e9195 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -336,7 +336,7 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
 		return 0;
 	}
 
-	pll_factors(freq_out*8, freq_in);
+	pll_factors(freq_out*4, freq_in);
 
 	wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
 	wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
@@ -367,7 +367,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		wm8510_write(codec, WM8510_GPIO, reg | div);
 		break;
 	case WM8510_MCLKDIV:
-		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f;
 		wm8510_write(codec, WM8510_CLOCK, reg | div);
 		break;
 	case WM8510_ADCCLK:
-- 
cgit v1.2.3


From 6335d05548eece40092000aa91b64a50310d69d5 Mon Sep 17 00:00:00 2001
From: Eric Miao <eric.y.miao@gmail.com>
Date: Tue, 3 Mar 2009 09:41:00 +0800
Subject: ASoC: make ops a pointer in 'struct snd_soc_dai'

Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.

The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.

Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h          |   2 +-
 sound/soc/atmel/atmel_ssc_dai.c  |  33 +++++--------
 sound/soc/au1x/psc-ac97.c        |  10 ++--
 sound/soc/au1x/psc-i2s.c         |  12 +++--
 sound/soc/blackfin/bf5xx-i2s.c   |  14 +++---
 sound/soc/codecs/ac97.c          |   7 ++-
 sound/soc/codecs/ak4535.c        |  14 +++---
 sound/soc/codecs/cs4270.c        |  14 +++---
 sound/soc/codecs/ssm2602.c       |  20 ++++----
 sound/soc/codecs/tlv320aic23.c   |  18 ++++---
 sound/soc/codecs/tlv320aic26.c   |  14 +++---
 sound/soc/codecs/tlv320aic3x.c   |  14 +++---
 sound/soc/codecs/uda134x.c       |  18 ++++---
 sound/soc/codecs/uda1380.c       |  46 ++++++++++--------
 sound/soc/codecs/wm8350.c        |  20 ++++----
 sound/soc/codecs/wm8510.c        |  16 +++---
 sound/soc/codecs/wm8580.c        |  30 +++++++-----
 sound/soc/codecs/wm8728.c        |  12 +++--
 sound/soc/codecs/wm8731.c        |  18 ++++---
 sound/soc/codecs/wm8750.c        |  14 +++---
 sound/soc/codecs/wm8753.c        |  90 +++++++++++++++++++---------------
 sound/soc/codecs/wm8900.c        |  16 +++---
 sound/soc/codecs/wm8903.c        |  18 ++++---
 sound/soc/codecs/wm8971.c        |  14 +++---
 sound/soc/codecs/wm8990.c        |  18 ++++---
 sound/soc/codecs/wm9705.c        |   8 +--
 sound/soc/codecs/wm9712.c        |  14 ++++--
 sound/soc/codecs/wm9713.c        |  40 +++++++++------
 sound/soc/davinci/davinci-i2s.c  |  14 +++---
 sound/soc/fsl/fsl_ssi.c          |  18 ++++---
 sound/soc/fsl/mpc5200_psc_i2s.c  |  20 ++++----
 sound/soc/omap/omap-mcbsp.c      |  20 ++++----
 sound/soc/pxa/pxa-ssp.c          |  65 +++++++------------------
 sound/soc/pxa/pxa2xx-ac97.c      |  13 ++---
 sound/soc/pxa/pxa2xx-i2s.c       |  18 ++++---
 sound/soc/s3c24xx/s3c2412-i2s.c  |  16 +++---
 sound/soc/s3c24xx/s3c2443-ac97.c |  13 ++---
 sound/soc/s3c24xx/s3c24xx-i2s.c  |  16 +++---
 sound/soc/sh/ssi.c               |  30 +++++-------
 sound/soc/soc-core.c             | 102 +++++++++++++++++++++------------------
 40 files changed, 481 insertions(+), 428 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 24247f763608..13676472ddfc 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -203,7 +203,7 @@ struct snd_soc_dai {
 	int (*resume)(struct snd_soc_dai *dai);
 
 	/* ops */
-	struct snd_soc_dai_ops ops;
+	struct snd_soc_dai_ops *ops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index ff0054b76502..e588e63f18d2 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
 			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops atmel_ssc_dai_ops = {
+	.startup	= atmel_ssc_startup,
+	.shutdown	= atmel_ssc_shutdown,
+	.prepare	= atmel_ssc_prepare,
+	.hw_params	= atmel_ssc_hw_params,
+	.set_fmt	= atmel_ssc_set_dai_fmt,
+	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
+};
+
 struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 	{	.name = "atmel-ssc0",
 		.id = 0,
@@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[0],
 	},
 #if NUM_SSC_DEVICES == 3
@@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[1],
 	},
 	{	.name = "atmel-ssc2",
@@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[2],
 	},
 #endif
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index f0e30aec7f23..479d7bdf1865 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
 	.ac97_control		= 1,
@@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.trigger	= au1xpsc_ac97_trigger,
-		.hw_params	= au1xpsc_ac97_hw_params,
-	},
+	.ops = &au1xpsc_ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index f916de4400ed..bb589327ee32 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.trigger	= au1xpsc_i2s_trigger,
+	.hw_params	= au1xpsc_i2s_hw_params,
+	.set_fmt	= au1xpsc_i2s_set_fmt,
+};
+
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
 	.probe			= au1xpsc_i2s_probe,
@@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 		.channels_min	= 2,
 		.channels_max	= 8,	/* 2 without external help */
 	},
-	.ops = {
-		.trigger	= au1xpsc_i2s_trigger,
-		.hw_params	= au1xpsc_i2s_hw_params,
-		.set_fmt	= au1xpsc_i2s_set_fmt,
-	},
+	.ops = &au1xpsc_i2s_dai_ops,
 };
 EXPORT_SYMBOL(au1xpsc_i2s_dai);
 
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index d1d95d2393fe..964824419678 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
 	SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
+	.startup	= bf5xx_i2s_startup,
+	.shutdown	= bf5xx_i2s_shutdown,
+	.hw_params	= bf5xx_i2s_hw_params,
+	.set_fmt	= bf5xx_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
@@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
-	.ops = {
-		.startup   = bf5xx_i2s_startup,
-		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,
-		.set_fmt = bf5xx_i2s_set_dai_fmt,
-	},
+	.ops = &bf5xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 11f84b6e5cb8..b0d4af145b87 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops ac97_dai_ops = {
+	.prepare	= ac97_prepare,
+};
+
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
 	.ac97_control = 1,
@@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = {
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
 
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index d56e6bb1fedb..1f63d387a2f4 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -421,6 +421,13 @@ 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)
 
+static struct snd_soc_dai_ops ak4535_dai_ops = {
+	.hw_params	= ak4535_hw_params,
+	.set_fmt	= ak4535_set_dai_fmt,
+	.digital_mute	= ak4535_mute,
+	.set_sysclk	= ak4535_set_dai_sysclk,
+};
+
 struct snd_soc_dai ak4535_dai = {
 	.name = "AK4535",
 	.playback = {
@@ -435,12 +442,7 @@ struct snd_soc_dai ak4535_dai = {
 		.channels_max = 2,
 		.rates = AK4535_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = ak4535_hw_params,
-		.set_fmt = ak4535_set_dai_fmt,
-		.digital_mute = ak4535_mute,
-		.set_sysclk = ak4535_set_dai_sysclk,
-	},
+	.ops = &ak4535_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ak4535_dai);
 
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f86f33cc1798..7ae3d6520e3f 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -503,6 +503,13 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
  */
 static struct snd_soc_codec *cs4270_codec;
 
+static struct snd_soc_dai_ops cs4270_dai_ops = {
+	.hw_params	= cs4270_hw_params,
+	.set_sysclk	= cs4270_set_dai_sysclk,
+	.set_fmt	= cs4270_set_dai_fmt,
+	.digital_mute	= cs4270_mute,
+};
+
 struct snd_soc_dai cs4270_dai = {
 	.name = "cs4270",
 	.playback = {
@@ -519,12 +526,7 @@ struct snd_soc_dai cs4270_dai = {
 		.rates = 0,
 		.formats = CS4270_FORMATS,
 	},
-	.ops = {
-		.hw_params = cs4270_hw_params,
-		.set_sysclk = cs4270_set_dai_sysclk,
-		.set_fmt = cs4270_set_dai_fmt,
-		.digital_mute = cs4270_mute,
-	},
+	.ops = &cs4270_dai_ops,
 };
 EXPORT_SYMBOL_GPL(cs4270_dai);
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 58e225dadc7e..87f606c76822 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -506,6 +506,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops ssm2602_dai_ops = {
+	.startup	= ssm2602_startup,
+	.prepare	= ssm2602_pcm_prepare,
+	.hw_params	= ssm2602_hw_params,
+	.shutdown	= ssm2602_shutdown,
+	.digital_mute	= ssm2602_mute,
+	.set_sysclk	= ssm2602_set_dai_sysclk,
+	.set_fmt	= ssm2602_set_dai_fmt,
+};
+
 struct snd_soc_dai ssm2602_dai = {
 	.name = "SSM2602",
 	.playback = {
@@ -520,15 +530,7 @@ struct snd_soc_dai ssm2602_dai = {
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
 		.formats = SSM2602_FORMATS,},
-	.ops = {
-		.startup = ssm2602_startup,
-		.prepare = ssm2602_pcm_prepare,
-		.hw_params = ssm2602_hw_params,
-		.shutdown = ssm2602_shutdown,
-		.digital_mute = ssm2602_mute,
-		.set_sysclk = ssm2602_set_dai_sysclk,
-		.set_fmt = ssm2602_set_dai_fmt,
-	}
+	.ops = &ssm2602_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ssm2602_dai);
 
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 8b20c360adf5..c3f4afb5d017 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -580,6 +580,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 #define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops tlv320aic23_dai_ops = {
+	.prepare	= tlv320aic23_pcm_prepare,
+	.hw_params	= tlv320aic23_hw_params,
+	.shutdown	= tlv320aic23_shutdown,
+	.digital_mute	= tlv320aic23_mute,
+	.set_fmt	= tlv320aic23_set_dai_fmt,
+	.set_sysclk	= tlv320aic23_set_dai_sysclk,
+};
+
 struct snd_soc_dai tlv320aic23_dai = {
 	.name = "tlv320aic23",
 	.playback = {
@@ -594,14 +603,7 @@ struct snd_soc_dai tlv320aic23_dai = {
 		    .channels_max = 2,
 		    .rates = AIC23_RATES,
 		    .formats = AIC23_FORMATS,},
-	.ops = {
-		.prepare = tlv320aic23_pcm_prepare,
-		.hw_params = tlv320aic23_hw_params,
-		.shutdown = tlv320aic23_shutdown,
-		.digital_mute = tlv320aic23_mute,
-		.set_fmt = tlv320aic23_set_dai_fmt,
-		.set_sysclk = tlv320aic23_set_dai_sysclk,
-	}
+	.ops = &tlv320aic23_dai_ops,
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 229e464cf713..a7f333fc579e 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 #define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
 			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
 
+static struct snd_soc_dai_ops aic26_dai_ops = {
+	.hw_params	= aic26_hw_params,
+	.digital_mute	= aic26_mute,
+	.set_sysclk	= aic26_set_sysclk,
+	.set_fmt	= aic26_set_fmt,
+};
+
 struct snd_soc_dai aic26_dai = {
 	.name = "tlv320aic26",
 	.playback = {
@@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = {
 		.rates = AIC26_RATES,
 		.formats = AIC26_FORMATS,
 	},
-	.ops = {
-		.hw_params = aic26_hw_params,
-		.digital_mute = aic26_mute,
-		.set_sysclk = aic26_set_sysclk,
-		.set_fmt = aic26_set_fmt,
-	},
+	.ops = &aic26_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic26_dai);
 
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index d638e3f0728b..ab099f482487 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1088,6 +1088,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed);
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops aic3x_dai_ops = {
+	.hw_params	= aic3x_hw_params,
+	.digital_mute	= aic3x_mute,
+	.set_sysclk	= aic3x_set_dai_sysclk,
+	.set_fmt	= aic3x_set_dai_fmt,
+};
+
 struct snd_soc_dai aic3x_dai = {
 	.name = "tlv320aic3x",
 	.playback = {
@@ -1102,12 +1109,7 @@ struct snd_soc_dai aic3x_dai = {
 		.channels_max = 2,
 		.rates = AIC3X_RATES,
 		.formats = AIC3X_FORMATS,},
-	.ops = {
-		.hw_params = aic3x_hw_params,
-		.digital_mute = aic3x_mute,
-		.set_sysclk = aic3x_set_dai_sysclk,
-		.set_fmt = aic3x_set_dai_fmt,
-	}
+	.ops = &aic3x_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic3x_dai);
 
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 661599295ca7..ddefb8f80145 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -431,6 +431,15 @@ 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,
+	.hw_params	= uda134x_hw_params,
+	.digital_mute	= uda134x_mute,
+	.set_sysclk	= uda134x_set_dai_sysclk,
+	.set_fmt	= uda134x_set_dai_fmt,
+};
+
 struct snd_soc_dai uda134x_dai = {
 	.name = "UDA134X",
 	/* playback capabilities */
@@ -450,14 +459,7 @@ struct snd_soc_dai uda134x_dai = {
 		.formats = UDA134X_FORMATS,
 	},
 	/* pcm operations */
-	.ops = {
-		.startup = uda134x_startup,
-		.shutdown = uda134x_shutdown,
-		.hw_params = uda134x_hw_params,
-		.digital_mute = uda134x_mute,
-		.set_sysclk = uda134x_set_dai_sysclk,
-		.set_fmt = uda134x_set_dai_fmt,
-	}
+	.ops = &uda134x_dai_ops,
 };
 EXPORT_SYMBOL(uda134x_dai);
 
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 5242b8156b38..cafa7684c0e7 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -583,6 +583,29 @@ 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)
 
+static struct snd_soc_dai_ops uda1380_dai_ops = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.prepare	= uda1380_pcm_prepare,
+	.digital_mute	= uda1380_mute,
+	.set_fmt	= uda1380_set_dai_fmt_both,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_playback = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.prepare	= uda1380_pcm_prepare,
+	.digital_mute	= uda1380_mute,
+	.set_fmt	= uda1380_set_dai_fmt_playback,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_capture = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.prepare	= uda1380_pcm_prepare,
+	.set_fmt	= uda1380_set_dai_fmt_capture,
+};
+
 struct snd_soc_dai uda1380_dai[] = {
 {
 	.name = "UDA1380",
@@ -598,13 +621,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.channels_max = 2,
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt_both,
-	},
+	.ops = &uda1380_dai_ops,
 },
 { /* playback only - dual interface */
 	.name = "UDA1380",
@@ -615,13 +632,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt_playback,
-	},
+	.ops = &uda1380_dai_ops_playback,
 },
 { /* capture only - dual interface*/
 	.name = "UDA1380",
@@ -632,12 +643,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.set_fmt = uda1380_set_dai_fmt_capture,
-	},
+	.ops = &uda1380_dai_ops_capture,
 },
 };
 EXPORT_SYMBOL_GPL(uda1380_dai);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 359e5cc86f34..3b1d0993bed9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1538,6 +1538,16 @@ static int wm8350_remove(struct platform_device *pdev)
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8350_dai_ops = {
+	 .hw_params	= wm8350_pcm_hw_params,
+	 .digital_mute	= wm8350_mute,
+	 .trigger	= wm8350_pcm_trigger,
+	 .set_fmt	= wm8350_set_dai_fmt,
+	 .set_sysclk	= wm8350_set_dai_sysclk,
+	 .set_pll	= wm8350_set_fll,
+	 .set_clkdiv	= wm8350_set_clkdiv,
+};
+
 struct snd_soc_dai wm8350_dai = {
 	.name = "WM8350",
 	.playback = {
@@ -1554,15 +1564,7 @@ struct snd_soc_dai wm8350_dai = {
 		 .rates = WM8350_RATES,
 		 .formats = WM8350_FORMATS,
 	 },
-	.ops = {
-		 .hw_params = wm8350_pcm_hw_params,
-		 .digital_mute = wm8350_mute,
-		 .trigger = wm8350_pcm_trigger,
-		 .set_fmt = wm8350_set_dai_fmt,
-		 .set_sysclk = wm8350_set_dai_sysclk,
-		 .set_pll = wm8350_set_fll,
-		 .set_clkdiv = wm8350_set_clkdiv,
-	 },
+	.ops = &wm8350_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8350_dai);
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index f01078cfbd72..cc975a62fa5c 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -554,6 +554,14 @@ 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)
 
+static struct snd_soc_dai_ops wm8510_dai_ops = {
+	.hw_params	= wm8510_pcm_hw_params,
+	.digital_mute	= wm8510_mute,
+	.set_fmt	= wm8510_set_dai_fmt,
+	.set_clkdiv	= wm8510_set_dai_clkdiv,
+	.set_pll	= wm8510_set_dai_pll,
+};
+
 struct snd_soc_dai wm8510_dai = {
 	.name = "WM8510 HiFi",
 	.playback = {
@@ -568,13 +576,7 @@ struct snd_soc_dai wm8510_dai = {
 		.channels_max = 2,
 		.rates = WM8510_RATES,
 		.formats = WM8510_FORMATS,},
-	.ops = {
-		.hw_params = wm8510_pcm_hw_params,
-		.digital_mute = wm8510_mute,
-		.set_fmt = wm8510_set_dai_fmt,
-		.set_clkdiv = wm8510_set_dai_clkdiv,
-		.set_pll = wm8510_set_dai_pll,
-	},
+	.ops = &wm8510_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8510_dai);
 
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d3c51ba5e6f9..ee0af23a1acc 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -771,6 +771,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops wm8580_dai_ops_playback = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+	.digital_mute	= wm8580_digital_mute,
+};
+
+static struct snd_soc_dai_ops wm8580_dai_ops_capture = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+};
+
 struct snd_soc_dai wm8580_dai[] = {
 	{
 		.name = "WM8580 PAIFRX",
@@ -782,13 +797,7 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-			 .digital_mute = wm8580_digital_mute,
-		 },
+		.ops = &wm8580_dai_ops_playback,
 	},
 	{
 		.name = "WM8580 PAIFTX",
@@ -800,12 +809,7 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-		 },
+		.ops = &wm8580_dai_ops_capture,
 	},
 };
 EXPORT_SYMBOL_GPL(wm8580_dai);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index f8363b308895..e7ff2121ede9 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -244,6 +244,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
 #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8728_dai_ops = {
+	.hw_params	= wm8728_hw_params,
+	.digital_mute	= wm8728_mute,
+	.set_fmt	= wm8728_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8728_dai = {
 	.name = "WM8728",
 	.playback = {
@@ -253,11 +259,7 @@ struct snd_soc_dai wm8728_dai = {
 		.rates = WM8728_RATES,
 		.formats = WM8728_FORMATS,
 	},
-	.ops = {
-		 .hw_params = wm8728_hw_params,
-		 .digital_mute = wm8728_mute,
-		 .set_fmt = wm8728_set_dai_fmt,
-	}
+	.ops = &wm8728_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8728_dai);
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9e7ebcc2c491..e043e3f60008 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -433,6 +433,15 @@ 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)
 
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8731_dai = {
 	.name = "WM8731",
 	.playback = {
@@ -447,14 +456,7 @@ struct snd_soc_dai wm8731_dai = {
 		.channels_max = 2,
 		.rates = WM8731_RATES,
 		.formats = WM8731_FORMATS,},
-	.ops = {
-		.prepare = wm8731_pcm_prepare,
-		.hw_params = wm8731_hw_params,
-		.shutdown = wm8731_shutdown,
-		.digital_mute = wm8731_mute,
-		.set_sysclk = wm8731_set_dai_sysclk,
-		.set_fmt = wm8731_set_dai_fmt,
-	}
+	.ops = &wm8731_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8731_dai);
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 96afb86addc6..b64509b01a49 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -679,6 +679,13 @@ 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)
 
+static struct snd_soc_dai_ops wm8750_dai_ops = {
+	.hw_params	= wm8750_pcm_hw_params,
+	.digital_mute	= wm8750_mute,
+	.set_fmt	= wm8750_set_dai_fmt,
+	.set_sysclk	= wm8750_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8750_dai = {
 	.name = "WM8750",
 	.playback = {
@@ -693,12 +700,7 @@ struct snd_soc_dai wm8750_dai = {
 		.channels_max = 2,
 		.rates = WM8750_RATES,
 		.formats = WM8750_FORMATS,},
-	.ops = {
-		.hw_params = wm8750_pcm_hw_params,
-		.digital_mute = wm8750_mute,
-		.set_fmt = wm8750_set_dai_fmt,
-		.set_sysclk = wm8750_set_dai_sysclk,
-	},
+	.ops = &wm8750_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8750_dai);
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 7f353e935d71..cc6e57f9acf8 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1306,6 +1306,51 @@ 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 struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1h_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1v_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode2_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
 static const struct snd_soc_dai wm8753_all_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "WM8753 HiFi",
@@ -1322,14 +1367,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1h_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode1,
 },
 /* DAI Voice mode 1 */
 {	.name = "WM8753 Voice",
@@ -1346,14 +1384,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1v_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode1,
 },
 /* DAI HiFi mode 2 - dummy */
 {	.name = "WM8753 HiFi",
@@ -1374,14 +1405,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode2_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode2,
 },
 /* DAI HiFi mode 3 */
 {	.name = "WM8753 HiFi",
@@ -1398,14 +1422,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode3,
 },
 /* DAI Voice mode 3 - dummy */
 {	.name = "WM8753 Voice",
@@ -1426,14 +1443,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode4,
 },
 /* DAI Voice mode 4 - dummy */
 {	.name = "WM8753 Voice",
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index da5ca64f89bb..46c5ea1ff921 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1088,6 +1088,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops wm8900_dai_ops = {
+	.hw_params	= wm8900_hw_params,
+	.set_clkdiv	= wm8900_set_dai_clkdiv,
+	.set_pll	= wm8900_set_dai_pll,
+	.set_fmt	= wm8900_set_dai_fmt,
+	.digital_mute	= wm8900_digital_mute,
+};
+
 struct snd_soc_dai wm8900_dai = {
 	.name = "WM8900 HiFi",
 	.playback = {
@@ -1104,13 +1112,7 @@ struct snd_soc_dai wm8900_dai = {
 		.rates = WM8900_RATES,
 		.formats = WM8900_PCM_FORMATS,
 	 },
-	.ops = {
-		.hw_params = wm8900_hw_params,
-		 .set_clkdiv = wm8900_set_dai_clkdiv,
-		 .set_pll = wm8900_set_dai_pll,
-		 .set_fmt = wm8900_set_dai_fmt,
-		 .digital_mute = wm8900_digital_mute,
-	 },
+	.ops = &wm8900_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8900_dai);
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index c6fa8a71b4dd..8cf571f1a803 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1497,6 +1497,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8903_dai_ops = {
+	.startup	= wm8903_startup,
+	.shutdown	= wm8903_shutdown,
+	.hw_params	= wm8903_hw_params,
+	.digital_mute	= wm8903_digital_mute,
+	.set_fmt	= wm8903_set_dai_fmt,
+	.set_sysclk	= wm8903_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8903_dai = {
 	.name = "WM8903",
 	.playback = {
@@ -1513,14 +1522,7 @@ struct snd_soc_dai wm8903_dai = {
 		 .rates = WM8903_CAPTURE_RATES,
 		 .formats = WM8903_FORMATS,
 	 },
-	.ops = {
-		 .startup = wm8903_startup,
-		 .shutdown = wm8903_shutdown,
-		 .hw_params = wm8903_hw_params,
-		 .digital_mute = wm8903_digital_mute,
-		 .set_fmt = wm8903_set_dai_fmt,
-		 .set_sysclk = wm8903_set_dai_sysclk
-	}
+	.ops = &wm8903_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8903_dai);
 
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 24d4c905a011..032dca22dbd3 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -604,6 +604,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8971_dai_ops = {
+	.hw_params	= wm8971_pcm_hw_params,
+	.digital_mute	= wm8971_mute,
+	.set_fmt	= wm8971_set_dai_fmt,
+	.set_sysclk	= wm8971_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8971_dai = {
 	.name = "WM8971",
 	.playback = {
@@ -618,12 +625,7 @@ struct snd_soc_dai wm8971_dai = {
 		.channels_max = 2,
 		.rates = WM8971_RATES,
 		.formats = WM8971_FORMATS,},
-	.ops = {
-		.hw_params = wm8971_pcm_hw_params,
-		.digital_mute = wm8971_mute,
-		.set_fmt = wm8971_set_dai_fmt,
-		.set_sysclk = wm8971_set_dai_sysclk,
-	},
+	.ops = &wm8971_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8971_dai);
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 1a38421f7594..c518c3e5aa3f 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1332,6 +1332,15 @@ 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
  */
+static struct snd_soc_dai_ops wm8990_dai_ops = {
+	.hw_params	= wm8990_hw_params,
+	.digital_mute	= wm8990_mute,
+	.set_fmt	= wm8990_set_dai_fmt,
+	.set_clkdiv	= wm8990_set_dai_clkdiv,
+	.set_pll	= wm8990_set_dai_pll,
+	.set_sysclk	= wm8990_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8990_dai = {
 /* ADC/DAC on primary */
 	.name = "WM8990 ADC/DAC Primary",
@@ -1348,14 +1357,7 @@ struct snd_soc_dai wm8990_dai = {
 		.channels_max = 2,
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
-	.ops = {
-		.hw_params = wm8990_hw_params,
-		.digital_mute = wm8990_mute,
-		.set_fmt = wm8990_set_dai_fmt,
-		.set_clkdiv = wm8990_set_dai_clkdiv,
-		.set_pll = wm8990_set_dai_pll,
-		.set_sysclk = wm8990_set_dai_sysclk,
-	},
+	.ops = &wm8990_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8990_dai);
 
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 2e9e06b2daaf..3265817c5c26 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -269,6 +269,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 			SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops wm9705_dai_ops = {
+	.prepare	= ac97_prepare,
+};
+
 struct snd_soc_dai wm9705_dai[] = {
 	{
 		.name = "AC97 HiFi",
@@ -287,9 +291,7 @@ struct snd_soc_dai wm9705_dai[] = {
 			.rates = WM9705_AC97_RATES,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
 		},
-		.ops = {
-			.prepare = ac97_prepare,
-		},
+		.ops = &wm9705_dai_ops,
 	},
 	{
 		.name = "AC97 Aux",
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index b3a8be77676e..765cf1e7369e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -517,6 +517,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
+	.prepare	= ac97_prepare,
+};
+
+static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+};
+
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -533,8 +541,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 2,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &wm9712_dai_ops_hifi,
 },
 {
 	.name = "AC97 Aux",
@@ -544,8 +551,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 1,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,},
+	.ops = &wm9712_dai_ops_aux,
 }
 };
 EXPORT_SYMBOL_GPL(wm9712_dai);
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a93aea5c1878..523bad077fa0 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1005,6 +1005,27 @@ 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)
 
+static struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
+	.prepare	= ac97_hifi_prepare,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+};
+
+static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+};
+
+static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
+	.hw_params	= wm9713_pcm_hw_params,
+	.shutdown	= wm9713_voiceshutdown,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+	.set_fmt	= wm9713_set_dai_fmt,
+	.set_tristate	= wm9713_set_dai_tristate,
+};
+
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -1021,10 +1042,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_hifi_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_hifi,
 	},
 	{
 	.name = "AC97 Aux",
@@ -1034,10 +1052,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 1,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_aux,
 	},
 	{
 	.name = "WM9713 Voice",
@@ -1053,14 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
-	.ops = {
-		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,
-		.set_fmt = wm9713_set_dai_fmt,
-		.set_tristate = wm9713_set_dai_tristate,
-	},
+	.ops = &wm9713_dai_ops_voice,
 	},
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 0fee779e3c76..ffdb9439d3d8 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
+static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.startup	= davinci_i2s_startup,
+	.trigger	= davinci_i2s_trigger,
+	.hw_params	= davinci_i2s_hw_params,
+	.set_fmt	= davinci_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
@@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = {
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = davinci_i2s_startup,
-		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,
-		.set_fmt = davinci_i2s_set_dai_fmt,
-	},
+	.ops = &davinci_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 6844009833db..0fddd437a7c9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -562,6 +562,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * fsl_ssi_dai_template: template CPU DAI for the SSI
  */
+static struct snd_soc_dai_ops fsl_ssi_dai_ops = {
+	.startup	= fsl_ssi_startup,
+	.hw_params	= fsl_ssi_hw_params,
+	.shutdown	= fsl_ssi_shutdown,
+	.trigger	= fsl_ssi_trigger,
+	.set_sysclk	= fsl_ssi_set_sysclk,
+	.set_fmt	= fsl_ssi_set_fmt,
+};
+
 static struct snd_soc_dai fsl_ssi_dai_template = {
 	.playback = {
 		/* The SSI does not support monaural audio. */
@@ -576,14 +585,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = fsl_ssi_startup,
-		.hw_params = fsl_ssi_hw_params,
-		.shutdown = fsl_ssi_shutdown,
-		.trigger = fsl_ssi_trigger,
-		.set_sysclk = fsl_ssi_set_sysclk,
-		.set_fmt = fsl_ssi_set_fmt,
-	},
+	.ops = &fsl_ssi_dai_ops,
 };
 
 /**
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9eb1ce185bd0..3aa729df27b5 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -468,6 +468,16 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
+static struct snd_soc_dai_ops psc_i2s_dai_ops = {
+	.startup	= psc_i2s_startup,
+	.hw_params	= psc_i2s_hw_params,
+	.hw_free	= psc_i2s_hw_free,
+	.shutdown	= psc_i2s_shutdown,
+	.trigger	= psc_i2s_trigger,
+	.set_sysclk	= psc_i2s_set_sysclk,
+	.set_fmt	= psc_i2s_set_fmt,
+};
+
 static struct snd_soc_dai psc_i2s_dai_template = {
 	.playback = {
 		.channels_min = 2,
@@ -481,15 +491,7 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.rates = PSC_I2S_RATES,
 		.formats = PSC_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = psc_i2s_startup,
-		.hw_params = psc_i2s_hw_params,
-		.hw_free = psc_i2s_hw_free,
-		.shutdown = psc_i2s_shutdown,
-		.trigger = psc_i2s_trigger,
-		.set_sysclk = psc_i2s_set_sysclk,
-		.set_fmt = psc_i2s_set_fmt,
-	},
+	.ops = &psc_i2s_dai_ops,
 };
 
 /* ---------------------------------------------------------------------
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 05dd5abcddf4..d6882be33452 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
+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,
+	.hw_params	= omap_mcbsp_dai_hw_params,
+	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
+	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
+	.set_sysclk	= omap_mcbsp_dai_set_dai_sysclk,
+};
+
 #define OMAP_MCBSP_DAI_BUILDER(link_id)				\
 {								\
 	.name = "omap-mcbsp-dai-"#link_id,			\
@@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
-	.ops = {						\
-		.startup = omap_mcbsp_dai_startup,		\
-		.shutdown = omap_mcbsp_dai_shutdown,		\
-		.trigger = omap_mcbsp_dai_trigger,		\
-		.hw_params = omap_mcbsp_dai_hw_params,		\
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
-	},							\
+	.ops = &omap_mcbsp_dai_ops,				\
 	.private_data = &mcbsp_data[(link_id)].bus_id,		\
 }
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 4a973ab710be..3e18064e86b2 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -784,6 +784,19 @@ static void pxa_ssp_remove(struct platform_device *pdev,
 			    SNDRV_PCM_FMTBIT_S24_LE |	\
 			    SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+	.startup	= pxa_ssp_startup,
+	.shutdown	= pxa_ssp_shutdown,
+	.trigger	= pxa_ssp_trigger,
+	.hw_params	= pxa_ssp_hw_params,
+	.set_sysclk	= pxa_ssp_set_dai_sysclk,
+	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
+	.set_pll	= pxa_ssp_set_dai_pll,
+	.set_fmt	= pxa_ssp_set_dai_fmt,
+	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
+	.set_tristate	= pxa_ssp_set_dai_tristate,
+};
+
 struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp1",
@@ -804,18 +817,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{	.name = "pxa2xx-ssp2",
 		.id = 1,
@@ -835,18 +837,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp3",
@@ -867,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp4",
@@ -899,18 +879,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 812c2b4d3e07..11cd0f289c16 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -164,6 +164,10 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops pxa_ac97_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_params,
+};
+
 /*
  * There is only 1 physical AC97 interface for pxa2xx, but it
  * has extra fifo's that can be used for aux DACs and ADCs.
@@ -189,8 +193,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_params,},
+	.ops = &pxa_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-aux",
@@ -208,8 +211,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_aux_params,},
+	.ops = &pxa_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -221,8 +223,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_mic_params,},
+	.ops = &pxa_ac97_dai_ops,
 },
 };
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 83b59d7fe96e..e6c24408c5f9 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -304,6 +304,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+	.startup	= pxa2xx_i2s_startup,
+	.shutdown	= pxa2xx_i2s_shutdown,
+	.trigger	= pxa2xx_i2s_trigger,
+	.hw_params	= pxa2xx_i2s_hw_params,
+	.set_fmt	= pxa2xx_i2s_set_dai_fmt,
+	.set_sysclk	= pxa2xx_i2s_set_dai_sysclk,
+};
+
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
@@ -319,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.channels_max = 2,
 		.rates = PXA2XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = pxa2xx_i2s_startup,
-		.shutdown = pxa2xx_i2s_shutdown,
-		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,
-		.set_fmt = pxa2xx_i2s_set_dai_fmt,
-		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
-	},
+	.ops = &pxa_i2s_dai_ops,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index f3fc0aba0aaf..382d7eee53ef 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -708,6 +708,14 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
+	.trigger	= s3c2412_i2s_trigger,
+	.hw_params	= s3c2412_i2s_hw_params,
+	.set_fmt	= s3c2412_i2s_set_fmt,
+	.set_clkdiv	= s3c2412_i2s_set_clkdiv,
+	.set_sysclk	= s3c2412_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c2412_i2s_dai = {
 	.name	= "s3c2412-i2s",
 	.id	= 0,
@@ -726,13 +734,7 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 		.rates		= S3C2412_I2S_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.trigger	= s3c2412_i2s_trigger,
-		.hw_params	= s3c2412_i2s_hw_params,
-		.set_fmt	= s3c2412_i2s_set_fmt,
-		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
-		.set_sysclk	= s3c2412_i2s_set_sysclk,
-	},
+	.ops = &s3c2412_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 5822d2dd49ba..83ea623234e7 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -355,6 +355,11 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = {
+	.hw_params	= s3c2443_ac97_hw_params,
+	.trigger	= s3c2443_ac97_trigger,
+};
+
 struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
@@ -374,9 +379,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_params,
-		.trigger = s3c2443_ac97_trigger},
+	.ops = &s3c2443_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -388,9 +391,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_mic_params,
-		.trigger = s3c2443_ac97_mic_trigger,},
+	.ops = &s3c2443_ac97_dai_ops,
 },
 };
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 1c2b05497107..4473fb584c4c 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -456,6 +456,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = {
+	.trigger	= s3c24xx_i2s_trigger,
+	.hw_params	= s3c24xx_i2s_hw_params,
+	.set_fmt	= s3c24xx_i2s_set_fmt,
+	.set_clkdiv	= s3c24xx_i2s_set_clkdiv,
+	.set_sysclk	= s3c24xx_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
@@ -472,13 +480,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = S3C24XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,
-		.set_fmt = s3c24xx_i2s_set_fmt,
-		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
-		.set_sysclk = s3c24xx_i2s_set_sysclk,
-	},
+	.ops = &s3c24xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index d1e5390fddeb..56fa0872abbb 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
 	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
 
+static struct snd_soc_dai_ops ssi_dai_ops = {
+	.startup	= ssi_startup,
+	.shutdown	= ssi_shutdown,
+	.trigger	= ssi_trigger,
+	.hw_params	= ssi_hw_params,
+	.set_sysclk	= ssi_set_sysclk,
+	.set_clkdiv	= ssi_set_clkdiv,
+	.set_fmt	= ssi_set_fmt,
+};
+
 struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
@@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
@@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #endif
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4b90d82a098..16518329f6b2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	mutex_lock(&pcm_mutex);
 
 	/* startup the audio subsystem */
-	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream, cpu_dai);
+	if (cpu_dai->ops->startup) {
+		ret = cpu_dai->ops->startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream, codec_dai);
+	if (codec_dai->ops->startup) {
+		ret = codec_dai->ops->startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -247,8 +247,8 @@ codec_dai_err:
 		platform->pcm_ops->close(substream);
 
 platform_err:
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 
-	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream, codec_dai);
+	if (codec_dai->ops->shutdown)
+		codec_dai->ops->shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream, codec_dai);
+	if (codec_dai->ops->prepare) {
+		ret = codec_dai->ops->prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
 		}
 	}
 
-	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream, cpu_dai);
+	if (cpu_dai->ops->prepare) {
+		ret = cpu_dai->ops->prepare(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
 			goto out;
@@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
+	if (codec_dai->ops->hw_params) {
+		ret = codec_dai->ops->hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
+	if (cpu_dai->ops->hw_params) {
+		ret = cpu_dai->ops->hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -526,12 +526,12 @@ out:
 	return ret;
 
 platform_err:
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 interface_err:
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		platform->pcm_ops->hw_free(substream);
 
 	/* now free hw params for the DAI's  */
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
-	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
+	if (codec_dai->ops->trigger) {
+		ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
+	if (cpu_dai->ops->trigger) {
+		ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	/* 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 (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 1);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
@@ -741,8 +741,8 @@ 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 (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 0);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 0);
 	}
 
 	for (i = 0; i < card->num_links; i++) {
@@ -2051,8 +2051,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -2071,8 +2071,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->ops.set_clkdiv)
-		return dai->ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops->set_clkdiv)
+		return dai->ops->set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -2090,8 +2090,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->ops.set_pll)
-		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops->set_pll)
+		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -2106,8 +2106,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->ops.set_fmt)
-		return dai->ops.set_fmt(dai, fmt);
+	if (dai->ops->set_fmt)
+		return dai->ops->set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -2125,8 +2125,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -2141,8 +2141,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tristate(dai, tristate);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -2157,8 +2157,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->ops.digital_mute)
-		return dai->ops.digital_mute(dai, mute);
+	if (dai->ops->digital_mute)
+		return dai->ops->digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
@@ -2211,6 +2211,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 	return 0;
 }
 
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
 /**
  * snd_soc_register_dai - Register a DAI with the ASoC core
  *
@@ -2225,6 +2228,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai)
 	if (!dai->dev)
 		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
 
+	if (!dai->ops)
+		dai->ops = &null_dai_ops;
+
 	INIT_LIST_HEAD(&dai->list);
 
 	mutex_lock(&client_mutex);
-- 
cgit v1.2.3


From 89492be88616aa20b3a6c3eb512f83c0c7d0c8a3 Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Date: Thu, 5 Mar 2009 12:48:49 +0200
Subject: ASoC: TWL4030: Make the HS ramp delay configurable

Enum type for selecting the desired ramp delay for the headset output.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 18 ++++++++++++++----
 1 file changed, 14 insertions(+), 4 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 535d8ce2c328..86bb15cc82ce 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -584,12 +584,11 @@ static int headsetl_event(struct snd_soc_dapm_widget *w,
 
 	/* Save the current volume */
 	hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
+	hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		/* Do the anti-pop/bias ramp enable according to the TRM */
-		hs_pop = TWL4030_RAMP_DELAY_645MS;
-		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
 		hs_pop |= TWL4030_VMID_EN;
 		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
 		/* Is this needed? Can we just use whatever gain here? */
@@ -603,8 +602,6 @@ static int headsetl_event(struct snd_soc_dapm_widget *w,
 		break;
 	case SND_SOC_DAPM_POST_PMD:
 		/* Do the anti-pop/bias ramp disable according to the TRM */
-		hs_pop = twl4030_read_reg_cache(w->codec,
-						TWL4030_REG_HS_POPN_SET);
 		hs_pop &= ~TWL4030_RAMP_EN;
 		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
 		/* Bypass the reg_cache to mute the headset */
@@ -847,6 +844,17 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
  */
 static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
 
+static const char *twl4030_rampdelay_texts[] = {
+	"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
+	"437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
+	"3495/2581/1748 ms"
+};
+
+static const struct soc_enum twl4030_rampdelay_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
+			ARRAY_SIZE(twl4030_rampdelay_texts),
+			twl4030_rampdelay_texts);
+
 static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 	/* Common playback gain controls */
 	SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
@@ -901,6 +909,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 
 	SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
 		0, 3, 5, 0, input_gain_tlv),
+
+	SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
 };
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
-- 
cgit v1.2.3


From 67a9c573b5bf39bc6b40c322c58640687c1b79fe Mon Sep 17 00:00:00 2001
From: Mike Frysinger <vapier.adi@gmail.com>
Date: Fri, 6 Mar 2009 15:53:30 +0800
Subject: ASoC: Blackfin: fix typo in MUTE definition

Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ad73311.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
index 507ce0c30edf..569573d2d4d7 100644
--- a/sound/soc/codecs/ad73311.h
+++ b/sound/soc/codecs/ad73311.h
@@ -70,7 +70,7 @@
 #define REGD_IGS(x)		(x & 0x7)
 #define REGD_RMOD		(1 << 3)
 #define REGD_OGS(x)		((x & 0x7) << 4)
-#define REGD_MUTE		(x << 7)
+#define REGD_MUTE		(1 << 7)
 
 /* Control register E */
 #define CTRL_REG_E	(4 << 8)
-- 
cgit v1.2.3


From b191f63c4fe9fbcfe583180228080d02b8dcdebc Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Sun, 8 Mar 2009 17:51:52 +0100
Subject: ASoC: bring cs4270 feature/limitations list in sync

Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c | 15 +++++++--------
 1 file changed, 7 insertions(+), 8 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f86f33cc1798..0e0c23ee6afc 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -12,14 +12,13 @@
  *
  * Current features/limitations:
  *
- * 1) Software mode is supported.  Stand-alone mode is not supported.
- * 2) Only I2C is supported, not SPI
- * 3) Only Master mode is supported, not Slave.
- * 4) The machine driver's 'startup' function must call
- *    cs4270_set_dai_sysclk() with the value of MCLK.
- * 5) Only I2S and left-justified modes are supported
- * 6) Power management is not supported
- * 7) The only supported control is volume and hardware mute (if enabled)
+ * - Software mode is supported.  Stand-alone mode is not supported.
+ * - Only I2C is supported, not SPI
+ * - Support for master and slave mode
+ * - The machine driver's 'startup' function must call
+ *   cs4270_set_dai_sysclk() with the value of MCLK.
+ * - Only I2S and left-justified modes are supported
+ * - Power management is not supported
  */
 
 #include <linux/module.h>
-- 
cgit v1.2.3


From 055a49b0c92c6282e7db22e9e6ebcae6cb74ebb4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 8 Mar 2009 18:57:34 +0000
Subject: ASoC: Remove unneeded forward reference to WM8753 SPI implementation

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8753.c | 5 -----
 1 file changed, 5 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 7f353e935d71..1d5eca89de60 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -51,11 +51,6 @@
 
 #include "wm8753.h"
 
-#ifdef CONFIG_SPI_MASTER
-static struct spi_driver wm8753_spi_driver;
-static int wm8753_spi_write(struct spi_device *spi, const char *data, int len);
-#endif
-
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
-- 
cgit v1.2.3


From a381934e5f9c0c3c292d780d61f5be9c22b2ef54 Mon Sep 17 00:00:00 2001
From: Daniel Mack <daniel@caiaq.de>
Date: Mon, 9 Mar 2009 02:13:17 +0100
Subject: ASoC: Add a driver for AK4104 S/PDIF transmitter

This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   4 +
 sound/soc/codecs/Makefile |   2 +
 sound/soc/codecs/ak4104.c | 363 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/ak4104.h |   7 +
 4 files changed, 376 insertions(+)
 create mode 100644 sound/soc/codecs/ak4104.c
 create mode 100644 sound/soc/codecs/ak4104.h

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 628a591c728f..a1af311e7f06 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -14,6 +14,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311 if I2C
+	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_PCM3008
@@ -60,6 +61,9 @@ config SND_SOC_AD1980
 config SND_SOC_AD73311
 	tristate
 
+config SND_SOC_AK4104
+	tristate
+
 config SND_SOC_AK4535
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 3664cdc300b2..4717c3c99040 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,6 +1,7 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
@@ -30,6 +31,7 @@ snd-soc-wm9713-objs := wm9713.o
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
new file mode 100644
index 000000000000..338381f4fe1e
--- /dev/null
+++ b/sound/soc/codecs/ak4104.c
@@ -0,0 +1,363 @@
+/*
+ * AK4104 ALSA SoC (ASoC) driver
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ *  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/module.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <linux/spi/spi.h>
+#include <sound/asoundef.h>
+
+#include "ak4104.h"
+
+/* AK4104 registers addresses */
+#define AK4104_REG_CONTROL1		0x00
+#define AK4104_REG_RESERVED		0x01
+#define AK4104_REG_CONTROL2		0x02
+#define AK4104_REG_TX			0x03
+#define AK4104_REG_CHN_STATUS(x)	((x) + 0x04)
+#define AK4104_NUM_REGS			10
+
+#define AK4104_REG_MASK			0x1f
+#define AK4104_READ			0xc0
+#define AK4104_WRITE			0xe0
+#define AK4104_RESERVED_VAL		0x5b
+
+/* Bit masks for AK4104 registers */
+#define AK4104_CONTROL1_RSTN		(1 << 0)
+#define AK4104_CONTROL1_PW		(1 << 1)
+#define AK4104_CONTROL1_DIF0		(1 << 2)
+#define AK4104_CONTROL1_DIF1		(1 << 3)
+
+#define AK4104_CONTROL2_SEL0		(1 << 0)
+#define AK4104_CONTROL2_SEL1		(1 << 1)
+#define AK4104_CONTROL2_MODE		(1 << 2)
+
+#define AK4104_TX_TXE			(1 << 0)
+#define AK4104_TX_V			(1 << 1)
+
+#define DRV_NAME "ak4104"
+
+struct ak4104_private {
+	struct snd_soc_codec codec;
+	u8 reg_cache[AK4104_NUM_REGS];
+};
+
+static int ak4104_fill_cache(struct snd_soc_codec *codec)
+{
+	int i;
+	u8 *reg_cache = codec->reg_cache;
+	struct spi_device *spi = codec->control_data;
+
+	for (i = 0; i < codec->reg_cache_size; i++) {
+		int ret = spi_w8r8(spi, i | AK4104_READ);
+		if (ret < 0) {
+			dev_err(&spi->dev, "SPI write failure\n");
+			return ret;
+		}
+
+		reg_cache[i] = ret;
+	}
+
+	return 0;
+}
+
+static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec,
+					  unsigned int reg)
+{
+	u8 *reg_cache = codec->reg_cache;
+
+	if (reg >= codec->reg_cache_size)
+		return -EINVAL;
+
+	return reg_cache[reg];
+}
+
+static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
+			    unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+	struct spi_device *spi = codec->control_data;
+
+	if (reg >= codec->reg_cache_size)
+		return -EINVAL;
+
+	reg &= AK4104_REG_MASK;
+	reg |= AK4104_WRITE;
+
+	/* only write to the hardware if value has changed */
+	if (cache[reg] != value) {
+		u8 tmp[2] = { reg, value };
+		if (spi_write(spi, tmp, sizeof(tmp))) {
+			dev_err(&spi->dev, "SPI write failed\n");
+			return -EIO;
+		}
+
+		cache[reg] = value;
+	}
+
+	return 0;
+}
+
+static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int val = 0;
+
+	val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+	if (val < 0)
+		return val;
+
+	val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1);
+
+	/* set DAI format */
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val |= AK4104_CONTROL1_DIF0;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1;
+		break;
+	default:
+		dev_err(codec->dev, "invalid dai format\n");
+		return -EINVAL;
+	}
+
+	/* This device can only be slave */
+	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
+	return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+}
+
+static int ak4104_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;
+	int val = 0;
+
+	/* set the IEC958 bits: consumer mode, no copyright bit */
+	val |= IEC958_AES0_CON_NOT_COPYRIGHT;
+	ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val);
+
+	val = 0;
+
+	switch (params_rate(params)) {
+	case 44100:
+		val |= IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		val |= IEC958_AES3_CON_FS_48000;
+		break;
+	case 32000:
+		val |= IEC958_AES3_CON_FS_32000;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported sampling rate\n");
+		return -EINVAL;
+	}
+
+	return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val);
+}
+
+struct snd_soc_dai ak4104_dai = {
+	.name = DRV_NAME,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000 |
+			 SNDRV_PCM_RATE_32000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE  |
+			   SNDRV_PCM_FMTBIT_S24_3LE |
+			   SNDRV_PCM_FMTBIT_S24_LE
+	},
+	.ops = {
+		.hw_params = ak4104_hw_params,
+		.set_fmt = ak4104_set_dai_fmt,
+	}
+};
+
+static struct snd_soc_codec *ak4104_codec;
+
+static int ak4104_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_codec *codec;
+	struct ak4104_private *ak4104;
+	int ret, val;
+
+	spi->bits_per_word = 8;
+	spi->mode = SPI_MODE_0;
+	ret = spi_setup(spi);
+	if (ret < 0)
+		return ret;
+
+	ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL);
+	if (!ak4104) {
+		dev_err(&spi->dev, "could not allocate codec\n");
+		return -ENOMEM;
+	}
+
+	codec = &ak4104->codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->dev = &spi->dev;
+	codec->name = DRV_NAME;
+	codec->owner = THIS_MODULE;
+	codec->dai = &ak4104_dai;
+	codec->num_dai = 1;
+	codec->private_data = ak4104;
+	codec->control_data = spi;
+	codec->reg_cache = ak4104->reg_cache;
+	codec->reg_cache_size = AK4104_NUM_REGS;
+
+	/* read all regs and fill the cache */
+	ret = ak4104_fill_cache(codec);
+	if (ret < 0) {
+		dev_err(&spi->dev, "failed to fill register cache\n");
+		return ret;
+	}
+
+	/* read the 'reserved' register - according to the datasheet, it
+	 * should contain 0x5b. Not a good way to verify the presence of
+	 * the device, but there is no hardware ID register. */
+	if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) !=
+					 AK4104_RESERVED_VAL) {
+		ret = -ENODEV;
+		goto error_free_codec;
+	}
+
+	/* set power-up and non-reset bits */
+	val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+	val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN;
+	ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+	if (ret < 0)
+		goto error_free_codec;
+
+	/* enable transmitter */
+	val = ak4104_read_reg_cache(codec, AK4104_REG_TX);
+	val |= AK4104_TX_TXE;
+	ret = ak4104_spi_write(codec, AK4104_REG_TX, val);
+	if (ret < 0)
+		goto error_free_codec;
+
+	ak4104_codec = codec;
+	ret = snd_soc_register_dai(&ak4104_dai);
+	if (ret < 0) {
+		dev_err(&spi->dev, "failed to register DAI\n");
+		goto error_free_codec;
+	}
+
+	spi_set_drvdata(spi, ak4104);
+	dev_info(&spi->dev, "SPI device initialized\n");
+	return 0;
+
+error_free_codec:
+	kfree(ak4104);
+	ak4104_dai.dev = NULL;
+	return ret;
+}
+
+static int __devexit ak4104_spi_remove(struct spi_device *spi)
+{
+	int ret, val;
+	struct ak4104_private *ak4104 = spi_get_drvdata(spi);
+
+	val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1);
+	if (val < 0)
+		return val;
+
+	/* clear power-up and non-reset bits */
+	val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+	ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val);
+	if (ret < 0)
+		return ret;
+
+	ak4104_codec = NULL;
+	kfree(ak4104);
+	return 0;
+}
+
+static int ak4104_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = ak4104_codec;
+	int ret;
+
+	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
+	socdev->card->codec = 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\n");
+		return ret;
+	}
+
+	/* Register the socdev */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card\n");
+		snd_soc_free_pcms(socdev);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int ak4104_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	snd_soc_free_pcms(socdev);
+	return 0;
+};
+
+struct snd_soc_codec_device soc_codec_device_ak4104 = {
+	.probe = 	ak4104_probe,
+	.remove = 	ak4104_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_ak4104);
+
+static struct spi_driver ak4104_spi_driver = {
+	.driver  = {
+		.name   = DRV_NAME,
+		.owner  = THIS_MODULE,
+	},
+	.probe  = ak4104_spi_probe,
+	.remove = __devexit_p(ak4104_spi_remove),
+};
+
+static int __init ak4104_init(void)
+{
+	pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
+	return spi_register_driver(&ak4104_spi_driver);
+}
+module_init(ak4104_init);
+
+static void __exit ak4104_exit(void)
+{
+	spi_unregister_driver(&ak4104_spi_driver);
+}
+module_exit(ak4104_exit);
+
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h
new file mode 100644
index 000000000000..eb88fe7e4def
--- /dev/null
+++ b/sound/soc/codecs/ak4104.h
@@ -0,0 +1,7 @@
+#ifndef _AK4104_H
+#define _AK4104_H
+
+extern struct snd_soc_dai ak4104_dai;
+extern struct snd_soc_codec_device soc_codec_device_ak4104;
+
+#endif
-- 
cgit v1.2.3


From aaf1e176fa9a96fe1eea33b710684bba066aedc1 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 10 Mar 2009 10:55:15 +0000
Subject: ASoC: Add initial driver for the WM8400 CODEC

The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application.  This driver supports the
primary audio CODEC features, including:

 - 1W speaker driver
 - Fully differential headphone output
 - Up to 4 differential microphone inputs

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/linux/mfd/wm8400-audio.h |    1 +
 sound/soc/codecs/Kconfig         |    4 +
 sound/soc/codecs/Makefile        |    2 +
 sound/soc/codecs/wm8400.c        | 1479 ++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8400.h        |   62 ++
 5 files changed, 1548 insertions(+)
 create mode 100644 sound/soc/codecs/wm8400.c
 create mode 100644 sound/soc/codecs/wm8400.h

(limited to 'sound/soc/codecs')

diff --git a/include/linux/mfd/wm8400-audio.h b/include/linux/mfd/wm8400-audio.h
index b6640e018046..e06ed3eb1d0a 100644
--- a/include/linux/mfd/wm8400-audio.h
+++ b/include/linux/mfd/wm8400-audio.h
@@ -1181,6 +1181,7 @@
 #define WM8400_FLL_OUTDIV_SHIFT                      0  /* FLL_OUTDIV - [2:0] */
 #define WM8400_FLL_OUTDIV_WIDTH                      3  /* FLL_OUTDIV - [2:0] */
 
+struct wm8400;
 void wm8400_reset_codec_reg_cache(struct wm8400 *wm8400);
 
 #endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a1af311e7f06..b6c7f7a01cb0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
+	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8580 if I2C
 	select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
@@ -110,6 +111,9 @@ config SND_SOC_UDA1380
 config SND_SOC_WM8350
 	tristate
 
+config SND_SOC_WM8400
+	tristate
+
 config SND_SOC_WM8510
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4717c3c99040..030d2454725f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-twl4030-objs := twl4030.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wm8350-objs := wm8350.o
+snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
 snd-soc-wm8728-objs := wm8728.o
@@ -44,6 +45,7 @@ obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(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
+obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
 obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
new file mode 100644
index 000000000000..9cb73d9d023d
--- /dev/null
+++ b/sound/soc/codecs/wm8400.c
@@ -0,0 +1,1479 @@
+/*
+ * wm8400.c  --  WM8400 ALSA Soc Audio driver
+ *
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * 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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mfd/wm8400-audio.h>
+#include <linux/mfd/wm8400-private.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 "wm8400.h"
+
+/* Fake register for internal state */
+#define WM8400_INTDRIVBITS      (WM8400_REGISTER_COUNT + 1)
+#define WM8400_INMIXL_PWR			0
+#define WM8400_AINLMUX_PWR			1
+#define WM8400_INMIXR_PWR			2
+#define WM8400_AINRMUX_PWR			3
+
+static struct regulator_bulk_data power[] = {
+	{
+		.supply = "I2S1VDD",
+	},
+	{
+		.supply = "I2S2VDD",
+	},
+	{
+		.supply = "DCVDD",
+	},
+	{
+		.supply = "FLLVDD",
+	},
+	{
+		.supply = "HPVDD",
+	},
+	{
+		.supply = "SPKVDD",
+	},
+};
+
+/* codec private data */
+struct wm8400_priv {
+	struct snd_soc_codec codec;
+	struct wm8400 *wm8400;
+	u16 fake_register;
+	unsigned int sysclk;
+	unsigned int pcmclk;
+	struct work_struct work;
+};
+
+static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	if (reg == WM8400_INTDRIVBITS)
+		return wm8400->fake_register;
+	else
+		return wm8400_reg_read(wm8400->wm8400, reg);
+}
+
+/*
+ * write to the wm8400 register space
+ */
+static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	if (reg == WM8400_INTDRIVBITS) {
+		wm8400->fake_register = value;
+		return 0;
+	} else
+		return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value);
+}
+
+static void wm8400_codec_reset(struct snd_soc_codec *codec)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	wm8400_reset_codec_reg_cache(wm8400->wm8400);
+}
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int reg = mc->reg;
+        int ret;
+        u16 val;
+
+        ret = snd_soc_put_volsw(kcontrol, ucontrol);
+        if (ret < 0)
+                return ret;
+
+        /* now hit the volume update bits (always bit 8) */
+        val = wm8400_read(codec, reg);
+        return wm8400_write(codec, reg, val | 0x0100);
+}
+
+#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, 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 = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8400_digital_sidetone[] =
+	{"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8400_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+		WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone);
+
+static const struct soc_enum wm8400_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+		WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone);
+
+static const char *wm8400_adcmode[] =
+	{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8400_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode);
+
+static const struct snd_kcontrol_new wm8400_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L12MNBST_SHIFT,
+	   1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L34MNBST_SHIFT,
+	   1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R12MNBST_SHIFT,
+	   1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R34MNBST_SHIFT,
+	   1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LLI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LR12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LL12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RRI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RL12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RR12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RLI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RLBROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RRBROVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* LOUT */
+WM8400_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8400_LEFT_OUTPUT_VOLUME,
+	WM8400_LOUTVOL_SHIFT, WM8400_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8400_LEFT_OUTPUT_VOLUME, WM8400_LOZC_SHIFT, 1, 0),
+
+/* ROUT */
+WM8400_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8400_RIGHT_OUTPUT_VOLUME,
+	WM8400_ROUTVOL_SHIFT, WM8400_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8400_RIGHT_OUTPUT_VOLUME, WM8400_ROZC_SHIFT, 1, 0),
+
+/* LOPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8400_LEFT_OPGA_VOLUME,
+	WM8400_LOPGAVOL_SHIFT, WM8400_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8400_LEFT_OPGA_VOLUME,
+	WM8400_LOPGAZC_SHIFT, 1, 0),
+
+/* ROPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8400_RIGHT_OPGA_VOLUME,
+	WM8400_ROPGAVOL_SHIFT, WM8400_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8400_RIGHT_OPGA_VOLUME,
+	WM8400_ROPGAZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LOPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LOATTN_SHIFT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_RONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_ROPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_ROATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT3MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT3ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT4MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT4ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8400_CLASSD1,
+	WM8400_CDMODE_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8400_SPEAKER_VOLUME,
+	WM8400_SPKATTN_SHIFT, WM8400_SPKATTN_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8400_CLASSD3,
+	WM8400_DCGAIN_SHIFT, 6, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8400_CLASSD3,
+	WM8400_ACGAIN_SHIFT, 6, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+	WM8400_LEFT_DAC_DIGITAL_VOLUME, WM8400_DACL_VOL_SHIFT,
+	127, 0, out_dac_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+	WM8400_RIGHT_DAC_DIGITAL_VOLUME, WM8400_DACR_VOL_SHIFT,
+	127, 0, out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8400_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8400_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+	WM8400_ADCL_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+	WM8400_ADCR_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8400_ADC_CTRL,
+	WM8400_ADC_HPF_ENA_SHIFT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8400_right_adcmode_enum),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+	WM8400_LEFT_ADC_DIGITAL_VOLUME,
+	WM8400_ADCL_VOL_SHIFT,
+	WM8400_ADCL_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+	WM8400_RIGHT_ADC_DIGITAL_VOLUME,
+	WM8400_ADCR_VOL_SHIFT,
+	WM8400_ADCR_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+	WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LIN12VOL_SHIFT,
+	WM8400_LIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+	WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LIN34VOL_SHIFT,
+	WM8400_LIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LI34MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+	WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RIN12VOL_SHIFT,
+	WM8400_RIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+	WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RIN34VOL_SHIFT,
+	WM8400_RIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RI34MUTE_SHIFT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8400_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8400_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&wm8400_snd_controls[i],codec,
+					NULL));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event (struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	u16 reg, fakepower;
+
+	reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2);
+	fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS);
+
+	if (fakepower & ((1 << WM8400_INMIXL_PWR) |
+		(1 << WM8400_AINLMUX_PWR))) {
+		reg |= WM8400_AINL_ENA;
+	} else {
+		reg &= ~WM8400_AINL_ENA;
+	}
+
+	if (fakepower & ((1 << WM8400_INMIXR_PWR) |
+		(1 << WM8400_AINRMUX_PWR))) {
+		reg |= WM8400_AINR_ENA;
+	} else {
+		reg &= ~WM8400_AINL_ENA;
+	}
+	wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+	return 0;
+}
+
+static int outmixer_event (struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol * kcontrol, int event)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 reg_shift = mc->shift;
+	int ret = 0;
+	u16 reg;
+
+	switch (reg_shift) {
+	case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
+		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1);
+		if (reg & WM8400_LDLO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 1 LDLO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
+		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2);
+		if (reg & WM8400_RDRO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 2 RDRO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
+		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		if (reg & WM8400_LDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer LDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
+		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		if (reg & WM8400_RDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer RDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0,7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8400_INPUT_MIXER2, WM8400_LMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8400_INPUT_MIXER2, WM8400_LMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8400_INPUT_MIXER2, WM8400_LMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8400_INPUT_MIXER2, WM8400_LMP4_SHIFT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8400_INPUT_MIXER2, WM8400_RMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8400_INPUT_MIXER2, WM8400_RMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8400_INPUT_MIXER2, WM8400_RMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8400_INPUT_MIXER2, WM8400_RMP4_SHIFT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8400_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8400_INPUT_MIXER3,
+	WM8400_LDBVOL_SHIFT, WM8400_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8400_INPUT_MIXER5, WM8400_LI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+		1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+		1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8400_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8400_INPUT_MIXER4,
+	WM8400_RDBVOL_SHIFT, WM8400_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8400_INPUT_MIXER6, WM8400_RI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+	1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+	1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8400_ainlmux[] =
+	{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8400_ainlmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT,
+	ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8400_ainrmux[] =
+	{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8400_ainrmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT,
+	ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT,
+			WM8400_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT,
+			WM8400_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LRBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LLBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LRI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LLI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LR12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LL12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LDLO_SHIFT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8400_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RLBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RRBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RLI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RRI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RL12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RR12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RDRO_SHIFT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LLOPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LROPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8400_LINE_MIXER1,
+	WM8400_LOPLON_SHIFT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER1,
+	WM8400_LR12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER1,
+	WM8400_LL12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LLOPGALOP_SHIFT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RROPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RLOPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8400_LINE_MIXER2,
+	WM8400_ROPRON_SHIFT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER2,
+	WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER2,
+	WM8400_RR12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RROPGAROP_SHIFT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+	WM8400_LI4O3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8400_OUT3_4_MIXER,
+	WM8400_LPGAO3_SHIFT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8400_OUT3_4_MIXER,
+	WM8400_RPGAO4_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+	WM8400_RI4O4_SHIFT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8400_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LI2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LB2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LOPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+	WM8400_ROPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RI2SPK_SHIFT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8400_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8400_POWER_MANAGEMENT_2,
+	WM8400_ADCL_ENA_SHIFT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8400_POWER_MANAGEMENT_2,
+	WM8400_ADCR_ENA_SHIFT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_LIN12_ENA_SHIFT,
+		   0, &wm8400_dapm_lin12_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_LIN34_ENA_SHIFT,
+		   0, &wm8400_dapm_lin34_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_RIN12_ENA_SHIFT,
+		   0, &wm8400_dapm_rin12_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_RIN34_ENA_SHIFT,
+		   0, &wm8400_dapm_rin34_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0,
+	&wm8400_dapm_inmixl_controls[0],
+	ARRAY_SIZE(wm8400_dapm_inmixl_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0,
+	&wm8400_dapm_ainlmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0,
+	&wm8400_dapm_inmixr_controls[0],
+	ARRAY_SIZE(wm8400_dapm_inmixr_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0,
+	&wm8400_dapm_ainrmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8400_POWER_MANAGEMENT_3,
+	WM8400_DACL_ENA_SHIFT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8400_POWER_MANAGEMENT_3,
+	WM8400_DACR_ENA_SHIFT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8400_POWER_MANAGEMENT_3,
+		     WM8400_LOMIX_ENA_SHIFT,
+		     0, &wm8400_dapm_lomix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_lomix_controls),
+		     outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LON_ENA_SHIFT,
+		   0, &wm8400_dapm_lonmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LOP_ENA_SHIFT,
+		   0, &wm8400_dapm_lopmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT3_ENA_SHIFT,
+		   0, &wm8400_dapm_out3mix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8400_POWER_MANAGEMENT_1, WM8400_SPK_ENA_SHIFT,
+		     0, &wm8400_dapm_spkmix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_spkmix_controls), outmixer_event,
+		     SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT4_ENA_SHIFT,
+	0, &wm8400_dapm_out4mix_controls[0],
+	ARRAY_SIZE(wm8400_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_ROP_ENA_SHIFT,
+		   0, &wm8400_dapm_ropmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_RON_ENA_SHIFT,
+		   0, &wm8400_dapm_ronmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8400_POWER_MANAGEMENT_3,
+		     WM8400_ROMIX_ENA_SHIFT,
+		     0, &wm8400_dapm_romix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_romix_controls),
+		     outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_LOUT_ENA_SHIFT,
+		 0, NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_ROUT_ENA_SHIFT,
+		 0, NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8400_POWER_MANAGEMENT_3, WM8400_LOPGA_ENA_SHIFT, 0,
+	NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0,
+	NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1,
+	WM8400_MIC1BIAS_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Make DACs turn on when playing even if not mixed into any outputs */
+	{"Internal DAC Sink", NULL, "Left DAC"},
+	{"Internal DAC Sink", NULL, "Right DAC"},
+
+	/* Make ADCs turn on when recording
+	 * even if not mixed from any inputs */
+	{"Left ADC", NULL, "Internal ADC Source"},
+	{"Right ADC", NULL, "Internal ADC Source"},
+
+	/* Input Side */
+	/* LIN12 PGA */
+	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
+	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
+	/* LIN34 PGA */
+	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
+	/* INMIXL */
+	{"INMIXL", "Record Left Volume", "LOMIX"},
+	{"INMIXL", "LIN2 Volume", "LIN2"},
+	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+	/* AILNMUX */
+	{"AILNMUX", "INMIXL Mix", "INMIXL"},
+	{"AILNMUX", "DIFFINL Mix", "LIN12 PGA"},
+	{"AILNMUX", "DIFFINL Mix", "LIN34 PGA"},
+	{"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Left ADC", NULL, "AILNMUX"},
+
+	/* RIN12 PGA */
+	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
+	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
+	/* RIN34 PGA */
+	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
+	/* INMIXL */
+	{"INMIXR", "Record Right Volume", "ROMIX"},
+	{"INMIXR", "RIN2 Volume", "RIN2"},
+	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+	/* AIRNMUX */
+	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"},
+	{"AIRNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Right ADC", NULL, "AIRNMUX"},
+
+	/* LOMIX */
+	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+	{"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+	{"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"LOMIX", "LOMIX Right ADC Bypass Switch", "AIRNMUX"},
+	{"LOMIX", "LOMIX Left ADC Bypass Switch", "AILNMUX"},
+	{"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+	/* ROMIX */
+	{"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+	{"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+	{"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"ROMIX", "ROMIX Right ADC Bypass Switch", "AIRNMUX"},
+	{"ROMIX", "ROMIX Left ADC Bypass Switch", "AILNMUX"},
+	{"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+	/* SPKMIX */
+	{"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+	{"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+	{"SPKMIX", "SPKMIX LADC Bypass Switch", "AILNMUX"},
+	{"SPKMIX", "SPKMIX RADC Bypass Switch", "AIRNMUX"},
+	{"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+	{"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+	{"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+	{"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+	/* LONMIX */
+	{"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+	/* LOPMIX */
+	{"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+	/* OUT3MIX */
+	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
+	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+	/* OUT4MIX */
+	{"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+	{"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+	/* RONMIX */
+	{"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+	/* ROPMIX */
+	{"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+	/* Out Mixer PGAs */
+	{"LOPGA", NULL, "LOMIX"},
+	{"ROPGA", NULL, "ROMIX"},
+
+	{"LOUT PGA", NULL, "LOMIX"},
+	{"ROUT PGA", NULL, "ROMIX"},
+
+	/* Output Pins */
+	{"LON", NULL, "LONMIX"},
+	{"LOP", NULL, "LOPMIX"},
+	{"OUT3", NULL, "OUT3MIX"},
+	{"LOUT", NULL, "LOUT PGA"},
+	{"SPKN", NULL, "SPKMIX"},
+	{"ROUT", NULL, "ROUT PGA"},
+	{"OUT4", NULL, "OUT4MIX"},
+	{"ROP", NULL, "ROPMIX"},
+	{"RON", NULL, "RONMIX"},
+};
+
+static int wm8400_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
+				  ARRAY_SIZE(wm8400_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+/*
+ * Clock after FLL and dividers
+ */
+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;
+
+	wm8400->sysclk = freq;
+	return 0;
+}
+
+/*
+ * Sets ADC and Voice DAC format.
+ */
+static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 audio1, audio3;
+
+	audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+	audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		audio3 &= ~WM8400_AIF_MSTR1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		audio3 |= WM8400_AIF_MSTR1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	audio1 &= ~WM8400_AIF_FMT_MASK;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		audio1 |= WM8400_AIF_FMT_I2S;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		audio1 |= WM8400_AIF_FMT_RIGHTJ;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		audio1 |= WM8400_AIF_FMT_LEFTJ;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		audio1 |= WM8400_AIF_FMT_DSP;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		audio1 |= WM8400_AIF_FMT_DSP | WM8400_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
+	return 0;
+}
+
+static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+		int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8400_MCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_MCLK_DIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_DACCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_DAC_CLKDIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_ADCCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_ADC_CLKDIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_BCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_1) &
+			~WM8400_BCLK_DIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_1, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8400_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;
+	u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+
+	audio1 &= ~WM8400_AIF_WL_MASK;
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		audio1 |= WM8400_AIF_WL_20BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		audio1 |= WM8400_AIF_WL_24BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		audio1 |= WM8400_AIF_WL_32BITS;
+		break;
+	}
+
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	return 0;
+}
+
+static int wm8400_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
+
+	if (mute)
+		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+	else
+		wm8400_write(codec, WM8400_DAC_CTRL, val);
+
+	return 0;
+}
+
+/* TODO: set bias for best performance at standby */
+static int wm8400_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+	u16 val;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*50k */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+			~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(power),
+						    &power[0]);
+			if (ret != 0) {
+				dev_err(wm8400->wm8400->dev,
+					"Failed to enable regulators: %d\n",
+					ret);
+				return ret;
+			}
+
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+				     WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
+
+			/* Enable all output discharge bits */
+			wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+				WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
+				WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
+				WM8400_DIS_ROUT);
+
+			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+				     WM8400_BUFDCOPEN | WM8400_POBCTRL);
+
+			msleep(500);
+
+			/* Enable outputs */
+			val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+			val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
+				WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+				WM8400_ROUT_ENA;
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+			/* disable all output discharge bits */
+			wm8400_write(codec, WM8400_ANTIPOP1, 0);
+
+			/* Enable VREF & VMID at 2x50k */
+			val |= 0x2 | WM8400_VREF_ENA;
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+			msleep(600);
+
+			/* Enable BUFIOEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+				     WM8400_BUFDCOPEN | WM8400_POBCTRL |
+				     WM8400_BUFIOEN);
+
+			/* Disable outputs */
+			val &= ~(WM8400_SPK_ENA | WM8400_OUT3_ENA |
+				 WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+				 WM8400_ROUT_ENA);
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
+		}
+
+		/* VMID=2*300k */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+			~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Enable POBCTRL and SOFT_ST */
+		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			WM8400_POBCTRL | WM8400_BUFIOEN);
+
+		/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			WM8400_BUFDCOPEN | WM8400_POBCTRL |
+			WM8400_BUFIOEN);
+
+		/* mute DAC */
+		val = wm8400_read(codec, WM8400_DAC_CTRL);
+		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+
+		/* Enable any disabled outputs */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+		val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
+			WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+			WM8400_ROUT_ENA;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		/* Disable VMID */
+		val &= ~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		msleep(300);
+
+		/* Enable all output discharge bits */
+		wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+			WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
+			WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
+			WM8400_DIS_ROUT);
+
+		/* Disable VREF */
+		val &= ~WM8400_VREF_ENA;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8400_write(codec, WM8400_ANTIPOP2, 0x0);
+
+		ret = regulator_bulk_disable(ARRAY_SIZE(power),
+					     &power[0]);
+		if (ret != 0)
+			return ret;
+
+		break;
+	}
+
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8400_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+/*
+ * The WM8400 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8400_dai = {
+/* ADC/DAC on primary */
+	.name = "WM8400 ADC/DAC Primary",
+	.id = 1,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8400_RATES,
+		.formats = WM8400_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8400_RATES,
+		.formats = WM8400_FORMATS,
+	},
+	.ops = {
+		.hw_params = wm8400_hw_params,
+		.digital_mute = wm8400_mute,
+		.set_fmt = wm8400_set_dai_fmt,
+		.set_clkdiv = wm8400_set_dai_clkdiv,
+		.set_sysclk = wm8400_set_dai_sysclk,
+	},
+};
+EXPORT_SYMBOL_GPL(wm8400_dai);
+
+static int wm8400_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;
+
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm8400_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8400_codec;
+
+static int wm8400_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (!wm8400_codec) {
+		dev_err(&pdev->dev, "wm8400 not yet discovered\n");
+		return -ENODEV;
+	}
+	codec = wm8400_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");
+		goto pcm_err;
+	}
+
+	wm8400_add_controls(codec);
+	wm8400_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	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);
+pcm_err:
+	return ret;
+}
+
+/* power down chip */
+static int wm8400_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_wm8400 = {
+	.probe =	wm8400_probe,
+	.remove =	wm8400_remove,
+	.suspend =	wm8400_suspend,
+	.resume =	wm8400_resume,
+};
+
+static void wm8400_probe_deferred(struct work_struct *work)
+{
+	struct wm8400_priv *priv = container_of(work, struct wm8400_priv,
+						work);
+	struct snd_soc_codec *codec = &priv->codec;
+	int ret;
+
+	/* charge output caps */
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* We're done, tell the subsystem. */
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(priv->wm8400->dev,
+			"Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8400_dai);
+	if (ret != 0) {
+		dev_err(priv->wm8400->dev,
+			"Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8400_codec_probe(struct platform_device *dev)
+{
+	struct wm8400_priv *priv;
+	int ret;
+	u16 reg;
+	struct snd_soc_codec *codec;
+
+	priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	codec = &priv->codec;
+	codec->private_data = priv;
+	codec->control_data = dev->dev.driver_data;
+	priv->wm8400 = dev->dev.driver_data;
+
+	ret = regulator_bulk_get(priv->wm8400->dev,
+				 ARRAY_SIZE(power), &power[0]);
+	if (ret != 0) {
+		dev_err(&dev->dev, "Failed to get regulators: %d\n", ret);
+	        goto err;
+	}
+
+	codec->dev = &dev->dev;
+	wm8400_dai.dev = &dev->dev;
+
+	codec->name = "WM8400";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8400_read;
+	codec->write = wm8400_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8400_set_bias_level;
+	codec->dai = &wm8400_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8400_REGISTER_COUNT;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	INIT_WORK(&priv->work, wm8400_probe_deferred);
+
+	wm8400_codec_reset(codec);
+
+	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+	wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
+
+	/* Latch volume update bits */
+	reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
+	wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+		     reg & WM8400_IPVU);
+	reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
+	wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+		     reg & WM8400_IPVU);
+
+	wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+	wm8400_codec = codec;
+
+	if (!schedule_work(&priv->work)) {
+		ret = -EINVAL;
+		goto err_regulator;
+	}
+
+	return 0;
+
+err_regulator:
+	wm8400_codec = NULL;
+	regulator_bulk_free(ARRAY_SIZE(power), power);
+err:
+	kfree(priv);
+	return ret;
+}
+
+static int __exit wm8400_codec_remove(struct platform_device *dev)
+{
+	struct wm8400_priv *priv = wm8400_codec->private_data;
+	u16 reg;
+
+	snd_soc_unregister_dai(&wm8400_dai);
+	snd_soc_unregister_codec(wm8400_codec);
+
+	reg = wm8400_read(wm8400_codec, WM8400_POWER_MANAGEMENT_1);
+	wm8400_write(wm8400_codec, WM8400_POWER_MANAGEMENT_1,
+		     reg & (~WM8400_CODEC_ENA));
+
+	regulator_bulk_free(ARRAY_SIZE(power), power);
+	kfree(priv);
+
+	wm8400_codec = NULL;
+
+	return 0;
+}
+
+static struct platform_driver wm8400_codec_driver = {
+	.driver = {
+		.name = "wm8400-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8400_codec_probe,
+	.remove	= __exit_p(wm8400_codec_remove),
+};
+
+static int __init wm8400_codec_init(void)
+{
+	return platform_driver_register(&wm8400_codec_driver);
+}
+module_init(wm8400_codec_init);
+
+static void __exit wm8400_codec_exit(void)
+{
+	platform_driver_unregister(&wm8400_codec_driver);
+}
+module_exit(wm8400_codec_exit);
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8400);
+
+MODULE_DESCRIPTION("ASoC WM8400 driver");
+MODULE_AUTHOR("Mark Brown");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8400-codec");
diff --git a/sound/soc/codecs/wm8400.h b/sound/soc/codecs/wm8400.h
new file mode 100644
index 000000000000..79c5934d4776
--- /dev/null
+++ b/sound/soc/codecs/wm8400.h
@@ -0,0 +1,62 @@
+/*
+ * wm8400.h  --  audio driver for WM8400
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * 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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef _WM8400_CODEC_H
+#define _WM8400_CODEC_H
+
+#define WM8400_MCLK_DIV 0
+#define WM8400_DACCLK_DIV 1
+#define WM8400_ADCCLK_DIV 2
+#define WM8400_BCLK_DIV 3
+
+#define WM8400_MCLK_DIV_1 0x400
+#define WM8400_MCLK_DIV_2 0x800
+
+#define WM8400_DAC_CLKDIV_1    0x00
+#define WM8400_DAC_CLKDIV_1_5  0x04
+#define WM8400_DAC_CLKDIV_2    0x08
+#define WM8400_DAC_CLKDIV_3    0x0c
+#define WM8400_DAC_CLKDIV_4    0x10
+#define WM8400_DAC_CLKDIV_5_5  0x14
+#define WM8400_DAC_CLKDIV_6    0x18
+
+#define WM8400_ADC_CLKDIV_1    0x00
+#define WM8400_ADC_CLKDIV_1_5  0x20
+#define WM8400_ADC_CLKDIV_2    0x40
+#define WM8400_ADC_CLKDIV_3    0x60
+#define WM8400_ADC_CLKDIV_4    0x80
+#define WM8400_ADC_CLKDIV_5_5  0xa0
+#define WM8400_ADC_CLKDIV_6    0xc0
+
+
+#define WM8400_BCLK_DIV_1                       (0x0 << 1)
+#define WM8400_BCLK_DIV_1_5                     (0x1 << 1)
+#define WM8400_BCLK_DIV_2                       (0x2 << 1)
+#define WM8400_BCLK_DIV_3                       (0x3 << 1)
+#define WM8400_BCLK_DIV_4                       (0x4 << 1)
+#define WM8400_BCLK_DIV_5_5                     (0x5 << 1)
+#define WM8400_BCLK_DIV_6                       (0x6 << 1)
+#define WM8400_BCLK_DIV_8                       (0x7 << 1)
+#define WM8400_BCLK_DIV_11                      (0x8 << 1)
+#define WM8400_BCLK_DIV_12                      (0x9 << 1)
+#define WM8400_BCLK_DIV_16                      (0xA << 1)
+#define WM8400_BCLK_DIV_22                      (0xB << 1)
+#define WM8400_BCLK_DIV_24                      (0xC << 1)
+#define WM8400_BCLK_DIV_32                      (0xD << 1)
+#define WM8400_BCLK_DIV_44                      (0xE << 1)
+#define WM8400_BCLK_DIV_48                      (0xF << 1)
+
+extern struct snd_soc_dai wm8400_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8400;
+
+#endif
-- 
cgit v1.2.3


From 02b7cbc3994622900e8fc201f5f229b591c43628 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 11 Mar 2009 14:12:28 +0000
Subject: ASoC: Remove version display from WM8580 driver

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8580.c | 4 ----
 1 file changed, 4 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d3c51ba5e6f9..6cab82a9c9d7 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -35,8 +35,6 @@
 
 #include "wm8580.h"
 
-#define WM8580_VERSION "0.1"
-
 struct pll_state {
 	unsigned int in;
 	unsigned int out;
@@ -972,8 +970,6 @@ static int wm8580_probe(struct platform_device *pdev)
 	struct wm8580_priv *wm8580;
 	int ret = 0;
 
-	pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
-- 
cgit v1.2.3


From 6f7cb44ba1a5195bf719f9ba1d57bd79e13262c1 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 11 Mar 2009 18:31:08 +0000
Subject: ASoC: Move WM8580 to normal I2C device probe

Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8580.c | 326 ++++++++++++++++++++++------------------------
 sound/soc/codecs/wm8580.h |   5 -
 2 files changed, 158 insertions(+), 173 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 27f9e231bf69..442ea6f160fc 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1,7 +1,7 @@
 /*
  * wm8580.c  --  WM8580 ALSA Soc Audio driver
  *
- * Copyright 2008 Wolfson Microelectronics PLC.
+ * Copyright 2008, 2009 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 as published by the
@@ -35,17 +35,6 @@
 
 #include "wm8580.h"
 
-struct pll_state {
-	unsigned int in;
-	unsigned int out;
-};
-
-/* codec private data */
-struct wm8580_priv {
-	struct pll_state a;
-	struct pll_state b;
-};
-
 /* WM8580 register space */
 #define WM8580_PLLA1                         0x00
 #define WM8580_PLLA2                         0x01
@@ -100,6 +89,8 @@ struct wm8580_priv {
 #define WM8580_READBACK                      0x34
 #define WM8580_RESET                         0x35
 
+#define WM8580_MAX_REGISTER                  0x35
+
 /* PLLB4 (register 7h) */
 #define WM8580_PLLB4_MCLKOUTSRC_MASK   0x60
 #define WM8580_PLLB4_MCLKOUTSRC_PLLA   0x20
@@ -191,6 +182,20 @@ static const u16 wm8580_reg[] = {
 	0x0000, 0x0000 /*R53*/
 };
 
+struct pll_state {
+	unsigned int in;
+	unsigned int out;
+};
+
+/* codec private data */
+struct wm8580_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8580_MAX_REGISTER + 1];
+	struct pll_state a;
+	struct pll_state b;
+};
+
+
 /*
  * read wm8580 register cache
  */
@@ -755,8 +760,22 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
+		break;
+
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Power up and get individual control of the DACs */
+			reg = wm8580_read(codec, WM8580_PWRDN1);
+			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
+			wm8580_write(codec, WM8580_PWRDN1, reg);
+
+			/* Make VMID high impedence */
+			reg = wm8580_read(codec,  WM8580_ADC_CONTROL1);
+			reg &= ~0x100;
+			wm8580_write(codec, WM8580_ADC_CONTROL1, reg);
+		}
 		break;
+
 	case SND_SOC_BIAS_OFF:
 		reg = wm8580_read(codec, WM8580_PWRDN1);
 		wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
@@ -812,100 +831,163 @@ struct snd_soc_dai wm8580_dai[] = {
 };
 EXPORT_SYMBOL_GPL(wm8580_dai);
 
-/*
- * initialise the WM8580 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8580_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8580_codec;
+
+static int wm8580_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->card->codec;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	codec->name = "WM8580";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8580_read_reg_cache;
-	codec->write = wm8580_write;
-	codec->set_bias_level = wm8580_set_bias_level;
-	codec->dai = wm8580_dai;
-	codec->num_dai = ARRAY_SIZE(wm8580_dai);
-	codec->reg_cache_size = ARRAY_SIZE(wm8580_reg);
-	codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg),
-				   GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
-
-	/* Get the codec into a known state */
-	wm8580_write(codec, WM8580_RESET, 0);
-
-	/* Power up and get individual control of the DACs */
-	wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) &
-		     ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD));
+	if (wm8580_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
 
-	/* Make VMID high impedence */
-	wm8580_write(codec, WM8580_ADC_CONTROL1,
-		     wm8580_read(codec,  WM8580_ADC_CONTROL1) & ~0x100);
+	socdev->card->codec = wm8580_codec;
+	codec = wm8580_codec;
 
 	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1,
-			       SNDRV_DEFAULT_STR1);
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8580: failed to create pcms\n");
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
 		goto pcm_err;
 	}
 
 	snd_soc_add_controls(codec, wm8580_snd_controls,
-				ARRAY_SIZE(wm8580_snd_controls));
+			     ARRAY_SIZE(wm8580_snd_controls));
 	wm8580_add_widgets(codec);
-
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8580: failed to register card\n");
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
 		goto card_err;
 	}
+
 	return ret;
 
 card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
 	return ret;
 }
 
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8580_socdev;
+/* power down chip */
+static int wm8580_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
 
-/*
- * WM8580 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8580 = {
+	.probe = 	wm8580_probe,
+	.remove = 	wm8580_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
+
+static int wm8580_register(struct wm8580_priv *wm8580)
+{
+	int ret, i;
+	struct snd_soc_codec *codec = &wm8580->codec;
+
+	if (wm8580_codec) {
+		dev_err(codec->dev, "Another WM8580 is registered\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm8580;
+	codec->name = "WM8580";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8580_read_reg_cache;
+	codec->write = wm8580_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8580_set_bias_level;
+	codec->dai = wm8580_dai;
+	codec->num_dai = ARRAY_SIZE(wm8580_dai);
+	codec->reg_cache_size = ARRAY_SIZE(wm8580->reg_cache);
+	codec->reg_cache = &wm8580->reg_cache;
+
+	memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg));
+
+	/* Get the codec into a known state */
+	ret = wm8580_write(codec, WM8580_RESET, 0);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to reset codec: %d\n", ret);
+		goto err;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++)
+		wm8580_dai[i].dev = codec->dev;
+
+	wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
+	wm8580_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(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8580);
+	return ret;
+}
+
+static void wm8580_unregister(struct wm8580_priv *wm8580)
+{
+	wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+	snd_soc_unregister_codec(&wm8580->codec);
+	kfree(wm8580);
+	wm8580_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int wm8580_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8580_socdev;
-	struct snd_soc_codec *codec = socdev->card->codec;
-	int ret;
+	struct wm8580_priv *wm8580;
+	struct snd_soc_codec *codec;
+
+	wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
+	if (wm8580 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8580->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
 
-	i2c_set_clientdata(i2c, codec);
+	i2c_set_clientdata(i2c, wm8580);
 	codec->control_data = i2c;
 
-	ret = wm8580_init(socdev);
-	if (ret < 0)
-		dev_err(&i2c->dev, "failed to initialise WM8580\n");
-	return ret;
+	codec->dev = &i2c->dev;
+
+	return wm8580_register(wm8580);
 }
 
 static int wm8580_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+	struct wm8580_priv *wm8580 = i2c_get_clientdata(client);
+	wm8580_unregister(wm8580);
 	return 0;
 }
 
@@ -917,127 +999,35 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
 static struct i2c_driver wm8580_i2c_driver = {
 	.driver = {
-		.name = "WM8580 I2C Codec",
+		.name = "wm8580",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8580_i2c_probe,
 	.remove =   wm8580_i2c_remove,
 	.id_table = wm8580_i2c_id,
 };
+#endif
 
-static int wm8580_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8580_setup_data *setup)
+static int __init wm8580_modinit(void)
 {
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
 	int ret;
 
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&wm8580_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, "wm8580", 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(&wm8580_i2c_driver);
-	return -ENODEV;
-}
-#endif
-
-static int wm8580_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8580_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8580_priv *wm8580;
-	int ret = 0;
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
-	if (wm8580 == NULL) {
-		kfree(codec);
-		return -ENOMEM;
+		pr_err("Failed to register WM8580 I2C driver: %d\n", ret);
 	}
-
-	codec->private_data = wm8580;
-	socdev->card->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8580_socdev = socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8580_add_i2c_device(pdev, setup);
-	}
-#else
-		/* Add other interfaces here */
-#endif
-	return ret;
-}
-
-/* power down chip */
-static int wm8580_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	if (codec->control_data)
-		wm8580_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(&wm8580_i2c_driver);
 #endif
-	kfree(codec->private_data);
-	kfree(codec);
 
 	return 0;
 }
-
-struct snd_soc_codec_device soc_codec_dev_wm8580 = {
-	.probe = 	wm8580_probe,
-	.remove = 	wm8580_remove,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
-
-static int __init wm8580_modinit(void)
-{
-	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
-}
 module_init(wm8580_modinit);
 
 static void __exit wm8580_exit(void)
 {
-	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8580_i2c_driver);
+#endif
 }
 module_exit(wm8580_exit);
 
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 09e4422f6f2f..0dfb5ddde6a2 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -28,11 +28,6 @@
 #define WM8580_CLKSRC_OSC  4
 #define WM8580_CLKSRC_NONE 5
 
-struct wm8580_setup_data {
-	int i2c_bus;
-	unsigned short i2c_address;
-};
-
 #define WM8580_DAI_PAIFRX 0
 #define WM8580_DAI_PAIFTX 1
 
-- 
cgit v1.2.3


From eb5f6d753e337834c7ceb07824ee472e43d9a7a2 Mon Sep 17 00:00:00 2001
From: Philipp Zabel <philipp.zabel@gmail.com>
Date: Thu, 12 Mar 2009 11:07:54 +0100
Subject: ASoC: Replace remaining uses of snd_soc_cnew with
 snd_soc_add_controls.

The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs4270.c          | 23 +++++------------------
 sound/soc/codecs/tlv320aic26.c     | 11 ++++-------
 sound/soc/codecs/wm8400.c          | 12 ++----------
 sound/soc/omap/n810.c              | 12 +++++-------
 sound/soc/pxa/corgi.c              | 12 +++++-------
 sound/soc/pxa/palm27x.c            | 13 +++++--------
 sound/soc/pxa/poodle.c             | 12 +++++-------
 sound/soc/pxa/spitz.c              | 12 +++++-------
 sound/soc/pxa/tosa.c               | 12 +++++-------
 sound/soc/s3c24xx/neo1973_wm8753.c | 13 +++++--------
 10 files changed, 46 insertions(+), 86 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 2137670c9b78..7fa09a387622 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -540,7 +540,6 @@ static int cs4270_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = cs4270_codec;
-	unsigned int i;
 	int ret;
 
 	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
@@ -554,23 +553,11 @@ static int cs4270_probe(struct platform_device *pdev)
 	}
 
 	/* Add the non-DAPM controls */
-	for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
-		struct snd_kcontrol *kctrl;
-
-		kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
-		if (!kctrl) {
-			dev_err(codec->dev, "error creating control '%s'\n",
-			       cs4270_snd_controls[i].name);
-			ret = -ENOMEM;
-			goto error_free_pcms;
-		}
-
-		ret = snd_ctl_add(codec->card, kctrl);
-		if (ret < 0) {
-			dev_err(codec->dev, "error adding control '%s'\n",
-			       cs4270_snd_controls[i].name);
-			goto error_free_pcms;
-		}
+	ret = snd_soc_add_controls(codec, cs4270_snd_controls,
+				ARRAY_SIZE(cs4270_snd_controls));
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to add controls\n");
+		goto error_free_pcms;
 	}
 
 	/* And finally, register the socdev */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index a7f333fc579e..3387d9e736ea 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -324,9 +324,8 @@ static int aic26_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
-	struct snd_kcontrol *kcontrol;
 	struct aic26 *aic26;
-	int i, ret, err;
+	int ret, err;
 
 	dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
 	dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
@@ -353,11 +352,9 @@ static int aic26_probe(struct platform_device *pdev)
 
 	/* register controls */
 	dev_dbg(&pdev->dev, "Registering controls\n");
-	for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
-		kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
-		err = snd_ctl_add(codec->card, kcontrol);
-		WARN_ON(err < 0);
-	}
+	err = snd_soc_add_controls(codec, aic26_snd_controls,
+			ARRAY_SIZE(aic26_snd_controls));
+	WARN_ON(err < 0);
 
 	/* CODEC is setup, we can register the card now */
 	dev_dbg(&pdev->dev, "Registering card\n");
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 4e1cefff8483..744e0dc73be4 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -351,16 +351,8 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
 /* add non dapm controls */
 static int wm8400_add_controls(struct snd_soc_codec *codec)
 {
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8400_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8400_snd_controls[i],codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
+	return snd_soc_add_controls(codec, wm8400_snd_controls,
+				ARRAY_SIZE(wm8400_snd_controls));
 }
 
 /*
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 9f037cd0191d..86471fd6340f 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -248,7 +248,7 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
 
 static int n810_aic33_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	/* Not connected */
 	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
@@ -256,12 +256,10 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "HPRCOM");
 
 	/* Add N810 specific controls */
-	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, aic33_n810_controls,
+				ARRAY_SIZE(aic33_n810_controls));
+	if (err < 0)
+		return err;
 
 	/* Add N810 specific widgets */
 	snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 146973ae0974..02263e5d8f03 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -276,18 +276,16 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
  */
 static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
 	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8731_corgi_controls,
+				ARRAY_SIZE(wm8731_corgi_controls));
+	if (err < 0)
+		return err;
 
 	/* Add corgi specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 29958cd9daec..48a73f64500b 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -146,19 +146,16 @@ static const struct snd_kcontrol_new palm27x_controls[] = {
 
 static int palm27x_ac97_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
 	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add palm27x specific controls */
-	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&palm27x_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, palm27x_controls,
+				ARRAY_SIZE(palm27x_controls));
+	if (err < 0)
+		return err;
 
 	/* add palm27x specific widgets */
 	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index fb17a0a5a093..ef7c6c8dc8f1 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -241,19 +241,17 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
  */
 static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
 	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8731_poodle_controls,
+				ARRAY_SIZE(wm8731_poodle_controls));
+	if (err < 0)
+		return err;
 
 	/* Add poodle specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 1aafd8c645a1..6ca9f53080c6 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -278,7 +278,7 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
  */
 static int spitz_wm8750_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	/* NC codec pins */
 	snd_soc_dapm_nc_pin(codec, "RINPUT1");
@@ -290,12 +290,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8750_spitz_controls,
+				ARRAY_SIZE(wm8750_spitz_controls));
+	if (err < 0)
+		return err;
 
 	/* Add spitz specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 09b5bada03b9..fc781374b1bf 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -188,18 +188,16 @@ static const struct snd_kcontrol_new tosa_controls[] = {
 
 static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
 	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
-	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&tosa_controls[i],codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, tosa_controls,
+				ARRAY_SIZE(tosa_controls));
+	if (err < 0)
+		return err;
 
 	/* add tosa specific widgets */
 	snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 5f6aeec0437d..289fadf60b10 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -498,7 +498,7 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
  */
 static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -518,13 +518,10 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
 
 	/* add neo1973 specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_neo1973_controls[i],
-				codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
+				ARRAY_SIZE(8753_neo1973_controls));
+	if (err < 0)
+		return err;
 
 	/* set up neo1973 specific audio routes */
 	err = snd_soc_dapm_add_routes(codec, dapm_routes,
-- 
cgit v1.2.3


From 10d9e3d99ee8332bb73a3d7f12a8cd8ffab8b136 Mon Sep 17 00:00:00 2001
From: Joonyoung Shim <jy0922.shim@samsung.com>
Date: Mon, 16 Mar 2009 21:23:35 +0900
Subject: ASoC: twl4030 - Fix build error

CC      sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl4030.c | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 86bb15cc82ce..97738e2ece04 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 #define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops twl4030_dai_ops = {
+	.hw_params	= twl4030_hw_params,
+	.set_sysclk	= twl4030_set_dai_sysclk,
+	.set_fmt	= twl4030_set_dai_fmt,
+};
+
 struct snd_soc_dai twl4030_dai = {
 	.name = "twl4030",
 	.playback = {
@@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = {
 		.channels_max = 2,
 		.rates = TWL4030_RATES,
 		.formats = TWL4030_FORMATS,},
-	.ops = {
-		.hw_params = twl4030_hw_params,
-		.set_sysclk = twl4030_set_dai_sysclk,
-		.set_fmt = twl4030_set_dai_fmt,
-	}
+	.ops = &twl4030_dai_ops,
 };
 EXPORT_SYMBOL_GPL(twl4030_dai);
 
-- 
cgit v1.2.3


From e3598f6e4218d1aad3369c97217266b2375e6aca Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Mar 2009 15:19:10 +0000
Subject: ASoC: Further optimise WM8400 bias configuration sequence

The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8400.c | 27 ++-------------------------
 1 file changed, 2 insertions(+), 25 deletions(-)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 744e0dc73be4..462f8b0d9ac7 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1096,45 +1096,22 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
 			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
 				     WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
 
-			/* Enable all output discharge bits */
-			wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
-				WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
-				WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
-				WM8400_DIS_ROUT);
-
 			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
 			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
 				     WM8400_BUFDCOPEN | WM8400_POBCTRL);
 
-			msleep(500);
-
-			/* Enable outputs */
-			val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
-			val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
-				WM8400_OUT4_ENA | WM8400_LOUT_ENA |
-				WM8400_ROUT_ENA;
-			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
-
-			/* disable all output discharge bits */
-			wm8400_write(codec, WM8400_ANTIPOP1, 0);
+			msleep(50);
 
 			/* Enable VREF & VMID at 2x50k */
+			val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
 			val |= 0x2 | WM8400_VREF_ENA;
 			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
 
-			msleep(600);
-
 			/* Enable BUFIOEN */
 			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
 				     WM8400_BUFDCOPEN | WM8400_POBCTRL |
 				     WM8400_BUFIOEN);
 
-			/* Disable outputs */
-			val &= ~(WM8400_SPK_ENA | WM8400_OUT3_ENA |
-				 WM8400_OUT4_ENA | WM8400_LOUT_ENA |
-				 WM8400_ROUT_ENA);
-			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
-
 			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
 		}
-- 
cgit v1.2.3


From 24a51029fc3055f33684e650b5e3a59f77c9b05c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Mar 2009 15:19:48 +0000
Subject: ASoC: Add separate AVDD for WM8400

There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8400.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 462f8b0d9ac7..b7350c25b61c 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -48,6 +48,9 @@ static struct regulator_bulk_data power[] = {
 	{
 		.supply = "DCVDD",
 	},
+	{
+		.supply = "AVDD",
+	},
 	{
 		.supply = "FLLVDD",
 	},
-- 
cgit v1.2.3


From e8523b641cddedec754ae5e44ec579dbceea5ef4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Mar 2009 18:28:01 +0000
Subject: ASoC: Add FLL support for WM8400

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8400.c | 129 ++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 129 insertions(+)

(limited to 'sound/soc/codecs')

diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index b7350c25b61c..510efa604008 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -70,6 +70,7 @@ struct wm8400_priv {
 	unsigned int sysclk;
 	unsigned int pcmclk;
 	struct work_struct work;
+	int fll_in, fll_out;
 };
 
 static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
@@ -931,6 +932,133 @@ static int wm8400_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
+struct fll_factors {
+	u16 n;
+	u16 k;
+	u16 outdiv;
+	u16 fratio;
+	u16 freq_ref;
+};
+
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
+		       unsigned int Fref, unsigned int Fout)
+{
+	u64 Kpart;
+	unsigned int K, Nmod, target;
+
+	factors->outdiv = 2;
+	while (Fout * factors->outdiv <  90000000 ||
+	       Fout * factors->outdiv > 100000000) {
+		factors->outdiv *= 2;
+		if (factors->outdiv > 32) {
+			dev_err(wm8400->wm8400->dev,
+				"Unsupported FLL output frequency %dHz\n",
+				Fout);
+			return -EINVAL;
+		}
+	}
+	target = Fout * factors->outdiv;
+	factors->outdiv = factors->outdiv >> 2;
+
+	if (Fref < 48000)
+		factors->freq_ref = 1;
+	else
+		factors->freq_ref = 0;
+
+	if (Fref < 1000000)
+		factors->fratio = 9;
+	else
+		factors->fratio = 0;
+
+	/* Ensure we have a fractional part */
+	do {
+		if (Fref < 1000000)
+			factors->fratio--;
+		else
+			factors->fratio++;
+
+		if (factors->fratio < 1 || factors->fratio > 8) {
+			dev_err(wm8400->wm8400->dev,
+				"Unable to calculate FRATIO\n");
+			return -EINVAL;
+		}
+
+		factors->n = target / (Fref * factors->fratio);
+		Nmod = target % (Fref * factors->fratio);
+	} while (Nmod == 0);
+
+	/* Calculate fractional part - scale up so we can round. */
+	Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, (Fref * factors->fratio));
+
+	K = Kpart & 0xFFFFFFFF;
+
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	factors->k = K / 10;
+
+	dev_dbg(wm8400->wm8400->dev,
+		"FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+		Fref, Fout,
+		factors->n, factors->k, factors->fratio, factors->outdiv);
+
+	return 0;
+}
+
+static int wm8400_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;
+	struct wm8400_priv *wm8400 = codec->private_data;
+	struct fll_factors factors;
+	int ret;
+	u16 reg;
+
+	if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out)
+		return 0;
+
+	if (freq_out != 0) {
+		ret = fll_factors(wm8400, &factors, freq_in, freq_out);
+		if (ret != 0)
+			return ret;
+	}
+
+	wm8400->fll_out = freq_out;
+	wm8400->fll_in = freq_in;
+
+	/* We *must* disable the FLL before any changes */
+	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2);
+	reg &= ~WM8400_FLL_ENA;
+	wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+	reg = wm8400_read(codec, WM8400_FLL_CONTROL_1);
+	reg &= ~WM8400_FLL_OSC_ENA;
+	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+
+	if (freq_out == 0)
+		return 0;
+
+	reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK);
+	reg |= WM8400_FLL_FRAC | factors.fratio;
+	reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT;
+	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+
+	wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k);
+	wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n);
+
+	reg = wm8400_read(codec, WM8400_FLL_CONTROL_4);
+	reg &= WM8400_FLL_OUTDIV_MASK;
+	reg |= factors.outdiv;
+	wm8400_write(codec, WM8400_FLL_CONTROL_4, reg);
+
+	return 0;
+}
+
 /*
  * Sets ADC and Voice DAC format.
  */
@@ -1188,6 +1316,7 @@ static struct snd_soc_dai_ops wm8400_dai_ops = {
 	.set_fmt = wm8400_set_dai_fmt,
 	.set_clkdiv = wm8400_set_dai_clkdiv,
 	.set_sysclk = wm8400_set_dai_sysclk,
+	.set_pll = wm8400_set_dai_pll,
 };
 
 /*
-- 
cgit v1.2.3