From 0bf79ef2c303cc70d036c9fb355aeb468e8efb62 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:08:52 -0600
Subject: ASoC: wm8903: init GPIOs during I2C probe not codec probe

This allows the GPIOs to be available as soon as the I2C device has
probed, which in turn enables machine drivers to request the GPIOs in
their probe(), rather than deferring this to their ASoC machine init
function, i.e. after the whole sound card has been constructed, and
hence the WM8903 codec is available.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8903.c | 48 ++++++++++++++++++++++-------------------------
 1 file changed, 22 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 86b8a2926591..f6a3fc5f09c0 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2,7 +2,7 @@
  * wm8903.c  --  WM8903 ALSA SoC Audio driver
  *
  * Copyright 2008 Wolfson Microelectronics
- * Copyright 2011 NVIDIA, Inc.
+ * Copyright 2011-2012 NVIDIA, Inc.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -116,6 +116,7 @@ static const struct reg_default wm8903_reg_defaults[] = {
 
 struct wm8903_priv {
 	struct wm8903_platform_data *pdata;
+	struct device *dev;
 	struct snd_soc_codec *codec;
 	struct regmap *regmap;
 
@@ -1774,7 +1775,6 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset)
 static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
 	unsigned int mask, val;
 	int ret;
 
@@ -1782,8 +1782,8 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 	val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) |
 		WM8903_GP1_DIR;
 
-	ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
-				  mask, val);
+	ret = regmap_update_bits(wm8903->regmap,
+				 WM8903_GPIO_CONTROL_1 + offset, mask, val);
 	if (ret < 0)
 		return ret;
 
@@ -1793,10 +1793,9 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
-	int reg;
+	unsigned int reg;
 
-	reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset);
+	regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg);
 
 	return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT;
 }
@@ -1805,7 +1804,6 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
 				     unsigned offset, int value)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
 	unsigned int mask, val;
 	int ret;
 
@@ -1813,8 +1811,8 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
 	val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) |
 		(value << WM8903_GP2_LVL_SHIFT);
 
-	ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
-				  mask, val);
+	ret = regmap_update_bits(wm8903->regmap,
+				 WM8903_GPIO_CONTROL_1 + offset, mask, val);
 	if (ret < 0)
 		return ret;
 
@@ -1824,11 +1822,10 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip,
 static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
 
-	snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
-			    WM8903_GP1_LVL_MASK,
-			    !!value << WM8903_GP1_LVL_SHIFT);
+	regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset,
+			   WM8903_GP1_LVL_MASK,
+			   !!value << WM8903_GP1_LVL_SHIFT);
 }
 
 static struct gpio_chip wm8903_template_chip = {
@@ -1842,15 +1839,14 @@ static struct gpio_chip wm8903_template_chip = {
 	.can_sleep		= 1,
 };
 
-static void wm8903_init_gpio(struct snd_soc_codec *codec)
+static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 {
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	struct wm8903_platform_data *pdata = wm8903->pdata;
 	int ret;
 
 	wm8903->gpio_chip = wm8903_template_chip;
 	wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO;
-	wm8903->gpio_chip.dev = codec->dev;
+	wm8903->gpio_chip.dev = wm8903->dev;
 
 	if (pdata->gpio_base)
 		wm8903->gpio_chip.base = pdata->gpio_base;
@@ -1859,24 +1855,23 @@ static void wm8903_init_gpio(struct snd_soc_codec *codec)
 
 	ret = gpiochip_add(&wm8903->gpio_chip);
 	if (ret != 0)
-		dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
+		dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret);
 }
 
-static void wm8903_free_gpio(struct snd_soc_codec *codec)
+static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 {
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	ret = gpiochip_remove(&wm8903->gpio_chip);
 	if (ret != 0)
-		dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+		dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret);
 }
 #else
-static void wm8903_init_gpio(struct snd_soc_codec *codec)
+static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 {
 }
 
-static void wm8903_free_gpio(struct snd_soc_codec *codec)
+static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 {
 }
 #endif
@@ -2000,8 +1995,6 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
 			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
 
-	wm8903_init_gpio(codec);
-
 	return ret;
 }
 
@@ -2010,7 +2003,6 @@ static int wm8903_remove(struct snd_soc_codec *codec)
 {
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
-	wm8903_free_gpio(codec);
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	if (wm8903->irq)
 		free_irq(wm8903->irq, codec);
@@ -2130,6 +2122,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 			      GFP_KERNEL);
 	if (wm8903 == NULL)
 		return -ENOMEM;
+	wm8903->dev = &i2c->dev;
 
 	wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap);
 	if (IS_ERR(wm8903->regmap)) {
@@ -2189,6 +2182,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 	/* Reset the device */
 	regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903);
 
+	wm8903_init_gpio(wm8903);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8903, &wm8903_dai, 1);
 	if (ret != 0)
@@ -2204,6 +2199,7 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct wm8903_priv *wm8903 = i2c_get_clientdata(client);
 
+	wm8903_free_gpio(wm8903);
 	regmap_exit(wm8903->regmap);
 	snd_soc_unregister_codec(&client->dev);
 
-- 
cgit v1.2.3


From f51022f1aedc4d1a02d0dfa8fde47f6a8291f618 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:08:54 -0600
Subject: ASoC: tegra+wm8903: move all GPIO setup into probe

Now that deferred probe exists, we can parse device tree and request
GPIOs from probe(), rather than deferring this to the DAI link's init().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 162 +++++++++++++++++++++--------------------
 1 file changed, 83 insertions(+), 79 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0b0df49d9d33..a8a3103ab4cb 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -245,80 +245,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
-	struct device_node *np = card->dev->of_node;
-	int ret;
-
-	if (card->dev->platform_data) {
-		memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
-	} else if (np) {
-		/*
-		 * This part must be in init() rather than probe() in order to
-		 * guarantee that the WM8903 has been probed, and hence its
-		 * GPIO controller registered, which is a pre-condition for
-		 * of_get_named_gpio() to be able to map the phandles in the
-		 * properties to the controller node. Given this, all
-		 * pdata handling is in init() for consistency.
-		 */
-		pdata->gpio_spkr_en = of_get_named_gpio(np,
-						"nvidia,spkr-en-gpios", 0);
-		pdata->gpio_hp_mute = of_get_named_gpio(np,
-						"nvidia,hp-mute-gpios", 0);
-		pdata->gpio_hp_det = of_get_named_gpio(np,
-						"nvidia,hp-det-gpios", 0);
-		pdata->gpio_int_mic_en = of_get_named_gpio(np,
-						"nvidia,int-mic-en-gpios", 0);
-		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
-						"nvidia,ext-mic-en-gpios", 0);
-	} else {
-		dev_err(card->dev, "No platform data supplied\n");
-		return -EINVAL;
-	}
-
-	if (gpio_is_valid(pdata->gpio_spkr_en)) {
-		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
-		if (ret) {
-			dev_err(card->dev, "cannot get spkr_en gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_SPKR_EN;
-
-		gpio_direction_output(pdata->gpio_spkr_en, 0);
-	}
-
-	if (gpio_is_valid(pdata->gpio_hp_mute)) {
-		ret = gpio_request(pdata->gpio_hp_mute, "hp_mute");
-		if (ret) {
-			dev_err(card->dev, "cannot get hp_mute gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_HP_MUTE;
-
-		gpio_direction_output(pdata->gpio_hp_mute, 1);
-	}
-
-	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
-		ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
-		if (ret) {
-			dev_err(card->dev, "cannot get int_mic_en gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_INT_MIC_EN;
-
-		/* Disable int mic; enable signal is active-high */
-		gpio_direction_output(pdata->gpio_int_mic_en, 0);
-	}
-
-	if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
-		ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
-		if (ret) {
-			dev_err(card->dev, "cannot get ext_mic_en gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_EXT_MIC_EN;
-
-		/* Enable ext mic; enable signal is active-low */
-		gpio_direction_output(pdata->gpio_ext_mic_en, 0);
-	}
 
 	if (gpio_is_valid(pdata->gpio_hp_det)) {
 		tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det;
@@ -372,8 +298,10 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
 
 static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_tegra_wm8903;
 	struct tegra_wm8903 *machine;
+	struct tegra_wm8903_platform_data *pdata;
 	int ret;
 
 	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
@@ -388,12 +316,42 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		ret = -ENOMEM;
 		goto err;
 	}
+	pdata = &machine->pdata;
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
-	if (pdev->dev.of_node) {
+	if (pdev->dev.platform_data) {
+		memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
+	} else if (np) {
+		pdata->gpio_spkr_en = of_get_named_gpio(np,
+						"nvidia,spkr-en-gpios", 0);
+		if (pdata->gpio_spkr_en == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_hp_mute = of_get_named_gpio(np,
+						"nvidia,hp-mute-gpios", 0);
+		if (pdata->gpio_hp_mute == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_hp_det = of_get_named_gpio(np,
+						"nvidia,hp-det-gpios", 0);
+		if (pdata->gpio_hp_det == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_int_mic_en = of_get_named_gpio(np,
+						"nvidia,int-mic-en-gpios", 0);
+		if (pdata->gpio_int_mic_en == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
+						"nvidia,ext-mic-en-gpios", 0);
+		if (pdata->gpio_ext_mic_en == -ENODEV)
+			return -EPROBE_DEFER;
+	}
+
+	if (np) {
 		ret = snd_soc_of_parse_card_name(card, "nvidia,model");
 		if (ret)
 			goto err;
@@ -404,8 +362,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 			goto err;
 
 		tegra_wm8903_dai.codec_name = NULL;
-		tegra_wm8903_dai.codec_of_node = of_parse_phandle(
-				pdev->dev.of_node, "nvidia,audio-codec", 0);
+		tegra_wm8903_dai.codec_of_node = of_parse_phandle(np,
+				"nvidia,audio-codec", 0);
 		if (!tegra_wm8903_dai.codec_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,audio-codec' missing or invalid\n");
@@ -414,8 +372,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 
 		tegra_wm8903_dai.cpu_dai_name = NULL;
-		tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(
-				pdev->dev.of_node, "nvidia,i2s-controller", 0);
+		tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(np,
+				"nvidia,i2s-controller", 0);
 		if (!tegra_wm8903_dai.cpu_dai_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,i2s-controller' missing or invalid\n");
@@ -442,6 +400,52 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 	}
 
+	if (gpio_is_valid(pdata->gpio_spkr_en)) {
+		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
+		if (ret) {
+			dev_err(card->dev, "cannot get spkr_en gpio\n");
+			return ret;
+		}
+		machine->gpio_requested |= GPIO_SPKR_EN;
+
+		gpio_direction_output(pdata->gpio_spkr_en, 0);
+	}
+
+	if (gpio_is_valid(pdata->gpio_hp_mute)) {
+		ret = gpio_request(pdata->gpio_hp_mute, "hp_mute");
+		if (ret) {
+			dev_err(card->dev, "cannot get hp_mute gpio\n");
+			return ret;
+		}
+		machine->gpio_requested |= GPIO_HP_MUTE;
+
+		gpio_direction_output(pdata->gpio_hp_mute, 1);
+	}
+
+	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
+		ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
+		if (ret) {
+			dev_err(card->dev, "cannot get int_mic_en gpio\n");
+			return ret;
+		}
+		machine->gpio_requested |= GPIO_INT_MIC_EN;
+
+		/* Disable int mic; enable signal is active-high */
+		gpio_direction_output(pdata->gpio_int_mic_en, 0);
+	}
+
+	if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
+		ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
+		if (ret) {
+			dev_err(card->dev, "cannot get ext_mic_en gpio\n");
+			return ret;
+		}
+		machine->gpio_requested |= GPIO_EXT_MIC_EN;
+
+		/* Enable ext mic; enable signal is active-low */
+		gpio_direction_output(pdata->gpio_ext_mic_en, 0);
+	}
+
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
 		goto err;
-- 
cgit v1.2.3


From e2d287c179a12a6069bc3b746e2e34edcddf81b3 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:08:55 -0600
Subject: ASoC: tegra+wm8903: Use devm_gpio_request_one

By using this function, the driver no longer needs to explicitly free
the GPIOs. Hence, we can also remove the flags we use to track whether
we allocated these GPIOs.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 42 ++++++++++--------------------------------
 1 file changed, 10 insertions(+), 32 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index a8a3103ab4cb..5ef2063e0ab1 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -50,10 +50,6 @@
 
 #define DRV_NAME "tegra-snd-wm8903"
 
-#define GPIO_SPKR_EN    BIT(0)
-#define GPIO_HP_MUTE    BIT(1)
-#define GPIO_INT_MIC_EN BIT(2)
-#define GPIO_EXT_MIC_EN BIT(3)
 #define GPIO_HP_DET     BIT(4)
 
 struct tegra_wm8903 {
@@ -401,49 +397,41 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 	}
 
 	if (gpio_is_valid(pdata->gpio_spkr_en)) {
-		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en,
+					    GPIOF_OUT_INIT_LOW, "spkr_en");
 		if (ret) {
 			dev_err(card->dev, "cannot get spkr_en gpio\n");
 			return ret;
 		}
-		machine->gpio_requested |= GPIO_SPKR_EN;
-
-		gpio_direction_output(pdata->gpio_spkr_en, 0);
 	}
 
 	if (gpio_is_valid(pdata->gpio_hp_mute)) {
-		ret = gpio_request(pdata->gpio_hp_mute, "hp_mute");
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute,
+					    GPIOF_OUT_INIT_HIGH, "hp_mute");
 		if (ret) {
 			dev_err(card->dev, "cannot get hp_mute gpio\n");
 			return ret;
 		}
-		machine->gpio_requested |= GPIO_HP_MUTE;
-
-		gpio_direction_output(pdata->gpio_hp_mute, 1);
 	}
 
 	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
-		ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
+		/* Disable int mic; enable signal is active-high */
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en,
+					    GPIOF_OUT_INIT_LOW, "int_mic_en");
 		if (ret) {
 			dev_err(card->dev, "cannot get int_mic_en gpio\n");
 			return ret;
 		}
-		machine->gpio_requested |= GPIO_INT_MIC_EN;
-
-		/* Disable int mic; enable signal is active-high */
-		gpio_direction_output(pdata->gpio_int_mic_en, 0);
 	}
 
 	if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
-		ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
+		/* Enable ext mic; enable signal is active-low */
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en,
+					    GPIOF_OUT_INIT_LOW, "ext_mic_en");
 		if (ret) {
 			dev_err(card->dev, "cannot get ext_mic_en gpio\n");
 			return ret;
 		}
-		machine->gpio_requested |= GPIO_EXT_MIC_EN;
-
-		/* Enable ext mic; enable signal is active-low */
-		gpio_direction_output(pdata->gpio_ext_mic_en, 0);
 	}
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
@@ -469,21 +457,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 {
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
 	if (machine->gpio_requested & GPIO_HP_DET)
 		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
 					1,
 					&tegra_wm8903_hp_jack_gpio);
-	if (machine->gpio_requested & GPIO_EXT_MIC_EN)
-		gpio_free(pdata->gpio_ext_mic_en);
-	if (machine->gpio_requested & GPIO_INT_MIC_EN)
-		gpio_free(pdata->gpio_int_mic_en);
-	if (machine->gpio_requested & GPIO_HP_MUTE)
-		gpio_free(pdata->gpio_hp_mute);
-	if (machine->gpio_requested & GPIO_SPKR_EN)
-		gpio_free(pdata->gpio_spkr_en);
-	machine->gpio_requested = 0;
 
 	snd_soc_unregister_card(card);
 
-- 
cgit v1.2.3


From e44fbbd45896e684d44391aaf881dd3e36bd1a16 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:08:56 -0600
Subject: ASoC: tegra+wm8903: unconditionally free jack GPIOs in remove

The headphone jack GPIOs are added/initialized in the DAI link's init()
method, and hence in theory may not always have been added before remove()
is called in some unusual cases. In order to prevent calling
snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the
code kept track of the initialization state to avoid the free call when
necessary.

However, it appears that snd_soc_jack_free_gpios() is robust in the face
of being called without snd_soc_jack_add_gpios() first succeeding, so
there is little point manually tracking this information. Hence, remove
the tracking code. Almost all other machine drivers already operate this
way.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 5ef2063e0ab1..9059525f3b08 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -50,12 +50,9 @@
 
 #define DRV_NAME "tegra-snd-wm8903"
 
-#define GPIO_HP_DET     BIT(4)
-
 struct tegra_wm8903 {
 	struct tegra_wm8903_platform_data pdata;
 	struct tegra_asoc_utils_data util_data;
-	int gpio_requested;
 };
 
 static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
@@ -252,7 +249,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 		snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
 					1,
 					&tegra_wm8903_hp_jack_gpio);
-		machine->gpio_requested |= GPIO_HP_DET;
 	}
 
 	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
@@ -458,10 +454,8 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev)
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 
-	if (machine->gpio_requested & GPIO_HP_DET)
-		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
-					1,
-					&tegra_wm8903_hp_jack_gpio);
+	snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1,
+				&tegra_wm8903_hp_jack_gpio);
 
 	snd_soc_unregister_card(card);
 
-- 
cgit v1.2.3


From aef9a37c01a63a132d43d65d231dfe515d0f918a Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:09:51 -0600
Subject: ASoC: tegra+alc5632: move all GPIO setup into probe

Now that deferred probe exists, we can parse device tree and request
GPIOs from probe(), rather than deferring this to the DAI link's init().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_alc5632.c | 10 ++++++----
 1 file changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 32de7006daf0..facf6f00c6b0 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -1,5 +1,5 @@
 /*
- * tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
+* tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
  *
  * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
  * Copyright (C) 2012 - NVIDIA, Inc.
@@ -110,7 +110,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	struct device_node *np = codec->card->dev->of_node;
 	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card);
 
 	snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
@@ -119,8 +118,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 			ARRAY_SIZE(tegra_alc5632_hs_jack_pins),
 			tegra_alc5632_hs_jack_pins);
 
-	machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
-
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det;
 		snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack,
@@ -159,6 +156,7 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = {
 
 static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_tegra_alc5632;
 	struct tegra_alc5632 *alc5632;
 	int ret;
@@ -181,6 +179,10 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 		goto err;
 	}
 
+	alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
+	if (alc5632->gpio_hp_det == -ENODEV)
+		return -EPROBE_DEFER;
+
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
 	if (ret)
 		goto err;
-- 
cgit v1.2.3


From 9f6328d910ef8df8176ed433aa2de037eba1f656 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:09:52 -0600
Subject: ASoC: tegra+alc5632: unconditionally free jack GPIOs in remove

The headphone jack GPIOs are added/initialized in the DAI link's init()
method, and hence in theory may not always have been added before remove()
is called in some unusual cases. In order to prevent calling
snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the
code kept track of the initialization state to avoid the free call when
necessary.

However, it appears that snd_soc_jack_free_gpios() is robust in the face
of being called without snd_soc_jack_add_gpios() first succeeding, so
there is little point manually tracking this information. Hence, remove
the tracking code. All other machine drivers already operate this way.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_alc5632.c | 11 ++---------
 1 file changed, 2 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index facf6f00c6b0..15669570ae31 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -33,11 +33,8 @@
 
 #define DRV_NAME "tegra-alc5632"
 
-#define GPIO_HP_DET     BIT(0)
-
 struct tegra_alc5632 {
 	struct tegra_asoc_utils_data util_data;
-	int gpio_requested;
 	int gpio_hp_det;
 };
 
@@ -123,7 +120,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 		snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack,
 						1,
 						&tegra_alc5632_hp_jack_gpio);
-		machine->gpio_requested |= GPIO_HP_DET;
 	}
 
 	snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1");
@@ -236,11 +232,8 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev)
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card);
 
-	if (machine->gpio_requested & GPIO_HP_DET)
-		snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack,
-					1,
-					&tegra_alc5632_hp_jack_gpio);
-	machine->gpio_requested = 0;
+	snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1,
+				&tegra_alc5632_hp_jack_gpio);
 
 	snd_soc_unregister_card(card);
 
-- 
cgit v1.2.3


From 14df415a38234aa483219335bc6c1ee899b85e10 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:08:53 -0600
Subject: ASoC: tegra+wm8903: simplify gpio tests in widget callbacks

By the time any widget callbacks could be called, if the GPIO ID they
will manipulate is valid, it must have already been requested, or the
card would have failed to probe or initialize. So, testing for GPIO
validity is equivalent to testing whether the GPIO was successfully
requested at this point in the code. Making this change will allow later
patches to remove the gpio_requested variable.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 9059525f3b08..1fd6a41b9162 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -153,7 +153,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w,
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (!(machine->gpio_requested & GPIO_SPKR_EN))
+	if (!gpio_is_valid(pdata->gpio_spkr_en))
 		return 0;
 
 	gpio_set_value_cansleep(pdata->gpio_spkr_en,
@@ -170,7 +170,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w,
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (!(machine->gpio_requested & GPIO_HP_MUTE))
+	if (!gpio_is_valid(pdata->gpio_hp_mute))
 		return 0;
 
 	gpio_set_value_cansleep(pdata->gpio_hp_mute,
-- 
cgit v1.2.3


From b350ecbe4c2e4639ed3a716ec67accb744e4417d Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Tue, 22 May 2012 16:11:19 -0600
Subject: ASoC: tegra+wm8903: remove non-DT support for Seaboard

In kernel 3.6, Seaboard will only be supported when booting using device
tree; the board files are being removed. Hence, remove the non-DT support
for Seaboard and derivatives Kaen and Aebl from the audio driver.

Harmony is the only remaining board supported by this driver when not
using DT. This support is currently scheduled for removal in 3.7.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 48 ++----------------------------------------
 1 file changed, 2 insertions(+), 46 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 1fd6a41b9162..b75e0e8db1d0 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -28,8 +28,6 @@
  *
  */
 
-#include <asm/mach-types.h>
-
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -196,37 +194,6 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = {
 	{"IN1L", NULL, "Mic Jack"},
 };
 
-static const struct snd_soc_dapm_route seaboard_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "ROP"},
-	{"Int Spk", NULL, "RON"},
-	{"Int Spk", NULL, "LOP"},
-	{"Int Spk", NULL, "LON"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN1R", NULL, "Mic Jack"},
-};
-
-static const struct snd_soc_dapm_route kaen_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "ROP"},
-	{"Int Spk", NULL, "RON"},
-	{"Int Spk", NULL, "LOP"},
-	{"Int Spk", NULL, "LON"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN2R", NULL, "Mic Jack"},
-};
-
-static const struct snd_soc_dapm_route aebl_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "LINEOUTR"},
-	{"Int Spk", NULL, "LINEOUTL"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN1R", NULL, "Mic Jack"},
-};
-
 static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Int Spk"),
 };
@@ -377,19 +344,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		tegra_wm8903_dai.platform_of_node =
 					tegra_wm8903_dai.cpu_dai_of_node;
 	} else {
-		if (machine_is_harmony()) {
-			card->dapm_routes = harmony_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
-		} else if (machine_is_seaboard()) {
-			card->dapm_routes = seaboard_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map);
-		} else if (machine_is_kaen()) {
-			card->dapm_routes = kaen_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map);
-		} else {
-			card->dapm_routes = aebl_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map);
-		}
+		card->dapm_routes = harmony_audio_map;
+		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
 	}
 
 	if (gpio_is_valid(pdata->gpio_spkr_en)) {
-- 
cgit v1.2.3


From 656baaebf92ae9b16644c7e10a273d8dfe1ba1f6 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 23 May 2012 12:39:07 +0100
Subject: ASoC: codecs: Refresh copyrights for Wolfson drivers

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm2000.c        | 2 +-
 sound/soc/codecs/wm5100-tables.c | 2 +-
 sound/soc/codecs/wm5100.c        | 2 +-
 sound/soc/codecs/wm8350.c        | 2 +-
 sound/soc/codecs/wm8400.c        | 2 +-
 sound/soc/codecs/wm8580.c        | 2 +-
 sound/soc/codecs/wm8731.c        | 1 +
 sound/soc/codecs/wm8741.c        | 2 +-
 sound/soc/codecs/wm8753.c        | 2 +-
 sound/soc/codecs/wm8776.c        | 2 +-
 sound/soc/codecs/wm8804.c        | 2 +-
 sound/soc/codecs/wm8903.c        | 2 +-
 sound/soc/codecs/wm8904.c        | 2 +-
 sound/soc/codecs/wm8960.c        | 2 ++
 sound/soc/codecs/wm8961.c        | 2 ++
 sound/soc/codecs/wm8962.c        | 2 +-
 sound/soc/codecs/wm8993.c        | 2 +-
 sound/soc/codecs/wm8994.c        | 2 +-
 sound/soc/codecs/wm8996.c        | 2 +-
 sound/soc/codecs/wm9081.c        | 2 +-
 sound/soc/codecs/wm9090.c        | 2 +-
 sound/soc/codecs/wm9712.c        | 2 +-
 sound/soc/codecs/wm9713.c        | 2 +-
 sound/soc/codecs/wm_hubs.c       | 2 +-
 24 files changed, 26 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index a75c3766aede..52f0a19217c4 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -1,7 +1,7 @@
 /*
  * wm2000.c  --  WM2000 ALSA Soc Audio driver
  *
- * Copyright 2008-2010 Wolfson Microelectronics PLC.
+ * Copyright 2008-2011 Wolfson Microelectronics PLC.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index e167207a19cc..e239f4bf2460 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -1,7 +1,7 @@
 /*
  * wm5100-tables.c  --  WM5100 ALSA SoC Audio driver data
  *
- * Copyright 2011 Wolfson Microelectronics plc
+ * Copyright 2011-2 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index cb6d5372103a..3823af362912 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1,7 +1,7 @@
 /*
  * wm5100.c  --  WM5100 ALSA SoC Audio driver
  *
- * Copyright 2011 Wolfson Microelectronics plc
+ * Copyright 2011-2 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 555ee146ae0d..e782a5aa2a31 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1,7 +1,7 @@
 /*
  * wm8350.c -- WM8350 ALSA SoC audio driver
  *
- * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ * Copyright (C) 2007-12 Wolfson Microelectronics PLC.
  *
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 5dc31ebcd0e7..5d277a915f81 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1,7 +1,7 @@
 /*
  * wm8400.c  --  WM8400 ALSA Soc Audio driver
  *
- * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Copyright 2008-11 Wolfson Microelectronics PLC.
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 211285164d70..7c68226376e4 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, 2009 Wolfson Microelectronics PLC.
+ * Copyright 2008-11 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
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9d1b9b0271f1..bb1d26919b10 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -2,6 +2,7 @@
  * wm8731.c  --  WM8731 ALSA SoC Audio driver
  *
  * Copyright 2005 Openedhand Ltd.
+ * Copyright 2006-12 Wolfson Microelectronics, plc
  *
  * Author: Richard Purdie <richard@openedhand.com>
  *
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 6e849cb04243..35f3d23200e0 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -1,7 +1,7 @@
 /*
  * wm8741.c  --  WM8741 ALSA SoC Audio driver
  *
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-1 Wolfson Microelectronics plc
  *
  * Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a26482cd7654..13bff87ddcf5 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1,7 +1,7 @@
 /*
  * wm8753.c  --  WM8753 ALSA Soc Audio driver
  *
- * Copyright 2003 Wolfson Microelectronics PLC.
+ * Copyright 2003-11 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index a19db5a0a17a..879c356a9045 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -1,7 +1,7 @@
 /*
  * wm8776.c  --  WM8776 ALSA SoC Audio driver
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6bd1b767b138..c088020172ab 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -1,7 +1,7 @@
 /*
  * wm8804.c  --  WM8804 S/PDIF transceiver driver
  *
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-11 Wolfson Microelectronics plc
  *
  * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index f6a3fc5f09c0..304b5cff3482 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1,7 +1,7 @@
 /*
  * wm8903.c  --  WM8903 ALSA SoC Audio driver
  *
- * Copyright 2008 Wolfson Microelectronics
+ * Copyright 2008-11 Wolfson Microelectronics
  * Copyright 2011-2012 NVIDIA, Inc.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 65d525d74c54..db94d10b5c1a 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1,7 +1,7 @@
 /*
  * wm8904.c  --  WM8904 ALSA SoC Audio driver
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8bc659d8dd2e..96518ac8e24c 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -1,6 +1,8 @@
 /*
  * wm8960.c  --  WM8960 ALSA SoC Audio driver
  *
+ * Copyright 2007-11 Wolfson Microelectronics, plc
+ *
  * Author: Liam Girdwood
  *
  * This program is free software; you can redistribute it and/or modify
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 05ea7c274093..01edbcc754d2 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1,6 +1,8 @@
 /*
  * wm8961.c  --  WM8961 ALSA SoC Audio driver
  *
+ * Copyright 2009-10 Wolfson Microelectronics, plc
+ *
  * Author: Mark Brown
  *
  * This program is free software; you can redistribute it and/or modify
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 0cfce9999c89..27da4d722edc 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1,7 +1,7 @@
 /*
  * wm8962.c  --  WM8962 ALSA SoC Audio driver
  *
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-2 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 36acfccab999..9fd80d688979 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1,7 +1,7 @@
 /*
  * wm8993.c -- WM8993 ALSA SoC audio driver
  *
- * Copyright 2009, 2010 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 993639d694ce..5d4d7df8d339 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1,7 +1,7 @@
 /*
  * wm8994.c  --  WM8994 ALSA SoC Audio driver
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 8af422e38fd0..efc4e9d0903b 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1,7 +1,7 @@
 /*
  * wm8996.c - WM8996 audio codec interface
  *
- * Copyright 2011 Wolfson Microelectronics PLC.
+ * Copyright 2011-2 Wolfson Microelectronics PLC.
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 9328270df16c..2de74e1ea225 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -3,7 +3,7 @@
  *
  * Author: Mark Brown
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 4b263b6edf13..2c2346fdd637 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -1,7 +1,7 @@
 /*
  * ALSA SoC WM9090 driver
  *
- * Copyright 2009, 2010 Wolfson Microelectronics
+ * Copyright 2009-12 Wolfson Microelectronics
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index a1541414d904..099e6ec32125 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -1,7 +1,7 @@
 /*
  * wm9712.c  --  ALSA Soc WM9712 codec support
  *
- * Copyright 2006 Wolfson Microelectronics PLC.
+ * Copyright 2006-12 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 2d22cc70d536..3eb19fb71d17 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1,7 +1,7 @@
 /*
  * wm9713.c  --  ALSA Soc WM9713 codec support
  *
- * Copyright 2006 Wolfson Microelectronics PLC.
+ * Copyright 2006-10 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index dfe957a47f29..61baa48823cb 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1,7 +1,7 @@
 /*
  * wm_hubs.c  --  WM8993/4 common code
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
-- 
cgit v1.2.3


From d7e7eb91551ad99244b989d71d092cb0375648fa Mon Sep 17 00:00:00 2001
From: Ola Lilja <ola.o.lilja@stericsson.com>
Date: Thu, 24 May 2012 15:26:25 +0200
Subject: ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY

Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.

Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 38 ++++++++++++++++++++++++++++++++++++++
 1 file changed, 38 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 90ee77d2409d..3bb7a6f058d0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -35,6 +35,7 @@
 #include <linux/debugfs.h>
 #include <linux/pm_runtime.h>
 #include <linux/regulator/consumer.h>
+#include <linux/clk.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -51,6 +52,7 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_pre] = 0,
 	[snd_soc_dapm_supply] = 1,
 	[snd_soc_dapm_regulator_supply] = 1,
+	[snd_soc_dapm_clock_supply] = 1,
 	[snd_soc_dapm_micbias] = 2,
 	[snd_soc_dapm_dai_link] = 2,
 	[snd_soc_dapm_dai] = 3,
@@ -92,6 +94,7 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_aif_out] = 10,
 	[snd_soc_dapm_dai] = 10,
 	[snd_soc_dapm_dai_link] = 11,
+	[snd_soc_dapm_clock_supply] = 12,
 	[snd_soc_dapm_regulator_supply] = 12,
 	[snd_soc_dapm_supply] = 12,
 	[snd_soc_dapm_post] = 13,
@@ -391,6 +394,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_vmid:
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai:
@@ -764,6 +768,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	switch (widget->id) {
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		return 0;
 	default:
 		break;
@@ -850,6 +855,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 	switch (widget->id) {
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		return 0;
 	default:
 		break;
@@ -996,6 +1002,24 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 }
 EXPORT_SYMBOL_GPL(dapm_regulator_event);
 
+/*
+ * Handler for clock supply widget.
+ */
+int dapm_clock_event(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol, int event)
+{
+	if (!w->clk)
+		return -EIO;
+
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		return clk_enable(w->clk);
+	} else {
+		clk_disable(w->clk);
+		return 0;
+	}
+}
+EXPORT_SYMBOL_GPL(dapm_clock_event);
+
 static int dapm_widget_power_check(struct snd_soc_dapm_widget *w)
 {
 	if (w->power_checked)
@@ -1487,6 +1511,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power,
 	switch (w->id) {
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		/* Supplies can't affect their outputs, only their inputs */
 		break;
 	default:
@@ -1587,6 +1612,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 				break;
 			case snd_soc_dapm_supply:
 			case snd_soc_dapm_regulator_supply:
+			case snd_soc_dapm_clock_supply:
 			case snd_soc_dapm_micbias:
 				if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
 					d->target_bias_level = SND_SOC_BIAS_STANDBY;
@@ -1941,6 +1967,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 		case snd_soc_dapm_mixer_named_ctl:
 		case snd_soc_dapm_supply:
 		case snd_soc_dapm_regulator_supply:
+		case snd_soc_dapm_clock_supply:
 			if (w->name)
 				count += sprintf(buf + count, "%s: %s\n",
 					w->name, w->power ? "On":"Off");
@@ -2187,6 +2214,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_post:
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai:
@@ -2873,6 +2901,15 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 			return NULL;
 		}
 		break;
+	case snd_soc_dapm_clock_supply:
+		w->clk = clk_get(dapm->dev, w->name);
+		if (IS_ERR(w->clk)) {
+			ret = PTR_ERR(w->clk);
+			dev_err(dapm->dev, "Failed to request %s: %d\n",
+				w->name, ret);
+			return NULL;
+		}
+		break;
 	default:
 		break;
 	}
@@ -2924,6 +2961,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		break;
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		w->power_check = dapm_supply_check_power;
 		break;
 	case snd_soc_dapm_dai:
-- 
cgit v1.2.3


From 01a0c1139c2bd075d005253093e7060022c5d0cb Mon Sep 17 00:00:00 2001
From: Ola Lilja <ola.o.lilja@stericsson.com>
Date: Thu, 24 May 2012 15:26:32 +0200
Subject: ASoC: Ux500: Add platform-driver

Add platform-driver handling all DMA-activities.

Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/ux500/Kconfig     |   7 +
 sound/soc/ux500/Makefile    |   3 +
 sound/soc/ux500/ux500_pcm.c | 318 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/ux500/ux500_pcm.h |  35 +++++
 4 files changed, 363 insertions(+)
 create mode 100644 sound/soc/ux500/ux500_pcm.c
 create mode 100644 sound/soc/ux500/ux500_pcm.h

(limited to 'sound')

diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
index 44cf43404cd9..1d385150064f 100644
--- a/sound/soc/ux500/Kconfig
+++ b/sound/soc/ux500/Kconfig
@@ -12,3 +12,10 @@ menuconfig SND_SOC_UX500
 config SND_SOC_UX500_PLAT_MSP_I2S
 	tristate
 	depends on SND_SOC_UX500
+
+config SND_SOC_UX500_PLAT_DMA
+	tristate "Platform - DB8500 (DMA)"
+	depends on SND_SOC_UX500
+	select SND_SOC_DMAENGINE_PCM
+	help
+		Say Y if you want to enable the Ux500 platform-driver.
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
index 19974c5a2ea1..4634bf015f62 100644
--- a/sound/soc/ux500/Makefile
+++ b/sound/soc/ux500/Makefile
@@ -2,3 +2,6 @@
 
 snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o
 obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o
+
+snd-soc-ux500-plat-dma-objs := ux500_pcm.o
+obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
new file mode 100644
index 000000000000..66b080e5de96
--- /dev/null
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -0,0 +1,318 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <asm/page.h>
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/slab.h>
+
+#include <plat/ste_dma40.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "ux500_msp_i2s.h"
+#include "ux500_pcm.h"
+
+static struct snd_pcm_hardware ux500_pcm_hw_playback = {
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_RESUME |
+		SNDRV_PCM_INFO_PAUSE,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE |
+		SNDRV_PCM_FMTBIT_U16_LE |
+		SNDRV_PCM_FMTBIT_S16_BE |
+		SNDRV_PCM_FMTBIT_U16_BE,
+	.rates = SNDRV_PCM_RATE_KNOT,
+	.rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK,
+	.rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK,
+	.channels_min = UX500_PLATFORM_MIN_CHANNELS,
+	.channels_max = UX500_PLATFORM_MAX_CHANNELS,
+	.buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
+	.period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
+	.period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
+	.periods_min = UX500_PLATFORM_PERIODS_MIN,
+	.periods_max = UX500_PLATFORM_PERIODS_MAX,
+};
+
+static struct snd_pcm_hardware ux500_pcm_hw_capture = {
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_RESUME |
+		SNDRV_PCM_INFO_PAUSE,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE |
+		SNDRV_PCM_FMTBIT_U16_LE |
+		SNDRV_PCM_FMTBIT_S16_BE |
+		SNDRV_PCM_FMTBIT_U16_BE,
+	.rates = SNDRV_PCM_RATE_KNOT,
+	.rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE,
+	.rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE,
+	.channels_min = UX500_PLATFORM_MIN_CHANNELS,
+	.channels_max = UX500_PLATFORM_MAX_CHANNELS,
+	.buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
+	.period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
+	.period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
+	.periods_min = UX500_PLATFORM_PERIODS_MIN,
+	.periods_max = UX500_PLATFORM_PERIODS_MAX,
+};
+
+static void ux500_pcm_dma_hw_free(struct device *dev,
+				struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+
+	if (runtime->dma_area == NULL)
+		return;
+
+	if (buf != &substream->dma_buffer) {
+		dma_free_coherent(buf->dev.dev, buf->bytes, buf->area,
+				buf->addr);
+		kfree(runtime->dma_buffer_p);
+	}
+
+	snd_pcm_set_runtime_buffer(substream, NULL);
+}
+
+static int ux500_pcm_open(struct snd_pcm_substream *substream)
+{
+	int stream_id = substream->pstr->stream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct device *dev = dai->dev;
+	int ret;
+	struct ux500_msp_dma_params *dma_params;
+	u16 per_data_width, mem_data_width;
+	struct stedma40_chan_cfg *dma_cfg;
+
+	dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id,
+		snd_pcm_stream_str(substream));
+
+	dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__);
+	if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_set_runtime_hwparams(substream,
+					&ux500_pcm_hw_playback);
+	else
+		snd_soc_set_runtime_hwparams(substream,
+					&ux500_pcm_hw_capture);
+
+	/* ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0) {
+		dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n",
+			__func__, ret);
+		return ret;
+	}
+
+	dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__,
+		snd_pcm_stream_str(substream));
+	runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ?
+		ux500_pcm_hw_playback : ux500_pcm_hw_capture;
+
+	mem_data_width = STEDMA40_HALFWORD_WIDTH;
+
+	dma_params = snd_soc_dai_get_dma_data(dai, substream);
+	switch (dma_params->data_size) {
+	case 32:
+		per_data_width = STEDMA40_WORD_WIDTH;
+		break;
+	case 16:
+		per_data_width = STEDMA40_HALFWORD_WIDTH;
+		break;
+	case 8:
+		per_data_width = STEDMA40_BYTE_WIDTH;
+		break;
+	default:
+		per_data_width = STEDMA40_WORD_WIDTH;
+		dev_warn(rtd->platform->dev,
+			"%s: Unknown data-size (%d)! Assuming 32 bits.\n",
+			__func__, dma_params->data_size);
+	}
+
+	dma_cfg = dma_params->dma_cfg;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dma_cfg->src_info.data_width = mem_data_width;
+		dma_cfg->dst_info.data_width = per_data_width;
+	} else {
+		dma_cfg->src_info.data_width = per_data_width;
+		dma_cfg->dst_info.data_width = mem_data_width;
+	}
+
+
+	ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg);
+	if (ret) {
+		dev_dbg(dai->dev,
+			"%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	snd_dmaengine_pcm_set_data(substream, dma_cfg);
+
+	return 0;
+}
+
+static int ux500_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+
+	dev_dbg(dai->dev, "%s: Enter\n", __func__);
+
+	snd_dmaengine_pcm_close(substream);
+
+	return 0;
+}
+
+static int ux500_pcm_hw_params(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int ret = 0;
+	int size;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__);
+
+	size = params_buffer_bytes(hw_params);
+
+	if (buf) {
+		if (buf->bytes >= size)
+			goto out;
+		ux500_pcm_dma_hw_free(NULL, substream);
+	}
+
+	if (substream->dma_buffer.area != NULL &&
+		substream->dma_buffer.bytes >= size) {
+		buf = &substream->dma_buffer;
+	} else {
+		buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL);
+		if (!buf)
+			goto nomem;
+
+		buf->dev.type = SNDRV_DMA_TYPE_DEV;
+		buf->dev.dev = NULL;
+		buf->area = dma_alloc_coherent(NULL, size, &buf->addr,
+					GFP_KERNEL);
+		buf->bytes = size;
+		buf->private_data = NULL;
+
+		if (!buf->area)
+			goto free;
+	}
+	snd_pcm_set_runtime_buffer(substream, buf);
+	ret = 1;
+ out:
+	runtime->dma_bytes = size;
+	return ret;
+
+ free:
+	kfree(buf);
+ nomem:
+	return -ENOMEM;
+}
+
+static int ux500_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__);
+
+	ux500_pcm_dma_hw_free(NULL, substream);
+
+	return 0;
+}
+
+static int ux500_pcm_mmap(struct snd_pcm_substream *substream,
+			struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__);
+
+	return dma_mmap_coherent(NULL, vma, runtime->dma_area,
+				runtime->dma_addr, runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops ux500_pcm_ops = {
+	.open		= ux500_pcm_open,
+	.close		= ux500_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= ux500_pcm_hw_params,
+	.hw_free	= ux500_pcm_hw_free,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= ux500_pcm_mmap
+};
+
+int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm *pcm = rtd->pcm;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__,
+		pcm->id);
+
+	pcm->info_flags = 0;
+
+	return 0;
+}
+
+static struct snd_soc_platform_driver ux500_pcm_soc_drv = {
+	.ops		= &ux500_pcm_ops,
+	.pcm_new        = ux500_pcm_new,
+};
+
+static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv);
+	if (ret < 0) {
+		dev_err(&pdev->dev,
+			"%s: ERROR: Failed to register platform '%s' (%d)!\n",
+			__func__, pdev->name, ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver ux500_pcm_driver = {
+	.driver = {
+		.name = "ux500-pcm",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = ux500_pcm_drv_probe,
+	.remove = __devexit_p(ux500_pcm_drv_remove),
+};
+module_platform_driver(ux500_pcm_driver);
+
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h
new file mode 100644
index 000000000000..77ed44d371e9
--- /dev/null
+++ b/sound/soc/ux500/ux500_pcm.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef UX500_PCM_H
+#define UX500_PCM_H
+
+#include <asm/page.h>
+
+#include <linux/workqueue.h>
+
+#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000
+#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000
+#define UX500_PLATFORM_MIN_RATE_CAPTURE	8000
+#define UX500_PLATFORM_MAX_RATE_CAPTURE	48000
+
+#define UX500_PLATFORM_MIN_CHANNELS 1
+#define UX500_PLATFORM_MAX_CHANNELS 8
+
+#define UX500_PLATFORM_PERIODS_BYTES_MIN	128
+#define UX500_PLATFORM_PERIODS_BYTES_MAX	(64 * PAGE_SIZE)
+#define UX500_PLATFORM_PERIODS_MIN		2
+#define UX500_PLATFORM_PERIODS_MAX		48
+#define UX500_PLATFORM_BUFFER_BYTES_MAX		(2048 * PAGE_SIZE)
+
+#endif
-- 
cgit v1.2.3


From 5514efdfe0384576ef38c66b1672b6826696fbf3 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 28 May 2012 23:29:36 -0700
Subject: ASoC: fsi: use dmaengine helper functions

This patch used dmaengine helper functions instead of using hand setting.
And reduced local variables

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/sh/fsi.c | 19 ++++---------------
 1 file changed, 4 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 2ef98536f1da..fcaa6b8abb0c 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1089,13 +1089,10 @@ static void fsi_dma_do_tasklet(unsigned long data)
 {
 	struct fsi_stream *io = (struct fsi_stream *)data;
 	struct fsi_priv *fsi = fsi_stream_to_priv(io);
-	struct dma_chan *chan;
 	struct snd_soc_dai *dai;
 	struct dma_async_tx_descriptor *desc;
-	struct scatterlist sg;
 	struct snd_pcm_runtime *runtime;
 	enum dma_data_direction dir;
-	dma_cookie_t cookie;
 	int is_play = fsi_stream_is_play(fsi, io);
 	int len;
 	dma_addr_t buf;
@@ -1104,7 +1101,6 @@ static void fsi_dma_do_tasklet(unsigned long data)
 		return;
 
 	dai	= fsi_get_dai(io->substream);
-	chan	= io->chan;
 	runtime	= io->substream->runtime;
 	dir	= is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
 	len	= samples_to_bytes(runtime, io->period_samples);
@@ -1112,14 +1108,8 @@ static void fsi_dma_do_tasklet(unsigned long data)
 
 	dma_sync_single_for_device(dai->dev, buf, len, dir);
 
-	sg_init_table(&sg, 1);
-	sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)),
-		    len , offset_in_page(buf));
-	sg_dma_address(&sg) = buf;
-	sg_dma_len(&sg) = len;
-
-	desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir,
-				       DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+	desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+					   DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
 		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
 		return;
@@ -1128,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data)
 	desc->callback		= fsi_dma_complete;
 	desc->callback_param	= io;
 
-	cookie = desc->tx_submit(desc);
-	if (cookie < 0) {
+	if (dmaengine_submit(desc) < 0) {
 		dev_err(dai->dev, "tx_submit() fail\n");
 		return;
 	}
 
-	dma_async_issue_pending(chan);
+	dma_async_issue_pending(io->chan);
 
 	/*
 	 * FIXME
-- 
cgit v1.2.3


From b1226dc59d55ecde7fc9a338d8cb2a313821fac0 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 24 May 2012 23:56:19 -0700
Subject: ASoC: fsi: use PIO handler if DMA handler was invalid

PIO handler is not good performance, but works on all platform.
So, switch to PIO handler if DMA handler was invalid case.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/sh/fsi.c | 29 ++++++++++++++++++++---------
 1 file changed, 20 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index fcaa6b8abb0c..53486ff9c2af 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -247,7 +247,7 @@ struct fsi_priv {
 struct fsi_stream_handler {
 	int (*init)(struct fsi_priv *fsi, struct fsi_stream *io);
 	int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io);
-	int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io);
+	int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev);
 	int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io);
 	int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io);
 	void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io,
@@ -571,16 +571,16 @@ static int fsi_stream_transfer(struct fsi_stream *io)
 #define fsi_stream_stop(fsi, io)\
 	fsi_stream_handler_call(io, start_stop, fsi, io, 0)
 
-static int fsi_stream_probe(struct fsi_priv *fsi)
+static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev)
 {
 	struct fsi_stream *io;
 	int ret1, ret2;
 
 	io = &fsi->playback;
-	ret1 = fsi_stream_handler_call(io, probe, fsi, io);
+	ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev);
 
 	io = &fsi->capture;
-	ret2 = fsi_stream_handler_call(io, probe, fsi, io);
+	ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev);
 
 	if (ret1 < 0)
 		return ret1;
@@ -1173,7 +1173,7 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io,
 		fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
 }
 
-static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
+static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev)
 {
 	dma_cap_mask_t mask;
 
@@ -1181,8 +1181,19 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
 	dma_cap_set(DMA_SLAVE, mask);
 
 	io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave);
-	if (!io->chan)
-		return -EIO;
+	if (!io->chan) {
+
+		/* switch to PIO handler */
+		if (fsi_stream_is_play(fsi, io))
+			fsi->playback.handler	= &fsi_pio_push_handler;
+		else
+			fsi->capture.handler	= &fsi_pio_pop_handler;
+
+		dev_info(dev, "switch handler (dma => pio)\n");
+
+		/* probe again */
+		return fsi_stream_probe(fsi, dev);
+	}
 
 	tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
 
@@ -1672,7 +1683,7 @@ static int fsi_probe(struct platform_device *pdev)
 	master->fsia.master	= master;
 	master->fsia.info	= &info->port_a;
 	fsi_handler_init(&master->fsia);
-	ret = fsi_stream_probe(&master->fsia);
+	ret = fsi_stream_probe(&master->fsia, &pdev->dev);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "FSIA stream probe failed\n");
 		goto exit_iounmap;
@@ -1683,7 +1694,7 @@ static int fsi_probe(struct platform_device *pdev)
 	master->fsib.master	= master;
 	master->fsib.info	= &info->port_b;
 	fsi_handler_init(&master->fsib);
-	ret = fsi_stream_probe(&master->fsib);
+	ret = fsi_stream_probe(&master->fsib, &pdev->dev);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "FSIB stream probe failed\n");
 		goto exit_fsia;
-- 
cgit v1.2.3


From 14a95fe865c0b2ede6f386f52413f6396c010833 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jarkko.nikula@bitmer.com>
Date: Mon, 28 May 2012 22:09:02 +0300
Subject: ASoC: tlv320aic3x: Change Class-D amplifier gain control name

ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output
gain control as a playback control. Fix this by changing the name as
"Class-D Playback Volume".

Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 64d2a4fa34b2..58ef59dfbae9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -368,7 +368,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
 
 static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
-	SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
+	SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
 
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
-- 
cgit v1.2.3


From 0561c1bf354c4a8230a1e0ada43362f54e60b2f0 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 30 May 2012 13:20:17 +0100
Subject: ASoC: ac97: Remove empty remove() function

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ac97.c | 6 ------
 1 file changed, 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2023c749f232..ea06b834a7de 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -91,11 +91,6 @@ static int ac97_soc_probe(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int ac97_soc_remove(struct snd_soc_codec *codec)
-{
-	return 0;
-}
-
 #ifdef CONFIG_PM
 static int ac97_soc_suspend(struct snd_soc_codec *codec)
 {
@@ -119,7 +114,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = {
 	.write =	ac97_write,
 	.read =		ac97_read,
 	.probe = 	ac97_soc_probe,
-	.remove = 	ac97_soc_remove,
 	.suspend =	ac97_soc_suspend,
 	.resume =	ac97_soc_resume,
 };
-- 
cgit v1.2.3


From 51cc7ed3e378a60a3413a7e424f536e4dec3f39d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 31 May 2012 14:48:07 +0100
Subject: ASoC: wm2000: Add register readability information

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm2000.c | 30 ++++++++++++++++++++++++++++++
 1 file changed, 30 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 52f0a19217c4..78a148f0a8ef 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -691,9 +691,39 @@ static int wm2000_resume(struct snd_soc_codec *codec)
 #define wm2000_resume NULL
 #endif
 
+static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case WM2000_REG_SYS_START:
+	case WM2000_REG_SPEECH_CLARITY:
+	case WM2000_REG_SYS_WATCHDOG:
+	case WM2000_REG_ANA_VMID_PD_TIME:
+	case WM2000_REG_ANA_VMID_PU_TIME:
+	case WM2000_REG_CAT_FLTR_INDX:
+	case WM2000_REG_CAT_GAIN_0:
+	case WM2000_REG_SYS_STATUS:
+	case WM2000_REG_SYS_MODE_CNTRL:
+	case WM2000_REG_SYS_START0:
+	case WM2000_REG_SYS_START1:
+	case WM2000_REG_ID1:
+	case WM2000_REG_ID2:
+	case WM2000_REG_REVISON:
+	case WM2000_REG_SYS_CTL1:
+	case WM2000_REG_SYS_CTL2:
+	case WM2000_REG_ANC_STAT:
+	case WM2000_REG_IF_CTL:
+		return true;
+	default:
+		return false;
+	}
+}
+
 static const struct regmap_config wm2000_regmap = {
 	.reg_bits = 8,
 	.val_bits = 8,
+
+	.max_register = WM2000_REG_IF_CTL,
+	.readable_reg = wm2000_readable_reg,
 };
 
 static int wm2000_probe(struct snd_soc_codec *codec)
-- 
cgit v1.2.3


From 210cb67cb5b9f9a23b7ce91de50bab357440ba9d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 8 May 2012 17:46:36 +0100
Subject: ASoC: io: Use dev_get_regmap() if driver doesn't provide a regmap

Less error prone and one less line of code in drivers.

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

(limited to 'sound')

diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 4d8dc6a27d4d..44d0174b4d97 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -142,6 +142,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 	case SND_SOC_REGMAP:
 		/* Device has made its own regmap arrangements */
 		codec->using_regmap = true;
+		if (!codec->control_data)
+			codec->control_data = dev_get_regmap(codec->dev, NULL);
 
 		ret = regmap_get_val_bytes(codec->control_data);
 		/* Errors are legitimate for non-integer byte multiples */
-- 
cgit v1.2.3


From bc92657a11c0982783979bbb84ceaf58ba222124 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Fri, 25 May 2012 18:22:11 -0600
Subject: ASoC: make snd_soc_dai_link more symmetrical

Prior to this patch, the CPU side of a DAI link was specified using a
single name. Often, this was the result of calling dev_name() on the
device providing the DAI, but in the case of a CPU DAI driver that
provided multiple DAIs, it needed to mix together both the device name
and some device-relative name, in order to form a single globally unique
name.

However, the CODEC side of the DAI link was specified using separate
fields for device (name or OF node) and device-relative DAI name.

This patch allows the CPU side of a DAI link to be specified in the same
way as the CODEC side, separating concepts of device and device-relative
DAI name.

I believe this will be important in multi-codec and/or dynamic PCM
scenarios, where a single CPU driver provides multiple DAIs, while also
booting using device tree, with accompanying desire not to hard-code the
CPU side device's name into the original .cpu_dai_name field.

Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link()
would now be identical. However, two things prevent that at present:

1) The need to save rtd->codec for the CODEC side, which means we have
to search for the CODEC explicitly, and not just the CODEC side DAI.

2) Since we know the CODEC side DAI is part of a codec, and not just
a standalone DAI, it's slightly more efficient to convert .codec_name/
.codec_of_node into a codec first, and then compare each DAI's .codec
field, since this avoids strcmp() on each DAI's CODEC's name within
the loop.

However, the two loops are essentially semantically equivalent.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/mxs/mxs-sgtl5000.c    |  2 +-
 sound/soc/soc-core.c            | 42 ++++++++++++++++++++++++++++++-----------
 sound/soc/tegra/tegra_alc5632.c |  6 +++---
 sound/soc/tegra/tegra_wm8753.c  |  6 +++---
 sound/soc/tegra/tegra_wm8903.c  |  6 +++---
 sound/soc/tegra/trimslice.c     |  6 +++---
 6 files changed, 44 insertions(+), 24 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 3e6e8764b2e6..215113b05f7d 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -133,7 +133,7 @@ static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev)
 		mxs_sgtl5000_dai[i].codec_name = NULL;
 		mxs_sgtl5000_dai[i].codec_of_node = codec_np;
 		mxs_sgtl5000_dai[i].cpu_dai_name = NULL;
-		mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i];
+		mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i];
 		mxs_sgtl5000_dai[i].platform_name = NULL;
 		mxs_sgtl5000_dai[i].platform_of_node = saif_np[i];
 	}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b37ee8077ed1..ec8350570346 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -812,13 +812,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 
 	/* Find CPU DAI from registered DAIs*/
 	list_for_each_entry(cpu_dai, &dai_list, list) {
-		if (dai_link->cpu_dai_of_node) {
-			if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node)
-				continue;
-		} else {
-			if (strcmp(cpu_dai->name, dai_link->cpu_dai_name))
-				continue;
-		}
+		if (dai_link->cpu_of_node &&
+		    (cpu_dai->dev->of_node != dai_link->cpu_of_node))
+			continue;
+		if (dai_link->cpu_name &&
+		    strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name))
+			continue;
+		if (dai_link->cpu_dai_name &&
+		    strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+			continue;
 
 		rtd->cpu_dai = cpu_dai;
 	}
@@ -3346,6 +3348,12 @@ int snd_soc_register_card(struct snd_soc_card *card)
 				link->name);
 			return -EINVAL;
 		}
+		/* Codec DAI name must be specified */
+		if (!link->codec_dai_name) {
+			dev_err(card->dev, "codec_dai_name not set for %s\n",
+				link->name);
+			return -EINVAL;
+		}
 
 		/*
 		 * Platform may be specified by either name or OF node, but
@@ -3358,12 +3366,24 @@ int snd_soc_register_card(struct snd_soc_card *card)
 		}
 
 		/*
-		 * CPU DAI must be specified by 1 of name or OF node,
-		 * not both or neither.
+		 * CPU device may be specified by either name or OF node, but
+		 * can be left unspecified, and will be matched based on DAI
+		 * name alone..
+		 */
+		if (link->cpu_name && link->cpu_of_node) {
+			dev_err(card->dev,
+				"Neither/both cpu name/of_node are set for %s\n",
+				link->name);
+			return -EINVAL;
+		}
+		/*
+		 * At least one of CPU DAI name or CPU device name/node must be
+		 * specified
 		 */
-		if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) {
+		if (!link->cpu_dai_name &&
+		    !(link->cpu_name || link->cpu_of_node)) {
 			dev_err(card->dev,
-				"Neither/both cpu_dai name/of_node are set for %s\n",
+				"Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
 				link->name);
 			return -EINVAL;
 		}
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 15669570ae31..417b09b83fdf 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -197,16 +197,16 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle(
+	tegra_alc5632_dai.cpu_of_node = of_parse_phandle(
 			pdev->dev.of_node, "nvidia,i2s-controller", 0);
-	if (!tegra_alc5632_dai.cpu_dai_of_node) {
+	if (!tegra_alc5632_dai.cpu_of_node) {
 		dev_err(&pdev->dev,
 		"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
 		goto err;
 	}
 
-	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node;
+	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
 	if (ret)
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index 4e77026807a2..02bd5a8e8544 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -157,9 +157,9 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle(
+	tegra_wm8753_dai.cpu_of_node = of_parse_phandle(
 			pdev->dev.of_node, "nvidia,i2s-controller", 0);
-	if (!tegra_wm8753_dai.cpu_dai_of_node) {
+	if (!tegra_wm8753_dai.cpu_of_node) {
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
@@ -167,7 +167,7 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev)
 	}
 
 	tegra_wm8753_dai.platform_of_node =
-				tegra_wm8753_dai.cpu_dai_of_node;
+				tegra_wm8753_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index b75e0e8db1d0..1fd71e5a9eb9 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -331,9 +331,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 
 		tegra_wm8903_dai.cpu_dai_name = NULL;
-		tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(np,
+		tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np,
 				"nvidia,i2s-controller", 0);
-		if (!tegra_wm8903_dai.cpu_dai_of_node) {
+		if (!tegra_wm8903_dai.cpu_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,i2s-controller' missing or invalid\n");
 			ret = -EINVAL;
@@ -342,7 +342,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 
 		tegra_wm8903_dai.platform_name = NULL;
 		tegra_wm8903_dai.platform_of_node =
-					tegra_wm8903_dai.cpu_dai_of_node;
+					tegra_wm8903_dai.cpu_of_node;
 	} else {
 		card->dapm_routes = harmony_audio_map;
 		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 4a8d5b672c9f..5815430e8521 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -162,9 +162,9 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
 		}
 
 		trimslice_tlv320aic23_dai.cpu_dai_name = NULL;
-		trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle(
+		trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle(
 				pdev->dev.of_node, "nvidia,i2s-controller", 0);
-		if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) {
+		if (!trimslice_tlv320aic23_dai.cpu_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,i2s-controller' missing or invalid\n");
 			ret = -EINVAL;
@@ -173,7 +173,7 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev)
 
 		trimslice_tlv320aic23_dai.platform_name = NULL;
 		trimslice_tlv320aic23_dai.platform_of_node =
-				trimslice_tlv320aic23_dai.cpu_dai_of_node;
+				trimslice_tlv320aic23_dai.cpu_of_node;
 	}
 
 	ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev);
-- 
cgit v1.2.3


From 6c9d8cf6372ed2995a3d982f5c1f966e842101cc Mon Sep 17 00:00:00 2001
From: Adam Thomson <Adam.Thomson@diasemi.com>
Date: Thu, 31 May 2012 15:18:01 +0100
Subject: ASoC: core: Add single controls with specified range of values

Control type added for cases where a specific range of values
within a register are required for control.

Added convenience macros:

SOC_SINGLE_RANGE
SOC_SINGLE_RANGE_TLV

Added accessor implementations:

snd_soc_info_volsw_range
snd_soc_put_volsw_range
snd_soc_get_volsw_range

Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com>
Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 98 ++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 98 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ec8350570346..3d803f3cd272 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2791,6 +2791,104 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 
+/**
+ * snd_soc_info_volsw_range - single mixer info callback with range.
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information, within a range, about a single
+ * mixer control.
+ *
+ * returns 0 for success.
+ */
+int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int platform_max;
+	int min = mc->min;
+
+	if (!mc->platform_max)
+		mc->platform_max = mc->max;
+	platform_max = mc->platform_max;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = platform_max - min;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range);
+
+/**
+ * snd_soc_put_volsw_range - single mixer put value callback with range.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to set the value, within a range, for a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	int min = mc->min;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
+	unsigned int val, val_mask;
+
+	val = ((ucontrol->value.integer.value[0] + min) & mask);
+	if (invert)
+		val = max - val;
+	val_mask = mask << shift;
+	val = val << shift;
+
+	return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
+
+/**
+ * snd_soc_get_volsw_range - single mixer get callback with range
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to get the value, within a range, of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	int min = mc->min;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	if (invert)
+		ucontrol->value.integer.value[0] =
+			max - ucontrol->value.integer.value[0];
+	ucontrol->value.integer.value[0] =
+		ucontrol->value.integer.value[0] - min;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
+
 /**
  * snd_soc_limit_volume - Set new limit to an existing volume control.
  *
-- 
cgit v1.2.3


From f59fef441753cdd07ffe7268b0801ec48cac7b1d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 30 Apr 2012 20:26:41 +0100
Subject: ASoC: wm8350: Convert to direct regmap API usage

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8350.c | 20 +++-----------------
 1 file changed, 3 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index e782a5aa2a31..d26c8ae4e6d9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -71,20 +71,6 @@ struct wm8350_data {
 	int fll_freq_in;
 };
 
-static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	struct wm8350 *wm8350 = codec->control_data;
-	return wm8350_reg_read(wm8350, reg);
-}
-
-static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	struct wm8350 *wm8350 = codec->control_data;
-	return wm8350_reg_write(wm8350, reg, value);
-}
-
 /*
  * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
  */
@@ -1519,7 +1505,9 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
-	codec->control_data = wm8350;
+	codec->control_data = wm8350->regmap;
+
+	snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
 
 	/* Put the codec into reset if it wasn't already */
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
@@ -1629,8 +1617,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8350 = {
 	.remove =	wm8350_codec_remove,
 	.suspend = 	wm8350_suspend,
 	.resume =	wm8350_resume,
-	.read = wm8350_codec_read,
-	.write = wm8350_codec_write,
 	.set_bias_level = wm8350_set_bias_level,
 
 	.controls = wm8350_snd_controls,
-- 
cgit v1.2.3


From 695594f1b79d3b88e99e28f06afaab32c4d65853 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 4 Jun 2012 08:14:13 +0100
Subject: ASoC: dapm: Use devm_clk_get()

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

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3bb7a6f058d0..a66379accec9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2902,7 +2902,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		}
 		break;
 	case snd_soc_dapm_clock_supply:
-		w->clk = clk_get(dapm->dev, w->name);
+		w->clk = devm_clk_get(dapm->dev, w->name);
 		if (IS_ERR(w->clk)) {
 			ret = PTR_ERR(w->clk);
 			dev_err(dapm->dev, "Failed to request %s: %d\n",
-- 
cgit v1.2.3


From ec02995adad5a7b428f46c1a87fae1bc93d6dfe3 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 4 Jun 2012 08:16:20 +0100
Subject: ASoC: dapm: Bodge for lack of a widely available clk API

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
 sound/soc/soc-dapm.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a66379accec9..39e8c2fdf50e 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1011,12 +1011,14 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w,
 	if (!w->clk)
 		return -EIO;
 
+#ifdef CONFIG_HAVE_CLK
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
 		return clk_enable(w->clk);
 	} else {
 		clk_disable(w->clk);
 		return 0;
 	}
+#endif
 }
 EXPORT_SYMBOL_GPL(dapm_clock_event);
 
@@ -2902,6 +2904,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		}
 		break;
 	case snd_soc_dapm_clock_supply:
+#ifdef CONFIG_HAVE_CLK
 		w->clk = devm_clk_get(dapm->dev, w->name);
 		if (IS_ERR(w->clk)) {
 			ret = PTR_ERR(w->clk);
@@ -2909,6 +2912,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 				w->name, ret);
 			return NULL;
 		}
+#else
+		return NULL;
+#endif
 		break;
 	default:
 		break;
-- 
cgit v1.2.3


From 014e5b56702575c5cd8ffc4b1a7924cfdfe0f1ea Mon Sep 17 00:00:00 2001
From: Richard Zhao <richard.zhao@freescale.com>
Date: Mon, 4 Jun 2012 09:42:53 +0800
Subject: ASoC: fsl_ssi: convert to use devm_clk_get

Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/fsl_ssi.c | 7 ++-----
 1 file changed, 2 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 4ed2afd47782..b10a427a8098 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -725,7 +725,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 		u32 dma_events[2];
 		ssi_private->ssi_on_imx = true;
 
-		ssi_private->clk = clk_get(&pdev->dev, NULL);
+		ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
 		if (IS_ERR(ssi_private->clk)) {
 			ret = PTR_ERR(ssi_private->clk);
 			dev_err(&pdev->dev, "could not get clock: %d\n", ret);
@@ -842,10 +842,8 @@ error_dev:
 	device_remove_file(&pdev->dev, dev_attr);
 
 error_clk:
-	if (ssi_private->ssi_on_imx) {
+	if (ssi_private->ssi_on_imx)
 		clk_disable_unprepare(ssi_private->clk);
-		clk_put(ssi_private->clk);
-	}
 
 error_irq:
 	free_irq(ssi_private->irq, ssi_private);
@@ -871,7 +869,6 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 	if (ssi_private->ssi_on_imx) {
 		platform_device_unregister(ssi_private->imx_pcm_pdev);
 		clk_disable_unprepare(ssi_private->clk);
-		clk_put(ssi_private->clk);
 	}
 	snd_soc_unregister_dai(&pdev->dev);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
-- 
cgit v1.2.3


From 7376bde8945fe20d35aa51f493a7e43b60a39dbe Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Sun, 3 Jun 2012 22:50:18 +0530
Subject: ASoC: cs42l52: Remove version.h header file inclusion

version.h header file is no longer needed.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs42l52.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index a7109413aef1..ec03abc79a9a 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -14,7 +14,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
-- 
cgit v1.2.3


From 2bce133c3b00020f4bc146cea94ff5d4de9a8a0f Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Sun, 3 Jun 2012 22:58:40 +0530
Subject: ASoC: lm49453: Remove version.h header file inclusion

version.h header file is no longer required.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/lm49453.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 802b9f176b16..c1bc9458906b 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -12,7 +12,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
-- 
cgit v1.2.3


From 2f989f7e9f5f9ba97535fa58f4240ec250d6b2df Mon Sep 17 00:00:00 2001
From: M R Swami Reddy <MR.Swami.Reddy@ti.com>
Date: Fri, 1 Jun 2012 22:21:54 +0530
Subject: ASoC: Support TI Isabelle Audio driver

ASoC: Support TI Isabelle Audio driver

The Isabelle Audio IC is a complete low power high fidelity CODEC with integrated
ADCs, DACs, decimation and interpolation filters, PLL, and power providers. This
device supports 2 analog and 2 digital microphone channels, a mono earpiece driver,
stereo class G headphone drivers with ultra low power and best SNR in the industry,
stereo Class D speaker drivers, and 2 high performance Line drivers.

The below patch is a basic driver code for TI Isabelle audio codec. The
functionalities like headset detection, etc., will be included incrementally
in the up-coming patches.

Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig    |    4 +
 sound/soc/codecs/Makefile   |    2 +
 sound/soc/codecs/isabelle.c | 1179 +++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/isabelle.h |  143 ++++++
 4 files changed, 1328 insertions(+)
 create mode 100644 sound/soc/codecs/isabelle.c
 create mode 100644 sound/soc/codecs/isabelle.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1e1613a438dd..8b879c71c23c 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
 	select SND_SOC_DFBMCS320
+	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
 	select SND_SOC_LM49453 if I2C
@@ -225,6 +226,9 @@ config SND_SOC_DFBMCS320
 config SND_SOC_DMIC
 	tristate
 
+config SND_SOC_ISABELLE
+        tristate
+
 config SND_SOC_LM49453
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fc27fec39487..e50811b182a4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -23,6 +23,7 @@ snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
 snd-soc-dfbmcs320-objs := dfbmcs320.o
 snd-soc-dmic-objs := dmic.o
+snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
 snd-soc-lm4857-objs := lm4857.o
@@ -134,6 +135,7 @@ obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
new file mode 100644
index 000000000000..b6921a82fbcc
--- /dev/null
+++ b/sound/soc/codecs/isabelle.c
@@ -0,0 +1,1179 @@
+/*
+ * isabelle.c - Low power high fidelity audio codec driver
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * 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.
+ *
+ *
+ * Initially based on sound/soc/codecs/twl6040.c
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+#include "isabelle.h"
+
+
+/* Register default values for ISABELLE driver. */
+static struct reg_default isabelle_reg_defs[] = {
+	{ 0, 0x00 },
+	{ 1, 0x00 },
+	{ 2, 0x00 },
+	{ 3, 0x00 },
+	{ 4, 0x00 },
+	{ 5, 0x00 },
+	{ 6, 0x00 },
+	{ 7, 0x00 },
+	{ 8, 0x00 },
+	{ 9, 0x00 },
+	{ 10, 0x00 },
+	{ 11, 0x00 },
+	{ 12, 0x00 },
+	{ 13, 0x00 },
+	{ 14, 0x00 },
+	{ 15, 0x00 },
+	{ 16, 0x00 },
+	{ 17, 0x00 },
+	{ 18, 0x00 },
+	{ 19, 0x00 },
+	{ 20, 0x00 },
+	{ 21, 0x02 },
+	{ 22, 0x02 },
+	{ 23, 0x02 },
+	{ 24, 0x02 },
+	{ 25, 0x0F },
+	{ 26, 0x8F },
+	{ 27, 0x0F },
+	{ 28, 0x8F },
+	{ 29, 0x00 },
+	{ 30, 0x00 },
+	{ 31, 0x00 },
+	{ 32, 0x00 },
+	{ 33, 0x00 },
+	{ 34, 0x00 },
+	{ 35, 0x00 },
+	{ 36, 0x00 },
+	{ 37, 0x00 },
+	{ 38, 0x00 },
+	{ 39, 0x00 },
+	{ 40, 0x00 },
+	{ 41, 0x00 },
+	{ 42, 0x00 },
+	{ 43, 0x00 },
+	{ 44, 0x00 },
+	{ 45, 0x00 },
+	{ 46, 0x00 },
+	{ 47, 0x00 },
+	{ 48, 0x00 },
+	{ 49, 0x00 },
+	{ 50, 0x00 },
+	{ 51, 0x00 },
+	{ 52, 0x00 },
+	{ 53, 0x00 },
+	{ 54, 0x00 },
+	{ 55, 0x00 },
+	{ 56, 0x00 },
+	{ 57, 0x00 },
+	{ 58, 0x00 },
+	{ 59, 0x00 },
+	{ 60, 0x00 },
+	{ 61, 0x00 },
+	{ 62, 0x00 },
+	{ 63, 0x00 },
+	{ 64, 0x00 },
+	{ 65, 0x00 },
+	{ 66, 0x00 },
+	{ 67, 0x00 },
+	{ 68, 0x00 },
+	{ 69, 0x90 },
+	{ 70, 0x90 },
+	{ 71, 0x90 },
+	{ 72, 0x00 },
+	{ 73, 0x00 },
+	{ 74, 0x00 },
+	{ 75, 0x00 },
+	{ 76, 0x00 },
+	{ 77, 0x00 },
+	{ 78, 0x00 },
+	{ 79, 0x00 },
+	{ 80, 0x00 },
+	{ 81, 0x00 },
+	{ 82, 0x00 },
+	{ 83, 0x00 },
+	{ 84, 0x00 },
+	{ 85, 0x07 },
+	{ 86, 0x00 },
+	{ 87, 0x00 },
+	{ 88, 0x00 },
+	{ 89, 0x07 },
+	{ 90, 0x80 },
+	{ 91, 0x07 },
+	{ 92, 0x07 },
+	{ 93, 0x00 },
+	{ 94, 0x00 },
+	{ 95, 0x00 },
+	{ 96, 0x00 },
+	{ 97, 0x00 },
+	{ 98, 0x00 },
+	{ 99, 0x00 },
+};
+
+static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"};
+static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"};
+
+static const struct soc_enum isabelle_rx1_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts),
+};
+
+static const struct soc_enum isabelle_rx2_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts),
+};
+
+/* Headset DAC playback switches */
+static const struct snd_kcontrol_new rx1_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_rx1_enum);
+
+static const struct snd_kcontrol_new rx2_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_rx2_enum);
+
+/* TX input selection */
+static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"};
+static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"};
+
+static const struct soc_enum isabelle_atx_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts),
+};
+
+static const struct soc_enum isabelle_vtx_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts),
+};
+
+static const struct snd_kcontrol_new atx_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_atx_enum);
+
+static const struct snd_kcontrol_new vtx_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_vtx_enum);
+
+/* Left analog microphone selection */
+static const char *isabelle_amic1_texts[] = {
+	"Main Mic", "Headset Mic", "Aux/FM Left"};
+
+/* Left analog microphone selection */
+static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"};
+
+static const struct soc_enum isabelle_amic1_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5,
+			ARRAY_SIZE(isabelle_amic1_texts),
+			isabelle_amic1_texts),
+};
+
+static const struct soc_enum isabelle_amic2_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4,
+			ARRAY_SIZE(isabelle_amic2_texts),
+			isabelle_amic2_texts),
+};
+
+static const struct snd_kcontrol_new amic1_control =
+	SOC_DAPM_ENUM("Route", isabelle_amic1_enum);
+
+static const struct snd_kcontrol_new amic2_control =
+	SOC_DAPM_ENUM("Route", isabelle_amic2_enum);
+
+static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"};
+
+static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"};
+
+static const struct soc_enum isabelle_st_audio_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1,
+			isabelle_st_audio_texts),
+	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1,
+			isabelle_st_audio_texts),
+};
+
+static const struct soc_enum isabelle_st_voice_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1,
+			isabelle_st_voice_texts),
+	SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1,
+			isabelle_st_voice_texts),
+};
+
+static const struct snd_kcontrol_new st_audio_control =
+	SOC_DAPM_ENUM("Route", isabelle_st_audio_enum);
+
+static const struct snd_kcontrol_new st_voice_control =
+	SOC_DAPM_ENUM("Route", isabelle_st_voice_enum);
+
+/* Mixer controls */
+static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new ep_path_enable_control =
+	SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0);
+
+/* TLV Declarations */
+static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0);
+static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0);
+
+/* from -63 to 0 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1);
+
+/* from -63 to 9 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1);
+static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0);
+
+static const struct snd_kcontrol_new isabelle_snd_controls[] = {
+	SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG,
+			4, 0, 0xF, 0, dac_tlv),
+	SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+			ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG,
+			0, 0x1F, 0, hf_tlv),
+	SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG,
+			4, 0, 0xF, 0, dac_tlv),
+	SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG,
+			0, 0xF, 0, dac_tlv),
+
+	SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0,
+			afm_amp_tlv),
+	SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F,
+			0, mic_amp_tlv),
+	SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F,
+			0, mic_amp_tlv),
+
+	SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG,
+			ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+	SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG,
+			ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+	SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG,
+			ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+
+	SOC_SINGLE_TLV("Sidetone Audio TX1 Volume",
+			ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv),
+	SOC_SINGLE_TLV("Sidetone Audio TX2 Volume",
+			ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv),
+	SOC_SINGLE_TLV("Sidetone Voice TX1 Volume",
+			ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv),
+	SOC_SINGLE_TLV("Sidetone Voice TX2 Volume",
+			ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv),
+
+	SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+	SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+	SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+	SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+
+	SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+
+	SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0),
+	SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0),
+
+	SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		7, 1, 0),
+	SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		6, 1, 0),
+	SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		5, 1, 0),
+	SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		4, 1, 0),
+	SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		3, 1, 0),
+	SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		2, 1, 0),
+	SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		1, 1, 0),
+	SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		0, 1, 0),
+	SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		3, 1, 0),
+	SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		2, 1, 0),
+
+	SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		7, 1, 0),
+	SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		6, 1, 0),
+	SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		5, 1, 0),
+	SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		4, 1, 0),
+	SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		5, 1, 0),
+	SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		4, 1, 0),
+	SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		3, 1, 0),
+	SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		2, 1, 0),
+	SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		1, 1, 0),
+	SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		0, 1, 0),
+
+	SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG,
+		7, 1, 0),
+
+	SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG,
+		7, 1, 0),
+	SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG,
+		7, 1, 0),
+	SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG,
+		7, 1, 0),
+
+	/* DMIC Switch */
+	SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = {
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MAINMIC"),
+	SND_SOC_DAPM_INPUT("HSMIC"),
+	SND_SOC_DAPM_INPUT("SUBMIC"),
+	SND_SOC_DAPM_INPUT("LINEIN1"),
+	SND_SOC_DAPM_INPUT("LINEIN2"),
+	SND_SOC_DAPM_INPUT("DMICDAT"),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HSOL"),
+	SND_SOC_DAPM_OUTPUT("HSOR"),
+	SND_SOC_DAPM_OUTPUT("HFL"),
+	SND_SOC_DAPM_OUTPUT("HFR"),
+	SND_SOC_DAPM_OUTPUT("EP"),
+	SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+	SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+
+	SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Analog input muxes for the capture amplifiers */
+	SND_SOC_DAPM_MUX("Analog Left Capture Route",
+			SND_SOC_NOPM, 0, 0, &amic1_control),
+	SND_SOC_DAPM_MUX("Analog Right Capture Route",
+			SND_SOC_NOPM, 0, 0, &amic2_control),
+
+	SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0,
+			&st_audio_control),
+	SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0,
+			&st_voice_control),
+
+	/* AIF */
+	SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0),
+	SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0),
+
+	SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0),
+	SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0),
+
+	SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Analog Capture PGAs */
+	SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0),
+
+	/* Auxiliary FM PGAs */
+	SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC1", "Left Front Capture",
+			ISABELLE_AMIC_CFG_REG, 7, 0),
+	SND_SOC_DAPM_ADC("ADC2", "Right Front Capture",
+			ISABELLE_AMIC_CFG_REG, 6, 0),
+
+	/* Microphone Bias */
+	SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG,
+			3, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG,
+			2, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias",
+			ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias",
+			ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0),
+
+	/* Mixers */
+	SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hs_left_mixer_controls,
+			ARRAY_SIZE(isabelle_hs_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hs_right_mixer_controls,
+			ARRAY_SIZE(isabelle_hs_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hf_left_mixer_controls,
+			ARRAY_SIZE(isabelle_hf_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hf_right_mixer_controls,
+			ARRAY_SIZE(isabelle_hf_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_aux_left_mixer_controls,
+			ARRAY_SIZE(isabelle_aux_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_aux_right_mixer_controls,
+			ARRAY_SIZE(isabelle_aux_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_ep_mixer_controls,
+			ARRAY_SIZE(isabelle_ep_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga1_left_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga1_right_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga2_left_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga2_right_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga3_left_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga3_right_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx1_mixer_controls,
+			ARRAY_SIZE(isabelle_rx1_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx2_mixer_controls,
+			ARRAY_SIZE(isabelle_rx2_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx3_mixer_controls,
+			ARRAY_SIZE(isabelle_rx3_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx4_mixer_controls,
+			ARRAY_SIZE(isabelle_rx4_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx5_mixer_controls,
+			ARRAY_SIZE(isabelle_rx5_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx6_mixer_controls,
+			ARRAY_SIZE(isabelle_rx6_mixer_controls)),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG,
+			5, 0),
+	SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG,
+			4, 0),
+	SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG,
+			3, 0),
+	SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG,
+			2, 0),
+	SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG,
+			1, 0),
+	SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG,
+			0, 0),
+
+	/* Analog Playback PGAs */
+	SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Analog Playback Mux */
+	SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0,
+			&rx1_mux_controls),
+	SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0,
+			&rx2_mux_controls),
+
+	/* TX Select */
+	SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG,
+			7, 0, &atx_mux_controls),
+	SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG,
+			6, 0, &vtx_mux_controls),
+
+	SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0,
+			&ep_path_enable_control),
+
+	/* Output Drivers */
+	SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG,
+			1, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG,
+			0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route isabelle_intercon[] = {
+	/* Interface mapping */
+	{ "DL1", "DL12 Playback Switch", "INTF1_SDI" },
+	{ "DL2", "DL12 Playback Switch", "INTF1_SDI" },
+	{ "DL3", "DL34 Playback Switch", "INTF1_SDI" },
+	{ "DL4", "DL34 Playback Switch", "INTF1_SDI" },
+	{ "DL5", "DL56 Playback Switch", "INTF1_SDI" },
+	{ "DL6", "DL56 Playback Switch", "INTF1_SDI" },
+
+	{ "DL1", "DL12 Playback Switch", "INTF2_SDI" },
+	{ "DL2", "DL12 Playback Switch", "INTF2_SDI" },
+	{ "DL3", "DL34 Playback Switch", "INTF2_SDI" },
+	{ "DL4", "DL34 Playback Switch", "INTF2_SDI" },
+	{ "DL5", "DL56 Playback Switch", "INTF2_SDI" },
+	{ "DL6", "DL56 Playback Switch", "INTF2_SDI" },
+
+	/* Input side mapping */
+	{ "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" },
+	{ "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" },
+
+	{ "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" },
+
+	{ "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX1 Mixer", "DL1 Playback Switch", "DL1" },
+
+	{ "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" },
+
+	{ "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX2 Mixer", "DL2 Playback Switch", "DL2" },
+
+	{ "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX3 Mixer", "DL3 Playback Switch", "DL3" },
+
+	{ "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX4 Mixer", "DL4 Playback Switch", "DL4" },
+
+	{ "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX5 Mixer", "DL5 Playback Switch", "DL5" },
+
+	{ "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX6 Mixer", "DL6 Playback Switch", "DL6" },
+
+	/* Capture path */
+	{ "Analog Left Capture Route", "Headset Mic", "HSMIC" },
+	{ "Analog Left Capture Route", "Main Mic", "MAINMIC" },
+	{ "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" },
+
+	{ "Analog Right Capture Route", "Sub Mic", "SUBMIC" },
+	{ "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" },
+
+	{ "MicAmp1", NULL, "Analog Left Capture Route" },
+	{ "MicAmp2", NULL, "Analog Right Capture Route" },
+
+	{ "ADC1", NULL, "MicAmp1" },
+	{ "ADC2", NULL, "MicAmp2" },
+
+	{ "ATX Select", "AMIC1", "ADC1" },
+	{ "ATX Select", "DMIC", "DMICDAT" },
+	{ "ATX Select", "AMIC2", "ADC2" },
+
+	{ "VTX Select", "AMIC1", "ADC1" },
+	{ "VTX Select", "DMIC", "DMICDAT" },
+	{ "VTX Select", "AMIC2", "ADC2" },
+
+	{ "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" },
+	{ "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" },
+	{ "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" },
+	{ "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" },
+
+	{ "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" },
+	{ "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" },
+	{ "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" },
+	{ "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" },
+
+	{ "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" },
+	{ "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" },
+	{ "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" },
+	{ "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" },
+
+	{ "INTF1_SDO", NULL, "ULVTX1" },
+	{ "INTF1_SDO", NULL, "ULVTX2" },
+	{ "INTF2_SDO", NULL, "ULVTX1" },
+	{ "INTF2_SDO", NULL, "ULVTX2" },
+
+	/* AFM Path */
+	{ "APGA1", NULL, "LINEIN1" },
+	{ "APGA2", NULL, "LINEIN2" },
+
+	{ "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" },
+	{ "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" },
+	{ "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" },
+
+	{ "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" },
+	{ "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" },
+	{ "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" },
+
+	{ "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" },
+	{ "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" },
+
+	{ "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" },
+	{ "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" },
+
+	{ "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" },
+	{ "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" },
+
+	{ "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" },
+	{ "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" },
+
+	{ "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+	{ "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+	{ "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+
+	{ "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA1L", NULL, "DPGA1L Mixer" },
+	{ "DPGA1R", NULL, "DPGA1R Mixer" },
+
+	{ "DAC1L", NULL, "DPGA1L" },
+	{ "DAC1R", NULL, "DPGA1R" },
+
+	{ "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+	{ "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+	{ "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+	{ "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA2L", NULL, "DPGA2L Mixer" },
+	{ "DPGA2R", NULL, "DPGA2R Mixer" },
+
+	{ "DAC2L", NULL, "DPGA2L" },
+	{ "DAC2R", NULL, "DPGA2R" },
+
+	{ "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+	{ "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+	{ "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+
+	{ "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA3L", NULL, "DPGA3L Mixer" },
+	{ "DPGA3R", NULL, "DPGA3R Mixer" },
+
+	{ "DAC3L", NULL, "DPGA3L" },
+	{ "DAC3R", NULL, "DPGA3R" },
+
+	{ "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" },
+	{ "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" },
+	{ "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" },
+
+	{ "HS Left Driver", NULL, "Headset Left Mixer" },
+	{ "HS Right Driver", NULL, "Headset Right Mixer" },
+
+	{ "HSOL", NULL, "HS Left Driver" },
+	{ "HSOR", NULL, "HS Right Driver" },
+
+	/* Earphone playback path */
+	{ "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" },
+	{ "Earphone Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "Earphone Playback", "Switch", "Earphone Mixer" },
+	{ "Earphone Driver", NULL, "Earphone Playback" },
+	{ "EP", NULL, "Earphone Driver" },
+
+	{ "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" },
+	{ "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" },
+	{ "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" },
+
+	{ "HF Left PGA", NULL, "Handsfree Left Mixer" },
+	{ "HF Right PGA", NULL, "Handsfree Right Mixer" },
+
+	{ "HF Left Driver", NULL, "HF Left PGA" },
+	{ "HF Right Driver", NULL, "HF Right PGA" },
+
+	{ "HFL", NULL, "HF Left Driver" },
+	{ "HFR", NULL, "HF Right Driver" },
+
+	{ "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" },
+	{ "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" },
+	{ "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" },
+
+	{ "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" },
+	{ "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" },
+
+	{ "LINEOUT1", NULL, "LINEOUT1 Driver" },
+	{ "LINEOUT2", NULL, "LINEOUT2 Driver" },
+};
+
+static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG,
+			BIT(4), (mute ? BIT(4) : 0));
+
+	return 0;
+}
+
+static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG,
+			BIT(4), (mute ? BIT(4) : 0));
+
+	return 0;
+}
+
+static int isabelle_line_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG,
+			BIT(4), (mute ? BIT(4) : 0));
+
+	return 0;
+}
+
+static int isabelle_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG,
+				ISABELLE_CHIP_EN, BIT(0));
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG,
+				ISABELLE_CHIP_EN, 0);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static int isabelle_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_codec *codec = rtd->codec;
+	u16 aif = 0;
+	unsigned int fs_val = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+		fs_val = ISABELLE_FS_RATE_8;
+		break;
+	case 11025:
+		fs_val = ISABELLE_FS_RATE_11;
+		break;
+	case 12000:
+		fs_val = ISABELLE_FS_RATE_12;
+		break;
+	case 16000:
+		fs_val = ISABELLE_FS_RATE_16;
+		break;
+	case 22050:
+		fs_val = ISABELLE_FS_RATE_22;
+		break;
+	case 24000:
+		fs_val = ISABELLE_FS_RATE_24;
+		break;
+	case 32000:
+		fs_val = ISABELLE_FS_RATE_32;
+		break;
+	case 44100:
+		fs_val = ISABELLE_FS_RATE_44;
+		break;
+	case 48000:
+		fs_val = ISABELLE_FS_RATE_48;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG,
+			ISABELLE_FS_RATE_MASK, fs_val);
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		aif |= ISABELLE_AIF_LENGTH_20;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		aif |= ISABELLE_AIF_LENGTH_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG,
+			ISABELLE_AIF_LENGTH_MASK, aif);
+
+	return 0;
+}
+
+static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	unsigned int aif_val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aif_val &= ~ISABELLE_AIF_MS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aif_val |= ISABELLE_AIF_MS;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		aif_val |= ISABELLE_I2S_MODE;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		aif_val |= ISABELLE_LEFT_J_MODE;
+		break;
+	case SND_SOC_DAIFMT_PDM:
+		aif_val |= ISABELLE_PDM_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG,
+			(ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val);
+
+	return 0;
+}
+
+/* Rates supported by Isabelle driver */
+#define ISABELLE_RATES		SNDRV_PCM_RATE_8000_48000
+
+/* Formates supported by Isabelle driver. */
+#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops isabelle_hs_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+	.digital_mute	= isabelle_hs_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_hf_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+	.digital_mute	= isabelle_hf_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_line_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+	.digital_mute	= isabelle_line_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_ul_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+};
+
+/* ISABELLE dai structure */
+struct snd_soc_dai_driver isabelle_dai[] = {
+	{
+		.name = "isabelle-dl1",
+		.playback = {
+			.stream_name = "Headset Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_hs_dai_ops,
+	},
+	{
+		.name = "isabelle-dl2",
+		.playback = {
+			.stream_name = "Handsfree Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_hf_dai_ops,
+	},
+	{
+		.name = "isabelle-lineout",
+		.playback = {
+			.stream_name = "Lineout Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_line_dai_ops,
+	},
+	{
+		.name = "isabelle-ul",
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_ul_dai_ops,
+	},
+};
+
+static int isabelle_probe(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+
+	codec->control_data = dev_get_regmap(codec->dev, NULL);
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_isabelle = {
+	.probe = isabelle_probe,
+	.set_bias_level = isabelle_set_bias_level,
+	.controls = isabelle_snd_controls,
+	.num_controls = ARRAY_SIZE(isabelle_snd_controls),
+	.dapm_widgets = isabelle_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets),
+	.dapm_routes = isabelle_intercon,
+	.num_dapm_routes = ARRAY_SIZE(isabelle_intercon),
+	.idle_bias_off = true,
+};
+
+static const struct regmap_config isabelle_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = ISABELLE_MAX_REGISTER,
+	.reg_defaults = isabelle_reg_defs,
+	.num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs),
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
+					const struct i2c_device_id *id)
+{
+	struct regmap *isabelle_regmap;
+	int ret = 0;
+
+	i2c_set_clientdata(i2c, isabelle_regmap);
+
+	isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config);
+	if (IS_ERR(isabelle_regmap)) {
+		ret = PTR_ERR(isabelle_regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+
+	ret =  snd_soc_register_codec(&i2c->dev,
+				&soc_codec_dev_isabelle, isabelle_dai,
+				ARRAY_SIZE(isabelle_dai));
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		regmap_exit(dev_get_regmap(&i2c->dev, NULL));
+		return ret;
+	}
+
+	return ret;
+}
+
+static int __devexit isabelle_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(dev_get_regmap(&client->dev, NULL));
+	return 0;
+}
+
+static const struct i2c_device_id isabelle_i2c_id[] = {
+	{ "isabelle", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id);
+
+static struct i2c_driver isabelle_i2c_driver = {
+	.driver = {
+		.name = "isabelle",
+		.owner = THIS_MODULE,
+	},
+	.probe = isabelle_i2c_probe,
+	.remove = __devexit_p(isabelle_i2c_remove),
+	.id_table = isabelle_i2c_id,
+};
+
+module_i2c_driver(isabelle_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ISABELLE driver");
+MODULE_AUTHOR("Vishwas A Deshpande <vishwas.a.deshpande@ti.com>");
+MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h
new file mode 100644
index 000000000000..96d839a8c956
--- /dev/null
+++ b/sound/soc/codecs/isabelle.h
@@ -0,0 +1,143 @@
+/*
+ * isabelle.h - Low power high fidelity audio codec driver header file
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * 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.
+ *
+ */
+
+#ifndef _ISABELLE_H
+#define _ISABELLE_H
+
+#include <linux/bitops.h>
+
+/* ISABELLE REGISTERS */
+
+#define ISABELLE_PWR_CFG_REG		0x01
+#define ISABELLE_PWR_EN_REG		0x02
+#define ISABELLE_PS_EN1_REG		0x03
+#define ISABELLE_INT1_STATUS_REG	0x04
+#define ISABELLE_INT1_MASK_REG		0x05
+#define ISABELLE_INT2_STATUS_REG	0x06
+#define ISABELLE_INT2_MASK_REG		0x07
+#define ISABELLE_HKCTL1_REG		0x08
+#define ISABELLE_HKCTL2_REG		0x09
+#define ISABELLE_HKCTL3_REG		0x0A
+#define ISABELLE_ACCDET_STATUS_REG	0x0B
+#define ISABELLE_BUTTON_ID_REG		0x0C
+#define ISABELLE_PLL_CFG_REG		0x10
+#define ISABELLE_PLL_EN_REG		0x11
+#define ISABELLE_FS_RATE_CFG_REG	0x12
+#define ISABELLE_INTF_CFG_REG		0x13
+#define ISABELLE_INTF_EN_REG		0x14
+#define ISABELLE_ULATX12_INTF_CFG_REG	0x15
+#define ISABELLE_DL12_INTF_CFG_REG	0x16
+#define ISABELLE_DL34_INTF_CFG_REG	0x17
+#define ISABELLE_DL56_INTF_CFG_REG	0x18
+#define ISABELLE_ATX_STPGA1_CFG_REG	0x19
+#define ISABELLE_ATX_STPGA2_CFG_REG	0x1A
+#define ISABELLE_VTX_STPGA1_CFG_REG	0x1B
+#define ISABELLE_VTX2_STPGA2_CFG_REG	0x1C
+#define ISABELLE_ATX1_DPGA_REG		0x1D
+#define ISABELLE_ATX2_DPGA_REG		0x1E
+#define ISABELLE_VTX1_DPGA_REG		0x1F
+#define ISABELLE_VTX2_DPGA_REG		0x20
+#define ISABELLE_TX_INPUT_CFG_REG	0x21
+#define ISABELLE_RX_INPUT_CFG_REG	0x22
+#define ISABELLE_RX_INPUT_CFG2_REG	0x23
+#define ISABELLE_VOICE_HPF_CFG_REG	0x24
+#define ISABELLE_AUDIO_HPF_CFG_REG	0x25
+#define ISABELLE_RX1_DPGA_REG		0x26
+#define ISABELLE_RX2_DPGA_REG		0x27
+#define ISABELLE_RX3_DPGA_REG		0x28
+#define ISABELLE_RX4_DPGA_REG		0x29
+#define ISABELLE_RX5_DPGA_REG		0x2A
+#define ISABELLE_RX6_DPGA_REG		0x2B
+#define ISABELLE_ALU_TX_EN_REG		0x2C
+#define ISABELLE_ALU_RX_EN_REG		0x2D
+#define ISABELLE_IIR_RESYNC_REG		0x2E
+#define ISABELLE_ABIAS_CFG_REG		0x30
+#define ISABELLE_DBIAS_CFG_REG		0x31
+#define ISABELLE_MIC1_GAIN_REG		0x32
+#define ISABELLE_MIC2_GAIN_REG		0x33
+#define ISABELLE_AMIC_CFG_REG		0x34
+#define ISABELLE_DMIC_CFG_REG		0x35
+#define ISABELLE_APGA_GAIN_REG		0x36
+#define ISABELLE_APGA_CFG_REG		0x37
+#define ISABELLE_TX_GAIN_DLY_REG	0x38
+#define ISABELLE_RX_GAIN_DLY_REG	0x39
+#define ISABELLE_RX_PWR_CTRL_REG	0x3A
+#define ISABELLE_DPGA1LR_IN_SEL_REG	0x3B
+#define ISABELLE_DPGA1L_GAIN_REG	0x3C
+#define ISABELLE_DPGA1R_GAIN_REG	0x3D
+#define ISABELLE_DPGA2L_IN_SEL_REG	0x3E
+#define ISABELLE_DPGA2R_IN_SEL_REG	0x3F
+#define ISABELLE_DPGA2L_GAIN_REG	0x40
+#define ISABELLE_DPGA2R_GAIN_REG	0x41
+#define ISABELLE_DPGA3LR_IN_SEL_REG	0x42
+#define ISABELLE_DPGA3L_GAIN_REG	0x43
+#define ISABELLE_DPGA3R_GAIN_REG	0x44
+#define ISABELLE_DAC1_SOFTRAMP_REG	0x45
+#define ISABELLE_DAC2_SOFTRAMP_REG	0x46
+#define ISABELLE_DAC3_SOFTRAMP_REG	0x47
+#define ISABELLE_DAC_CFG_REG		0x48
+#define ISABELLE_EARDRV_CFG1_REG	0x49
+#define ISABELLE_EARDRV_CFG2_REG	0x4A
+#define ISABELLE_HSDRV_GAIN_REG		0x4B
+#define ISABELLE_HSDRV_CFG1_REG		0x4C
+#define ISABELLE_HSDRV_CFG2_REG		0x4D
+#define ISABELLE_HS_NG_CFG1_REG		0x4E
+#define ISABELLE_HS_NG_CFG2_REG		0x4F
+#define ISABELLE_LINEAMP_GAIN_REG	0x50
+#define ISABELLE_LINEAMP_CFG_REG	0x51
+#define ISABELLE_HFL_VOL_CTRL_REG	0x52
+#define ISABELLE_HFL_SFTVOL_CTRL_REG	0x53
+#define ISABELLE_HFL_LIM_CTRL_1_REG	0x54
+#define ISABELLE_HFL_LIM_CTRL_2_REG	0x55
+#define ISABELLE_HFR_VOL_CTRL_REG	0x56
+#define ISABELLE_HFR_SFTVOL_CTRL_REG	0x57
+#define ISABELLE_HFR_LIM_CTRL_1_REG	0x58
+#define ISABELLE_HFR_LIM_CTRL_2_REG	0x59
+#define ISABELLE_HF_MODE_REG		0x5A
+#define ISABELLE_HFLPGA_CFG_REG		0x5B
+#define ISABELLE_HFRPGA_CFG_REG		0x5C
+#define ISABELLE_HFDRV_CFG_REG		0x5D
+#define ISABELLE_PDMOUT_CFG1_REG	0x5E
+#define ISABELLE_PDMOUT_CFG2_REG	0x5F
+#define ISABELLE_PDMOUT_L_WM_REG	0x60
+#define ISABELLE_PDMOUT_R_WM_REG	0x61
+#define ISABELLE_HF_NG_CFG1_REG		0x62
+#define ISABELLE_HF_NG_CFG2_REG		0x63
+
+/* ISABELLE_PWR_EN_REG (0x02h) */
+#define ISABELLE_CHIP_EN		BIT(0)
+
+/* ISABELLE DAI FORMATS */
+#define ISABELLE_AIF_FMT_MASK		0x70
+#define ISABELLE_I2S_MODE		0x0
+#define ISABELLE_LEFT_J_MODE		0x1
+#define ISABELLE_PDM_MODE		0x2
+
+#define ISABELLE_AIF_LENGTH_MASK	0x30
+#define ISABELLE_AIF_LENGTH_20		0x00
+#define ISABELLE_AIF_LENGTH_32		0x10
+
+#define ISABELLE_AIF_MS			0x80
+
+#define ISABELLE_FS_RATE_MASK		0xF
+#define ISABELLE_FS_RATE_8		0x0
+#define ISABELLE_FS_RATE_11		0x1
+#define ISABELLE_FS_RATE_12		0x2
+#define ISABELLE_FS_RATE_16		0x4
+#define ISABELLE_FS_RATE_22		0x5
+#define ISABELLE_FS_RATE_24		0x6
+#define ISABELLE_FS_RATE_32		0x8
+#define ISABELLE_FS_RATE_44		0x9
+#define ISABELLE_FS_RATE_48		0xA
+
+#define ISABELLE_MAX_REGISTER		0xFF
+
+#endif
-- 
cgit v1.2.3


From 5eba8ec37fe8cfed4cacff56f9025b756cc43faa Mon Sep 17 00:00:00 2001
From: MR Swami Reddy <MR.Swami.Reddy@ti.com>
Date: Mon, 4 Jun 2012 17:44:54 +0530
Subject: ASoC: isabelle: Remove regmap_exit()

With devm_ APIs regmap_exit() not needed, so remove regmap_exit().

Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/isabelle.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index b6921a82fbcc..bcc77ef0eda2 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -1141,7 +1141,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
 				ARRAY_SIZE(isabelle_dai));
 	if (ret < 0) {
 		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
-		regmap_exit(dev_get_regmap(&i2c->dev, NULL));
 		return ret;
 	}
 
@@ -1151,7 +1150,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
 static int __devexit isabelle_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(dev_get_regmap(&client->dev, NULL));
 	return 0;
 }
 
-- 
cgit v1.2.3


From 165961efc03159631eadc086877704c7778ac356 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 5 Jun 2012 10:44:23 +0100
Subject: ASoC: dapm: The clock API is even less consistent than thought

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

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 39e8c2fdf50e..7365fed1ba74 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2904,7 +2904,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		}
 		break;
 	case snd_soc_dapm_clock_supply:
-#ifdef CONFIG_HAVE_CLK
+#ifdef CONFIG_CLKDEV_LOOKUP
 		w->clk = devm_clk_get(dapm->dev, w->name);
 		if (IS_ERR(w->clk)) {
 			ret = PTR_ERR(w->clk);
-- 
cgit v1.2.3


From 571f6a7f07e9dda6c9795398747278e52368c88a Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Mon, 4 Jun 2012 13:19:41 -0500
Subject: ASoC: cs42l73: Convert to devm_regmap_init_i2c()

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs42l73.c | 20 +++++---------------
 1 file changed, 5 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index e0d45fdaa750..2c08c4cb465a 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1362,11 +1362,11 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 
 	i2c_set_clientdata(i2c_client, cs42l73);
 
-	cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap);
+	cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap);
 	if (IS_ERR(cs42l73->regmap)) {
 		ret = PTR_ERR(cs42l73->regmap);
 		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
-		goto err;
+		return ret;
 	}
 	/* initialize codec */
 	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg);
@@ -1384,13 +1384,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 		dev_err(&i2c_client->dev,
 			"CS42L73 Device ID (%X). Expected %X\n",
 			devid, CS42L73_DEVID);
-		goto err_regmap;
+		return ret;
 	}
 
 	ret = regmap_read(cs42l73->regmap, CS42L73_REVID, &reg);
 	if (ret < 0) {
 		dev_err(&i2c_client->dev, "Get Revision ID failed\n");
-		goto err_regmap;
+		return ret;;
 	}
 
 	dev_info(&i2c_client->dev,
@@ -1402,23 +1402,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client,
 			&soc_codec_dev_cs42l73, cs42l73_dai,
 			ARRAY_SIZE(cs42l73_dai));
 	if (ret < 0)
-		goto err_regmap;
+		return ret;
 	return 0;
-
-err_regmap:
-	regmap_exit(cs42l73->regmap);
-
-err:
-	return ret;
 }
 
 static __devexit int cs42l73_i2c_remove(struct i2c_client *client)
 {
-	struct cs42l73_private *cs42l73 = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(cs42l73->regmap);
-
 	return 0;
 }
 
-- 
cgit v1.2.3


From 134b2f576b9144223dd5b59a496218e3aadaf56b Mon Sep 17 00:00:00 2001
From: Brian Austin <brian.austin@cirrus.com>
Date: Mon, 4 Jun 2012 13:19:42 -0500
Subject: ASoC: cs42l52: Convert to devm_regmap_init_i2c()

Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cs42l52.c | 18 ++++--------------
 1 file changed, 4 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index ec03abc79a9a..628daf6a1d97 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1216,11 +1216,11 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 		return -ENOMEM;
 	cs42l52->dev = &i2c_client->dev;
 
-	cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap);
+	cs42l52->regmap = devm_regmap_init_i2c(i2c_client, &cs42l52_regmap);
 	if (IS_ERR(cs42l52->regmap)) {
 		ret = PTR_ERR(cs42l52->regmap);
 		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
-		goto err;
+		return ret;
 	}
 
 	i2c_set_clientdata(i2c_client, cs42l52);
@@ -1242,7 +1242,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 		dev_err(&i2c_client->dev,
 			"CS42L52 Device ID (%X). Expected %X\n",
 			devid, CS42L52_CHIP_ID);
-		goto err_regmap;
+		return ret;
 	}
 
 	regcache_cache_only(cs42l52->regmap, true);
@@ -1250,23 +1250,13 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client,
 	ret =  snd_soc_register_codec(&i2c_client->dev,
 			&soc_codec_dev_cs42l52, &cs42l52_dai, 1);
 	if (ret < 0)
-		goto err_regmap;
+		return ret;
 	return 0;
-
-err_regmap:
-	regmap_exit(cs42l52->regmap);
-
-err:
-	return ret;
 }
 
 static int cs42l52_i2c_remove(struct i2c_client *client)
 {
-	struct cs42l52_private *cs42l52 = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(cs42l52->regmap);
-
 	return 0;
 }
 
-- 
cgit v1.2.3


From 61dc479e99d4d74c6113656dc50babed90a384c5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 5 Jun 2012 11:08:45 +0100
Subject: Revert "ASoC: fsl_ssi: convert to use devm_clk_get"

This reverts commit 014e5b56702575c5cd8ffc4b1a7924cfdfe0f1ea since
PowerPC doesn't use clkdev and hasn't implemented devm_clk_get() itself.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/fsl_ssi.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index b10a427a8098..4ed2afd47782 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -725,7 +725,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 		u32 dma_events[2];
 		ssi_private->ssi_on_imx = true;
 
-		ssi_private->clk = devm_clk_get(&pdev->dev, NULL);
+		ssi_private->clk = clk_get(&pdev->dev, NULL);
 		if (IS_ERR(ssi_private->clk)) {
 			ret = PTR_ERR(ssi_private->clk);
 			dev_err(&pdev->dev, "could not get clock: %d\n", ret);
@@ -842,8 +842,10 @@ error_dev:
 	device_remove_file(&pdev->dev, dev_attr);
 
 error_clk:
-	if (ssi_private->ssi_on_imx)
+	if (ssi_private->ssi_on_imx) {
 		clk_disable_unprepare(ssi_private->clk);
+		clk_put(ssi_private->clk);
+	}
 
 error_irq:
 	free_irq(ssi_private->irq, ssi_private);
@@ -869,6 +871,7 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 	if (ssi_private->ssi_on_imx) {
 		platform_device_unregister(ssi_private->imx_pcm_pdev);
 		clk_disable_unprepare(ssi_private->clk);
+		clk_put(ssi_private->clk);
 	}
 	snd_soc_unregister_dai(&pdev->dev);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
-- 
cgit v1.2.3


From cd86e3ce304189fbdb144622245d0da9189551a1 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 4 Jun 2012 18:20:21 +0100
Subject: ASoC: lm59453: Unconstify dai_driver

The core fills in some blanks which makes it annoying to do the right thing
and constify the calls in the core.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/lm49453.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index c1bc9458906b..99b0a9dcff34 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1357,7 +1357,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
 };
 
 /* LM49453 dai structure. */
-static const struct snd_soc_dai_driver lm49453_dai[] = {
+static struct snd_soc_dai_driver lm49453_dai[] = {
 	{
 		.name = "LM49453 Headset",
 		.playback = {
-- 
cgit v1.2.3


From 223f18e4482fd90b7cdabb222fad5ca502dc0028 Mon Sep 17 00:00:00 2001
From: Raymond Yau <superquad.vortex2@gmail.com>
Date: Mon, 4 Jun 2012 14:59:55 +0800
Subject: ALSA: au88x0 - Remove unused "Master Mono" Playback Volume and
 Playback Switch of ac97 codec

Remove "Master Mono Playback Volume" and "Master Mono Playback Switch"
of ac97 mixer since au88x0 does no use "Master Mono Pin" of AC97 codec
even au88x0 support mono playback

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0_mixer.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c
index 557c782ae4fc..fa13efbebdaf 100644
--- a/sound/pci/au88x0/au88x0_mixer.c
+++ b/sound/pci/au88x0/au88x0_mixer.c
@@ -10,6 +10,15 @@
 #include <sound/core.h>
 #include "au88x0.h"
 
+static int remove_ctl(struct snd_card *card, const char *name)
+{
+	struct snd_ctl_elem_id id;
+	memset(&id, 0, sizeof(id));
+	strcpy(id.name, name);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	return snd_ctl_remove_id(card, &id);
+}
+
 static int __devinit snd_vortex_mixer(vortex_t * vortex)
 {
 	struct snd_ac97_bus *pbus;
@@ -28,5 +37,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex)
 	ac97.scaps = AC97_SCAP_NO_SPDIF;
 	err = snd_ac97_mixer(pbus, &ac97, &vortex->codec);
 	vortex->isquad = ((vortex->codec == NULL) ?  0 : (vortex->codec->ext_id&0x80));
+	remove_ctl(vortex->card, "Master Mono Playback Volume");
+	remove_ctl(vortex->card, "Master Mono Playback Switch");
 	return err;
 }
-- 
cgit v1.2.3


From ce63f3ba256a48023b425a4a6792f8da03d00948 Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Wed, 6 Jun 2012 13:49:44 +0800
Subject: ALSA: hda - add power states information in proc

add more power states information:
- reset status
- clock stop ok
- power states error

Output like:
Power: setting=D0, actual=D0, Error, Clock-stop-OK, Setting-reset

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.h | 3 +++
 sound/pci/hda/hda_proc.c  | 9 ++++++++-
 2 files changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 4fc3960c8591..71864cddcb9d 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -323,6 +323,9 @@ enum {
 #define AC_PWRST_D1			0x01
 #define AC_PWRST_D2			0x02
 #define AC_PWRST_D3			0x03
+#define AC_PWRST_ERROR                  (1<<8)
+#define AC_PWRST_CLK_STOP_OK            (1<<9)
+#define AC_PWRST_SETTING_RESET          (1<<10)
 
 /* Processing capabilies */
 #define AC_PCAP_BENIGN			(1<<0)
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index e59e2f059b6e..fb9ef132f13b 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -455,10 +455,17 @@ static void print_power_state(struct snd_info_buffer *buffer,
 		snd_iprintf(buffer, "  Power states: %s\n",
 			    bits_names(sup, names, ARRAY_SIZE(names)));
 
-	snd_iprintf(buffer, "  Power: setting=%s, actual=%s\n",
+	snd_iprintf(buffer, "  Power: setting=%s, actual=%s",
 		    get_pwr_state(pwr & AC_PWRST_SETTING),
 		    get_pwr_state((pwr & AC_PWRST_ACTUAL) >>
 				  AC_PWRST_ACTUAL_SHIFT));
+	if (pwr & AC_PWRST_ERROR)
+		snd_iprintf(buffer, ", Error");
+	if (pwr & AC_PWRST_CLK_STOP_OK)
+		snd_iprintf(buffer, ", Clock-stop-OK");
+	if (pwr & AC_PWRST_SETTING_RESET)
+		snd_iprintf(buffer, ", Setting-reset");
+	snd_iprintf(buffer, "\n");
 }
 
 static void print_unsol_cap(struct snd_info_buffer *buffer,
-- 
cgit v1.2.3


From 167d2d55bfb4628169a57e3adbb1e5b097dca0f5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 6 Jun 2012 12:17:20 +0200
Subject: ALSA: hda - Show D3cold state in proc files

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_proc.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index fb9ef132f13b..1dd9dbeeef6e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -426,10 +426,10 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 
 static const char *get_pwr_state(u32 state)
 {
-	static const char * const buf[4] = {
-		"D0", "D1", "D2", "D3"
+	static const char * const buf[] = {
+		"D0", "D1", "D2", "D3", "D3cold"
 	};
-	if (state < 4)
+	if (state < ARRAY_SIZE(buf))
 		return buf[state];
 	return "UNKNOWN";
 }
-- 
cgit v1.2.3


From 0c7f46ad927cbd29965d4971730de713b478d270 Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Wed, 6 Jun 2012 22:02:48 +0800
Subject: ALSA: hda - check supported power states

Add function to check whether power states supported by specific
codec node.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 16 ++++++++++++++++
 1 file changed, 16 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 41ca803a1fff..b89c8ecc819a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3504,6 +3504,22 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg,
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all);
 
+/*
+ *  supported power states check
+ */
+static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state)
+{
+	int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
+
+	if (sup < 0)
+		return false;
+	if (sup & power_state)
+		return true;
+	else
+		return false;
+}
+
 /*
  * set power state of the codec
  */
-- 
cgit v1.2.3


From 0f4ccbb02533276ab750961e150aeee06492ed7c Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Thu, 7 Jun 2012 16:51:33 +0800
Subject: ALSA: hda - reduce msleep time if EPSS power states supported

if EPSS supported, transition from D3 state to D0 state in less
than 10ms

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b89c8ecc819a..fedbfae978af 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3532,8 +3532,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	}
 
 	/* this delay seems necessary to avoid click noise at power-down */
-	if (power_state == AC_PWRST_D3)
-		msleep(100);
+	if (power_state == AC_PWRST_D3) {
+		/* transition time less than 10ms for power down */
+		bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
+		msleep(epss ? 10 : 100);
+	}
 	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
 			    power_state);
 	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
-- 
cgit v1.2.3


From e076eb5c952c0f724980b44a77902a2ff93d098c Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Thu, 7 Jun 2012 16:52:07 +0800
Subject: ALSA: hda - check proper return value

snd_hda_param_read() return value -1 means error, others are responses

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_proc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 1dd9dbeeef6e..7e46258fc700 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -451,7 +451,7 @@ static void print_power_state(struct snd_info_buffer *buffer,
 	int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE);
 	int pwr = snd_hda_codec_read(codec, nid, 0,
 				     AC_VERB_GET_POWER_STATE, 0);
-	if (sup)
+	if (sup != -1)
 		snd_iprintf(buffer, "  Power states: %s\n",
 			    bits_names(sup, names, ARRAY_SIZE(names)));
 
-- 
cgit v1.2.3


From a265367ccbe72010757a56e5776fcf9a49370181 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 31 May 2012 14:47:46 +0100
Subject: ASoC: max98095: Staticise non-exported functions and export jack
 detect

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/max98095.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 35179e2c23c9..7cd508e16a5c 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -2216,7 +2216,7 @@ static irqreturn_t max98095_report_jack(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
-int max98095_jack_detect_enable(struct snd_soc_codec *codec)
+static int max98095_jack_detect_enable(struct snd_soc_codec *codec)
 {
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
@@ -2245,7 +2245,7 @@ int max98095_jack_detect_enable(struct snd_soc_codec *codec)
 	return ret;
 }
 
-int max98095_jack_detect_disable(struct snd_soc_codec *codec)
+static int max98095_jack_detect_disable(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
@@ -2286,6 +2286,7 @@ int max98095_jack_detect(struct snd_soc_codec *codec,
 	max98095_report_jack(client->irq, codec);
 	return 0;
 }
+EXPORT_SYMBOL_GPL(max98095_jack_detect);
 
 #ifdef CONFIG_PM
 static int max98095_suspend(struct snd_soc_codec *codec)
-- 
cgit v1.2.3


From 40820105d4caef1489edd56e9dc2b85871b65308 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Thu, 7 Jun 2012 15:38:37 +0300
Subject: ASoC: isabelle: using an uninitialized variable

We should set "isabelle_regmap" before using it.  GCC complains.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/isabelle.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index bcc77ef0eda2..0d62f3b0f474 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -1126,8 +1126,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
 	struct regmap *isabelle_regmap;
 	int ret = 0;
 
-	i2c_set_clientdata(i2c, isabelle_regmap);
-
 	isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config);
 	if (IS_ERR(isabelle_regmap)) {
 		ret = PTR_ERR(isabelle_regmap);
@@ -1135,6 +1133,7 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
 			ret);
 		return ret;
 	}
+	i2c_set_clientdata(i2c, isabelle_regmap);
 
 	ret =  snd_soc_register_codec(&i2c->dev,
 				&soc_codec_dev_isabelle, isabelle_dai,
-- 
cgit v1.2.3


From 9515c1010c98347ec92d923bd3e23793fa6dc6fe Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 6 Jun 2012 17:15:07 -0600
Subject: ASoC: tegra: add .stream_name to CPU DAIs

This is certainly required if the I2S and SPDIF controllers are converted
to be CODECs, and is probably good practice irrespective.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra20_i2s.c   | 2 ++
 sound/soc/tegra/tegra20_spdif.c | 1 +
 sound/soc/tegra/tegra30_i2s.c   | 2 ++
 3 files changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 0c7af63d444b..9d5d4704da29 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -261,12 +261,14 @@ static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = {
 static const struct snd_soc_dai_driver tegra20_i2s_dai_template = {
 	.probe = tegra20_i2s_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
+		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index f9b57418bd08..ffbd99c4106e 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -181,6 +181,7 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = {
 	.name = DRV_NAME,
 	.probe = tegra20_spdif_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 8596032985dc..9c5c0e6819eb 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -320,12 +320,14 @@ static struct snd_soc_dai_ops tegra30_i2s_dai_ops = {
 static const struct snd_soc_dai_driver tegra30_i2s_dai_template = {
 	.probe = tegra30_i2s_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
+		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
-- 
cgit v1.2.3


From 408dafc4235e393036708126057e4d643f579486 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 6 Jun 2012 17:15:48 -0600
Subject: ASoC: tegra: statically define DAI link format

Define the DAI format statically in the dai_link, rather than executing
code to set it each time the hw params are set.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_wm8903.c | 22 +++-------------------
 sound/soc/tegra/trimslice.c    | 22 +++-------------------
 2 files changed, 6 insertions(+), 38 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 1fd71e5a9eb9..087d3d8d6c06 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -58,7 +58,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
@@ -86,24 +85,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
 		return err;
 	}
 
-	err = snd_soc_dai_set_fmt(codec_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "codec_dai fmt not set\n");
-		return err;
-	}
-
-	err = snd_soc_dai_set_fmt(cpu_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "cpu_dai fmt not set\n");
-		return err;
-	}
-
 	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
 					SND_SOC_CLOCK_IN);
 	if (err < 0) {
@@ -240,6 +221,9 @@ static struct snd_soc_dai_link tegra_wm8903_dai = {
 	.codec_dai_name = "wm8903-hifi",
 	.init = tegra_wm8903_init,
 	.ops = &tegra_wm8903_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S |
+		   SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
 };
 
 static struct snd_soc_card snd_soc_tegra_wm8903 = {
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 5815430e8521..62bb805022dd 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -52,7 +52,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
@@ -68,24 +67,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
 		return err;
 	}
 
-	err = snd_soc_dai_set_fmt(codec_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "codec_dai fmt not set\n");
-		return err;
-	}
-
-	err = snd_soc_dai_set_fmt(cpu_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "cpu_dai fmt not set\n");
-		return err;
-	}
-
 	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
 					SND_SOC_CLOCK_IN);
 	if (err < 0) {
@@ -121,6 +102,9 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
 	.cpu_dai_name = "tegra20-i2s.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.ops = &trimslice_asoc_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S |
+		   SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
 };
 
 static struct snd_soc_card snd_soc_trimslice = {
-- 
cgit v1.2.3


From 40db77a0c4223d0b87c4b61ae38760d47593b7a5 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 6 Jun 2012 17:15:49 -0600
Subject: ASoC: tegra: remove usage of rtd->codec

rtd->codec_dai->codec can be used instead.

This is a slight step along the way to not needing the rtd->codec field
any more.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra_alc5632.c | 5 +++--
 sound/soc/tegra/tegra_wm8753.c  | 2 +-
 sound/soc/tegra/tegra_wm8903.c  | 5 +++--
 sound/soc/tegra/trimslice.c     | 2 +-
 4 files changed, 8 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 417b09b83fdf..d684df294c0c 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -43,7 +43,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
@@ -105,7 +105,8 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = {
 
 static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card);
 
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index 02bd5a8e8544..ea9166d5c4eb 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -57,7 +57,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 087d3d8d6c06..08b5fef67b31 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -58,7 +58,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
@@ -181,7 +181,8 @@ static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
 
 static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 62bb805022dd..e69a4f7000d6 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -52,7 +52,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
-- 
cgit v1.2.3


From c92a40e3a163b6708e0dd82ba4612f79df846912 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 6 Jun 2012 17:15:05 -0600
Subject: ASoC: tegra: use DAI's not card's dev for dev_err

This is the actual device of the I2S or SPDIF controller reporting the
problem. If a future change converts these controllers to be CODECs, then
there may be no pcm associated with the substream, so this change avoids
a crash.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra20_i2s.c   | 2 +-
 sound/soc/tegra/tegra20_spdif.c | 2 +-
 sound/soc/tegra/tegra30_i2s.c   | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 9d5d4704da29..647daf610e4a 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -138,7 +138,7 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct device *dev = substream->pcm->card->dev;
+	struct device *dev = dai->dev;
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	u32 reg;
 	int ret, sample_size, srate, i2sclock, bitcnt;
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index ffbd99c4106e..f774a2d5e585 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -77,7 +77,7 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct device *dev = substream->pcm->card->dev;
+	struct device *dev = dai->dev;
 	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
 	int ret, spdifclock;
 
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 9c5c0e6819eb..2327f62e8a8c 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -181,7 +181,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct device *dev = substream->pcm->card->dev;
+	struct device *dev = dai->dev;
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
 	u32 val;
 	int ret, sample_size, srate, i2sclock, bitcnt;
-- 
cgit v1.2.3


From 0f163546a772d62250f59bad6a9338a0e3a2605c Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 6 Jun 2012 17:15:06 -0600
Subject: ASoC: tegra: use regmap more directly

Stop open-coding the caching of the ctrl registers; instead, use
regmap_update_bits() to update parts of the register from different
places. The removal of the open-coded cache will allow controls to be
created which touch registers, which will be necessary if any of these
modules are converted to CODECs.

Get rid of tegra*_read/write; just call regmap_read/write directly.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/tegra20_i2s.c   | 90 ++++++++++++++++++++---------------------
 sound/soc/tegra/tegra20_i2s.h   |  1 -
 sound/soc/tegra/tegra20_spdif.c | 33 ++++++---------
 sound/soc/tegra/tegra20_spdif.h |  1 -
 sound/soc/tegra/tegra30_i2s.c   | 81 ++++++++++++++++++-------------------
 sound/soc/tegra/tegra30_i2s.h   |  1 -
 6 files changed, 95 insertions(+), 112 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 647daf610e4a..c5fc6b1404f6 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -46,18 +46,6 @@
 
 #define DRV_NAME "tegra20-i2s"
 
-static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val)
-{
-	regmap_write(i2s->regmap, reg, val);
-}
-
-static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg)
-{
-	u32 val;
-	regmap_read(i2s->regmap, reg, &val);
-	return val;
-}
-
 static int tegra20_i2s_runtime_suspend(struct device *dev)
 {
 	struct tegra20_i2s *i2s = dev_get_drvdata(dev);
@@ -85,6 +73,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -93,10 +82,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE;
+	mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
+		val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
@@ -104,33 +93,35 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
-			   TEGRA20_I2S_CTRL_LRCK_MASK);
+	mask |= TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
+		TEGRA20_I2S_CTRL_LRCK_MASK;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_A:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+		val |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
 		break;
 	case SND_SOC_DAIFMT_I2S:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val);
+
 	return 0;
 }
 
@@ -140,27 +131,32 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct device *dev = dai->dev;
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	u32 reg;
+	unsigned int mask, val;
 	int ret, sample_size, srate, i2sclock, bitcnt;
 
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
+	mask = TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16;
+		val = TEGRA20_I2S_CTRL_BIT_SIZE_16;
 		sample_size = 16;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24;
+		val = TEGRA20_I2S_CTRL_BIT_SIZE_24;
 		sample_size = 24;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32;
+		val = TEGRA20_I2S_CTRL_BIT_SIZE_32;
 		sample_size = 32;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	mask |= TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK;
+	val |= TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
+
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val);
+
 	srate = params_rate(params);
 
 	/* Final "* 2" required by Tegra hardware */
@@ -175,42 +171,44 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream,
 	bitcnt = (i2sclock / (2 * srate)) - 1;
 	if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
 		return -EINVAL;
-	reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
+	val = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
 
 	if (i2sclock % (2 * srate))
-		reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
+		val |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
 
-	tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg);
+	regmap_write(i2s->regmap, TEGRA20_I2S_TIMING, val);
 
-	tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR,
-		TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
-		TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+	regmap_write(i2s->regmap, TEGRA20_I2S_FIFO_SCR,
+		     TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
+		     TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
 	return 0;
 }
 
 static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO1_ENABLE,
+			   TEGRA20_I2S_CTRL_FIFO1_ENABLE);
 }
 
 static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO1_ENABLE, 0);
 }
 
 static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO2_ENABLE,
+			   TEGRA20_I2S_CTRL_FIFO2_ENABLE);
 }
 
 static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO2_ENABLE, 0);
 }
 
 static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -414,8 +412,6 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev)
 	i2s->playback_dma_data.width = 32;
 	i2s->playback_dma_data.req_sel = dma_ch;
 
-	i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
-
 	pm_runtime_enable(&pdev->dev);
 	if (!pm_runtime_enabled(&pdev->dev)) {
 		ret = tegra20_i2s_runtime_resume(&pdev->dev);
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index a57efc6a597e..c27069d24d77 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -158,7 +158,6 @@ struct tegra20_i2s {
 	struct tegra_pcm_dma_params capture_dma_data;
 	struct tegra_pcm_dma_params playback_dma_data;
 	struct regmap *regmap;
-	u32 reg_ctrl;
 };
 
 #endif
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index f774a2d5e585..5c33c618929d 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -37,19 +37,6 @@
 
 #define DRV_NAME "tegra20-spdif"
 
-static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg,
-					u32 val)
-{
-	regmap_write(spdif->regmap, reg, val);
-}
-
-static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg)
-{
-	u32 val;
-	regmap_read(spdif->regmap, reg, &val);
-	return val;
-}
-
 static int tegra20_spdif_runtime_suspend(struct device *dev)
 {
 	struct tegra20_spdif *spdif = dev_get_drvdata(dev);
@@ -79,19 +66,22 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
 {
 	struct device *dev = dai->dev;
 	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
 	int ret, spdifclock;
 
-	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK;
-	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+	mask = TEGRA20_SPDIF_CTRL_PACK |
+	       TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK;
-		spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+		val = TEGRA20_SPDIF_CTRL_PACK |
+		      TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, mask, val);
+
 	switch (params_rate(params)) {
 	case 32000:
 		spdifclock = 4096000;
@@ -129,14 +119,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
 
 static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif)
 {
-	spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN;
-	tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+	regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL,
+			   TEGRA20_SPDIF_CTRL_TX_EN,
+			   TEGRA20_SPDIF_CTRL_TX_EN);
 }
 
 static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif)
 {
-	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN;
-	tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+	regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL,
+			   TEGRA20_SPDIF_CTRL_TX_EN, 0);
 }
 
 static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h
index ed756527efea..b48d699fd583 100644
--- a/sound/soc/tegra/tegra20_spdif.h
+++ b/sound/soc/tegra/tegra20_spdif.h
@@ -465,7 +465,6 @@ struct tegra20_spdif {
 	struct tegra_pcm_dma_params capture_dma_data;
 	struct tegra_pcm_dma_params playback_dma_data;
 	struct regmap *regmap;
-	u32 reg_ctrl;
 };
 
 #endif
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 2327f62e8a8c..b68e27a14608 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -44,18 +44,6 @@
 
 #define DRV_NAME "tegra30-i2s"
 
-static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val)
-{
-	regmap_write(i2s->regmap, reg, val);
-}
-
-static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg)
-{
-	u32 val;
-	regmap_read(i2s->regmap, reg, &val);
-	return val;
-}
-
 static int tegra30_i2s_runtime_suspend(struct device *dev)
 {
 	struct tegra30_i2s *i2s = dev_get_drvdata(dev);
@@ -128,6 +116,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -136,10 +125,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE;
+	mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
+		val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
@@ -147,33 +136,37 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
-			   TEGRA30_I2S_CTRL_LRCK_MASK);
+	mask |= TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
+		TEGRA30_I2S_CTRL_LRCK_MASK;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_A:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+		val |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
 		break;
 	case SND_SOC_DAIFMT_I2S:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(dai->dev);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val);
+	pm_runtime_put(dai->dev);
+
 	return 0;
 }
 
@@ -183,22 +176,24 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct device *dev = dai->dev;
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	u32 val;
+	unsigned int mask, val, reg;
 	int ret, sample_size, srate, i2sclock, bitcnt;
 
 	if (params_channels(params) != 2)
 		return -EINVAL;
 
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
+	mask = TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16;
+		val = TEGRA30_I2S_CTRL_BIT_SIZE_16;
 		sample_size = 16;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val);
+
 	srate = params_rate(params);
 
 	/* Final "* 2" required by Tegra hardware */
@@ -219,7 +214,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (i2sclock % (2 * srate))
 		val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE;
 
-	tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val);
+	regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val);
 
 	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
 	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
@@ -229,15 +224,17 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
-		tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val);
+		reg = TEGRA30_I2S_CIF_RX_CTRL;
 	} else {
 		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
-		tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val);
+		reg = TEGRA30_I2S_CIF_RX_CTRL;
 	}
 
+	regmap_write(i2s->regmap, reg, val);
+
 	val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) |
 	      (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT);
-	tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val);
+	regmap_write(i2s->regmap, TEGRA30_I2S_OFFSET, val);
 
 	return 0;
 }
@@ -245,29 +242,31 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
 static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif);
-	i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_TX,
+			   TEGRA30_I2S_CTRL_XFER_EN_TX);
 }
 
 static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif);
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_TX, 0);
 }
 
 static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif);
-	i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_RX,
+			   TEGRA30_I2S_CTRL_XFER_EN_RX);
 }
 
 static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif);
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_RX, 0);
 }
 
 static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 91adf29c7a87..34dc47b9581c 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -236,7 +236,6 @@ struct tegra30_i2s {
 	enum tegra30_ahub_txcif playback_fifo_cif;
 	struct tegra_pcm_dma_params playback_dma_data;
 	struct regmap *regmap;
-	u32 reg_ctrl;
 };
 
 #endif
-- 
cgit v1.2.3


From 09617ce4774ebf30a55b8451f4b35031f626f763 Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Fri, 8 Jun 2012 10:26:08 +0800
Subject: ALSA: hda - power setting error check

codec may reject power state transition requests(reporting PS-ERROR set),
in that case we re-issue a power state setting and check error bit again.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 17 ++++++++++++++---
 1 file changed, 14 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index fedbfae978af..851e6ecfa011 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3526,6 +3526,9 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg
 static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state)
 {
+	int count;
+	unsigned int state;
+
 	if (codec->patch_ops.set_power_state) {
 		codec->patch_ops.set_power_state(codec, fg, power_state);
 		return;
@@ -3537,9 +3540,17 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 		bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
 		msleep(epss ? 10 : 100);
 	}
-	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
-			    power_state);
-	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+
+	/* repeat power states setting at most 10 times*/
+	for (count = 0; count < 10; count++) {
+		snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+				    power_state);
+		snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+		state = snd_hda_codec_read(codec, fg, 0,
+					   AC_VERB_GET_POWER_STATE, 0);
+		if (!(state & AC_PWRST_ERROR))
+			break;
+	}
 }
 
 #ifdef CONFIG_SND_HDA_HWDEP
-- 
cgit v1.2.3


From 7d116684945459e98538c797dca37c54ddd89906 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 10:21:16 +0800
Subject: ASoC: wm8903: Convert to devm_regmap_init_i2c()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8903.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 304b5cff3482..3abd450842ee 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -2124,7 +2124,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 	wm8903->dev = &i2c->dev;
 
-	wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap);
+	wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap);
 	if (IS_ERR(wm8903->regmap)) {
 		ret = PTR_ERR(wm8903->regmap);
 		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2191,7 +2191,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 	return 0;
 err:
-	regmap_exit(wm8903->regmap);
 	return ret;
 }
 
@@ -2200,7 +2199,6 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 	struct wm8903_priv *wm8903 = i2c_get_clientdata(client);
 
 	wm8903_free_gpio(wm8903);
-	regmap_exit(wm8903->regmap);
 	snd_soc_unregister_codec(&client->dev);
 
 	return 0;
-- 
cgit v1.2.3


From 8cb28fd6d1e98fe4cf232d7803093a3b7b46e969 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 11:57:50 +0800
Subject: ASoC: wm8904: Convert to module_i2c_driver()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8904.c | 18 +-----------------
 1 file changed, 1 insertion(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index db94d10b5c1a..02bc2caac83a 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2313,23 +2313,7 @@ static struct i2c_driver wm8904_i2c_driver = {
 	.id_table = wm8904_i2c_id,
 };
 
-static int __init wm8904_modinit(void)
-{
-	int ret = 0;
-	ret = i2c_add_driver(&wm8904_i2c_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n",
-		       ret);
-	}
-	return ret;
-}
-module_init(wm8904_modinit);
-
-static void __exit wm8904_exit(void)
-{
-	i2c_del_driver(&wm8904_i2c_driver);
-}
-module_exit(wm8904_exit);
+module_i2c_driver(wm8904_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC WM8904 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
-- 
cgit v1.2.3


From d633edd95dc938f3f5f0d4e431932f4ca042bffb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 11:26:01 +0800
Subject: ASoC: wm8904: Convert to devm_regmap_init_i2c()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8904.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 02bc2caac83a..560a9a47596b 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2263,7 +2263,7 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
 	if (wm8904 == NULL)
 		return -ENOMEM;
 
-	wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap);
+	wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap);
 	if (IS_ERR(wm8904->regmap)) {
 		ret = PTR_ERR(wm8904->regmap);
 		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2283,15 +2283,12 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
 	return 0;
 
 err:
-	regmap_exit(wm8904->regmap);
 	return ret;
 }
 
 static __devexit int wm8904_i2c_remove(struct i2c_client *client)
 {
-	struct wm8904_priv *wm8904 = i2c_get_clientdata(client);
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(wm8904->regmap);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 679d7abdc7543e56abc41b8f4858f31a91259b29 Mon Sep 17 00:00:00 2001
From: Ola Lilja <ola.o.lilja@stericsson.com>
Date: Thu, 7 Jun 2012 14:00:21 +0200
Subject: ASoC: codecs: Add AB8500 codec-driver

Add codec-driver for ST-Ericsson AB8500 mixed-signal ASIC.

Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig        |    4 +
 sound/soc/codecs/Makefile       |    2 +
 sound/soc/codecs/ab8500-codec.c | 2521 +++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/ab8500-codec.h |  590 +++++++++
 4 files changed, 3117 insertions(+)
 create mode 100644 sound/soc/codecs/ab8500-codec.c
 create mode 100644 sound/soc/codecs/ab8500-codec.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8b879c71c23c..f63776d422b3 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
 	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
+	select SND_SOC_AB8500_CODEC if ABX500_CORE
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1836 if SPI_MASTER
 	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
@@ -132,6 +133,9 @@ config SND_SOC_WM_HUBS
 	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
 	default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m
 
+config SND_SOC_AB8500_CODEC
+	tristate
+
 config SND_SOC_AC97_CODEC
 	tristate
 	select SND_AC97_CODEC
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index e50811b182a4..fc93b4b0c2c5 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,4 +1,5 @@
 snd-soc-88pm860x-objs := 88pm860x-codec.o
+snd-soc-ab8500-codec-objs := ab8500-codec.o
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad193x-objs := ad193x.o
@@ -109,6 +110,7 @@ snd-soc-max9877-objs := max9877.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 
 obj-$(CONFIG_SND_SOC_88PM860X)	+= snd-soc-88pm860x.o
+obj-$(CONFIG_SND_SOC_AB8500_CODEC)	+= snd-soc-ab8500-codec.o
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
new file mode 100644
index 000000000000..95dc7d5bb076
--- /dev/null
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -0,0 +1,2521 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ *         for ST-Ericsson.
+ *
+ *         Based on the early work done by:
+ *         Mikko J. Lehto <mikko.lehto@symbio.com>,
+ *         Mikko Sarmanne <mikko.sarmanne@symbio.com>,
+ *         Jarmo K. Kuronen <jarmo.kuronen@symbio.com>,
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mutex.h>
+#include <linux/mfd/abx500/ab8500.h>
+#include <linux/mfd/abx500.h>
+#include <linux/mfd/abx500/ab8500-sysctrl.h>
+#include <linux/mfd/abx500/ab8500-codec.h>
+#include <linux/regulator/consumer.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "ab8500-codec.h"
+
+/* Macrocell value definitions */
+#define CLK_32K_OUT2_DISABLE			0x01
+#define INACTIVE_RESET_AUDIO			0x02
+#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK		0x10
+#define ENABLE_VINTCORE12_SUPPLY		0x04
+#define GPIO27_DIR_OUTPUT			0x04
+#define GPIO29_DIR_OUTPUT			0x10
+#define GPIO31_DIR_OUTPUT			0x40
+
+/* Macrocell register definitions */
+#define AB8500_CTRL3_REG			0x0200
+#define AB8500_GPIO_DIR4_REG			0x1013
+
+/* Nr of FIR/IIR-coeff banks in ANC-block */
+#define AB8500_NR_OF_ANC_COEFF_BANKS		2
+
+/* Minimum duration to keep ANC IIR Init bit high or
+low before proceeding with the configuration sequence */
+#define AB8500_ANC_SM_DELAY			2000
+
+#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.info = filter_control_info, \
+	.get = filter_control_get, .put = filter_control_put, \
+	.private_value = (unsigned long)&(struct filter_control) \
+		{.count = xcount, .min = xmin, .max = xmax} }
+
+struct filter_control {
+	long min, max;
+	unsigned int count;
+	long value[128];
+};
+
+/* Sidetone states */
+static const char * const enum_sid_state[] = {
+	"Unconfigured",
+	"Apply FIR",
+	"FIR is configured",
+};
+enum sid_state {
+	SID_UNCONFIGURED = 0,
+	SID_APPLY_FIR = 1,
+	SID_FIR_CONFIGURED = 2,
+};
+
+static const char * const enum_anc_state[] = {
+	"Unconfigured",
+	"Apply FIR and IIR",
+	"FIR and IIR are configured",
+	"Apply FIR",
+	"FIR is configured",
+	"Apply IIR",
+	"IIR is configured"
+};
+enum anc_state {
+	ANC_UNCONFIGURED = 0,
+	ANC_APPLY_FIR_IIR = 1,
+	ANC_FIR_IIR_CONFIGURED = 2,
+	ANC_APPLY_FIR = 3,
+	ANC_FIR_CONFIGURED = 4,
+	ANC_APPLY_IIR = 5,
+	ANC_IIR_CONFIGURED = 6
+};
+
+/* Analog microphones */
+enum amic_idx {
+	AMIC_IDX_1A,
+	AMIC_IDX_1B,
+	AMIC_IDX_2
+};
+
+struct ab8500_codec_drvdata_dbg {
+	struct regulator *vaud;
+	struct regulator *vamic1;
+	struct regulator *vamic2;
+	struct regulator *vdmic;
+};
+
+/* Private data for AB8500 device-driver */
+struct ab8500_codec_drvdata {
+	/* Sidetone */
+	long *sid_fir_values;
+	enum sid_state sid_status;
+
+	/* ANC */
+	struct mutex anc_lock;
+	long *anc_fir_values;
+	long *anc_iir_values;
+	enum anc_state anc_status;
+};
+
+static inline const char *amic_micbias_str(enum amic_micbias micbias)
+{
+	switch (micbias) {
+	case AMIC_MICBIAS_VAMIC1:
+		return "VAMIC1";
+	case AMIC_MICBIAS_VAMIC2:
+		return "VAMIC2";
+	default:
+		return "Unknown";
+	}
+}
+
+static inline const char *amic_type_str(enum amic_type type)
+{
+	switch (type) {
+	case AMIC_TYPE_DIFFERENTIAL:
+		return "DIFFERENTIAL";
+	case AMIC_TYPE_SINGLE_ENDED:
+		return "SINGLE ENDED";
+	default:
+		return "Unknown";
+	}
+}
+
+/*
+ * Read'n'write functions
+ */
+
+/* Read a register from the audio-bank of AB8500 */
+static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec,
+					unsigned int reg)
+{
+	int status;
+	unsigned int value = 0;
+
+	u8 value8;
+	status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO,
+						reg, &value8);
+	if (status < 0) {
+		dev_err(codec->dev,
+			"%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n",
+			__func__, (u8)AB8500_AUDIO, (u8)reg, status);
+	} else {
+		dev_dbg(codec->dev,
+			"%s: Read 0x%02x from register 0x%02x:0x%02x\n",
+			__func__, value8, (u8)AB8500_AUDIO, (u8)reg);
+		value = (unsigned int)value8;
+	}
+
+	return value;
+}
+
+/* Write to a register in the audio-bank of AB8500 */
+static int ab8500_codec_write_reg(struct snd_soc_codec *codec,
+				unsigned int reg, unsigned int value)
+{
+	int status;
+
+	status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO,
+						reg, value);
+	if (status < 0)
+		dev_err(codec->dev,
+			"%s: ERROR: Register (%02x:%02x) write failed (%d).\n",
+			__func__, (u8)AB8500_AUDIO, (u8)reg, status);
+	else
+		dev_dbg(codec->dev,
+			"%s: Wrote 0x%02x into register %02x:%02x\n",
+			__func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg);
+
+	return status;
+}
+
+/*
+ * Controls - DAPM
+ */
+
+/* Earpiece */
+
+/* Earpiece source selector */
+static const char * const enum_ear_lineout_source[] = {"Headset Left",
+						"Speaker Left"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source);
+static const struct snd_kcontrol_new dapm_ear_lineout_source =
+	SOC_DAPM_ENUM("Earpiece or LineOut Mono Source",
+		dapm_enum_ear_lineout_source);
+
+/* LineOut */
+
+/* LineOut source selector */
+static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"};
+static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, AB8500_ANACONF5,
+			AB8500_ANACONF5_HSLDACTOLOL,
+			AB8500_ANACONF5_HSRDACTOLOR, enum_lineout_source);
+static const struct snd_kcontrol_new dapm_lineout_source[] = {
+	SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source),
+};
+
+/* Handsfree */
+
+/* Speaker Left - ANC selector */
+static const char * const enum_HFx_sel[] = {"Audio Path", "ANC"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_HFl_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_HFLSEL, enum_HFx_sel);
+static const struct snd_kcontrol_new dapm_HFl_select[] = {
+	SOC_DAPM_ENUM("Speaker Left Source", dapm_enum_HFl_sel),
+};
+
+/* Speaker Right - ANC selector */
+static SOC_ENUM_SINGLE_DECL(dapm_enum_HFr_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_HFRSEL, enum_HFx_sel);
+static const struct snd_kcontrol_new dapm_HFr_select[] = {
+	SOC_DAPM_ENUM("Speaker Right Source", dapm_enum_HFr_sel),
+};
+
+/* Mic 1 */
+
+/* Mic 1 - Mic 1a or 1b selector */
+static const char * const enum_mic1ab_sel[] = {"Mic 1b", "Mic 1a"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, AB8500_ANACONF3,
+			AB8500_ANACONF3_MIC1SEL, enum_mic1ab_sel);
+static const struct snd_kcontrol_new dapm_mic1ab_mux[] = {
+	SOC_DAPM_ENUM("Mic 1a or 1b Select", dapm_enum_mic1ab_sel),
+};
+
+/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */
+static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD3SEL, enum_ad3_sel);
+static const struct snd_kcontrol_new dapm_ad3_select[] = {
+	SOC_DAPM_ENUM("AD3 Source Select", dapm_enum_ad3_sel),
+};
+
+/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */
+static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD6SEL, enum_ad6_sel);
+static const struct snd_kcontrol_new dapm_ad6_select[] = {
+	SOC_DAPM_ENUM("AD6 Source Select", dapm_enum_ad6_sel),
+};
+
+/* Mic 2 */
+
+/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */
+static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD5SEL, enum_ad5_sel);
+static const struct snd_kcontrol_new dapm_ad5_select[] = {
+	SOC_DAPM_ENUM("AD5 Source Select", dapm_enum_ad5_sel),
+};
+
+/* LineIn */
+
+/* LineIn left - AD1 - LineIn Left or DMic 1 selector */
+static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD1SEL, enum_ad1_sel);
+static const struct snd_kcontrol_new dapm_ad1_select[] = {
+	SOC_DAPM_ENUM("AD1 Source Select", dapm_enum_ad1_sel),
+};
+
+/* LineIn right - Mic 2 or LineIn Right selector */
+static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, AB8500_ANACONF3,
+			AB8500_ANACONF3_LINRSEL, enum_mic2lr_sel);
+static const struct snd_kcontrol_new dapm_mic2lr_select[] = {
+	SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel),
+};
+
+/* LineIn right - AD2 - LineIn Right or DMic2 selector */
+static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD2SEL, enum_ad2_sel);
+static const struct snd_kcontrol_new dapm_ad2_select[] = {
+	SOC_DAPM_ENUM("AD2 Source Select", dapm_enum_ad2_sel),
+};
+
+
+/* ANC */
+
+static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6",
+					"Mic 2 / DMic 5"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_ANCINSEL, enum_anc_in_sel);
+static const struct snd_kcontrol_new dapm_anc_in_select[] = {
+	SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel),
+};
+
+/* ANC - Enable/Disable */
+static const struct snd_kcontrol_new dapm_anc_enable[] = {
+	SOC_DAPM_SINGLE("Switch", AB8500_ANCCONF1,
+			AB8500_ANCCONF1_ENANC, 0, 0),
+};
+
+/* ANC to Earpiece - Mute */
+static const struct snd_kcontrol_new dapm_anc_ear_mute[] = {
+	SOC_DAPM_SINGLE("Switch", AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_ANCSEL, 1, 0),
+};
+
+
+
+/* Sidetone left */
+
+/* Sidetone left - Input selector */
+static const char * const enum_stfir1_in_sel[] = {
+	"LineIn Left", "LineIn Right", "Mic 1", "Headset Left"
+};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel);
+static const struct snd_kcontrol_new dapm_stfir1_in_select[] = {
+	SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel),
+};
+
+/* Sidetone right path */
+
+/* Sidetone right - Input selector */
+static const char * const enum_stfir2_in_sel[] = {
+	"LineIn Right", "Mic 1", "DMic 4", "Headset Right"
+};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel);
+static const struct snd_kcontrol_new dapm_stfir2_in_select[] = {
+	SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel),
+};
+
+/* Vibra */
+
+static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"};
+
+static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, AB8500_PWMGENCONF1,
+			AB8500_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx);
+
+static const struct snd_kcontrol_new dapm_pwm2vib1[] = {
+	SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1),
+};
+
+static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, AB8500_PWMGENCONF1,
+			AB8500_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx);
+
+static const struct snd_kcontrol_new dapm_pwm2vib2[] = {
+	SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2),
+};
+
+/*
+ * DAPM-widgets
+ */
+
+static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = {
+
+	/* Clocks */
+	SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"),
+
+	/* Regulators */
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0),
+
+	/* Power */
+	SND_SOC_DAPM_SUPPLY("Audio Power",
+			AB8500_POWERUP, AB8500_POWERUP_POWERUP, 0,
+			NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("Audio Analog Power",
+			AB8500_POWERUP, AB8500_POWERUP_ENANA, 0,
+			NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Main supply node */
+	SND_SOC_DAPM_SUPPLY("Main Supply", SND_SOC_NOPM, 0, 0,
+			NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* DA/AD */
+
+	SND_SOC_DAPM_INPUT("ADC Input"),
+	SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUTPUT("DAC Output"),
+
+	SND_SOC_DAPM_AIF_IN("DA_IN1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN6", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT57", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT68", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+	/* Headset path */
+
+	SND_SOC_DAPM_SUPPLY("Charge Pump", AB8500_ANACONF5,
+			AB8500_ANACONF5_ENCPHS, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA1, 0),
+	SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA2, 0),
+
+	SND_SOC_DAPM_PGA("HSL Digital Volume", SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("HSR Digital Volume", SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSL, 0),
+	SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSR, 0),
+	SND_SOC_DAPM_MIXER("HSL DAC Mute", AB8500_MUTECONF,
+			AB8500_MUTECONF_MUTDACHSL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSR DAC Mute", AB8500_MUTECONF,
+			AB8500_MUTECONF_MUTDACHSR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p",
+			AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSL, 0),
+	SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p",
+			AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSR, 0),
+
+	SND_SOC_DAPM_MIXER("HSL Mute",
+			AB8500_MUTECONF, AB8500_MUTECONF_MUTHSL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSR Mute",
+			AB8500_MUTECONF, AB8500_MUTECONF_MUTHSR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSL Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHSL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSR Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHSR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("HSL Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("HSR Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Headset Left"),
+	SND_SOC_DAPM_OUTPUT("Headset Right"),
+
+	/* LineOut path */
+
+	SND_SOC_DAPM_MUX("LineOut Source",
+			SND_SOC_NOPM, 0, 0, dapm_lineout_source),
+
+	SND_SOC_DAPM_MIXER("LOL Disable HFL",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LOR Disable HFR",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 1,
+			NULL, 0),
+
+	SND_SOC_DAPM_MIXER("LOL Enable",
+			AB8500_ANACONF5, AB8500_ANACONF5_ENLOL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LOR Enable",
+			AB8500_ANACONF5, AB8500_ANACONF5_ENLOR, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LineOut Left"),
+	SND_SOC_DAPM_OUTPUT("LineOut Right"),
+
+	/* Earpiece path */
+
+	SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source",
+			SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source),
+	SND_SOC_DAPM_MIXER("EAR DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACEAR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("EAR Mute",
+			AB8500_MUTECONF, AB8500_MUTECONF_MUTEAR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("EAR Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENEAR, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Earpiece"),
+
+	/* Handsfree path */
+
+	SND_SOC_DAPM_MIXER("DA3 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA3, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA4 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA4, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("Speaker Left Source",
+			SND_SOC_NOPM, 0, 0, dapm_HFl_select),
+	SND_SOC_DAPM_MUX("Speaker Right Source",
+			SND_SOC_NOPM, 0, 0, dapm_HFr_select),
+	SND_SOC_DAPM_MIXER("HFL DAC", AB8500_DAPATHCONF,
+			AB8500_DAPATHCONF_ENDACHFL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HFR DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHFR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR",
+			AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFREN, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL",
+			AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFLEN, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HFL Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HFR Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Speaker Left"),
+	SND_SOC_DAPM_OUTPUT("Speaker Right"),
+
+	/* Vibrator path */
+
+	SND_SOC_DAPM_INPUT("PWMGEN1"),
+	SND_SOC_DAPM_INPUT("PWMGEN2"),
+
+	SND_SOC_DAPM_MIXER("DA5 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA5, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA6 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA6, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("VIB1 DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("VIB2 DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB2, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("Vibra 1 Controller",
+			SND_SOC_NOPM, 0, 0, dapm_pwm2vib1),
+	SND_SOC_DAPM_MUX("Vibra 2 Controller",
+			SND_SOC_NOPM, 0, 0, dapm_pwm2vib2),
+	SND_SOC_DAPM_MIXER("VIB1 Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENVIB1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("VIB2 Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENVIB2, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Vibra 1"),
+	SND_SOC_DAPM_OUTPUT("Vibra 2"),
+
+	/* Mic 1 */
+
+	SND_SOC_DAPM_INPUT("Mic 1"),
+
+	SND_SOC_DAPM_MUX("Mic 1a or 1b Select",
+			SND_SOC_NOPM, 0, 0, dapm_mic1ab_mux),
+	SND_SOC_DAPM_MIXER("MIC1 Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC1, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC1A V-AMICx Enable",
+			AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC1B V-AMICx Enable",
+			AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC1 ADC",
+			AB8500_ANACONF3, AB8500_ANACONF3_ENADCMIC, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("AD3 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad3_select),
+	SND_SOC_DAPM_MIXER("AD3 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD3 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, 0,
+			NULL, 0),
+
+	/* Mic 2 */
+
+	SND_SOC_DAPM_INPUT("Mic 2"),
+
+	SND_SOC_DAPM_MIXER("MIC2 Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC2, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC2 V-AMICx Enable", AB8500_ANACONF2,
+			AB8500_ANACONF2_ENMIC2, 0,
+			NULL, 0),
+
+	/* LineIn */
+
+	SND_SOC_DAPM_INPUT("LineIn Left"),
+	SND_SOC_DAPM_INPUT("LineIn Right"),
+
+	SND_SOC_DAPM_MIXER("LINL Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTLINL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTLINR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINL Enable", AB8500_ANACONF2,
+			AB8500_ANACONF2_ENLINL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR Enable", AB8500_ANACONF2,
+			AB8500_ANACONF2_ENLINR, 0,
+			NULL, 0),
+
+	/* LineIn Bypass path */
+	SND_SOC_DAPM_MIXER("LINL to HSL Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR to HSR Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	/* LineIn, Mic 2 */
+	SND_SOC_DAPM_MUX("Mic 2 or LINR Select",
+			SND_SOC_NOPM, 0, 0, dapm_mic2lr_select),
+	SND_SOC_DAPM_MIXER("LINL ADC", AB8500_ANACONF3,
+			AB8500_ANACONF3_ENADCLINL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR ADC", AB8500_ANACONF3,
+			AB8500_ANACONF3_ENADCLINR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("AD1 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad1_select),
+	SND_SOC_DAPM_MUX("AD2 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad2_select),
+	SND_SOC_DAPM_MIXER("AD1 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD2 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_MIXER("AD12 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD12, 0,
+			NULL, 0),
+
+	/* HD Capture path */
+
+	SND_SOC_DAPM_MUX("AD5 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad5_select),
+	SND_SOC_DAPM_MUX("AD6 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad6_select),
+	SND_SOC_DAPM_MIXER("AD5 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD6 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD57 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD68 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0,
+			NULL, 0),
+
+	/* Digital Microphone path */
+
+	SND_SOC_DAPM_INPUT("DMic 1"),
+	SND_SOC_DAPM_INPUT("DMic 2"),
+	SND_SOC_DAPM_INPUT("DMic 3"),
+	SND_SOC_DAPM_INPUT("DMic 4"),
+	SND_SOC_DAPM_INPUT("DMic 5"),
+	SND_SOC_DAPM_INPUT("DMic 6"),
+
+	SND_SOC_DAPM_MIXER("DMIC1",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC2",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC2, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC3",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC3, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC4",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC4, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC5",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC5, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC6",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC6, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD4 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD4 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34,
+			0, NULL, 0),
+
+	/* Acoustical Noise Cancellation path */
+
+	SND_SOC_DAPM_INPUT("ANC Configure Input"),
+	SND_SOC_DAPM_OUTPUT("ANC Configure Output"),
+
+	SND_SOC_DAPM_MUX("ANC Source",
+			SND_SOC_NOPM, 0, 0,
+			dapm_anc_in_select),
+	SND_SOC_DAPM_SWITCH("ANC",
+			SND_SOC_NOPM, 0, 0,
+			dapm_anc_enable),
+	SND_SOC_DAPM_SWITCH("ANC to Earpiece",
+			SND_SOC_NOPM, 0, 0,
+			dapm_anc_ear_mute),
+
+	/* Sidetone Filter path */
+
+	SND_SOC_DAPM_MUX("Sidetone Left Source",
+			SND_SOC_NOPM, 0, 0,
+			dapm_stfir1_in_select),
+	SND_SOC_DAPM_MUX("Sidetone Right Source",
+			SND_SOC_NOPM, 0, 0,
+			dapm_stfir2_in_select),
+	SND_SOC_DAPM_MIXER("STFIR1 Control",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("STFIR2 Control",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("STFIR1 Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("STFIR2 Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+};
+
+/*
+ * DAPM-routes
+ */
+static const struct snd_soc_dapm_route ab8500_dapm_routes[] = {
+	/* Power AB8500 audio-block when AD/DA is active */
+	{"Main Supply", NULL, "V-AUD"},
+	{"Main Supply", NULL, "audioclk"},
+	{"Main Supply", NULL, "Audio Power"},
+	{"Main Supply", NULL, "Audio Analog Power"},
+
+	{"DAC", NULL, "ab8500_0p"},
+	{"DAC", NULL, "Main Supply"},
+	{"ADC", NULL, "ab8500_0c"},
+	{"ADC", NULL, "Main Supply"},
+
+	/* ANC Configure */
+	{"ANC Configure Input", NULL, "Main Supply"},
+	{"ANC Configure Output", NULL, "ANC Configure Input"},
+
+	/* AD/DA */
+	{"ADC", NULL, "ADC Input"},
+	{"DAC Output", NULL, "DAC"},
+
+	/* Powerup charge pump if DA1/2 is in use */
+
+	{"DA_IN1", NULL, "ab8500_0p"},
+	{"DA_IN1", NULL, "Charge Pump"},
+	{"DA_IN2", NULL, "ab8500_0p"},
+	{"DA_IN2", NULL, "Charge Pump"},
+
+	/* Headset path */
+
+	{"DA1 Enable", NULL, "DA_IN1"},
+	{"DA2 Enable", NULL, "DA_IN2"},
+
+	{"HSL Digital Volume", NULL, "DA1 Enable"},
+	{"HSR Digital Volume", NULL, "DA2 Enable"},
+
+	{"HSL DAC", NULL, "HSL Digital Volume"},
+	{"HSR DAC", NULL, "HSR Digital Volume"},
+
+	{"HSL DAC Mute", NULL, "HSL DAC"},
+	{"HSR DAC Mute", NULL, "HSR DAC"},
+
+	{"HSL DAC Driver", NULL, "HSL DAC Mute"},
+	{"HSR DAC Driver", NULL, "HSR DAC Mute"},
+
+	{"HSL Mute", NULL, "HSL DAC Driver"},
+	{"HSR Mute", NULL, "HSR DAC Driver"},
+
+	{"HSL Enable", NULL, "HSL Mute"},
+	{"HSR Enable", NULL, "HSR Mute"},
+
+	{"HSL Volume", NULL, "HSL Enable"},
+	{"HSR Volume", NULL, "HSR Enable"},
+
+	{"Headset Left", NULL, "HSL Volume"},
+	{"Headset Right", NULL, "HSR Volume"},
+
+	/* HF or LineOut path */
+
+	{"DA_IN3", NULL, "ab8500_0p"},
+	{"DA3 Channel Volume", NULL, "DA_IN3"},
+	{"DA_IN4", NULL, "ab8500_0p"},
+	{"DA4 Channel Volume", NULL, "DA_IN4"},
+
+	{"Speaker Left Source", "Audio Path", "DA3 Channel Volume"},
+	{"Speaker Right Source", "Audio Path", "DA4 Channel Volume"},
+
+	{"DA3 or ANC path to HfL", NULL, "Speaker Left Source"},
+	{"DA4 or ANC path to HfR", NULL, "Speaker Right Source"},
+
+	/* HF path */
+
+	{"HFL DAC", NULL, "DA3 or ANC path to HfL"},
+	{"HFR DAC", NULL, "DA4 or ANC path to HfR"},
+
+	{"HFL Enable", NULL, "HFL DAC"},
+	{"HFR Enable", NULL, "HFR DAC"},
+
+	{"Speaker Left", NULL, "HFL Enable"},
+	{"Speaker Right", NULL, "HFR Enable"},
+
+	/* Earpiece path */
+
+	{"Earpiece or LineOut Mono Source", "Headset Left",
+		"HSL Digital Volume"},
+	{"Earpiece or LineOut Mono Source", "Speaker Left",
+		"DA3 or ANC path to HfL"},
+
+	{"EAR DAC", NULL, "Earpiece or LineOut Mono Source"},
+
+	{"EAR Mute", NULL, "EAR DAC"},
+
+	{"EAR Enable", NULL, "EAR Mute"},
+
+	{"Earpiece", NULL, "EAR Enable"},
+
+	/* LineOut path stereo */
+
+	{"LineOut Source", "Stereo Path", "HSL DAC Driver"},
+	{"LineOut Source", "Stereo Path", "HSR DAC Driver"},
+
+	/* LineOut path mono */
+
+	{"LineOut Source", "Mono Path", "EAR DAC"},
+
+	/* LineOut path */
+
+	{"LOL Disable HFL", NULL, "LineOut Source"},
+	{"LOR Disable HFR", NULL, "LineOut Source"},
+
+	{"LOL Enable", NULL, "LOL Disable HFL"},
+	{"LOR Enable", NULL, "LOR Disable HFR"},
+
+	{"LineOut Left", NULL, "LOL Enable"},
+	{"LineOut Right", NULL, "LOR Enable"},
+
+	/* Vibrator path */
+
+	{"DA_IN5", NULL, "ab8500_0p"},
+	{"DA5 Channel Volume", NULL, "DA_IN5"},
+	{"DA_IN6", NULL, "ab8500_0p"},
+	{"DA6 Channel Volume", NULL, "DA_IN6"},
+
+	{"VIB1 DAC", NULL, "DA5 Channel Volume"},
+	{"VIB2 DAC", NULL, "DA6 Channel Volume"},
+
+	{"Vibra 1 Controller", "Audio Path", "VIB1 DAC"},
+	{"Vibra 2 Controller", "Audio Path", "VIB2 DAC"},
+	{"Vibra 1 Controller", "PWM Generator", "PWMGEN1"},
+	{"Vibra 2 Controller", "PWM Generator", "PWMGEN2"},
+
+	{"VIB1 Enable", NULL, "Vibra 1 Controller"},
+	{"VIB2 Enable", NULL, "Vibra 2 Controller"},
+
+	{"Vibra 1", NULL, "VIB1 Enable"},
+	{"Vibra 2", NULL, "VIB2 Enable"},
+
+
+	/* Mic 2 */
+
+	{"MIC2 V-AMICx Enable", NULL, "Mic 2"},
+
+	/* LineIn */
+	{"LINL Mute", NULL, "LineIn Left"},
+	{"LINR Mute", NULL, "LineIn Right"},
+
+	{"LINL Enable", NULL, "LINL Mute"},
+	{"LINR Enable", NULL, "LINR Mute"},
+
+	/* LineIn, Mic 2 */
+	{"Mic 2 or LINR Select", "LineIn Right", "LINR Enable"},
+	{"Mic 2 or LINR Select", "Mic 2", "MIC2 V-AMICx Enable"},
+
+	{"LINL ADC", NULL, "LINL Enable"},
+	{"LINR ADC", NULL, "Mic 2 or LINR Select"},
+
+	{"AD1 Source Select", "LineIn Left", "LINL ADC"},
+	{"AD2 Source Select", "LineIn Right", "LINR ADC"},
+
+	{"AD1 Channel Volume", NULL, "AD1 Source Select"},
+	{"AD2 Channel Volume", NULL, "AD2 Source Select"},
+
+	{"AD12 Enable", NULL, "AD1 Channel Volume"},
+	{"AD12 Enable", NULL, "AD2 Channel Volume"},
+
+	{"AD_OUT1", NULL, "ab8500_0c"},
+	{"AD_OUT1", NULL, "AD12 Enable"},
+	{"AD_OUT2", NULL, "ab8500_0c"},
+	{"AD_OUT2", NULL, "AD12 Enable"},
+
+	/* Mic 1 */
+
+	{"MIC1 Mute", NULL, "Mic 1"},
+
+	{"MIC1A V-AMICx Enable", NULL, "MIC1 Mute"},
+	{"MIC1B V-AMICx Enable", NULL, "MIC1 Mute"},
+
+	{"Mic 1a or 1b Select", "Mic 1a", "MIC1A V-AMICx Enable"},
+	{"Mic 1a or 1b Select", "Mic 1b", "MIC1B V-AMICx Enable"},
+
+	{"MIC1 ADC", NULL, "Mic 1a or 1b Select"},
+
+	{"AD3 Source Select", "Mic 1", "MIC1 ADC"},
+
+	{"AD3 Channel Volume", NULL, "AD3 Source Select"},
+
+	{"AD3 Enable", NULL, "AD3 Channel Volume"},
+
+	{"AD_OUT3", NULL, "ab8500_0c"},
+	{"AD_OUT3", NULL, "AD3 Enable"},
+
+	/* HD Capture path */
+
+	{"AD5 Source Select", "Mic 2", "LINR ADC"},
+	{"AD6 Source Select", "Mic 1", "MIC1 ADC"},
+
+	{"AD5 Channel Volume", NULL, "AD5 Source Select"},
+	{"AD6 Channel Volume", NULL, "AD6 Source Select"},
+
+	{"AD57 Enable", NULL, "AD5 Channel Volume"},
+	{"AD68 Enable", NULL, "AD6 Channel Volume"},
+
+	{"AD_OUT57", NULL, "ab8500_0c"},
+	{"AD_OUT57", NULL, "AD57 Enable"},
+	{"AD_OUT68", NULL, "ab8500_0c"},
+	{"AD_OUT68", NULL, "AD68 Enable"},
+
+	/* Digital Microphone path */
+
+	{"DMic 1", NULL, "V-DMIC"},
+	{"DMic 2", NULL, "V-DMIC"},
+	{"DMic 3", NULL, "V-DMIC"},
+	{"DMic 4", NULL, "V-DMIC"},
+	{"DMic 5", NULL, "V-DMIC"},
+	{"DMic 6", NULL, "V-DMIC"},
+
+	{"AD1 Source Select", NULL, "DMic 1"},
+	{"AD2 Source Select", NULL, "DMic 2"},
+	{"AD3 Source Select", NULL, "DMic 3"},
+	{"AD5 Source Select", NULL, "DMic 5"},
+	{"AD6 Source Select", NULL, "DMic 6"},
+
+	{"AD4 Channel Volume", NULL, "DMic 4"},
+	{"AD4 Enable", NULL, "AD4 Channel Volume"},
+
+	{"AD_OUT4", NULL, "ab8500_0c"},
+	{"AD_OUT4", NULL, "AD4 Enable"},
+
+	/* LineIn Bypass path */
+
+	{"LINL to HSL Volume", NULL, "LINL Enable"},
+	{"LINR to HSR Volume", NULL, "LINR Enable"},
+
+	{"HSL DAC Driver", NULL, "LINL to HSL Volume"},
+	{"HSR DAC Driver", NULL, "LINR to HSR Volume"},
+
+	/* ANC path (Acoustic Noise Cancellation) */
+
+	{"ANC Source", "Mic 2 / DMic 5", "AD5 Channel Volume"},
+	{"ANC Source", "Mic 1 / DMic 6", "AD6 Channel Volume"},
+
+	{"ANC", "Switch", "ANC Source"},
+
+	{"Speaker Left Source", "ANC", "ANC"},
+	{"Speaker Right Source", "ANC", "ANC"},
+	{"ANC to Earpiece", "Switch", "ANC"},
+
+	{"HSL Digital Volume", NULL, "ANC to Earpiece"},
+
+	/* Sidetone Filter path */
+
+	{"Sidetone Left Source", "LineIn Left", "AD12 Enable"},
+	{"Sidetone Left Source", "LineIn Right", "AD12 Enable"},
+	{"Sidetone Left Source", "Mic 1", "AD3 Enable"},
+	{"Sidetone Left Source", "Headset Left", "DA_IN1"},
+	{"Sidetone Right Source", "LineIn Right", "AD12 Enable"},
+	{"Sidetone Right Source", "Mic 1", "AD3 Enable"},
+	{"Sidetone Right Source", "DMic 4", "AD4 Enable"},
+	{"Sidetone Right Source", "Headset Right", "DA_IN2"},
+
+	{"STFIR1 Control", NULL, "Sidetone Left Source"},
+	{"STFIR2 Control", NULL, "Sidetone Right Source"},
+
+	{"STFIR1 Volume", NULL, "STFIR1 Control"},
+	{"STFIR2 Volume", NULL, "STFIR2 Control"},
+
+	{"DA1 Enable", NULL, "STFIR1 Volume"},
+	{"DA2 Enable", NULL, "STFIR2 Volume"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1a_vamicx[] = {
+	{"MIC1A V-AMICx Enable", NULL, "V-AMIC1"},
+	{"MIC1A V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1b_vamicx[] = {
+	{"MIC1B V-AMICx Enable", NULL, "V-AMIC1"},
+	{"MIC1B V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = {
+	{"MIC2 V-AMICx Enable", NULL, "V-AMIC1"},
+	{"MIC2 V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+/* ANC FIR-coefficients configuration sequence */
+static void anc_fir(struct snd_soc_codec *codec,
+		unsigned int bnk, unsigned int par, unsigned int val)
+{
+	if (par == 0 && bnk == 0)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ANCFIRUPDATE),
+			BIT(AB8500_ANCCONF1_ANCFIRUPDATE));
+
+	snd_soc_write(codec, AB8500_ANCCONF5, val >> 8 & 0xff);
+	snd_soc_write(codec, AB8500_ANCCONF6, val &  0xff);
+
+	if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0);
+}
+
+/* ANC IIR-coefficients configuration sequence */
+static void anc_iir(struct snd_soc_codec *codec, unsigned int bnk,
+		unsigned int par, unsigned int val)
+{
+	if (par == 0) {
+		if (bnk == 0) {
+			snd_soc_update_bits(codec, AB8500_ANCCONF1,
+					BIT(AB8500_ANCCONF1_ANCIIRINIT),
+					BIT(AB8500_ANCCONF1_ANCIIRINIT));
+			usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);
+			snd_soc_update_bits(codec, AB8500_ANCCONF1,
+					BIT(AB8500_ANCCONF1_ANCIIRINIT), 0);
+			usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);
+		} else {
+			snd_soc_update_bits(codec, AB8500_ANCCONF1,
+					BIT(AB8500_ANCCONF1_ANCIIRUPDATE),
+					BIT(AB8500_ANCCONF1_ANCIIRUPDATE));
+		}
+	} else if (par > 3) {
+		snd_soc_write(codec, AB8500_ANCCONF7, 0);
+		snd_soc_write(codec, AB8500_ANCCONF8, val >> 16 & 0xff);
+	}
+
+	snd_soc_write(codec, AB8500_ANCCONF7, val >> 8 & 0xff);
+	snd_soc_write(codec, AB8500_ANCCONF8, val & 0xff);
+
+	if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0);
+}
+
+/* ANC IIR-/FIR-coefficients configuration sequence */
+static void anc_configure(struct snd_soc_codec *codec,
+			bool apply_fir, bool apply_iir)
+{
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+	unsigned int bnk, par, val;
+
+	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+	if (apply_fir)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ENANC), 0);
+
+	snd_soc_update_bits(codec, AB8500_ANCCONF1,
+		BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC));
+
+	if (apply_fir)
+		for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
+			for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) {
+				val = snd_soc_read(codec,
+						drvdata->anc_fir_values[par]);
+				anc_fir(codec, bnk, par, val);
+			}
+
+	if (apply_iir)
+		for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
+			for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) {
+				val = snd_soc_read(codec,
+						drvdata->anc_iir_values[par]);
+				anc_iir(codec, bnk, par, val);
+			}
+
+	dev_dbg(codec->dev, "%s: Exit.\n", __func__);
+}
+
+/*
+ * Control-events
+ */
+
+static int sid_status_control_get(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+
+	mutex_lock(&codec->mutex);
+	ucontrol->value.integer.value[0] = drvdata->sid_status;
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+/* Write sidetone FIR-coefficients configuration sequence */
+static int sid_status_control_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+	unsigned int param, sidconf, val;
+	int status = 1;
+
+	dev_dbg(codec->dev, "%s: Enter\n", __func__);
+
+	if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) {
+		dev_err(codec->dev,
+			"%s: ERROR: This control supports '%s' only!\n",
+			__func__, enum_sid_state[SID_APPLY_FIR]);
+		return -EIO;
+	}
+
+	mutex_lock(&codec->mutex);
+
+	sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF);
+	if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) {
+		if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) {
+			dev_err(codec->dev, "%s: Sidetone busy while off!\n",
+				__func__);
+			status = -EPERM;
+		} else {
+			status = -EBUSY;
+		}
+		goto out;
+	}
+
+	snd_soc_write(codec, AB8500_SIDFIRADR, 0);
+
+	for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) {
+		val = snd_soc_read(codec, drvdata->sid_fir_values[param]);
+		snd_soc_write(codec, AB8500_SIDFIRCOEF1, val >> 8 & 0xff);
+		snd_soc_write(codec, AB8500_SIDFIRCOEF2, val & 0xff);
+	}
+
+	snd_soc_update_bits(codec, AB8500_SIDFIRADR,
+		BIT(AB8500_SIDFIRADR_FIRSIDSET),
+		BIT(AB8500_SIDFIRADR_FIRSIDSET));
+	snd_soc_update_bits(codec, AB8500_SIDFIRADR,
+		BIT(AB8500_SIDFIRADR_FIRSIDSET), 0);
+
+	drvdata->sid_status = SID_FIR_CONFIGURED;
+
+out:
+	mutex_unlock(&codec->mutex);
+
+	dev_dbg(codec->dev, "%s: Exit\n", __func__);
+
+	return status;
+}
+
+static int anc_status_control_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+
+	mutex_lock(&codec->mutex);
+	ucontrol->value.integer.value[0] = drvdata->anc_status;
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+static int anc_status_control_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+	struct device *dev = codec->dev;
+	bool apply_fir, apply_iir;
+	int req, status;
+
+	dev_dbg(dev, "%s: Enter.\n", __func__);
+
+	mutex_lock(&drvdata->anc_lock);
+
+	req = ucontrol->value.integer.value[0];
+	if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
+		req != ANC_APPLY_IIR) {
+		dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
+			__func__, enum_anc_state[req]);
+		return -EINVAL;
+	}
+	apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
+	apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
+
+	status = snd_soc_dapm_force_enable_pin(&codec->dapm,
+					"ANC Configure Input");
+	if (status < 0) {
+		dev_err(dev,
+			"%s: ERROR: Failed to enable power (status = %d)!\n",
+			__func__, status);
+		goto cleanup;
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+
+	mutex_lock(&codec->mutex);
+	anc_configure(codec, apply_fir, apply_iir);
+	mutex_unlock(&codec->mutex);
+
+	if (apply_fir) {
+		if (drvdata->anc_status == ANC_IIR_CONFIGURED)
+			drvdata->anc_status = ANC_FIR_IIR_CONFIGURED;
+		else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED)
+			drvdata->anc_status =  ANC_FIR_CONFIGURED;
+	}
+	if (apply_iir) {
+		if (drvdata->anc_status == ANC_FIR_CONFIGURED)
+			drvdata->anc_status = ANC_FIR_IIR_CONFIGURED;
+		else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED)
+			drvdata->anc_status =  ANC_IIR_CONFIGURED;
+	}
+
+	status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+	snd_soc_dapm_sync(&codec->dapm);
+
+cleanup:
+	mutex_unlock(&drvdata->anc_lock);
+
+	if (status < 0)
+		dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n",
+			__func__, status);
+
+	dev_dbg(dev, "%s: Exit.\n", __func__);
+
+	return (status < 0) ? status : 1;
+}
+
+static int filter_control_info(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_info *uinfo)
+{
+	struct filter_control *fc =
+			(struct filter_control *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = fc->count;
+	uinfo->value.integer.min = fc->min;
+	uinfo->value.integer.max = fc->max;
+
+	return 0;
+}
+
+static int filter_control_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct filter_control *fc =
+			(struct filter_control *)kcontrol->private_value;
+	unsigned int i;
+
+	mutex_lock(&codec->mutex);
+	for (i = 0; i < fc->count; i++)
+		ucontrol->value.integer.value[i] = fc->value[i];
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+static int filter_control_put(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct filter_control *fc =
+			(struct filter_control *)kcontrol->private_value;
+	unsigned int i;
+
+	mutex_lock(&codec->mutex);
+	for (i = 0; i < fc->count; i++)
+		fc->value[i] = ucontrol->value.integer.value[i];
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+/*
+ * Controls - Non-DAPM ASoC
+ */
+
+static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1);
+/* -32dB = Mute */
+
+static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1);
+/* -63dB = Mute */
+
+static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1);
+/* -1dB = Mute */
+
+static const unsigned int hs_gain_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0),
+	4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0);
+
+static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1);
+/* -38dB = Mute */
+
+static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms",
+					"5ms"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed,
+	AB8500_DIGMICCONF, AB8500_DIGMICCONF_HSFADSPEED, enum_hsfadspeed);
+
+static const char * const enum_envdetthre[] = {
+	"250mV", "300mV", "350mV", "400mV",
+	"450mV", "500mV", "550mV", "600mV",
+	"650mV", "700mV", "750mV", "800mV",
+	"850mV", "900mV", "950mV", "1.00V" };
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre,
+	AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETHTHRE, enum_envdetthre);
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre,
+	AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETLTHRE, enum_envdetthre);
+static const char * const enum_envdettime[] = {
+	"26.6us", "53.2us", "106us",  "213us",
+	"426us",  "851us",  "1.70ms", "3.40ms",
+	"6.81ms", "13.6ms", "27.2ms", "54.5ms",
+	"109ms",  "218ms",  "436ms",  "872ms" };
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime,
+	AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETTIME, enum_envdettime);
+
+static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, AB8500_HSLEARDIGGAIN,
+			AB8500_HSLEARDIGGAIN_HSSINC1, enum_sinc31);
+
+static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, AB8500_HSRDIGGAIN,
+			AB8500_HSRDIGGAIN_FADESPEED, enum_fadespeed);
+
+/* Earpiece */
+
+static const char * const enum_lowpow[] = {"Normal", "Low Power"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, AB8500_ANACONF1,
+			AB8500_ANACONF1_EARDACLOWPOW, enum_lowpow);
+static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, AB8500_ANACONF1,
+			AB8500_ANACONF1_EARDRVLOWPOW, enum_lowpow);
+
+static const char * const enum_av_mode[] = {"Audio", "Voice"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, AB8500_ADFILTCONF,
+	AB8500_ADFILTCONF_AD1VOICE, AB8500_ADFILTCONF_AD2VOICE, enum_av_mode);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, AB8500_ADFILTCONF,
+	AB8500_ADFILTCONF_AD3VOICE, AB8500_ADFILTCONF_AD4VOICE, enum_av_mode);
+
+/* DA */
+
+static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice,
+			AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DA12VOICE,
+			enum_av_mode);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice,
+			AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DA34VOICE,
+			enum_av_mode);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice,
+			AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DA56VOICE,
+			enum_av_mode);
+
+static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_DATOHSLEN,
+			AB8500_DIGMULTCONF1_DATOHSREN, enum_da2hslr);
+
+static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DMIC1SINC3,
+			AB8500_DMICFILTCONF_DMIC2SINC3, enum_sinc53);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DMIC3SINC3,
+			AB8500_DMICFILTCONF_DMIC4SINC3, enum_sinc53);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DMIC5SINC3,
+			AB8500_DMICFILTCONF_DMIC6SINC3, enum_sinc53);
+
+/* Digital interface - DA from slot mapping */
+static const char * const enum_da_from_slot_map[] = {"SLOT0",
+					"SLOT1",
+					"SLOT2",
+					"SLOT3",
+					"SLOT4",
+					"SLOT5",
+					"SLOT6",
+					"SLOT7",
+					"SLOT8",
+					"SLOT9",
+					"SLOT10",
+					"SLOT11",
+					"SLOT12",
+					"SLOT13",
+					"SLOT14",
+					"SLOT15",
+					"SLOT16",
+					"SLOT17",
+					"SLOT18",
+					"SLOT19",
+					"SLOT20",
+					"SLOT21",
+					"SLOT22",
+					"SLOT23",
+					"SLOT24",
+					"SLOT25",
+					"SLOT26",
+					"SLOT27",
+					"SLOT28",
+					"SLOT29",
+					"SLOT30",
+					"SLOT31"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap,
+			AB8500_DASLOTCONF1, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap,
+			AB8500_DASLOTCONF2, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap,
+			AB8500_DASLOTCONF3, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap,
+			AB8500_DASLOTCONF4, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap,
+			AB8500_DASLOTCONF5, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap,
+			AB8500_DASLOTCONF6, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap,
+			AB8500_DASLOTCONF7, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap,
+			AB8500_DASLOTCONF8, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+
+/* Digital interface - AD to slot mapping */
+static const char * const enum_ad_to_slot_map[] = {"AD_OUT1",
+					"AD_OUT2",
+					"AD_OUT3",
+					"AD_OUT4",
+					"AD_OUT5",
+					"AD_OUT6",
+					"AD_OUT7",
+					"AD_OUT8",
+					"zeroes",
+					"tristate"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map,
+			AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map,
+			AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map,
+			AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map,
+			AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map,
+			AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map,
+			AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map,
+			AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map,
+			AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map,
+			AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map,
+			AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map,
+			AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map,
+			AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map,
+			AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map,
+			AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map,
+			AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map,
+			AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map,
+			AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map,
+			AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map,
+			AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map,
+			AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map,
+			AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map,
+			AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map,
+			AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map,
+			AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map,
+			AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map,
+			AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map,
+			AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map,
+			AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map,
+			AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map,
+			AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map,
+			AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map,
+			AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+
+/* Digital interface - Burst mode */
+static const char * const enum_mask[] = {"Unmasked", "Masked"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask,
+			AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOMASK,
+			enum_mask);
+static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2,
+			AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFO19M2,
+			enum_bitclk0);
+static const char * const enum_slavemaster[] = {"Slave", "Master"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast,
+			AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT,
+			enum_slavemaster);
+
+/* Sidetone */
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state);
+
+/* ANC */
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state);
+
+static struct snd_kcontrol_new ab8500_ctrls[] = {
+	/* Charge pump */
+	SOC_ENUM("Charge Pump High Threshold For Low Voltage",
+		soc_enum_envdeththre),
+	SOC_ENUM("Charge Pump Low Threshold For Low Voltage",
+		soc_enum_envdetlthre),
+	SOC_SINGLE("Charge Pump Envelope Detection Switch",
+		AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETCPEN,
+		1, 0),
+	SOC_ENUM("Charge Pump Envelope Detection Decay Time",
+		soc_enum_envdettime),
+
+	/* Headset */
+	SOC_ENUM("Headset Mode", soc_enum_da12voice),
+	SOC_SINGLE("Headset High Pass Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_HSHPEN,
+		1, 0),
+	SOC_SINGLE("Headset Low Power Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_HSLOWPOW,
+		1, 0),
+	SOC_SINGLE("Headset DAC Low Power Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW1,
+		1, 0),
+	SOC_SINGLE("Headset DAC Drv Low Power Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW0,
+		1, 0),
+	SOC_ENUM("Headset Fade Speed", soc_enum_hsfadspeed),
+	SOC_ENUM("Headset Source", soc_enum_da2hslr),
+	SOC_ENUM("Headset Filter", soc_enum_hsesinc),
+	SOC_DOUBLE_R_TLV("Headset Master Volume",
+		AB8500_DADIGGAIN1, AB8500_DADIGGAIN2,
+		0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+	SOC_DOUBLE_R_TLV("Headset Digital Volume",
+		AB8500_HSLEARDIGGAIN, AB8500_HSRDIGGAIN,
+		0, AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX, 1, hs_ear_dig_gain_tlv),
+	SOC_DOUBLE_TLV("Headset Volume",
+		AB8500_ANAGAIN3,
+		AB8500_ANAGAIN3_HSLGAIN, AB8500_ANAGAIN3_HSRGAIN,
+		AB8500_ANAGAIN3_HSXGAIN_MAX, 1, hs_gain_tlv),
+
+	/* Earpiece */
+	SOC_ENUM("Earpiece DAC Mode",
+		soc_enum_eardaclowpow),
+	SOC_ENUM("Earpiece DAC Drv Mode",
+		soc_enum_eardrvlowpow),
+
+	/* HandsFree */
+	SOC_ENUM("HF Mode", soc_enum_da34voice),
+	SOC_SINGLE("HF and Headset Swap Switch",
+		AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_SWAPDA12_34,
+		1, 0),
+	SOC_DOUBLE("HF Low EMI Mode Switch",
+		AB8500_CLASSDCONF1,
+		AB8500_CLASSDCONF1_HFLSWAPEN, AB8500_CLASSDCONF1_HFRSWAPEN,
+		1, 0),
+	SOC_DOUBLE("HF FIR Bypass Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_FIRBYP0, AB8500_CLASSDCONF2_FIRBYP1,
+		1, 0),
+	SOC_DOUBLE("HF High Volume Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_HIGHVOLEN0, AB8500_CLASSDCONF2_HIGHVOLEN1,
+		1, 0),
+	SOC_SINGLE("HF L and R Bridge Switch",
+		AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLHF,
+		1, 0),
+	SOC_DOUBLE_R_TLV("HF Master Volume",
+		AB8500_DADIGGAIN3, AB8500_DADIGGAIN4,
+		0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+
+	/* Vibra */
+	SOC_DOUBLE("Vibra High Volume Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_HIGHVOLEN2, AB8500_CLASSDCONF2_HIGHVOLEN3,
+		1, 0),
+	SOC_DOUBLE("Vibra Low EMI Mode Switch",
+		AB8500_CLASSDCONF1,
+		AB8500_CLASSDCONF1_VIB1SWAPEN, AB8500_CLASSDCONF1_VIB2SWAPEN,
+		1, 0),
+	SOC_DOUBLE("Vibra FIR Bypass Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_FIRBYP2, AB8500_CLASSDCONF2_FIRBYP3,
+		1, 0),
+	SOC_ENUM("Vibra Mode", soc_enum_da56voice),
+	SOC_DOUBLE_R("Vibra PWM Duty Cycle N",
+		AB8500_PWMGENCONF3, AB8500_PWMGENCONF5,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0),
+	SOC_DOUBLE_R("Vibra PWM Duty Cycle P",
+		AB8500_PWMGENCONF2, AB8500_PWMGENCONF4,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0),
+	SOC_SINGLE("Vibra 1 and 2 Bridge Switch",
+		AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLVIB,
+		1, 0),
+	SOC_DOUBLE_R_TLV("Vibra Master Volume",
+		AB8500_DADIGGAIN5, AB8500_DADIGGAIN6,
+		0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+
+	/* HandsFree, Vibra */
+	SOC_SINGLE("ClassD High Pass Volume",
+		AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHHPGAIN,
+		AB8500_CLASSDCONF3_DITHHPGAIN_MAX, 0),
+	SOC_SINGLE("ClassD White Volume",
+		AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHWGAIN,
+		AB8500_CLASSDCONF3_DITHWGAIN_MAX, 0),
+
+	/* Mic 1, Mic 2, LineIn */
+	SOC_DOUBLE_R_TLV("Mic Master Volume",
+		AB8500_ADDIGGAIN3, AB8500_ADDIGGAIN4,
+		0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+
+	/* Mic 1 */
+	SOC_SINGLE_TLV("Mic 1",
+		AB8500_ANAGAIN1,
+		AB8500_ANAGAINX_MICXGAIN,
+		AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv),
+	SOC_SINGLE("Mic 1 Low Power Switch",
+		AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX,
+		1, 0),
+
+	/* Mic 2 */
+	SOC_DOUBLE("Mic High Pass Switch",
+		AB8500_ADFILTCONF,
+		AB8500_ADFILTCONF_AD3NH, AB8500_ADFILTCONF_AD4NH,
+		1, 1),
+	SOC_ENUM("Mic Mode", soc_enum_ad34voice),
+	SOC_ENUM("Mic Filter", soc_enum_dmic34sinc),
+	SOC_SINGLE_TLV("Mic 2",
+		AB8500_ANAGAIN2,
+		AB8500_ANAGAINX_MICXGAIN,
+		AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv),
+	SOC_SINGLE("Mic 2 Low Power Switch",
+		AB8500_ANAGAIN2, AB8500_ANAGAINX_LOWPOWMICX,
+		1, 0),
+
+	/* LineIn */
+	SOC_DOUBLE("LineIn High Pass Switch",
+		AB8500_ADFILTCONF,
+		AB8500_ADFILTCONF_AD1NH, AB8500_ADFILTCONF_AD2NH,
+		1, 1),
+	SOC_ENUM("LineIn Filter", soc_enum_dmic12sinc),
+	SOC_ENUM("LineIn Mode", soc_enum_ad12voice),
+	SOC_DOUBLE_R_TLV("LineIn Master Volume",
+		AB8500_ADDIGGAIN1, AB8500_ADDIGGAIN2,
+		0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+	SOC_DOUBLE_TLV("LineIn",
+		AB8500_ANAGAIN4,
+		AB8500_ANAGAIN4_LINLGAIN, AB8500_ANAGAIN4_LINRGAIN,
+		AB8500_ANAGAIN4_LINXGAIN_MAX, 0, lin_gain_tlv),
+	SOC_DOUBLE_R_TLV("LineIn to Headset Volume",
+		AB8500_DIGLINHSLGAIN, AB8500_DIGLINHSRGAIN,
+		AB8500_DIGLINHSXGAIN_LINTOHSXGAIN,
+		AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX,
+		1, lin2hs_gain_tlv),
+
+	/* DMic */
+	SOC_ENUM("DMic Filter", soc_enum_dmic56sinc),
+	SOC_DOUBLE_R_TLV("DMic Master Volume",
+		AB8500_ADDIGGAIN5, AB8500_ADDIGGAIN6,
+		0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+
+	/* Digital gains */
+	SOC_ENUM("Digital Gain Fade Speed", soc_enum_fadespeed),
+
+	/* Analog loopback */
+	SOC_DOUBLE_R_TLV("Analog Loopback Volume",
+		AB8500_ADDIGLOOPGAIN1, AB8500_ADDIGLOOPGAIN2,
+		0, AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX, 1, dax_dig_gain_tlv),
+
+	/* Digital interface - DA from slot mapping */
+	SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap),
+	SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap),
+	SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap),
+	SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap),
+	SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap),
+	SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap),
+	SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap),
+	SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap),
+
+	/* Digital interface - AD to slot mapping */
+	SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map),
+	SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map),
+	SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map),
+	SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map),
+	SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map),
+	SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map),
+	SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map),
+	SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map),
+	SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map),
+	SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map),
+	SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map),
+	SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map),
+	SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map),
+	SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map),
+	SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map),
+	SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map),
+	SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map),
+	SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map),
+	SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map),
+	SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map),
+	SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map),
+	SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map),
+	SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map),
+	SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map),
+	SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map),
+	SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map),
+	SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map),
+	SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map),
+	SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map),
+	SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map),
+	SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map),
+	SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map),
+
+	/* Digital interface - Loopback */
+	SOC_SINGLE("Digital Interface AD 1 Loopback Switch",
+		AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DAI7TOADO1,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 2 Loopback Switch",
+		AB8500_DASLOTCONF2, AB8500_DASLOTCONF2_DAI8TOADO2,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 3 Loopback Switch",
+		AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DAI7TOADO3,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 4 Loopback Switch",
+		AB8500_DASLOTCONF4, AB8500_DASLOTCONF4_DAI8TOADO4,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 5 Loopback Switch",
+		AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DAI7TOADO5,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 6 Loopback Switch",
+		AB8500_DASLOTCONF6, AB8500_DASLOTCONF6_DAI8TOADO6,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 7 Loopback Switch",
+		AB8500_DASLOTCONF7, AB8500_DASLOTCONF7_DAI8TOADO7,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 8 Loopback Switch",
+		AB8500_DASLOTCONF8, AB8500_DASLOTCONF8_DAI7TOADO8,
+		1, 0),
+
+	/* Digital interface - Burst FIFO */
+	SOC_SINGLE("Digital Interface 0 FIFO Enable Switch",
+		AB8500_DIGIFCONF3, AB8500_DIGIFCONF3_IF0BFIFOEN,
+		1, 0),
+	SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask),
+	SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2),
+	SOC_SINGLE("Burst FIFO Threshold",
+		AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOINT_SHIFT,
+		AB8500_FIFOCONF1_BFIFOINT_MAX, 0),
+	SOC_SINGLE("Burst FIFO Length",
+		AB8500_FIFOCONF2, AB8500_FIFOCONF2_BFIFOTX_SHIFT,
+		AB8500_FIFOCONF2_BFIFOTX_MAX, 0),
+	SOC_SINGLE("Burst FIFO EOS Extra Slots",
+		AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOEXSL_SHIFT,
+		AB8500_FIFOCONF3_BFIFOEXSL_MAX, 0),
+	SOC_SINGLE("Burst FIFO FS Extra Bit-clocks",
+		AB8500_FIFOCONF3, AB8500_FIFOCONF3_PREBITCLK0_SHIFT,
+		AB8500_FIFOCONF3_PREBITCLK0_MAX, 0),
+	SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast),
+
+	SOC_SINGLE("Burst FIFO Interface Switch",
+		AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFORUN_SHIFT,
+		1, 0),
+	SOC_SINGLE("Burst FIFO Switch Frame Number",
+		AB8500_FIFOCONF4, AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT,
+		AB8500_FIFOCONF4_BFIFOFRAMSW_MAX, 0),
+	SOC_SINGLE("Burst FIFO Wake Up Delay",
+		AB8500_FIFOCONF5, AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT,
+		AB8500_FIFOCONF5_BFIFOWAKEUP_MAX, 0),
+	SOC_SINGLE("Burst FIFO Samples In FIFO",
+		AB8500_FIFOCONF6, AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT,
+		AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0),
+
+	/* ANC */
+	SOC_ENUM_EXT("ANC Status", soc_enum_ancstate,
+		anc_status_control_get, anc_status_control_put),
+	SOC_SINGLE_XR_SX("ANC Warp Delay Shift",
+		AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT,
+		AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0),
+	SOC_SINGLE_XR_SX("ANC FIR Output Shift",
+		AB8500_ANCCONF3, 1, AB8500_ANCCONF3_SHIFT,
+		AB8500_ANCCONF3_MIN, AB8500_ANCCONF3_MAX, 0),
+	SOC_SINGLE_XR_SX("ANC IIR Output Shift",
+		AB8500_ANCCONF4, 1, AB8500_ANCCONF4_SHIFT,
+		AB8500_ANCCONF4_MIN, AB8500_ANCCONF4_MAX, 0),
+	SOC_SINGLE_XR_SX("ANC Warp Delay",
+		AB8500_ANCCONF9, 2, AB8500_ANC_WARP_DELAY_SHIFT,
+		AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0),
+
+	/* Sidetone */
+	SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate,
+		sid_status_control_get, sid_status_control_put),
+	SOC_SINGLE_STROBE("Sidetone Reset",
+		AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0),
+};
+
+static struct snd_kcontrol_new ab8500_filter_controls[] = {
+	AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS,
+		AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX),
+	AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS,
+		AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX),
+	AB8500_FILTER_CONTROL("Sidetone FIR Coefficients",
+			AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN,
+			AB8500_SID_FIR_COEFF_MAX)
+};
+enum ab8500_filter {
+	AB8500_FILTER_ANC_FIR = 0,
+	AB8500_FILTER_ANC_IIR = 1,
+	AB8500_FILTER_SID_FIR = 2,
+};
+
+/*
+ * Extended interface for codec-driver
+ */
+
+static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec)
+{
+	int status;
+
+	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+	/* Reset audio-registers and disable 32kHz-clock output 2 */
+	status = ab8500_sysctrl_write(AB8500_STW4500CTRL3,
+				AB8500_STW4500CTRL3_CLK32KOUT2DIS |
+					AB8500_STW4500CTRL3_RESETAUDN,
+				AB8500_STW4500CTRL3_RESETAUDN);
+	if (status < 0)
+		return status;
+
+	return 0;
+}
+
+static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
+			struct amic_settings *amics)
+{
+	u8 value8;
+	unsigned int value;
+	int status;
+	const struct snd_soc_dapm_route *route;
+
+	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+	/* Set DMic-clocks to outputs */
+	status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC,
+						(u8)AB8500_GPIO_DIR4_REG,
+						&value8);
+	if (status < 0)
+		return status;
+	value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT |
+		GPIO31_DIR_OUTPUT;
+	status = abx500_set_register_interruptible(codec->dev,
+						(u8)AB8500_MISC,
+						(u8)AB8500_GPIO_DIR4_REG,
+						value);
+	if (status < 0)
+		return status;
+
+	/* Attach regulators to AMic DAPM-paths */
+	dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__,
+		amic_micbias_str(amics->mic1a_micbias));
+	route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias];
+	status = snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__,
+		amic_micbias_str(amics->mic1b_micbias));
+	route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias];
+	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__,
+		amic_micbias_str(amics->mic2_micbias));
+	route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias];
+	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	if (status < 0) {
+		dev_err(codec->dev,
+			"%s: Failed to add AMic-regulator DAPM-routes (%d).\n",
+			__func__, status);
+		return status;
+	}
+
+	/* Set AMic-configuration */
+	dev_dbg(codec->dev, "%s: Mic 1 mic-type: %s\n", __func__,
+		amic_type_str(amics->mic1_type));
+	snd_soc_update_bits(codec, AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX,
+			amics->mic1_type == AMIC_TYPE_DIFFERENTIAL ?
+				0 : AB8500_ANAGAINX_ENSEMICX);
+	dev_dbg(codec->dev, "%s: Mic 2 mic-type: %s\n", __func__,
+		amic_type_str(amics->mic2_type));
+	snd_soc_update_bits(codec, AB8500_ANAGAIN2, AB8500_ANAGAINX_ENSEMICX,
+			amics->mic2_type == AMIC_TYPE_DIFFERENTIAL ?
+				0 : AB8500_ANAGAINX_ENSEMICX);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics);
+
+static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec,
+				enum ear_cm_voltage ear_cmv)
+{
+	char *cmv_str;
+
+	switch (ear_cmv) {
+	case EAR_CMV_0_95V:
+		cmv_str = "0.95V";
+		break;
+	case EAR_CMV_1_10V:
+		cmv_str = "1.10V";
+		break;
+	case EAR_CMV_1_27V:
+		cmv_str = "1.27V";
+		break;
+	case EAR_CMV_1_58V:
+		cmv_str = "1.58V";
+		break;
+	default:
+		dev_err(codec->dev,
+			"%s: Unknown earpiece CM-voltage (%d)!\n",
+			__func__, (int)ear_cmv);
+		return -EINVAL;
+	}
+	dev_dbg(codec->dev, "%s: Earpiece CM-voltage: %s\n", __func__,
+		cmv_str);
+	snd_soc_update_bits(codec, AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM,
+			ear_cmv);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv);
+
+static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai,
+				unsigned int delay)
+{
+	unsigned int mask, val;
+	struct snd_soc_codec *codec = dai->codec;
+
+	mask = BIT(AB8500_DIGIFCONF2_IF0DEL);
+	val = 0;
+
+	switch (delay) {
+	case 0:
+		break;
+	case 1:
+		val |= BIT(AB8500_DIGIFCONF2_IF0DEL);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported bit-delay (0x%x)!\n",
+			__func__, delay);
+		return -EINVAL;
+	}
+
+	dev_dbg(dai->codec->dev, "%s: IF0 Bit-delay: %d bits.\n",
+		__func__, delay);
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+	return 0;
+}
+
+/* Gates clocking according format mask */
+static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec,
+					unsigned int fmt)
+{
+	unsigned int mask;
+	unsigned int val;
+
+	mask = BIT(AB8500_DIGIFCONF1_ENMASTGEN) |
+			BIT(AB8500_DIGIFCONF1_ENFSBITCLK0);
+
+	val = BIT(AB8500_DIGIFCONF1_ENMASTGEN);
+
+	switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+	case SND_SOC_DAIFMT_CONT: /* continuous clock */
+		dev_dbg(codec->dev, "%s: IF0 Clock is continuous.\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF1_ENFSBITCLK0);
+		break;
+	case SND_SOC_DAIFMT_GATED: /* clock is gated */
+		dev_dbg(codec->dev, "%s: IF0 Clock is gated.\n",
+			__func__);
+		break;
+	default:
+		dev_err(codec->dev,
+			"%s: ERROR: Unsupported clock mask (0x%x)!\n",
+			__func__, fmt & SND_SOC_DAIFMT_CLOCK_MASK);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val);
+
+	return 0;
+}
+
+static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	unsigned int mask;
+	unsigned int val;
+	struct snd_soc_codec *codec = dai->codec;
+	int status;
+
+	dev_dbg(codec->dev, "%s: Enter (fmt = 0x%x)\n", __func__, fmt);
+
+	mask = BIT(AB8500_DIGIFCONF3_IF1DATOIF0AD) |
+			BIT(AB8500_DIGIFCONF3_IF1CLKTOIF0CLK) |
+			BIT(AB8500_DIGIFCONF3_IF0BFIFOEN) |
+			BIT(AB8500_DIGIFCONF3_IF0MASTER);
+	val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Master-mode: AB8500 master.\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF3_IF0MASTER);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Master-mode: AB8500 slave.\n", __func__);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */
+	case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
+		dev_err(dai->codec->dev,
+			"%s: ERROR: The device is either a master or a slave.\n",
+			__func__);
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupporter master mask 0x%x\n",
+			__func__, fmt & SND_SOC_DAIFMT_MASTER_MASK);
+		return -EINVAL;
+		break;
+	}
+
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val);
+
+	/* Set clock gating */
+	status = ab8500_codec_set_dai_clock_gate(codec, fmt);
+	if (status) {
+		dev_err(dai->codec->dev,
+			"%s: ERRROR: Failed to set clock gate (%d).\n",
+			__func__, status);
+		return status;
+	}
+
+	/* Setting data transfer format */
+
+	mask = BIT(AB8500_DIGIFCONF2_IF0FORMAT0) |
+		BIT(AB8500_DIGIFCONF2_IF0FORMAT1) |
+		BIT(AB8500_DIGIFCONF2_FSYNC0P) |
+		BIT(AB8500_DIGIFCONF2_BITCLK0P);
+	val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S: /* I2S mode */
+		dev_dbg(dai->codec->dev, "%s: IF0 Protocol: I2S\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT1);
+		ab8500_audio_set_bit_delay(dai, 0);
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Protocol: DSP A (TDM)\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0);
+		ab8500_audio_set_bit_delay(dai, 1);
+		break;
+
+	case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Protocol: DSP B (TDM)\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0);
+		ab8500_audio_set_bit_delay(dai, 0);
+		break;
+
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported format (0x%x)!\n",
+			__func__, fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Normal bit clock, normal frame\n",
+			__func__);
+		break;
+	case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Normal bit clock, inverted frame\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF2_FSYNC0P);
+		break;
+	case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Inverted bit clock, normal frame\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF2_BITCLK0P);
+		break;
+	case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Inverted bit clock, inverted frame\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF2_FSYNC0P);
+		val |= BIT(AB8500_DIGIFCONF2_BITCLK0P);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported INV mask 0x%x\n",
+			__func__, fmt & SND_SOC_DAIFMT_INV_MASK);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+	return 0;
+}
+
+static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai,
+		unsigned int tx_mask, unsigned int rx_mask,
+		int slots, int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val, mask, slots_active;
+
+	mask = BIT(AB8500_DIGIFCONF2_IF0WL0) |
+		BIT(AB8500_DIGIFCONF2_IF0WL1);
+	val = 0;
+
+	switch (slot_width) {
+	case 16:
+		break;
+	case 20:
+		val |= BIT(AB8500_DIGIFCONF2_IF0WL0);
+		break;
+	case 24:
+		val |= BIT(AB8500_DIGIFCONF2_IF0WL1);
+		break;
+	case 32:
+		val |= BIT(AB8500_DIGIFCONF2_IF0WL1) |
+			BIT(AB8500_DIGIFCONF2_IF0WL0);
+		break;
+	default:
+		dev_err(dai->codec->dev, "%s: Unsupported slot-width 0x%x\n",
+			__func__, slot_width);
+		return -EINVAL;
+	}
+
+	dev_dbg(dai->codec->dev, "%s: IF0 slot-width: %d bits.\n",
+		__func__, slot_width);
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+	/* Setup TDM clocking according to slot count */
+	dev_dbg(dai->codec->dev, "%s: Slots, total: %d\n", __func__, slots);
+	mask = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) |
+			BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+	switch (slots) {
+	case 2:
+		val = AB8500_MASK_NONE;
+		break;
+	case 4:
+		val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0);
+		break;
+	case 8:
+		val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+		break;
+	case 16:
+		val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) |
+			BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported number of slots (%d)!\n",
+			__func__, slots);
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val);
+
+	/* Setup TDM DA according to active tx slots */
+	mask = AB8500_DASLOTCONFX_SLTODAX_MASK;
+	slots_active = hweight32(tx_mask);
+	dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__,
+		slots_active);
+	switch (slots_active) {
+	case 0:
+		break;
+	case 1:
+		/* Slot 9 -> DA_IN1 & DA_IN3 */
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11);
+		break;
+	case 2:
+		/* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11);
+
+		break;
+	case 8:
+		dev_dbg(dai->codec->dev,
+			"%s: In 8-channel mode DA-from-slot mapping is set manually.",
+			__func__);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: Unsupported number of active TX-slots (%d)!\n",
+			__func__, slots_active);
+		return -EINVAL;
+	}
+
+	/* Setup TDM AD according to active RX-slots */
+	slots_active = hweight32(rx_mask);
+	dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__,
+		slots_active);
+	switch (slots_active) {
+	case 0:
+		break;
+	case 1:
+		/* AD_OUT3 -> slot 0 & 1 */
+		snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL,
+				AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN |
+				AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD);
+		break;
+	case 2:
+		/* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */
+		snd_soc_update_bits(codec,
+				AB8500_ADSLOTSEL1,
+				AB8500_MASK_ALL,
+				AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN |
+				AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD);
+		break;
+	case 8:
+		dev_dbg(dai->codec->dev,
+			"%s: In 8-channel mode AD-to-slot mapping is set manually.",
+			__func__);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: Unsupported number of active RX-slots (%d)!\n",
+			__func__, slots_active);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+struct snd_soc_dai_driver ab8500_codec_dai[] = {
+	{
+		.name = "ab8500-codec-dai.0",
+		.id = 0,
+		.playback = {
+			.stream_name = "ab8500_0p",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = AB8500_SUPPORTED_RATE,
+			.formats = AB8500_SUPPORTED_FMT,
+		},
+		.ops = (struct snd_soc_dai_ops[]) {
+			{
+				.set_tdm_slot = ab8500_codec_set_dai_tdm_slot,
+				.set_fmt = ab8500_codec_set_dai_fmt,
+			}
+		},
+		.symmetric_rates = 1
+	},
+	{
+		.name = "ab8500-codec-dai.1",
+		.id = 1,
+		.capture = {
+			.stream_name = "ab8500_0c",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = AB8500_SUPPORTED_RATE,
+			.formats = AB8500_SUPPORTED_FMT,
+		},
+		.ops = (struct snd_soc_dai_ops[]) {
+			{
+				.set_tdm_slot = ab8500_codec_set_dai_tdm_slot,
+				.set_fmt = ab8500_codec_set_dai_fmt,
+			}
+		},
+		.symmetric_rates = 1
+	}
+};
+
+static int ab8500_codec_probe(struct snd_soc_codec *codec)
+{
+	struct device *dev = codec->dev;
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
+	struct ab8500_platform_data *pdata;
+	struct filter_control *fc;
+	int status;
+
+	dev_dbg(dev, "%s: Enter.\n", __func__);
+
+	/* Setup AB8500 according to board-settings */
+	pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
+	status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
+	if (status < 0) {
+		pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
+		return status;
+	}
+	status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv);
+	if (status < 0) {
+		pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n",
+			__func__, status);
+		return status;
+	}
+
+	status = ab8500_audio_init_audioblock(codec);
+	if (status < 0) {
+		dev_err(dev, "%s: failed to init audio-block (%d)!\n",
+			__func__, status);
+		return status;
+	}
+
+	/* Override HW-defaults */
+	ab8500_codec_write_reg(codec,
+				AB8500_ANACONF5,
+				BIT(AB8500_ANACONF5_HSAUTOEN));
+	ab8500_codec_write_reg(codec,
+				AB8500_SHORTCIRCONF,
+				BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
+
+	/* Add filter controls */
+	status = snd_soc_add_codec_controls(codec, ab8500_filter_controls,
+				ARRAY_SIZE(ab8500_filter_controls));
+	if (status < 0) {
+		dev_err(dev,
+			"%s: failed to add ab8500 filter controls (%d).\n",
+			__func__, status);
+		return status;
+	}
+	fc = (struct filter_control *)
+		&ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value;
+	drvdata->anc_fir_values = (long *)fc->value;
+	fc = (struct filter_control *)
+		&ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value;
+	drvdata->anc_iir_values = (long *)fc->value;
+	fc = (struct filter_control *)
+		&ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value;
+	drvdata->sid_fir_values = (long *)fc->value;
+
+	(void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+
+	mutex_init(&drvdata->anc_lock);
+
+	return status;
+}
+
+static struct snd_soc_codec_driver ab8500_codec_driver = {
+	.probe =		ab8500_codec_probe,
+	.read =			ab8500_codec_read_reg,
+	.write =		ab8500_codec_write_reg,
+	.reg_word_size =	sizeof(u8),
+	.controls =		ab8500_ctrls,
+	.num_controls =		ARRAY_SIZE(ab8500_ctrls),
+	.dapm_widgets =		ab8500_dapm_widgets,
+	.num_dapm_widgets =	ARRAY_SIZE(ab8500_dapm_widgets),
+	.dapm_routes =		ab8500_dapm_routes,
+	.num_dapm_routes =	ARRAY_SIZE(ab8500_dapm_routes),
+};
+
+static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev)
+{
+	int status;
+	struct ab8500_codec_drvdata *drvdata;
+
+	dev_dbg(&pdev->dev, "%s: Enter.\n", __func__);
+
+	/* Create driver private-data struct */
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata),
+			GFP_KERNEL);
+	drvdata->sid_status = SID_UNCONFIGURED;
+	drvdata->anc_status = ANC_UNCONFIGURED;
+	dev_set_drvdata(&pdev->dev, drvdata);
+
+	dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__);
+	status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver,
+				ab8500_codec_dai,
+				ARRAY_SIZE(ab8500_codec_dai));
+	if (status < 0)
+		dev_err(&pdev->dev,
+			"%s: Error: Failed to register codec (%d).\n",
+			__func__, status);
+
+	return status;
+}
+
+static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev)
+{
+	dev_info(&pdev->dev, "%s Enter.\n", __func__);
+
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver ab8500_codec_platform_driver = {
+	.driver	= {
+		.name	= "ab8500-codec",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= ab8500_codec_driver_probe,
+	.remove		= __devexit_p(ab8500_codec_driver_remove),
+	.suspend	= NULL,
+	.resume		= NULL,
+};
+module_platform_driver(ab8500_codec_platform_driver);
+
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h
new file mode 100644
index 000000000000..114f69a0c629
--- /dev/null
+++ b/sound/soc/codecs/ab8500-codec.h
@@ -0,0 +1,590 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ *         for ST-Ericsson.
+ *
+ *         Based on the early work done by:
+ *         Mikko J. Lehto <mikko.lehto@symbio.com>,
+ *         Mikko Sarmanne <mikko.sarmanne@symbio.com>,
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef AB8500_CODEC_REGISTERS_H
+#define AB8500_CODEC_REGISTERS_H
+
+#define AB8500_SUPPORTED_RATE			(SNDRV_PCM_RATE_48000)
+#define AB8500_SUPPORTED_FMT			(SNDRV_PCM_FMTBIT_S16_LE)
+
+/* AB8500 audio bank (0x0d) register definitions */
+
+#define AB8500_POWERUP				0x00
+#define AB8500_AUDSWRESET			0x01
+#define AB8500_ADPATHENA			0x02
+#define AB8500_DAPATHENA			0x03
+#define AB8500_ANACONF1				0x04
+#define AB8500_ANACONF2				0x05
+#define AB8500_DIGMICCONF			0x06
+#define AB8500_ANACONF3				0x07
+#define AB8500_ANACONF4				0x08
+#define AB8500_DAPATHCONF			0x09
+#define AB8500_MUTECONF				0x0A
+#define AB8500_SHORTCIRCONF			0x0B
+#define AB8500_ANACONF5				0x0C
+#define AB8500_ENVCPCONF			0x0D
+#define AB8500_SIGENVCONF			0x0E
+#define AB8500_PWMGENCONF1			0x0F
+#define AB8500_PWMGENCONF2			0x10
+#define AB8500_PWMGENCONF3			0x11
+#define AB8500_PWMGENCONF4			0x12
+#define AB8500_PWMGENCONF5			0x13
+#define AB8500_ANAGAIN1				0x14
+#define AB8500_ANAGAIN2				0x15
+#define AB8500_ANAGAIN3				0x16
+#define AB8500_ANAGAIN4				0x17
+#define AB8500_DIGLINHSLGAIN			0x18
+#define AB8500_DIGLINHSRGAIN			0x19
+#define AB8500_ADFILTCONF			0x1A
+#define AB8500_DIGIFCONF1			0x1B
+#define AB8500_DIGIFCONF2			0x1C
+#define AB8500_DIGIFCONF3			0x1D
+#define AB8500_DIGIFCONF4			0x1E
+#define AB8500_ADSLOTSEL1			0x1F
+#define AB8500_ADSLOTSEL2			0x20
+#define AB8500_ADSLOTSEL3			0x21
+#define AB8500_ADSLOTSEL4			0x22
+#define AB8500_ADSLOTSEL5			0x23
+#define AB8500_ADSLOTSEL6			0x24
+#define AB8500_ADSLOTSEL7			0x25
+#define AB8500_ADSLOTSEL8			0x26
+#define AB8500_ADSLOTSEL9			0x27
+#define AB8500_ADSLOTSEL10			0x28
+#define AB8500_ADSLOTSEL11			0x29
+#define AB8500_ADSLOTSEL12			0x2A
+#define AB8500_ADSLOTSEL13			0x2B
+#define AB8500_ADSLOTSEL14			0x2C
+#define AB8500_ADSLOTSEL15			0x2D
+#define AB8500_ADSLOTSEL16			0x2E
+#define AB8500_ADSLOTHIZCTRL1			0x2F
+#define AB8500_ADSLOTHIZCTRL2			0x30
+#define AB8500_ADSLOTHIZCTRL3			0x31
+#define AB8500_ADSLOTHIZCTRL4			0x32
+#define AB8500_DASLOTCONF1			0x33
+#define AB8500_DASLOTCONF2			0x34
+#define AB8500_DASLOTCONF3			0x35
+#define AB8500_DASLOTCONF4			0x36
+#define AB8500_DASLOTCONF5			0x37
+#define AB8500_DASLOTCONF6			0x38
+#define AB8500_DASLOTCONF7			0x39
+#define AB8500_DASLOTCONF8			0x3A
+#define AB8500_CLASSDCONF1			0x3B
+#define AB8500_CLASSDCONF2			0x3C
+#define AB8500_CLASSDCONF3			0x3D
+#define AB8500_DMICFILTCONF			0x3E
+#define AB8500_DIGMULTCONF1			0x3F
+#define AB8500_DIGMULTCONF2			0x40
+#define AB8500_ADDIGGAIN1			0x41
+#define AB8500_ADDIGGAIN2			0x42
+#define AB8500_ADDIGGAIN3			0x43
+#define AB8500_ADDIGGAIN4			0x44
+#define AB8500_ADDIGGAIN5			0x45
+#define AB8500_ADDIGGAIN6			0x46
+#define AB8500_DADIGGAIN1			0x47
+#define AB8500_DADIGGAIN2			0x48
+#define AB8500_DADIGGAIN3			0x49
+#define AB8500_DADIGGAIN4			0x4A
+#define AB8500_DADIGGAIN5			0x4B
+#define AB8500_DADIGGAIN6			0x4C
+#define AB8500_ADDIGLOOPGAIN1			0x4D
+#define AB8500_ADDIGLOOPGAIN2			0x4E
+#define AB8500_HSLEARDIGGAIN			0x4F
+#define AB8500_HSRDIGGAIN			0x50
+#define AB8500_SIDFIRGAIN1			0x51
+#define AB8500_SIDFIRGAIN2			0x52
+#define AB8500_ANCCONF1				0x53
+#define AB8500_ANCCONF2				0x54
+#define AB8500_ANCCONF3				0x55
+#define AB8500_ANCCONF4				0x56
+#define AB8500_ANCCONF5				0x57
+#define AB8500_ANCCONF6				0x58
+#define AB8500_ANCCONF7				0x59
+#define AB8500_ANCCONF8				0x5A
+#define AB8500_ANCCONF9				0x5B
+#define AB8500_ANCCONF10			0x5C
+#define AB8500_ANCCONF11			0x5D
+#define AB8500_ANCCONF12			0x5E
+#define AB8500_ANCCONF13			0x5F
+#define AB8500_ANCCONF14			0x60
+#define AB8500_SIDFIRADR			0x61
+#define AB8500_SIDFIRCOEF1			0x62
+#define AB8500_SIDFIRCOEF2			0x63
+#define AB8500_SIDFIRCONF			0x64
+#define AB8500_AUDINTMASK1			0x65
+#define AB8500_AUDINTSOURCE1			0x66
+#define AB8500_AUDINTMASK2			0x67
+#define AB8500_AUDINTSOURCE2			0x68
+#define AB8500_FIFOCONF1			0x69
+#define AB8500_FIFOCONF2			0x6A
+#define AB8500_FIFOCONF3			0x6B
+#define AB8500_FIFOCONF4			0x6C
+#define AB8500_FIFOCONF5			0x6D
+#define AB8500_FIFOCONF6			0x6E
+#define AB8500_AUDREV				0x6F
+
+#define AB8500_FIRST_REG			AB8500_POWERUP
+#define AB8500_LAST_REG				AB8500_AUDREV
+#define AB8500_CACHEREGNUM			(AB8500_LAST_REG + 1)
+
+#define AB8500_MASK_ALL				0xFF
+#define AB8500_MASK_NONE			0x00
+
+/* AB8500_POWERUP */
+#define AB8500_POWERUP_POWERUP			7
+#define AB8500_POWERUP_ENANA			3
+
+/* AB8500_AUDSWRESET */
+#define AB8500_AUDSWRESET_SWRESET		7
+
+/* AB8500_ADPATHENA */
+#define AB8500_ADPATHENA_ENAD12			7
+#define AB8500_ADPATHENA_ENAD34			5
+#define AB8500_ADPATHENA_ENAD5768		3
+
+/* AB8500_DAPATHENA */
+#define AB8500_DAPATHENA_ENDA1			7
+#define AB8500_DAPATHENA_ENDA2			6
+#define AB8500_DAPATHENA_ENDA3			5
+#define AB8500_DAPATHENA_ENDA4			4
+#define AB8500_DAPATHENA_ENDA5			3
+#define AB8500_DAPATHENA_ENDA6			2
+
+/* AB8500_ANACONF1 */
+#define AB8500_ANACONF1_HSLOWPOW		7
+#define AB8500_ANACONF1_DACLOWPOW1		6
+#define AB8500_ANACONF1_DACLOWPOW0		5
+#define AB8500_ANACONF1_EARDACLOWPOW		4
+#define AB8500_ANACONF1_EARSELCM		2
+#define AB8500_ANACONF1_HSHPEN			1
+#define AB8500_ANACONF1_EARDRVLOWPOW		0
+
+/* AB8500_ANACONF2 */
+#define AB8500_ANACONF2_ENMIC1			7
+#define AB8500_ANACONF2_ENMIC2			6
+#define AB8500_ANACONF2_ENLINL			5
+#define AB8500_ANACONF2_ENLINR			4
+#define AB8500_ANACONF2_MUTMIC1			3
+#define AB8500_ANACONF2_MUTMIC2			2
+#define AB8500_ANACONF2_MUTLINL			1
+#define AB8500_ANACONF2_MUTLINR			0
+
+/* AB8500_DIGMICCONF */
+#define AB8500_DIGMICCONF_ENDMIC1		7
+#define AB8500_DIGMICCONF_ENDMIC2		6
+#define AB8500_DIGMICCONF_ENDMIC3		5
+#define AB8500_DIGMICCONF_ENDMIC4		4
+#define AB8500_DIGMICCONF_ENDMIC5		3
+#define AB8500_DIGMICCONF_ENDMIC6		2
+#define AB8500_DIGMICCONF_HSFADSPEED		0
+
+/* AB8500_ANACONF3 */
+#define AB8500_ANACONF3_MIC1SEL			7
+#define AB8500_ANACONF3_LINRSEL			6
+#define AB8500_ANACONF3_ENDRVHSL		5
+#define AB8500_ANACONF3_ENDRVHSR		4
+#define AB8500_ANACONF3_ENADCMIC		2
+#define AB8500_ANACONF3_ENADCLINL		1
+#define AB8500_ANACONF3_ENADCLINR		0
+
+/* AB8500_ANACONF4 */
+#define AB8500_ANACONF4_DISPDVSS		7
+#define AB8500_ANACONF4_ENEAR			6
+#define AB8500_ANACONF4_ENHSL			5
+#define AB8500_ANACONF4_ENHSR			4
+#define AB8500_ANACONF4_ENHFL			3
+#define AB8500_ANACONF4_ENHFR			2
+#define AB8500_ANACONF4_ENVIB1			1
+#define AB8500_ANACONF4_ENVIB2			0
+
+/* AB8500_DAPATHCONF */
+#define AB8500_DAPATHCONF_ENDACEAR		6
+#define AB8500_DAPATHCONF_ENDACHSL		5
+#define AB8500_DAPATHCONF_ENDACHSR		4
+#define AB8500_DAPATHCONF_ENDACHFL		3
+#define AB8500_DAPATHCONF_ENDACHFR		2
+#define AB8500_DAPATHCONF_ENDACVIB1		1
+#define AB8500_DAPATHCONF_ENDACVIB2		0
+
+/* AB8500_MUTECONF */
+#define AB8500_MUTECONF_MUTEAR			6
+#define AB8500_MUTECONF_MUTHSL			5
+#define AB8500_MUTECONF_MUTHSR			4
+#define AB8500_MUTECONF_MUTDACEAR		2
+#define AB8500_MUTECONF_MUTDACHSL		1
+#define AB8500_MUTECONF_MUTDACHSR		0
+
+/* AB8500_SHORTCIRCONF */
+#define AB8500_SHORTCIRCONF_ENSHORTPWD		7
+#define AB8500_SHORTCIRCONF_EARSHORTDIS		6
+#define AB8500_SHORTCIRCONF_HSSHORTDIS		5
+#define AB8500_SHORTCIRCONF_HSPULLDEN		4
+#define AB8500_SHORTCIRCONF_HSOSCEN		2
+#define AB8500_SHORTCIRCONF_HSFADDIS		1
+#define AB8500_SHORTCIRCONF_HSZCDDIS		0
+/* Zero cross should be disabled */
+
+/* AB8500_ANACONF5 */
+#define AB8500_ANACONF5_ENCPHS			7
+#define AB8500_ANACONF5_HSLDACTOLOL		5
+#define AB8500_ANACONF5_HSRDACTOLOR		4
+#define AB8500_ANACONF5_ENLOL			3
+#define AB8500_ANACONF5_ENLOR			2
+#define AB8500_ANACONF5_HSAUTOEN		0
+
+/* AB8500_ENVCPCONF */
+#define AB8500_ENVCPCONF_ENVDETHTHRE		4
+#define AB8500_ENVCPCONF_ENVDETLTHRE		0
+#define AB8500_ENVCPCONF_ENVDETHTHRE_MAX	0x0F
+#define AB8500_ENVCPCONF_ENVDETLTHRE_MAX	0x0F
+
+/* AB8500_SIGENVCONF */
+#define AB8500_SIGENVCONF_CPLVEN		5
+#define AB8500_SIGENVCONF_ENVDETCPEN		4
+#define AB8500_SIGENVCONF_ENVDETTIME		0
+#define AB8500_SIGENVCONF_ENVDETTIME_MAX	0x0F
+
+/* AB8500_PWMGENCONF1 */
+#define AB8500_PWMGENCONF1_PWMTOVIB1		7
+#define AB8500_PWMGENCONF1_PWMTOVIB2		6
+#define AB8500_PWMGENCONF1_PWM1CTRL		5
+#define AB8500_PWMGENCONF1_PWM2CTRL		4
+#define AB8500_PWMGENCONF1_PWM1NCTRL		3
+#define AB8500_PWMGENCONF1_PWM1PCTRL		2
+#define AB8500_PWMGENCONF1_PWM2NCTRL		1
+#define AB8500_PWMGENCONF1_PWM2PCTRL		0
+
+/* AB8500_PWMGENCONF2 */
+/* AB8500_PWMGENCONF3 */
+/* AB8500_PWMGENCONF4 */
+/* AB8500_PWMGENCONF5 */
+#define AB8500_PWMGENCONFX_PWMVIBXPOL		7
+#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC	0
+#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX	0x64
+
+/* AB8500_ANAGAIN1 */
+/* AB8500_ANAGAIN2 */
+#define AB8500_ANAGAINX_ENSEMICX		7
+#define AB8500_ANAGAINX_LOWPOWMICX		6
+#define AB8500_ANAGAINX_MICXGAIN		0
+#define AB8500_ANAGAINX_MICXGAIN_MAX		0x1F
+
+/* AB8500_ANAGAIN3 */
+#define AB8500_ANAGAIN3_HSLGAIN			4
+#define AB8500_ANAGAIN3_HSRGAIN			0
+#define AB8500_ANAGAIN3_HSXGAIN_MAX		0x0F
+
+/* AB8500_ANAGAIN4 */
+#define AB8500_ANAGAIN4_LINLGAIN		4
+#define AB8500_ANAGAIN4_LINRGAIN		0
+#define AB8500_ANAGAIN4_LINXGAIN_MAX		0x0F
+
+/* AB8500_DIGLINHSLGAIN */
+/* AB8500_DIGLINHSRGAIN */
+#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN	0
+#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX	0x13
+
+/* AB8500_ADFILTCONF */
+#define AB8500_ADFILTCONF_AD1NH			7
+#define AB8500_ADFILTCONF_AD2NH			6
+#define AB8500_ADFILTCONF_AD3NH			5
+#define AB8500_ADFILTCONF_AD4NH			4
+#define AB8500_ADFILTCONF_AD1VOICE		3
+#define AB8500_ADFILTCONF_AD2VOICE		2
+#define AB8500_ADFILTCONF_AD3VOICE		1
+#define AB8500_ADFILTCONF_AD4VOICE		0
+
+/* AB8500_DIGIFCONF1 */
+#define AB8500_DIGIFCONF1_ENMASTGEN		7
+#define AB8500_DIGIFCONF1_IF1BITCLKOS1		6
+#define AB8500_DIGIFCONF1_IF1BITCLKOS0		5
+#define AB8500_DIGIFCONF1_ENFSBITCLK1		4
+#define AB8500_DIGIFCONF1_IF0BITCLKOS1		2
+#define AB8500_DIGIFCONF1_IF0BITCLKOS0		1
+#define AB8500_DIGIFCONF1_ENFSBITCLK0		0
+
+/* AB8500_DIGIFCONF2 */
+#define AB8500_DIGIFCONF2_FSYNC0P		6
+#define AB8500_DIGIFCONF2_BITCLK0P		5
+#define AB8500_DIGIFCONF2_IF0DEL		4
+#define AB8500_DIGIFCONF2_IF0FORMAT1		3
+#define AB8500_DIGIFCONF2_IF0FORMAT0		2
+#define AB8500_DIGIFCONF2_IF0WL1		1
+#define AB8500_DIGIFCONF2_IF0WL0		0
+
+/* AB8500_DIGIFCONF3 */
+#define AB8500_DIGIFCONF3_IF0DATOIF1AD		7
+#define AB8500_DIGIFCONF3_IF0CLKTOIF1CLK	6
+#define AB8500_DIGIFCONF3_IF1MASTER		5
+#define AB8500_DIGIFCONF3_IF1DATOIF0AD		3
+#define AB8500_DIGIFCONF3_IF1CLKTOIF0CLK	2
+#define AB8500_DIGIFCONF3_IF0MASTER		1
+#define AB8500_DIGIFCONF3_IF0BFIFOEN		0
+
+/* AB8500_DIGIFCONF4 */
+#define AB8500_DIGIFCONF4_FSYNC1P		6
+#define AB8500_DIGIFCONF4_BITCLK1P		5
+#define AB8500_DIGIFCONF4_IF1DEL		4
+#define AB8500_DIGIFCONF4_IF1FORMAT1		3
+#define AB8500_DIGIFCONF4_IF1FORMAT0		2
+#define AB8500_DIGIFCONF4_IF1WL1		1
+#define AB8500_DIGIFCONF4_IF1WL0		0
+
+/* AB8500_ADSLOTSELX */
+#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD	0x00
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD	0x01
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD	0x02
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD	0x03
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD	0x04
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD	0x05
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD	0x06
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD	0x07
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD	0x08
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD	0x0F
+#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN	0x00
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN	0x10
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN	0x20
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN	0x30
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN	0x40
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN	0x50
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN	0x60
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN	0x70
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN	0x80
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN	0xF0
+#define AB8500_ADSLOTSELX_EVEN_SHIFT		0
+#define AB8500_ADSLOTSELX_ODD_SHIFT		4
+
+/* AB8500_ADSLOTHIZCTRL1 */
+/* AB8500_ADSLOTHIZCTRL2 */
+/* AB8500_ADSLOTHIZCTRL3 */
+/* AB8500_ADSLOTHIZCTRL4 */
+/* AB8500_DASLOTCONF1 */
+#define AB8500_DASLOTCONF1_DA12VOICE		7
+#define AB8500_DASLOTCONF1_SWAPDA12_34		6
+#define AB8500_DASLOTCONF1_DAI7TOADO1		5
+
+/* AB8500_DASLOTCONF2 */
+#define AB8500_DASLOTCONF2_DAI8TOADO2		5
+
+/* AB8500_DASLOTCONF3 */
+#define AB8500_DASLOTCONF3_DA34VOICE		7
+#define AB8500_DASLOTCONF3_DAI7TOADO3		5
+
+/* AB8500_DASLOTCONF4 */
+#define AB8500_DASLOTCONF4_DAI8TOADO4		5
+
+/* AB8500_DASLOTCONF5 */
+#define AB8500_DASLOTCONF5_DA56VOICE		7
+#define AB8500_DASLOTCONF5_DAI7TOADO5		5
+
+/* AB8500_DASLOTCONF6 */
+#define AB8500_DASLOTCONF6_DAI8TOADO6		5
+
+/* AB8500_DASLOTCONF7 */
+#define AB8500_DASLOTCONF7_DAI8TOADO7		5
+
+/* AB8500_DASLOTCONF8 */
+#define AB8500_DASLOTCONF8_DAI7TOADO8		5
+
+#define AB8500_DASLOTCONFX_SLTODAX_SHIFT	0
+#define AB8500_DASLOTCONFX_SLTODAX_MASK		0x1F
+
+/* AB8500_CLASSDCONF1 */
+#define AB8500_CLASSDCONF1_PARLHF		7
+#define AB8500_CLASSDCONF1_PARLVIB		6
+#define AB8500_CLASSDCONF1_VIB1SWAPEN		3
+#define AB8500_CLASSDCONF1_VIB2SWAPEN		2
+#define AB8500_CLASSDCONF1_HFLSWAPEN		1
+#define AB8500_CLASSDCONF1_HFRSWAPEN		0
+
+/* AB8500_CLASSDCONF2 */
+#define AB8500_CLASSDCONF2_FIRBYP3		7
+#define AB8500_CLASSDCONF2_FIRBYP2		6
+#define AB8500_CLASSDCONF2_FIRBYP1		5
+#define AB8500_CLASSDCONF2_FIRBYP0		4
+#define AB8500_CLASSDCONF2_HIGHVOLEN3		3
+#define AB8500_CLASSDCONF2_HIGHVOLEN2		2
+#define AB8500_CLASSDCONF2_HIGHVOLEN1		1
+#define AB8500_CLASSDCONF2_HIGHVOLEN0		0
+
+/* AB8500_CLASSDCONF3 */
+#define AB8500_CLASSDCONF3_DITHHPGAIN		4
+#define AB8500_CLASSDCONF3_DITHHPGAIN_MAX	0x0A
+#define AB8500_CLASSDCONF3_DITHWGAIN		0
+#define AB8500_CLASSDCONF3_DITHWGAIN_MAX	0x0A
+
+/* AB8500_DMICFILTCONF */
+#define AB8500_DMICFILTCONF_ANCINSEL		7
+#define AB8500_DMICFILTCONF_DA3TOEAR		6
+#define AB8500_DMICFILTCONF_DMIC1SINC3		5
+#define AB8500_DMICFILTCONF_DMIC2SINC3		4
+#define AB8500_DMICFILTCONF_DMIC3SINC3		3
+#define AB8500_DMICFILTCONF_DMIC4SINC3		2
+#define AB8500_DMICFILTCONF_DMIC5SINC3		1
+#define AB8500_DMICFILTCONF_DMIC6SINC3		0
+
+/* AB8500_DIGMULTCONF1 */
+#define AB8500_DIGMULTCONF1_DATOHSLEN		7
+#define AB8500_DIGMULTCONF1_DATOHSREN		6
+#define AB8500_DIGMULTCONF1_AD1SEL		5
+#define AB8500_DIGMULTCONF1_AD2SEL		4
+#define AB8500_DIGMULTCONF1_AD3SEL		3
+#define AB8500_DIGMULTCONF1_AD5SEL		2
+#define AB8500_DIGMULTCONF1_AD6SEL		1
+#define AB8500_DIGMULTCONF1_ANCSEL		0
+
+/* AB8500_DIGMULTCONF2 */
+#define AB8500_DIGMULTCONF2_DATOHFREN		7
+#define AB8500_DIGMULTCONF2_DATOHFLEN		6
+#define AB8500_DIGMULTCONF2_HFRSEL		5
+#define AB8500_DIGMULTCONF2_HFLSEL		4
+#define AB8500_DIGMULTCONF2_FIRSID1SEL		2
+#define AB8500_DIGMULTCONF2_FIRSID2SEL		0
+
+/* AB8500_ADDIGGAIN1 */
+/* AB8500_ADDIGGAIN2 */
+/* AB8500_ADDIGGAIN3 */
+/* AB8500_ADDIGGAIN4 */
+/* AB8500_ADDIGGAIN5 */
+/* AB8500_ADDIGGAIN6 */
+#define AB8500_ADDIGGAINX_FADEDISADX		6
+#define AB8500_ADDIGGAINX_ADXGAIN_MAX		0x3F
+
+/* AB8500_DADIGGAIN1 */
+/* AB8500_DADIGGAIN2 */
+/* AB8500_DADIGGAIN3 */
+/* AB8500_DADIGGAIN4 */
+/* AB8500_DADIGGAIN5 */
+/* AB8500_DADIGGAIN6 */
+#define AB8500_DADIGGAINX_FADEDISDAX		6
+#define AB8500_DADIGGAINX_DAXGAIN_MAX		0x3F
+
+/* AB8500_ADDIGLOOPGAIN1 */
+/* AB8500_ADDIGLOOPGAIN2 */
+#define AB8500_ADDIGLOOPGAINX_FADEDISADXL	6
+#define AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX	0x3F
+
+/* AB8500_HSLEARDIGGAIN */
+#define AB8500_HSLEARDIGGAIN_HSSINC1		7
+#define AB8500_HSLEARDIGGAIN_FADEDISHSL		4
+#define AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX	0x09
+
+/* AB8500_HSRDIGGAIN */
+#define AB8500_HSRDIGGAIN_FADESPEED		6
+#define AB8500_HSRDIGGAIN_FADEDISHSR		4
+#define AB8500_HSRDIGGAIN_HSRDGAIN_MAX		0x09
+
+/* AB8500_SIDFIRGAIN1 */
+/* AB8500_SIDFIRGAIN2 */
+#define AB8500_SIDFIRGAINX_FIRSIDXGAIN_MAX	0x1F
+
+/* AB8500_ANCCONF1 */
+#define AB8500_ANCCONF1_ANCIIRUPDATE		3
+#define AB8500_ANCCONF1_ENANC			2
+#define AB8500_ANCCONF1_ANCIIRINIT		1
+#define AB8500_ANCCONF1_ANCFIRUPDATE		0
+
+/* AB8500_ANCCONF2 */
+#define AB8500_ANCCONF2_SHIFT			5
+#define AB8500_ANCCONF2_MIN			-0x10
+#define AB8500_ANCCONF2_MAX			0xF
+
+/* AB8500_ANCCONF3 */
+#define AB8500_ANCCONF3_SHIFT			5
+#define AB8500_ANCCONF3_MIN			-0x10
+#define AB8500_ANCCONF3_MAX			0xF
+
+/* AB8500_ANCCONF4 */
+#define AB8500_ANCCONF4_SHIFT			5
+#define AB8500_ANCCONF4_MIN			-0x10
+#define AB8500_ANCCONF4_MAX			0xF
+
+/* AB8500_ANC_FIR_COEFFS */
+#define AB8500_ANC_FIR_COEFF_MIN		-0x8000
+#define AB8500_ANC_FIR_COEFF_MAX		0x7FFF
+#define AB8500_ANC_FIR_COEFFS			15
+
+/* AB8500_ANC_IIR_COEFFS */
+#define AB8500_ANC_IIR_COEFF_MIN		-0x800000
+#define AB8500_ANC_IIR_COEFF_MAX		0x7FFFFF
+#define AB8500_ANC_IIR_COEFFS			24
+/* AB8500_ANC_WARP_DELAY */
+#define AB8500_ANC_WARP_DELAY_SHIFT		16
+#define AB8500_ANC_WARP_DELAY_MIN		0x0000
+#define AB8500_ANC_WARP_DELAY_MAX		0xFFFF
+
+/* AB8500_ANCCONF11 */
+/* AB8500_ANCCONF12 */
+/* AB8500_ANCCONF13 */
+/* AB8500_ANCCONF14 */
+
+/* AB8500_SIDFIRADR */
+#define AB8500_SIDFIRADR_FIRSIDSET		7
+#define AB8500_SIDFIRADR_ADDRESS_SHIFT		0
+#define AB8500_SIDFIRADR_ADDRESS_MAX		0x7F
+
+/* AB8500_SIDFIRCOEF1 */
+/* AB8500_SIDFIRCOEF2 */
+#define AB8500_SID_FIR_COEFF_MIN		0
+#define AB8500_SID_FIR_COEFF_MAX		0xFFFF
+#define AB8500_SID_FIR_COEFFS			128
+
+/* AB8500_SIDFIRCONF */
+#define AB8500_SIDFIRCONF_ENFIRSIDS		2
+#define AB8500_SIDFIRCONF_FIRSIDSTOIF1		1
+#define AB8500_SIDFIRCONF_FIRSIDBUSY		0
+
+/* AB8500_AUDINTMASK1 */
+/* AB8500_AUDINTSOURCE1 */
+/* AB8500_AUDINTMASK2 */
+/* AB8500_AUDINTSOURCE2 */
+
+/* AB8500_FIFOCONF1 */
+#define AB8500_FIFOCONF1_BFIFOMASK		0x80
+#define AB8500_FIFOCONF1_BFIFO19M2		0x40
+#define AB8500_FIFOCONF1_BFIFOINT_SHIFT		0
+#define AB8500_FIFOCONF1_BFIFOINT_MAX		0x3F
+
+/* AB8500_FIFOCONF2 */
+#define AB8500_FIFOCONF2_BFIFOTX_SHIFT		0
+#define AB8500_FIFOCONF2_BFIFOTX_MAX		0xFF
+
+/* AB8500_FIFOCONF3 */
+#define AB8500_FIFOCONF3_BFIFOEXSL_SHIFT	5
+#define AB8500_FIFOCONF3_BFIFOEXSL_MAX		0x5
+#define AB8500_FIFOCONF3_PREBITCLK0_SHIFT	2
+#define AB8500_FIFOCONF3_PREBITCLK0_MAX		0x7
+#define AB8500_FIFOCONF3_BFIFOMAST_SHIFT	1
+#define AB8500_FIFOCONF3_BFIFORUN_SHIFT		0
+
+/* AB8500_FIFOCONF4 */
+#define AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT	0
+#define AB8500_FIFOCONF4_BFIFOFRAMSW_MAX	0xFF
+
+/* AB8500_FIFOCONF5 */
+#define AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT	0
+#define AB8500_FIFOCONF5_BFIFOWAKEUP_MAX	0xFF
+
+/* AB8500_FIFOCONF6 */
+#define AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT	0
+#define AB8500_FIFOCONF6_BFIFOSAMPLE_MAX	0xFF
+
+/* AB8500_AUDREV */
+
+#endif
-- 
cgit v1.2.3


From 754fdff86f956a91834887ad56ea292f5d2fa114 Mon Sep 17 00:00:00 2001
From: Annie Liu <annieliu@viatech.com.cn>
Date: Fri, 8 Jun 2012 19:18:39 +0800
Subject: ALSA: hda - add support for HD-Audio of VIA HDMI GFX Cards

This is patch supporting HD-Audio function of VIA GFX cards which
support HDMI.
Those are integrated graphics of chipsets VX900 and VX11 separately.

Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2b6392be451c..d49926e4d19f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3338,6 +3338,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* VIA VT8251/VT8237A */
 	{ PCI_DEVICE(0x1106, 0x3288),
 	  .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA },
+	/* VIA GFX VT7122/VX900 */
+	{ PCI_DEVICE(0x1106, 0x9170), .driver_data = AZX_DRIVER_GENERIC },
+	/* VIA GFX VT6122/VX11 */
+	{ PCI_DEVICE(0x1106, 0x9140), .driver_data = AZX_DRIVER_GENERIC },
 	/* SIS966 */
 	{ PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS },
 	/* ULI M5461 */
-- 
cgit v1.2.3


From 3de5ff88773d9f106b668937da2f36c97801b332 Mon Sep 17 00:00:00 2001
From: Annie Liu <annieliu@viatech.com.cn>
Date: Fri, 8 Jun 2012 19:18:42 +0800
Subject: ALSA: hda - add support for HD-Audio CODECes of VIA HDMI GFX Cards

This is patch supporting the CODECes of HD-Audio function of VIA GFX
cards which support HDMI.
For CODECes 0x9f80/0x9f81, which belong to VX900, since the hardware
is not fully compliant to HD-Audio 1.3, simple_i*() is adopted
temporarily.

Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 64 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index ad319d4dc32f..696681826b01 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1863,6 +1863,62 @@ static int patch_atihdmi(struct hda_codec *codec)
 	return 0;
 }
 
+/* VIA HDMI Implementation */
+#define VIAHDMI_CVT_NID	0x02	/* audio converter1 */
+#define VIAHDMI_PIN_NID	0x03	/* HDMI output pin1 */
+
+static struct hda_verb viahdmi_basic_init[] = {
+	/* enable digital output on pin widget */
+	{ VIAHDMI_PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{} /* terminator */
+};
+
+static int via_hdmi_init(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, viahdmi_basic_init);
+	return 0;
+}
+
+static const struct hda_codec_ops via_hdmi_patch_ops = {
+	.build_controls = simple_playback_build_controls,
+	.build_pcms = simple_playback_build_pcms,
+	.init = via_hdmi_init,
+	.free = simple_playback_free,
+};
+
+static struct hda_pcm_stream via_hdmi_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = VIAHDMI_CVT_NID, /* NID to query formats and rates*/
+	.ops = {
+		.open = simple_playback_pcm_open,
+		.close = simple_playback_pcm_close,
+		.prepare = simple_playback_pcm_prepare
+	},
+};
+
+static int patch_via_hdmi(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	spec->multiout.num_dacs = 0;	  /* no analog */
+	spec->multiout.max_channels = 2;
+	spec->multiout.dig_out_nid = VIAHDMI_CVT_NID; /* pure-digital case */
+	spec->num_cvts = 1;
+	spec->cvts[0].cvt_nid = VIAHDMI_CVT_NID;
+	spec->pins[0].pin_nid = VIAHDMI_PIN_NID;
+	spec->pcm_playback = &via_hdmi_digital_playback;
+
+	codec->spec = spec;
+	codec->patch_ops = via_hdmi_patch_ops;
+
+	return 0;
+}
 
 /*
  * patch entries
@@ -1904,6 +1960,10 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
+{ .id = 0x11069f80, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
+{ .id = 0x11069f81, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
+{ .id = 0x11069f84, .name = "VX11 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x11069f85, .name = "VX11 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x80860054, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862801, .name = "Bearlake HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862802, .name = "Cantiga HDMI",	.patch = patch_generic_hdmi },
@@ -1950,6 +2010,10 @@ MODULE_ALIAS("snd-hda-codec-id:10de0043");
 MODULE_ALIAS("snd-hda-codec-id:10de0044");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
 MODULE_ALIAS("snd-hda-codec-id:10de8001");
+MODULE_ALIAS("snd-hda-codec-id:11069f80");
+MODULE_ALIAS("snd-hda-codec-id:11069f81");
+MODULE_ALIAS("snd-hda-codec-id:11069f84");
+MODULE_ALIAS("snd-hda-codec-id:11069f85");
 MODULE_ALIAS("snd-hda-codec-id:17e80047");
 MODULE_ALIAS("snd-hda-codec-id:80860054");
 MODULE_ALIAS("snd-hda-codec-id:80862801");
-- 
cgit v1.2.3


From b71dad181a55d2ad90bd03cd3216a5a8a31d9468 Mon Sep 17 00:00:00 2001
From: Mark Hills <mark@pogo.org.uk>
Date: Sat, 9 Jun 2012 13:16:38 +0100
Subject: ALSA: usb-audio: Use a table of mixer controls

Allow mixer controls to be provided clearly in a table, to avoid
quantity of error checking at each use.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_quirks.c | 105 ++++++++++++++++++++++-------------------------
 1 file changed, 49 insertions(+), 56 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 41f4b6911920..ce7d96f91578 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -42,6 +42,13 @@
 
 extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
 
+struct std_mono_table {
+	unsigned int unitid, control, cmask;
+	int val_type;
+	const char *name;
+	snd_kcontrol_tlv_rw_t *tlv_callback;
+};
+
 /* private_free callback */
 static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
 {
@@ -113,6 +120,25 @@ static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer,
 	return 0;
 }
 
+/*
+ * Create a set of standard UAC controls from a table
+ */
+static int snd_create_std_mono_table(struct usb_mixer_interface *mixer,
+				struct std_mono_table *t)
+{
+	int err;
+
+	while (t->name != NULL) {
+		err = snd_create_std_mono_ctl(mixer, t->unitid, t->control,
+				t->cmask, t->val_type, t->name, t->tlv_callback);
+		if (err < 0)
+			return err;
+		t++;
+	}
+
+	return 0;
+}
+
 /*
  * Sound Blaster remote control configuration
  *
@@ -916,61 +942,6 @@ static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
-
-/*
- * Create mixer for Electrix Ebox-44
- *
- * The mixer units from this device are corrupt, and even where they
- * are valid they presents mono controls as L and R channels of
- * stereo. So we create a good mixer in code.
- */
-
-static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer)
-{
-	int err;
-
-	err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Headphone Playback Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16,
-				"Headphone A Mix Playback Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16,
-				"Headphone B Mix Playback Volume", NULL);
-	if (err < 0)
-		return err;
-
-	err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Output Playback Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16,
-				"Output A Playback Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16,
-				"Output B Playback Volume", NULL);
-	if (err < 0)
-		return err;
-
-	err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Input Capture Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16,
-				"Input A Capture Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16,
-				"Input B Capture Volume", NULL);
-	if (err < 0)
-		return err;
-
-	return 0;
-}
-
 void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 			       unsigned char samplerate_id)
 {
@@ -990,6 +961,27 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 	}
 }
 
+/*
+ * The mixer units for Ebox-44 are corrupt, and even where they
+ * are valid they presents mono controls as L and R channels of
+ * stereo. So we provide a good mixer here.
+ */
+struct std_mono_table ebox44_table[] = {
+	{ 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Headphone Playback Switch", NULL },
+	{ 4, 2, 0x1, USB_MIXER_S16, "Headphone A Mix Playback Volume", NULL },
+	{ 4, 2, 0x2, USB_MIXER_S16, "Headphone B Mix Playback Volume", NULL },
+
+	{ 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Output Playback Switch", NULL },
+	{ 7, 2, 0x1, USB_MIXER_S16, "Output A Playback Volume", NULL },
+	{ 7, 2, 0x2, USB_MIXER_S16, "Output B Playback Volume", NULL },
+
+	{ 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Input Capture Switch", NULL },
+	{ 10, 2, 0x1, USB_MIXER_S16, "Input A Capture Volume", NULL },
+	{ 10, 2, 0x2, USB_MIXER_S16, "Input B Capture Volume", NULL },
+
+	{ }
+};
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -1035,7 +1027,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 		break;
 
 	case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */
-		err = snd_ebox44_create_mixer(mixer);
+		/* detection is disabled in mixer_maps.c */
+		err = snd_create_std_mono_table(mixer, ebox44_table);
 		break;
 	}
 
-- 
cgit v1.2.3


From 989b01385fa3cc4eaa488068a0868ae4de5198a9 Mon Sep 17 00:00:00 2001
From: Mark Hills <mark@pogo.org.uk>
Date: Sat, 9 Jun 2012 13:16:39 +0100
Subject: ALSA: usb-audio: Convert table to preferred C99 format

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer_quirks.c | 74 +++++++++++++++++++++++++++++++++++++++++-------
 1 file changed, 64 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index ce7d96f91578..690000db0ec0 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -967,19 +967,73 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
  * stereo. So we provide a good mixer here.
  */
 struct std_mono_table ebox44_table[] = {
-	{ 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Headphone Playback Switch", NULL },
-	{ 4, 2, 0x1, USB_MIXER_S16, "Headphone A Mix Playback Volume", NULL },
-	{ 4, 2, 0x2, USB_MIXER_S16, "Headphone B Mix Playback Volume", NULL },
+	{
+		.unitid = 4,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Headphone Playback Switch"
+	},
+	{
+		.unitid = 4,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Headphone A Mix Playback Volume"
+	},
+	{
+		.unitid = 4,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Headphone B Mix Playback Volume"
+	},
 
-	{ 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Output Playback Switch", NULL },
-	{ 7, 2, 0x1, USB_MIXER_S16, "Output A Playback Volume", NULL },
-	{ 7, 2, 0x2, USB_MIXER_S16, "Output B Playback Volume", NULL },
+	{
+		.unitid = 7,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Output Playback Switch"
+	},
+	{
+		.unitid = 7,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Output A Playback Volume"
+	},
+	{
+		.unitid = 7,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Output B Playback Volume"
+	},
 
-	{ 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Input Capture Switch", NULL },
-	{ 10, 2, 0x1, USB_MIXER_S16, "Input A Capture Volume", NULL },
-	{ 10, 2, 0x2, USB_MIXER_S16, "Input B Capture Volume", NULL },
+	{
+		.unitid = 10,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Input Capture Switch"
+	},
+	{
+		.unitid = 10,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Input A Capture Volume"
+	},
+	{
+		.unitid = 10,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Input B Capture Volume"
+	},
 
-	{ }
+	{}
 };
 
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
-- 
cgit v1.2.3


From 66c2b7377a7cf22c48ebba7fdff5340ab492b7bc Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 11 Jun 2012 18:09:46 +0800
Subject: ASoC: wm8996: Remove write sequencer registers from the defaults
 table

They aren't marked as readable and the feature is never used so they'll
never get referenced.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8996.c | 178 ----------------------------------------------
 1 file changed, 178 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index efc4e9d0903b..f24989f090ca 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -296,184 +296,6 @@ static struct reg_default wm8996_reg[] = {
 	{ WM8996_RIGHT_PDM_SPEAKER, 0x1 },
 	{ WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 },
 	{ WM8996_PDM_SPEAKER_VOLUME, 0x66 },
-	{ WM8996_WRITE_SEQUENCER_0, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_1, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_3, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_4, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_5, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_6, 0xf },
-	{ WM8996_WRITE_SEQUENCER_7, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_8, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_9, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_10, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_12, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_13, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_14, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_16, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_17, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_18, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_20, 0x51 },
-	{ WM8996_WRITE_SEQUENCER_21, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_22, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_23, 0xa },
-	{ WM8996_WRITE_SEQUENCER_24, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_25, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_26, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_27, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_28, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_32, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_36, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_40, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_44, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_48, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_52, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_56, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_60, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_64, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_65, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_67, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_68, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_69, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_70, 0xf },
-	{ WM8996_WRITE_SEQUENCER_71, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_72, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_73, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_74, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_76, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_77, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_78, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_80, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_81, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_82, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_84, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_85, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_86, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_87, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_88, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_92, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_96, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_100, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_104, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_108, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_112, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_116, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_120, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_124, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_128, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_129, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_131, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_132, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_133, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_134, 0xf },
-	{ WM8996_WRITE_SEQUENCER_135, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_136, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_137, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_138, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_140, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_141, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_142, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_144, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_145, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_146, 0x102 },
-	{ WM8996_WRITE_SEQUENCER_148, 0x51 },
-	{ WM8996_WRITE_SEQUENCER_149, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_150, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_151, 0xa },
-	{ WM8996_WRITE_SEQUENCER_152, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_153, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_154, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_155, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_156, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_160, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_164, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_168, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_172, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_176, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_180, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_184, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_188, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_192, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_193, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_195, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_196, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_197, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_198, 0xf },
-	{ WM8996_WRITE_SEQUENCER_199, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_200, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_201, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_202, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_204, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_205, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_206, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_208, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_209, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_210, 0x102 },
-	{ WM8996_WRITE_SEQUENCER_212, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_213, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_214, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_215, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_216, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_220, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_224, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_228, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_232, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_236, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_240, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_244, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_248, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_252, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_256, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_258, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_260, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_262, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_264, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_266, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_267, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_268, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_272, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_276, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_280, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_284, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_288, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_292, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_296, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_300, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_304, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_308, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_312, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_316, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_320, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_322, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_324, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_326, 0x102 },
-	{ WM8996_WRITE_SEQUENCER_328, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_330, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_331, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_332, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_336, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_340, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_344, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_348, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_352, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_356, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_360, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_364, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_368, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_372, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_376, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_380, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_384, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_386, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_388, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_390, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_392, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_394, 0x300 },
-	{ WM8996_WRITE_SEQUENCER_396, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_398, 0x304 },
-	{ WM8996_WRITE_SEQUENCER_400, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_402, 0xf },
-	{ WM8996_WRITE_SEQUENCER_404, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_407, 0x100 },
 };
 
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0);
-- 
cgit v1.2.3


From af691fb62c626fe374955ab306092b09f672e27d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 11 Jun 2012 18:20:48 +0800
Subject: ASoC: wm8996: Convert to devm_regmap_init_i2c()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8996.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index f24989f090ca..a6b5cffa498a 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -3000,7 +3000,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 		msleep(5);
 	}
 
-	wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap);
+	wm8996->regmap = devm_regmap_init_i2c(i2c, &wm8996_regmap);
 	if (IS_ERR(wm8996->regmap)) {
 		ret = PTR_ERR(wm8996->regmap);
 		dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
@@ -3049,7 +3049,6 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 err_gpiolib:
 	wm8996_free_gpio(wm8996);
 err_regmap:
-	regmap_exit(wm8996->regmap);
 err_enable:
 	if (wm8996->pdata.ldo_ena > 0)
 		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
@@ -3068,7 +3067,6 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client)
 
 	snd_soc_unregister_codec(&client->dev);
 	wm8996_free_gpio(wm8996);
-	regmap_exit(wm8996->regmap);
 	if (wm8996->pdata.ldo_ena > 0) {
 		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
 		gpio_free(wm8996->pdata.ldo_ena);
-- 
cgit v1.2.3


From 48e278746070b5fc62ec3da2e65f7cd511f6bbf4 Mon Sep 17 00:00:00 2001
From: Adam Thomson <Adam.Thomson@diasemi.com>
Date: Mon, 11 Jun 2012 13:15:27 +0100
Subject: ASoC: codecs: Add DA732x codec driver

This patch adds support for Dialog DA732x audio codecs.

Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com>
Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig      |    4 +
 sound/soc/codecs/Makefile     |    2 +
 sound/soc/codecs/da732x.c     | 1627 +++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/da732x.h     |  133 ++++
 sound/soc/codecs/da732x_reg.h |  654 +++++++++++++++++
 5 files changed, 2420 insertions(+)
 create mode 100644 sound/soc/codecs/da732x.c
 create mode 100644 sound/soc/codecs/da732x.h
 create mode 100644 sound/soc/codecs/da732x_reg.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index f63776d422b3..43f5240e6942 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
+	select SND_SOC_DA732X if I2C
 	select SND_SOC_DFBMCS320
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
@@ -224,6 +225,9 @@ config SND_SOC_L3
 config SND_SOC_DA7210
         tristate
 
+config SND_SOC_DA732X
+        tristate
+
 config SND_SOC_DFBMCS320
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fc93b4b0c2c5..3d30654f6fcc 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -22,6 +22,7 @@ snd-soc-cs4270-objs := cs4270.o
 snd-soc-cs4271-objs := cs4271.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
+snd-soc-da732x-objs := da732x.o
 snd-soc-dfbmcs320-objs := dfbmcs320.o
 snd-soc-dmic-objs := dmic.o
 snd-soc-isabelle-objs := isabelle.o
@@ -135,6 +136,7 @@ obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
new file mode 100644
index 000000000000..04af369f228c
--- /dev/null
+++ b/sound/soc/codecs/da732x.c
@@ -0,0 +1,1627 @@
+/*
+ * da732x.c --- Dialog DA732X ALSA SoC Audio Driver
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/sysfs.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 <asm/div64.h>
+
+#include "da732x.h"
+#include "da732x_reg.h"
+
+
+struct da732x_priv {
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+
+	unsigned int sysclk;
+	bool pll_en;
+};
+
+/*
+ * da732x register cache - default settings
+ */
+static struct reg_default da732x_reg_cache[] = {
+	{ DA732X_REG_REF1		, 0x02 },
+	{ DA732X_REG_BIAS_EN		, 0x80 },
+	{ DA732X_REG_BIAS1		, 0x00 },
+	{ DA732X_REG_BIAS2		, 0x00 },
+	{ DA732X_REG_BIAS3		, 0x00 },
+	{ DA732X_REG_BIAS4		, 0x00 },
+	{ DA732X_REG_MICBIAS2		, 0x00 },
+	{ DA732X_REG_MICBIAS1		, 0x00 },
+	{ DA732X_REG_MICDET		, 0x00 },
+	{ DA732X_REG_MIC1_PRE		, 0x01 },
+	{ DA732X_REG_MIC1		, 0x40 },
+	{ DA732X_REG_MIC2_PRE		, 0x01 },
+	{ DA732X_REG_MIC2		, 0x40 },
+	{ DA732X_REG_AUX1L		, 0x75 },
+	{ DA732X_REG_AUX1R		, 0x75 },
+	{ DA732X_REG_MIC3_PRE		, 0x01 },
+	{ DA732X_REG_MIC3		, 0x40 },
+	{ DA732X_REG_INP_PINBIAS	, 0x00 },
+	{ DA732X_REG_INP_ZC_EN		, 0x00 },
+	{ DA732X_REG_INP_MUX		, 0x50 },
+	{ DA732X_REG_HP_DET		, 0x00 },
+	{ DA732X_REG_HPL_DAC_OFFSET	, 0x00 },
+	{ DA732X_REG_HPL_DAC_OFF_CNTL	, 0x00 },
+	{ DA732X_REG_HPL_OUT_OFFSET	, 0x00 },
+	{ DA732X_REG_HPL		, 0x40 },
+	{ DA732X_REG_HPL_VOL		, 0x0F },
+	{ DA732X_REG_HPR_DAC_OFFSET	, 0x00 },
+	{ DA732X_REG_HPR_DAC_OFF_CNTL	, 0x00 },
+	{ DA732X_REG_HPR_OUT_OFFSET	, 0x00 },
+	{ DA732X_REG_HPR		, 0x40 },
+	{ DA732X_REG_HPR_VOL		, 0x0F },
+	{ DA732X_REG_LIN2		, 0x4F },
+	{ DA732X_REG_LIN3		, 0x4F },
+	{ DA732X_REG_LIN4		, 0x4F },
+	{ DA732X_REG_OUT_ZC_EN		, 0x00 },
+	{ DA732X_REG_HP_LIN1_GNDSEL	, 0x00 },
+	{ DA732X_REG_CP_HP1		, 0x0C },
+	{ DA732X_REG_CP_HP2		, 0x03 },
+	{ DA732X_REG_CP_CTRL1		, 0x00 },
+	{ DA732X_REG_CP_CTRL2		, 0x99 },
+	{ DA732X_REG_CP_CTRL3		, 0x25 },
+	{ DA732X_REG_CP_LEVEL_MASK	, 0x3F },
+	{ DA732X_REG_CP_DET		, 0x00 },
+	{ DA732X_REG_CP_STATUS		, 0x00 },
+	{ DA732X_REG_CP_THRESH1		, 0x00 },
+	{ DA732X_REG_CP_THRESH2		, 0x00 },
+	{ DA732X_REG_CP_THRESH3		, 0x00 },
+	{ DA732X_REG_CP_THRESH4		, 0x00 },
+	{ DA732X_REG_CP_THRESH5		, 0x00 },
+	{ DA732X_REG_CP_THRESH6		, 0x00 },
+	{ DA732X_REG_CP_THRESH7		, 0x00 },
+	{ DA732X_REG_CP_THRESH8		, 0x00 },
+	{ DA732X_REG_PLL_DIV_LO		, 0x00 },
+	{ DA732X_REG_PLL_DIV_MID	, 0x00 },
+	{ DA732X_REG_PLL_DIV_HI		, 0x00 },
+	{ DA732X_REG_PLL_CTRL		, 0x02 },
+	{ DA732X_REG_CLK_CTRL		, 0xaa },
+	{ DA732X_REG_CLK_DSP		, 0x07 },
+	{ DA732X_REG_CLK_EN1		, 0x00 },
+	{ DA732X_REG_CLK_EN2		, 0x00 },
+	{ DA732X_REG_CLK_EN3		, 0x00 },
+	{ DA732X_REG_CLK_EN4		, 0x00 },
+	{ DA732X_REG_CLK_EN5		, 0x00 },
+	{ DA732X_REG_AIF_MCLK		, 0x00 },
+	{ DA732X_REG_AIFA1		, 0x02 },
+	{ DA732X_REG_AIFA2		, 0x00 },
+	{ DA732X_REG_AIFA3		, 0x08 },
+	{ DA732X_REG_AIFB1		, 0x02 },
+	{ DA732X_REG_AIFB2		, 0x00 },
+	{ DA732X_REG_AIFB3		, 0x08 },
+	{ DA732X_REG_PC_CTRL		, 0xC0 },
+	{ DA732X_REG_DATA_ROUTE		, 0x00 },
+	{ DA732X_REG_DSP_CTRL		, 0x00 },
+	{ DA732X_REG_CIF_CTRL2		, 0x00 },
+	{ DA732X_REG_HANDSHAKE		, 0x00 },
+	{ DA732X_REG_SPARE1_OUT		, 0x00 },
+	{ DA732X_REG_SPARE2_OUT		, 0x00 },
+	{ DA732X_REG_SPARE1_IN		, 0x00 },
+	{ DA732X_REG_ADC1_PD		, 0x00 },
+	{ DA732X_REG_ADC1_HPF		, 0x00 },
+	{ DA732X_REG_ADC1_SEL		, 0x00 },
+	{ DA732X_REG_ADC1_EQ12		, 0x00 },
+	{ DA732X_REG_ADC1_EQ34		, 0x00 },
+	{ DA732X_REG_ADC1_EQ5		, 0x00 },
+	{ DA732X_REG_ADC2_PD		, 0x00 },
+	{ DA732X_REG_ADC2_HPF		, 0x00 },
+	{ DA732X_REG_ADC2_SEL		, 0x00 },
+	{ DA732X_REG_ADC2_EQ12		, 0x00 },
+	{ DA732X_REG_ADC2_EQ34		, 0x00 },
+	{ DA732X_REG_ADC2_EQ5		, 0x00 },
+	{ DA732X_REG_DAC1_HPF		, 0x00 },
+	{ DA732X_REG_DAC1_L_VOL		, 0x00 },
+	{ DA732X_REG_DAC1_R_VOL		, 0x00 },
+	{ DA732X_REG_DAC1_SEL		, 0x00 },
+	{ DA732X_REG_DAC1_SOFTMUTE	, 0x00 },
+	{ DA732X_REG_DAC1_EQ12		, 0x00 },
+	{ DA732X_REG_DAC1_EQ34		, 0x00 },
+	{ DA732X_REG_DAC1_EQ5		, 0x00 },
+	{ DA732X_REG_DAC2_HPF		, 0x00 },
+	{ DA732X_REG_DAC2_L_VOL		, 0x00 },
+	{ DA732X_REG_DAC2_R_VOL		, 0x00 },
+	{ DA732X_REG_DAC2_SEL		, 0x00 },
+	{ DA732X_REG_DAC2_SOFTMUTE	, 0x00 },
+	{ DA732X_REG_DAC2_EQ12		, 0x00 },
+	{ DA732X_REG_DAC2_EQ34		, 0x00 },
+	{ DA732X_REG_DAC2_EQ5		, 0x00 },
+	{ DA732X_REG_DAC3_HPF		, 0x00 },
+	{ DA732X_REG_DAC3_VOL		, 0x00 },
+	{ DA732X_REG_DAC3_SEL		, 0x00 },
+	{ DA732X_REG_DAC3_SOFTMUTE	, 0x00 },
+	{ DA732X_REG_DAC3_EQ12		, 0x00 },
+	{ DA732X_REG_DAC3_EQ34		, 0x00 },
+	{ DA732X_REG_DAC3_EQ5		, 0x00 },
+	{ DA732X_REG_BIQ_BYP		, 0x00 },
+	{ DA732X_REG_DMA_CMD		, 0x00 },
+	{ DA732X_REG_DMA_ADDR0		, 0x00 },
+	{ DA732X_REG_DMA_ADDR1		, 0x00 },
+	{ DA732X_REG_DMA_DATA0		, 0x00 },
+	{ DA732X_REG_DMA_DATA1		, 0x00 },
+	{ DA732X_REG_DMA_DATA2		, 0x00 },
+	{ DA732X_REG_DMA_DATA3		, 0x00 },
+	{ DA732X_REG_UNLOCK		, 0x00 },
+};
+
+static inline int da732x_get_input_div(struct snd_soc_codec *codec, int sysclk)
+{
+	int val;
+	int ret;
+
+	if (sysclk < DA732X_MCLK_10MHZ) {
+		val = DA732X_MCLK_RET_0_10MHZ;
+		ret = DA732X_MCLK_VAL_0_10MHZ;
+	} else if ((sysclk >= DA732X_MCLK_10MHZ) &&
+	    (sysclk < DA732X_MCLK_20MHZ)) {
+		val = DA732X_MCLK_RET_10_20MHZ;
+		ret = DA732X_MCLK_VAL_10_20MHZ;
+	} else if ((sysclk >= DA732X_MCLK_20MHZ) &&
+	    (sysclk < DA732X_MCLK_40MHZ)) {
+		val = DA732X_MCLK_RET_20_40MHZ;
+		ret = DA732X_MCLK_VAL_20_40MHZ;
+	} else if ((sysclk >= DA732X_MCLK_40MHZ) &&
+	    (sysclk <= DA732X_MCLK_54MHZ)) {
+		val = DA732X_MCLK_RET_40_54MHZ;
+		ret = DA732X_MCLK_VAL_40_54MHZ;
+	} else {
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, DA732X_REG_PLL_CTRL, val);
+
+	return ret;
+}
+
+static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state)
+{
+	switch (state) {
+	case DA732X_ENABLE_CP:
+		snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_EN);
+		snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_EN |
+			      DA732X_HP_CP_REG | DA732X_HP_CP_PULSESKIP);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA732X_CP_EN |
+			      DA732X_CP_CTRL_CPVDD1);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL2,
+			      DA732X_CP_MANAGE_MAGNITUDE | DA732X_CP_BOOST);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL3, DA732X_CP_1MHZ);
+		break;
+	case DA732X_DISABLE_CP:
+		snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_DIS);
+		snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_DIS);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS);
+		break;
+	default:
+		pr_err(KERN_ERR "Wrong charge pump state\n");
+		break;
+	}
+}
+
+static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, DA732X_MIC_PRE_VOL_DB_MIN,
+				  DA732X_MIC_PRE_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, DA732X_MIC_VOL_DB_MIN,
+				  DA732X_MIC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(aux_pga_tlv, DA732X_AUX_VOL_DB_MIN,
+				  DA732X_AUX_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(hp_pga_tlv, DA732X_HP_VOL_DB_MIN,
+				  DA732X_AUX_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin2_pga_tlv, DA732X_LIN2_VOL_DB_MIN,
+				  DA732X_LIN2_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin3_pga_tlv, DA732X_LIN3_VOL_DB_MIN,
+				  DA732X_LIN3_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin4_pga_tlv, DA732X_LIN4_VOL_DB_MIN,
+				  DA732X_LIN4_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(adc_pga_tlv, DA732X_ADC_VOL_DB_MIN,
+				  DA732X_ADC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(dac_pga_tlv, DA732X_DAC_VOL_DB_MIN,
+				  DA732X_DAC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_band_pga_tlv, DA732X_EQ_BAND_VOL_DB_MIN,
+				  DA732X_EQ_BAND_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_overall_tlv, DA732X_EQ_OVERALL_VOL_DB_MIN,
+				  DA732X_EQ_OVERALL_VOL_DB_INC, 0);
+
+/* High Pass Filter */
+static const char *da732x_hpf_mode[] = {
+	"Disable", "Music", "Voice",
+};
+
+static const char *da732x_hpf_music[] = {
+	"1.8Hz", "3.75Hz", "7.5Hz", "15Hz",
+};
+
+static const char *da732x_hpf_voice[] = {
+	"2.5Hz", "25Hz", "50Hz", "100Hz",
+	"150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da732x_dac1_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac2_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac3_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_adc1_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_adc2_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac1_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac2_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac3_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_adc1_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_adc2_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac1_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_dac2_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_dac3_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_adc1_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_adc2_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+
+static int da732x_hpf_set(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value;
+	unsigned int reg = enum_ctrl->reg;
+	unsigned int sel = ucontrol->value.integer.value[0];
+	unsigned int bits;
+
+	switch (sel) {
+	case DA732X_HPF_DISABLED:
+		bits = DA732X_HPF_DIS;
+		break;
+	case DA732X_HPF_VOICE:
+		bits = DA732X_HPF_VOICE_EN;
+		break;
+	case DA732X_HPF_MUSIC:
+		bits = DA732X_HPF_MUSIC_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, reg, DA732X_HPF_MASK, bits);
+
+	return 0;
+}
+
+static int da732x_hpf_get(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value;
+	unsigned int reg = enum_ctrl->reg;
+	int val;
+
+	val = snd_soc_read(codec, reg) & DA732X_HPF_MASK;
+
+	switch (val) {
+	case DA732X_HPF_VOICE_EN:
+		ucontrol->value.integer.value[0] = DA732X_HPF_VOICE;
+		break;
+	case DA732X_HPF_MUSIC_EN:
+		ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC;
+		break;
+	default:
+		ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED;
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new da732x_snd_controls[] = {
+	/* Input PGAs */
+	SOC_SINGLE_RANGE_TLV("MIC1 Boost Volume", DA732X_REG_MIC1_PRE,
+			     DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+			     DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+	SOC_SINGLE_RANGE_TLV("MIC2 Boost Volume", DA732X_REG_MIC2_PRE,
+			     DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+			     DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+	SOC_SINGLE_RANGE_TLV("MIC3 Boost Volume", DA732X_REG_MIC3_PRE,
+			     DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+			     DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+
+	/* MICs */
+	SOC_SINGLE("MIC1 Switch", DA732X_REG_MIC1, DA732X_MIC_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_RANGE_TLV("MIC1 Volume", DA732X_REG_MIC1,
+			     DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+			     DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+	SOC_SINGLE("MIC2 Switch", DA732X_REG_MIC2, DA732X_MIC_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_RANGE_TLV("MIC2 Volume", DA732X_REG_MIC2,
+			     DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+			     DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+	SOC_SINGLE("MIC3 Switch", DA732X_REG_MIC3, DA732X_MIC_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_RANGE_TLV("MIC3 Volume", DA732X_REG_MIC3,
+			     DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+			     DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+
+	/* AUXs */
+	SOC_SINGLE("AUX1L Switch", DA732X_REG_AUX1L, DA732X_AUX_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("AUX1L Volume", DA732X_REG_AUX1L,
+		       DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, aux_pga_tlv),
+	SOC_SINGLE("AUX1R Switch", DA732X_REG_AUX1R, DA732X_AUX_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("AUX1R Volume", DA732X_REG_AUX1R,
+		       DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, aux_pga_tlv),
+
+	/* ADCs */
+	SOC_DOUBLE_TLV("ADC1 Volume", DA732X_REG_ADC1_SEL,
+		       DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT,
+		       DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv),
+
+	SOC_DOUBLE_TLV("ADC2 Volume", DA732X_REG_ADC2_SEL,
+		       DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT,
+		       DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv),
+
+	/* DACs */
+	SOC_DOUBLE("Digital Playback DAC12 Switch", DA732X_REG_DAC1_SEL,
+		   DA732X_DACL_MUTE_SHIFT, DA732X_DACR_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_DOUBLE_R_TLV("Digital Playback DAC12 Volume", DA732X_REG_DAC1_L_VOL,
+			 DA732X_REG_DAC1_R_VOL, DA732X_DAC_VOL_SHIFT,
+			 DA732X_DAC_VOL_VAL_MAX, DA732X_INVERT, dac_pga_tlv),
+	SOC_SINGLE("Digital Playback DAC3 Switch", DA732X_REG_DAC2_SEL,
+		   DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Digital Playback DAC3 Volume", DA732X_REG_DAC2_L_VOL,
+			DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+			DA732X_INVERT, dac_pga_tlv),
+	SOC_SINGLE("Digital Playback DAC4 Switch", DA732X_REG_DAC2_SEL,
+		   DA732X_DACR_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Digital Playback DAC4 Volume", DA732X_REG_DAC2_R_VOL,
+		       DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+		       DA732X_INVERT, dac_pga_tlv),
+	SOC_SINGLE("Digital Playback DAC5 Switch", DA732X_REG_DAC3_SEL,
+		   DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Digital Playback DAC5 Volume", DA732X_REG_DAC3_VOL,
+		       DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+		       DA732X_INVERT, dac_pga_tlv),
+
+	/* High Pass Filters */
+	SOC_ENUM_EXT("DAC1 High Pass Filter Mode",
+		     da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("DAC1 High Pass Filter", da732x_dac1_hp_filter_enum),
+	SOC_ENUM("DAC1 Voice Filter", da732x_dac1_voice_filter_enum),
+
+	SOC_ENUM_EXT("DAC2 High Pass Filter Mode",
+		     da732x_dac2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("DAC2 High Pass Filter", da732x_dac2_hp_filter_enum),
+	SOC_ENUM("DAC2 Voice Filter", da732x_dac2_voice_filter_enum),
+
+	SOC_ENUM_EXT("DAC3 High Pass Filter Mode",
+		     da732x_dac3_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("DAC3 High Pass Filter", da732x_dac3_hp_filter_enum),
+	SOC_ENUM("DAC3 Filter Mode", da732x_dac3_voice_filter_enum),
+
+	SOC_ENUM_EXT("ADC1 High Pass Filter Mode",
+		     da732x_adc1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("ADC1 High Pass Filter", da732x_adc1_hp_filter_enum),
+	SOC_ENUM("ADC1 Voice Filter", da732x_adc1_voice_filter_enum),
+
+	SOC_ENUM_EXT("ADC2 High Pass Filter Mode",
+		     da732x_adc2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("ADC2 High Pass Filter", da732x_adc2_hp_filter_enum),
+	SOC_ENUM("ADC2 Voice Filter", da732x_adc2_voice_filter_enum),
+
+	/* Equalizers */
+	SOC_SINGLE("ADC1 EQ Switch", DA732X_REG_ADC1_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 2 Volume", DA732X_REG_ADC1_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 3 Volume", DA732X_REG_ADC1_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 4 Volume", DA732X_REG_ADC1_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 5 Volume", DA732X_REG_ADC1_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Overall Volume", DA732X_REG_ADC1_EQ5,
+		       DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_overall_tlv),
+
+	SOC_SINGLE("ADC2 EQ Switch", DA732X_REG_ADC2_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("ADC2 EQ Band 1 Volume", DA732X_REG_ADC2_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC2 EQ Band 2 Volume", DA732X_REG_ADC2_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC2 EQ Band 3 Volume", DA732X_REG_ADC2_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ACD2 EQ Band 4 Volume", DA732X_REG_ADC2_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ACD2 EQ Band 5 Volume", DA732X_REG_ADC2_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC2 EQ Overall Volume", DA732X_REG_ADC1_EQ5,
+		       DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_overall_tlv),
+
+	SOC_SINGLE("DAC1 EQ Switch", DA732X_REG_DAC1_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("DAC1 EQ Band 1 Volume", DA732X_REG_DAC1_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 2 Volume", DA732X_REG_DAC1_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 3 Volume", DA732X_REG_DAC1_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 4 Volume", DA732X_REG_DAC1_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 5 Volume", DA732X_REG_DAC1_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+
+	SOC_SINGLE("DAC2 EQ Switch", DA732X_REG_DAC2_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("DAC2 EQ Band 1 Volume", DA732X_REG_DAC2_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 2 Volume", DA732X_REG_DAC2_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 3 Volume", DA732X_REG_DAC2_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 4 Volume", DA732X_REG_DAC2_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 5 Volume", DA732X_REG_DAC2_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+
+	SOC_SINGLE("DAC3 EQ Switch", DA732X_REG_DAC3_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("DAC3 EQ Band 1 Volume", DA732X_REG_DAC3_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 2 Volume", DA732X_REG_DAC3_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 3 Volume", DA732X_REG_DAC3_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 4 Volume", DA732X_REG_DAC3_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 5 Volume", DA732X_REG_DAC3_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+
+	/* Lineout 2 Reciever*/
+	SOC_SINGLE("Lineout 2 Switch", DA732X_REG_LIN2, DA732X_LOUT_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Lineout 2 Volume", DA732X_REG_LIN2,
+		       DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, lin2_pga_tlv),
+
+	/* Lineout 3 SPEAKER*/
+	SOC_SINGLE("Lineout 3 Switch", DA732X_REG_LIN3, DA732X_LOUT_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Lineout 3 Volume", DA732X_REG_LIN3,
+		       DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, lin3_pga_tlv),
+
+	/* Lineout 4 */
+	SOC_SINGLE("Lineout 4 Switch", DA732X_REG_LIN4, DA732X_LOUT_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Lineout 4 Volume", DA732X_REG_LIN4,
+		       DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, lin4_pga_tlv),
+
+	/* Headphones */
+	SOC_DOUBLE_R("Headphone Switch", DA732X_REG_HPR, DA732X_REG_HPL,
+		     DA732X_HP_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_DOUBLE_R_TLV("Headphone Volume", DA732X_REG_HPL_VOL,
+			 DA732X_REG_HPR_VOL, DA732X_HP_VOL_SHIFT,
+			 DA732X_HP_VOL_VAL_MAX, DA732X_NO_INVERT, hp_pga_tlv),
+};
+
+static int da732x_adc_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		switch (w->reg) {
+		case DA732X_REG_ADC1_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCA_BB_CLK_EN,
+					    DA732X_ADCA_BB_CLK_EN);
+			break;
+		case DA732X_REG_ADC2_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCC_BB_CLK_EN,
+					    DA732X_ADCC_BB_CLK_EN);
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK,
+				    DA732X_ADC_SET_ACT);
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK,
+				    DA732X_ADC_ON);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK,
+				    DA732X_ADC_OFF);
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK,
+				    DA732X_ADC_SET_RST);
+
+		switch (w->reg) {
+		case DA732X_REG_ADC1_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCA_BB_CLK_EN, 0);
+			break;
+		case DA732X_REG_ADC2_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCC_BB_CLK_EN, 0);
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int da732x_out_pga_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_update_bits(codec, w->reg,
+				    (1 << w->shift) | DA732X_OUT_HIZ_EN,
+				    (1 << w->shift) | DA732X_OUT_HIZ_EN);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, w->reg,
+				    (1 << w->shift) | DA732X_OUT_HIZ_EN,
+				    (1 << w->shift) | DA732X_OUT_HIZ_DIS);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const char *adcl_text[] = {
+	"AUX1L", "MIC1"
+};
+
+static const char *adcr_text[] = {
+	"AUX1R", "MIC2", "MIC3"
+};
+
+static const char *enable_text[] = {
+	"Disabled",
+	"Enabled"
+};
+
+/* ADC1LMUX */
+static const struct soc_enum adc1l_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
+			DA732X_ADCL_MUX_MAX, adcl_text);
+static const struct snd_kcontrol_new adc1l_mux =
+	SOC_DAPM_ENUM("ADC Route", adc1l_enum);
+
+/* ADC1RMUX */
+static const struct soc_enum adc1r_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
+			DA732X_ADCR_MUX_MAX, adcr_text);
+static const struct snd_kcontrol_new adc1r_mux =
+	SOC_DAPM_ENUM("ADC Route", adc1r_enum);
+
+/* ADC2LMUX */
+static const struct soc_enum adc2l_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
+			DA732X_ADCL_MUX_MAX, adcl_text);
+static const struct snd_kcontrol_new adc2l_mux =
+	SOC_DAPM_ENUM("ADC Route", adc2l_enum);
+
+/* ADC2RMUX */
+static const struct soc_enum adc2r_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
+			DA732X_ADCR_MUX_MAX, adcr_text);
+
+static const struct snd_kcontrol_new adc2r_mux =
+	SOC_DAPM_ENUM("ADC Route", adc2r_enum);
+
+static const struct soc_enum da732x_hp_left_output =
+	SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new hpl_mux =
+	SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output);
+
+static const struct soc_enum da732x_hp_right_output =
+	SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new hpr_mux =
+	SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output);
+
+static const struct soc_enum da732x_speaker_output =
+	SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new spk_mux =
+	SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output);
+
+static const struct soc_enum da732x_lout4_output =
+	SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new lout4_mux =
+	SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output);
+
+static const struct soc_enum da732x_lout2_output =
+	SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new lout2_mux =
+	SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output);
+
+static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = {
+	/* Supplies */
+	SND_SOC_DAPM_SUPPLY("ADC1 Supply", DA732X_REG_ADC1_PD, 0,
+			    DA732X_NO_INVERT, da732x_adc_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("ADC2 Supply", DA732X_REG_ADC2_PD, 0,
+			    DA732X_NO_INVERT, da732x_adc_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("DAC1 CLK", DA732X_REG_CLK_EN4,
+			    DA732X_DACA_BB_CLK_SHIFT, DA732X_NO_INVERT,
+			    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC2 CLK", DA732X_REG_CLK_EN4,
+			    DA732X_DACC_BB_CLK_SHIFT, DA732X_NO_INVERT,
+			    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC3 CLK", DA732X_REG_CLK_EN5,
+			    DA732X_DACE_BB_CLK_SHIFT, DA732X_NO_INVERT,
+			    NULL, 0),
+
+	/* Micbias */
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", DA732X_REG_MICBIAS1,
+			    DA732X_MICBIAS_EN_SHIFT,
+			    DA732X_NO_INVERT, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS2", DA732X_REG_MICBIAS2,
+			    DA732X_MICBIAS_EN_SHIFT,
+			    DA732X_NO_INVERT, NULL, 0),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_INPUT("MIC2"),
+	SND_SOC_DAPM_INPUT("MIC3"),
+	SND_SOC_DAPM_INPUT("AUX1L"),
+	SND_SOC_DAPM_INPUT("AUX1R"),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("LOUTL"),
+	SND_SOC_DAPM_OUTPUT("LOUTR"),
+	SND_SOC_DAPM_OUTPUT("ClassD"),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC1L", NULL, DA732X_REG_ADC1_SEL,
+			 DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_ADC("ADC1R", NULL, DA732X_REG_ADC1_SEL,
+			 DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_ADC("ADC2L", NULL, DA732X_REG_ADC2_SEL,
+			 DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_ADC("ADC2R", NULL, DA732X_REG_ADC2_SEL,
+			 DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC1L", NULL, DA732X_REG_DAC1_SEL,
+			 DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC1R", NULL, DA732X_REG_DAC1_SEL,
+			 DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC2L", NULL, DA732X_REG_DAC2_SEL,
+			 DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC2R", NULL, DA732X_REG_DAC2_SEL,
+			 DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC3", NULL, DA732X_REG_DAC3_SEL,
+			 DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+
+	/* Input Pgas */
+	SND_SOC_DAPM_PGA("MIC1 PGA", DA732X_REG_MIC1, DA732X_MIC_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC2 PGA", DA732X_REG_MIC2, DA732X_MIC_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC3 PGA", DA732X_REG_MIC3, DA732X_MIC_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX1L PGA", DA732X_REG_AUX1L, DA732X_AUX_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX1R PGA", DA732X_REG_AUX1R, DA732X_AUX_EN_SHIFT,
+			 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA_E("HP Left", DA732X_REG_HPL, DA732X_HP_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("HP Right", DA732X_REG_HPR, DA732X_HP_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("LIN2", DA732X_REG_LIN2, DA732X_LIN_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("LIN3", DA732X_REG_LIN3, DA732X_LIN_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("LIN4", DA732X_REG_LIN4, DA732X_LIN_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* MUXs */
+	SND_SOC_DAPM_MUX("ADC1 Left MUX", SND_SOC_NOPM, 0, 0, &adc1l_mux),
+	SND_SOC_DAPM_MUX("ADC1 Right MUX", SND_SOC_NOPM, 0, 0, &adc1r_mux),
+	SND_SOC_DAPM_MUX("ADC2 Left MUX", SND_SOC_NOPM, 0, 0, &adc2l_mux),
+	SND_SOC_DAPM_MUX("ADC2 Right MUX", SND_SOC_NOPM, 0, 0, &adc2r_mux),
+
+	SND_SOC_DAPM_MUX("HP Left MUX", SND_SOC_NOPM, 0, 0, &hpl_mux),
+	SND_SOC_DAPM_MUX("HP Right MUX", SND_SOC_NOPM, 0, 0, &hpr_mux),
+	SND_SOC_DAPM_MUX("Speaker MUX", SND_SOC_NOPM, 0, 0, &spk_mux),
+	SND_SOC_DAPM_MUX("LOUT2 MUX", SND_SOC_NOPM, 0, 0, &lout2_mux),
+	SND_SOC_DAPM_MUX("LOUT4 MUX", SND_SOC_NOPM, 0, 0, &lout4_mux),
+
+	/* AIF interfaces */
+	SND_SOC_DAPM_AIF_OUT("AIFA Output", "AIFA Capture", 0, DA732X_REG_AIFA3,
+			     DA732X_AIF_EN_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("AIFA Input", "AIFA Playback", 0, DA732X_REG_AIFA3,
+			    DA732X_AIF_EN_SHIFT, 0),
+
+	SND_SOC_DAPM_AIF_OUT("AIFB Output", "AIFB Capture", 0, DA732X_REG_AIFB3,
+			     DA732X_AIF_EN_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("AIFB Input", "AIFB Playback", 0, DA732X_REG_AIFB3,
+			    DA732X_AIF_EN_SHIFT, 0),
+};
+
+static const struct snd_soc_dapm_route da732x_dapm_routes[] = {
+	/* Inputs */
+	{"AUX1L PGA", "NULL", "AUX1L"},
+	{"AUX1R PGA", "NULL", "AUX1R"},
+	{"MIC1 PGA", NULL, "MIC1"},
+	{"MIC2 PGA", "NULL", "MIC2"},
+	{"MIC3 PGA", "NULL", "MIC3"},
+
+	/* Capture Path */
+	{"ADC1 Left MUX", "MIC1", "MIC1 PGA"},
+	{"ADC1 Left MUX", "AUX1L", "AUX1L PGA"},
+
+	{"ADC1 Right MUX", "AUX1R", "AUX1R PGA"},
+	{"ADC1 Right MUX", "MIC2", "MIC2 PGA"},
+	{"ADC1 Right MUX", "MIC3", "MIC3 PGA"},
+
+	{"ADC2 Left MUX", "AUX1L", "AUX1L PGA"},
+	{"ADC2 Left MUX", "MIC1", "MIC1 PGA"},
+
+	{"ADC2 Right MUX", "AUX1R", "AUX1R PGA"},
+	{"ADC2 Right MUX", "MIC2", "MIC2 PGA"},
+	{"ADC2 Right MUX", "MIC3", "MIC3 PGA"},
+
+	{"ADC1L", NULL, "ADC1 Supply"},
+	{"ADC1R", NULL, "ADC1 Supply"},
+	{"ADC2L", NULL, "ADC2 Supply"},
+	{"ADC2R", NULL, "ADC2 Supply"},
+
+	{"ADC1L", NULL, "ADC1 Left MUX"},
+	{"ADC1R", NULL, "ADC1 Right MUX"},
+	{"ADC2L", NULL, "ADC2 Left MUX"},
+	{"ADC2R", NULL, "ADC2 Right MUX"},
+
+	{"AIFA Output", NULL, "ADC1L"},
+	{"AIFA Output", NULL, "ADC1R"},
+	{"AIFB Output", NULL, "ADC2L"},
+	{"AIFB Output", NULL, "ADC2R"},
+
+	{"HP Left MUX", "Enabled", "AIFA Input"},
+	{"HP Right MUX", "Enabled", "AIFA Input"},
+	{"Speaker MUX", "Enabled", "AIFB Input"},
+	{"LOUT2 MUX", "Enabled", "AIFB Input"},
+	{"LOUT4 MUX", "Enabled", "AIFB Input"},
+
+	{"DAC1L", NULL, "DAC1 CLK"},
+	{"DAC1R", NULL, "DAC1 CLK"},
+	{"DAC2L", NULL, "DAC2 CLK"},
+	{"DAC2R", NULL, "DAC2 CLK"},
+	{"DAC3", NULL, "DAC3 CLK"},
+
+	{"DAC1L", NULL, "HP Left MUX"},
+	{"DAC1R", NULL, "HP Right MUX"},
+	{"DAC2L", NULL, "Speaker MUX"},
+	{"DAC2R", NULL, "LOUT4 MUX"},
+	{"DAC3", NULL, "LOUT2 MUX"},
+
+	/* Output Pgas */
+	{"HP Left", NULL, "DAC1L"},
+	{"HP Right", NULL, "DAC1R"},
+	{"LIN3", NULL, "DAC2L"},
+	{"LIN4", NULL, "DAC2R"},
+	{"LIN2", NULL, "DAC3"},
+
+	/* Outputs */
+	{"ClassD", NULL, "LIN3"},
+	{"LOUTL", NULL, "LIN2"},
+	{"LOUTR", NULL, "LIN4"},
+	{"HPL", NULL, "HP Left"},
+	{"HPR", NULL, "HP Right"},
+};
+
+static int da732x_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u32 aif = 0;
+	u32 reg_aif;
+	u32 fs;
+
+	reg_aif = dai->driver->base;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		aif |= DA732X_AIF_WORD_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		aif |= DA732X_AIF_WORD_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		aif |= DA732X_AIF_WORD_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		aif |= DA732X_AIF_WORD_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 8000:
+		fs = DA732X_SR_8KHZ;
+		break;
+	case 11025:
+		fs = DA732X_SR_11_025KHZ;
+		break;
+	case 12000:
+		fs = DA732X_SR_12KHZ;
+		break;
+	case 16000:
+		fs = DA732X_SR_16KHZ;
+		break;
+	case 22050:
+		fs = DA732X_SR_22_05KHZ;
+		break;
+	case 24000:
+		fs = DA732X_SR_24KHZ;
+		break;
+	case 32000:
+		fs = DA732X_SR_32KHZ;
+		break;
+	case 44100:
+		fs = DA732X_SR_44_1KHZ;
+		break;
+	case 48000:
+		fs = DA732X_SR_48KHZ;
+		break;
+	case 88100:
+		fs = DA732X_SR_88_1KHZ;
+		break;
+	case 96000:
+		fs = DA732X_SR_96KHZ;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, reg_aif, DA732X_AIF_WORD_MASK, aif);
+	snd_soc_update_bits(codec, DA732X_REG_CLK_CTRL, DA732X_SR1_MASK, fs);
+
+	return 0;
+}
+
+static int da732x_set_dai_fmt(struct snd_soc_dai *dai, u32 fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u32 aif_mclk, pc_count;
+	u32 reg_aif1, aif1;
+	u32 reg_aif3, aif3;
+
+	switch (dai->id) {
+	case DA732X_DAI_ID1:
+		reg_aif1 = DA732X_REG_AIFA1;
+		reg_aif3 = DA732X_REG_AIFA3;
+		pc_count = DA732X_PC_PULSE_AIFA | DA732X_PC_RESYNC_NOT_AUT |
+			   DA732X_PC_SAME;
+		break;
+	case DA732X_DAI_ID2:
+		reg_aif1 = DA732X_REG_AIFB1;
+		reg_aif3 = DA732X_REG_AIFB3;
+		pc_count = DA732X_PC_PULSE_AIFB | DA732X_PC_RESYNC_NOT_AUT |
+			   DA732X_PC_SAME;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aif1 = DA732X_AIF_SLAVE;
+		aif_mclk = DA732X_AIFM_FRAME_64 | DA732X_AIFM_SRC_SEL_AIFA;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aif1 = DA732X_AIF_CLK_FROM_SRC;
+		aif_mclk = DA732X_CLK_GENERATION_AIF_A;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		aif3 = DA732X_AIF_I2S_MODE;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		aif3 = DA732X_AIF_RIGHT_J_MODE;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		aif3 = DA732X_AIF_LEFT_J_MODE;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		aif3 = DA732X_AIF_DSP_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_B:
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			aif3 |= DA732X_AIF_BCLK_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			aif3 |= DA732X_AIF_BCLK_INV | DA732X_AIF_WCLK_INV;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			aif3 |= DA732X_AIF_BCLK_INV;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			aif3 |= DA732X_AIF_WCLK_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, DA732X_REG_AIF_MCLK, aif_mclk);
+	snd_soc_update_bits(codec, reg_aif1, DA732X_AIF1_CLK_MASK, aif1);
+	snd_soc_update_bits(codec, reg_aif3, DA732X_AIF_BCLK_INV |
+			    DA732X_AIF_WCLK_INV | DA732X_AIF_MODE_MASK, aif3);
+	snd_soc_write(codec, DA732X_REG_PC_CTRL, pc_count);
+
+	return 0;
+}
+
+
+
+static int da732x_set_dai_pll(struct snd_soc_codec *codec, int pll_id,
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out)
+{
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+	int fref, indiv;
+	u8 div_lo, div_mid, div_hi;
+	u64 frac_div;
+
+	/* Disable PLL */
+	if (freq_out == 0) {
+		snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL,
+				    DA732X_PLL_EN, 0);
+		da732x->pll_en = false;
+		return 0;
+	}
+
+	if (da732x->pll_en)
+		return -EBUSY;
+
+	if (source == DA732X_SRCCLK_MCLK) {
+		/* Validate Sysclk rate */
+		switch (da732x->sysclk) {
+		case 11290000:
+		case 12288000:
+		case 22580000:
+		case 24576000:
+		case 45160000:
+		case 49152000:
+			snd_soc_write(codec, DA732X_REG_PLL_CTRL,
+				      DA732X_PLL_BYPASS);
+			return 0;
+		default:
+			dev_err(codec->dev,
+				"Cannot use PLL Bypass, invalid SYSCLK rate\n");
+			return -EINVAL;
+		}
+	}
+
+	indiv = da732x_get_input_div(codec, da732x->sysclk);
+	if (indiv < 0)
+		return indiv;
+
+	fref = (da732x->sysclk / indiv);
+	div_hi = freq_out / fref;
+	frac_div = (u64)(freq_out % fref) * 8192ULL;
+	do_div(frac_div, fref);
+	div_mid = (frac_div >> DA732X_1BYTE_SHIFT) & DA732X_U8_MASK;
+	div_lo = (frac_div) & DA732X_U8_MASK;
+
+	snd_soc_write(codec, DA732X_REG_PLL_DIV_LO, div_lo);
+	snd_soc_write(codec, DA732X_REG_PLL_DIV_MID, div_mid);
+	snd_soc_write(codec, DA732X_REG_PLL_DIV_HI, div_hi);
+
+	snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, DA732X_PLL_EN,
+			    DA732X_PLL_EN);
+
+	da732x->pll_en = true;
+
+	return 0;
+}
+
+static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				 unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+
+	da732x->sysclk = freq;
+
+	return 0;
+}
+
+#define DA732X_RATES	SNDRV_PCM_RATE_8000_96000
+
+#define	DA732X_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 da732x_dai1_ops = {
+	.hw_params	= da732x_hw_params,
+	.set_fmt	= da732x_set_dai_fmt,
+	.set_sysclk	= da732x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops da732x_dai2_ops = {
+	.hw_params	= da732x_hw_params,
+	.set_fmt	= da732x_set_dai_fmt,
+	.set_sysclk	= da732x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver da732x_dai[] = {
+	{
+		.name	= "DA732X_AIFA",
+		.id	= DA732X_DAI_ID1,
+		.base	= DA732X_REG_AIFA1,
+		.playback = {
+			.stream_name = "AIFA Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIFA Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.ops = &da732x_dai1_ops,
+	},
+	{
+		.name	= "DA732X_AIFB",
+		.id	= DA732X_DAI_ID2,
+		.base	= DA732X_REG_AIFB1,
+		.playback = {
+			.stream_name = "AIFB Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIFB Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.ops = &da732x_dai2_ops,
+	},
+};
+
+static const struct regmap_config da732x_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+
+	.max_register		= DA732X_MAX_REG,
+	.reg_defaults		= da732x_reg_cache,
+	.num_reg_defaults	= ARRAY_SIZE(da732x_reg_cache),
+	.cache_type		= REGCACHE_RBTREE,
+};
+
+
+static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
+{
+	u8 offset[DA732X_HP_DACS];
+	u8 sign[DA732X_HP_DACS];
+	u8 step = DA732X_DAC_OFFSET_STEP;
+
+	/* Initialize DAC offset calibration circuits and registers */
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+		      DA732X_HP_DAC_OFFSET_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+		      DA732X_HP_DAC_OFFSET_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL,
+		      DA732X_HP_DAC_OFF_CALIBRATION |
+		      DA732X_HP_DAC_OFF_SCALE_STEPS);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL,
+		      DA732X_HP_DAC_OFF_CALIBRATION |
+		      DA732X_HP_DAC_OFF_SCALE_STEPS);
+
+	/* Wait for voltage stabilization */
+	msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+	/* Check DAC offset sign */
+	sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+				DA732X_HP_DAC_OFF_CNTL_COMPO);
+	sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+				DA732X_HP_DAC_OFF_CNTL_COMPO);
+
+	/* Binary search DAC offset values (both channels at once) */
+	offset[DA732X_HPL_DAC] = sign[DA732X_HPL_DAC] << DA732X_HP_DAC_COMPO_SHIFT;
+	offset[DA732X_HPR_DAC] = sign[DA732X_HPR_DAC] << DA732X_HP_DAC_COMPO_SHIFT;
+
+	do {
+		offset[DA732X_HPL_DAC] |= step;
+		offset[DA732X_HPR_DAC] |= step;
+		snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+			      ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK);
+		snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+			      ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK);
+
+		msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+		if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+		     DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC])
+			offset[DA732X_HPL_DAC] &= ~step;
+		if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+		     DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC])
+			offset[DA732X_HPR_DAC] &= ~step;
+
+		step >>= 1;
+	} while (step);
+
+	/* Write final DAC offsets to registers */
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+		      ~offset[DA732X_HPL_DAC] &	DA732X_HP_DAC_OFF_MASK);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+		      ~offset[DA732X_HPR_DAC] &	DA732X_HP_DAC_OFF_MASK);
+
+	/* End DAC calibration mode */
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL,
+		DA732X_HP_DAC_OFF_SCALE_STEPS);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL,
+		DA732X_HP_DAC_OFF_SCALE_STEPS);
+}
+
+static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
+{
+	u8 offset[DA732X_HP_AMPS];
+	u8 sign[DA732X_HP_AMPS];
+	u8 step = DA732X_OUTPUT_OFFSET_STEP;
+
+	offset[DA732X_HPL_AMP] = DA732X_HP_OUT_TRIM_VAL;
+	offset[DA732X_HPR_AMP] = DA732X_HP_OUT_TRIM_VAL;
+
+	/* Initialize output offset calibration circuits and registers  */
+	snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPL,
+		      DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN);
+	snd_soc_write(codec, DA732X_REG_HPR,
+		      DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN);
+
+	/* Wait for voltage stabilization */
+	msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+	/* Check output offset sign */
+	sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) &
+			       DA732X_HP_OUT_COMPO;
+	sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) &
+			       DA732X_HP_OUT_COMPO;
+
+	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP |
+		      (sign[DA732X_HPL_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) |
+		      DA732X_HP_OUT_EN);
+	snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_COMP |
+		      (sign[DA732X_HPR_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) |
+		      DA732X_HP_OUT_EN);
+
+	/* Binary search output offset values (both channels at once) */
+	do {
+		offset[DA732X_HPL_AMP] |= step;
+		offset[DA732X_HPR_AMP] |= step;
+		snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET,
+			      offset[DA732X_HPL_AMP]);
+		snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET,
+			      offset[DA732X_HPR_AMP]);
+
+		msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+		if ((codec->hw_read(codec, DA732X_REG_HPL) &
+		     DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP])
+			offset[DA732X_HPL_AMP] &= ~step;
+		if ((codec->hw_read(codec, DA732X_REG_HPR) &
+		     DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP])
+			offset[DA732X_HPR_AMP] &= ~step;
+
+		step >>= 1;
+	} while (step);
+
+	/* Write final DAC offsets to registers */
+	snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, offset[DA732X_HPL_AMP]);
+	snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, offset[DA732X_HPR_AMP]);
+}
+
+static void da732x_hp_dc_offset_cancellation(struct snd_soc_codec *codec)
+{
+	/* Make sure that we have Soft Mute enabled */
+	snd_soc_write(codec, DA732X_REG_DAC1_SOFTMUTE, DA732X_SOFTMUTE_EN |
+		      DA732X_GAIN_RAMPED | DA732X_16_SAMPLES);
+	snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACL_EN |
+		      DA732X_DACR_EN | DA732X_DACL_SDM | DA732X_DACR_SDM |
+		      DA732X_DACL_MUTE | DA732X_DACR_MUTE);
+	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN |
+		      DA732X_HP_OUT_MUTE | DA732X_HP_OUT_EN);
+	snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_EN |
+		      DA732X_HP_OUT_MUTE | DA732X_HP_OUT_DAC_EN);
+
+	da732x_dac_offset_adjust(codec);
+	da732x_output_offset_adjust(codec);
+
+	snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACS_DIS);
+	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_DIS);
+	snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_DIS);
+}
+
+static int da732x_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
+				    DA732X_BIAS_BOOST_MASK,
+				    DA732X_BIAS_BOOST_100PC);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			/* Init Codec */
+			snd_soc_write(codec, DA732X_REG_REF1,
+				      DA732X_VMID_FASTCHG);
+			snd_soc_write(codec, DA732X_REG_BIAS_EN,
+				      DA732X_BIAS_EN);
+
+			mdelay(DA732X_STARTUP_DELAY);
+
+			/* Disable Fast Charge and enable DAC ref voltage */
+			snd_soc_write(codec, DA732X_REG_REF1,
+				      DA732X_REFBUFX2_EN);
+
+			/* Enable bypass DSP routing */
+			snd_soc_write(codec, DA732X_REG_DATA_ROUTE,
+				      DA732X_BYPASS_DSP);
+
+			/* Enable Digital subsystem */
+			snd_soc_write(codec, DA732X_REG_DSP_CTRL,
+				      DA732X_DIGITAL_EN);
+
+			snd_soc_write(codec, DA732X_REG_SPARE1_OUT,
+				      DA732X_HP_DRIVER_EN |
+				      DA732X_HP_GATE_LOW |
+				      DA732X_HP_LOOP_GAIN_CTRL);
+			snd_soc_write(codec, DA732X_REG_HP_LIN1_GNDSEL,
+				      DA732X_HP_OUT_GNDSEL);
+
+			da732x_set_charge_pump(codec, DA732X_ENABLE_CP);
+
+			snd_soc_write(codec, DA732X_REG_CLK_EN1,
+			      DA732X_SYS3_CLK_EN | DA732X_PC_CLK_EN);
+
+			/* Enable Zero Crossing */
+			snd_soc_write(codec, DA732X_REG_INP_ZC_EN,
+				      DA732X_MIC1_PRE_ZC_EN |
+				      DA732X_MIC1_ZC_EN |
+				      DA732X_MIC2_PRE_ZC_EN |
+				      DA732X_MIC2_ZC_EN |
+				      DA732X_AUXL_ZC_EN |
+				      DA732X_AUXR_ZC_EN |
+				      DA732X_MIC3_PRE_ZC_EN |
+				      DA732X_MIC3_ZC_EN);
+			snd_soc_write(codec, DA732X_REG_OUT_ZC_EN,
+				      DA732X_HPL_ZC_EN | DA732X_HPR_ZC_EN |
+				      DA732X_LIN2_ZC_EN | DA732X_LIN3_ZC_EN |
+				      DA732X_LIN4_ZC_EN);
+
+			da732x_hp_dc_offset_cancellation(codec);
+
+			regcache_cache_only(codec->control_data, false);
+			regcache_sync(codec->control_data);
+		} else {
+			snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
+					    DA732X_BIAS_BOOST_MASK,
+					    DA732X_BIAS_BOOST_50PC);
+			snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL,
+					    DA732X_PLL_EN, 0);
+			da732x->pll_en = false;
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		regcache_cache_only(codec->control_data, true);
+		da732x_set_charge_pump(codec, DA732X_DISABLE_CP);
+		snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN,
+				    DA732X_BIAS_DIS);
+		da732x->pll_en = false;
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static int da732x_probe(struct snd_soc_codec *codec)
+{
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret = 0;
+
+	da732x->codec = codec;
+
+	dapm->idle_bias_off = false;
+
+	codec->control_data = da732x->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec.\n");
+		goto err;
+	}
+
+	da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+err:
+	return ret;
+}
+
+static int da732x_remove(struct snd_soc_codec *codec)
+{
+
+	da732x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+struct snd_soc_codec_driver soc_codec_dev_da732x = {
+	.probe			= da732x_probe,
+	.remove			= da732x_remove,
+	.set_bias_level		= da732x_set_bias_level,
+	.controls		= da732x_snd_controls,
+	.num_controls		= ARRAY_SIZE(da732x_snd_controls),
+	.dapm_widgets		= da732x_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(da732x_dapm_widgets),
+	.dapm_routes		= da732x_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(da732x_dapm_routes),
+	.set_pll		= da732x_set_dai_pll,
+	.reg_cache_size		= ARRAY_SIZE(da732x_reg_cache),
+};
+
+static __devinit int da732x_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct da732x_priv *da732x;
+	unsigned int reg;
+	int ret;
+
+	da732x = devm_kzalloc(&i2c->dev, sizeof(struct da732x_priv),
+			      GFP_KERNEL);
+	if (!da732x)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, da732x);
+
+	da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap);
+	if (IS_ERR(da732x->regmap)) {
+		ret = PTR_ERR(da732x->regmap);
+		dev_err(&i2c->dev, "Failed to initialize regmap\n");
+		goto err;
+	}
+
+	ret = regmap_read(da732x->regmap, DA732X_REG_ID, &reg);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
+		goto err;
+	}
+
+	dev_info(&i2c->dev, "Revision: %d.%d\n",
+		 (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK));
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x,
+				     da732x_dai, ARRAY_SIZE(da732x_dai));
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to register codec.\n");
+
+err:
+	return ret;
+}
+
+static __devexit int da732x_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+
+	return 0;
+}
+
+static const struct i2c_device_id da732x_i2c_id[] = {
+	{ "da7320", 0},
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, da732x_i2c_id);
+
+static struct i2c_driver da732x_i2c_driver = {
+	.driver		= {
+		.name	= "da7320",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= da732x_i2c_probe,
+	.remove		= __devexit_p(da732x_i2c_remove),
+	.id_table	= da732x_i2c_id,
+};
+
+module_i2c_driver(da732x_i2c_driver);
+
+
+MODULE_DESCRIPTION("ASoC DA732X driver");
+MODULE_AUTHOR("Michal Hajduk <michal.hajduk@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
new file mode 100644
index 000000000000..c8ce5475de22
--- /dev/null
+++ b/sound/soc/codecs/da732x.h
@@ -0,0 +1,133 @@
+/*
+ * da732x.h -- Dialog DA732X ALSA SoC Audio Driver Header File
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __DA732X_H_
+#define __DA732X_H
+
+#include <sound/soc.h>
+
+/* General */
+#define	DA732X_U8_MASK			0xFF
+#define	DA732X_4BYTES			4
+#define	DA732X_3BYTES			3
+#define	DA732X_2BYTES			2
+#define	DA732X_1BYTE			1
+#define	DA732X_1BYTE_SHIFT		8
+#define	DA732X_2BYTES_SHIFT		16
+#define	DA732X_3BYTES_SHIFT		24
+#define	DA732X_4BYTES_SHIFT		32
+
+#define	DA732X_DACS_DIS			0x0
+#define	DA732X_HP_DIS			0x0
+#define	DA732X_CLEAR_REG		0x0
+
+/* Calibration */
+#define	DA732X_DAC_OFFSET_STEP		0x20
+#define	DA732X_OUTPUT_OFFSET_STEP	0x80
+#define	DA732X_HP_OUT_TRIM_VAL		0x0
+#define	DA732X_WAIT_FOR_STABILIZATION	1
+#define	DA732X_HPL_DAC			0
+#define	DA732X_HPR_DAC			1
+#define	DA732X_HP_DACS			2
+#define	DA732X_HPL_AMP			0
+#define	DA732X_HPR_AMP			1
+#define	DA732X_HP_AMPS			2
+
+/* Clock settings */
+#define DA732X_STARTUP_DELAY		100
+#define	DA732X_PLL_OUT_196608		196608000
+#define	DA732X_PLL_OUT_180634		180633600
+#define	DA732X_PLL_OUT_SRM		188620800
+#define	DA732X_MCLK_10MHZ		10000000
+#define	DA732X_MCLK_20MHZ		20000000
+#define	DA732X_MCLK_40MHZ		40000000
+#define	DA732X_MCLK_54MHZ		54000000
+#define	DA732X_MCLK_RET_0_10MHZ		0
+#define	DA732X_MCLK_VAL_0_10MHZ		1
+#define	DA732X_MCLK_RET_10_20MHZ	1
+#define	DA732X_MCLK_VAL_10_20MHZ	2
+#define	DA732X_MCLK_RET_20_40MHZ	2
+#define	DA732X_MCLK_VAL_20_40MHZ	4
+#define	DA732X_MCLK_RET_40_54MHZ	3
+#define	DA732X_MCLK_VAL_40_54MHZ	8
+#define	DA732X_DAI_ID1			0
+#define	DA732X_DAI_ID2			1
+#define	DA732X_SRCCLK_PLL		0
+#define	DA732X_SRCCLK_MCLK		1
+
+#define	DA732X_LIN_LP_VOL		0x4F
+#define	DA732X_LP_VOL			0x40
+
+/* Kcontrols */
+#define	DA732X_DAC_EN_MAX		2
+#define	DA732X_ADCL_MUX_MAX		2
+#define	DA732X_ADCR_MUX_MAX		3
+#define	DA732X_HPF_MODE_MAX		3
+#define	DA732X_HPF_MODE_SHIFT		4
+#define	DA732X_HPF_MUSIC_SHIFT		0
+#define	DA732X_HPF_MUSIC_MAX		4
+#define	DA732X_HPF_VOICE_SHIFT		4
+#define	DA732X_HPF_VOICE_MAX		8
+#define	DA732X_EQ_EN_MAX		1
+#define	DA732X_HPF_VOICE		1
+#define	DA732X_HPF_MUSIC		2
+#define	DA732X_HPF_DISABLED		0
+#define	DA732X_NO_INVERT		0
+#define	DA732X_INVERT			1
+#define	DA732X_SWITCH_MAX		1
+#define	DA732X_ENABLE_CP		1
+#define	DA732X_DISABLE_CP		0
+#define	DA732X_DISABLE_ALL_CLKS		0
+#define	DA732X_RESET_ADCS		0
+
+/* dB values */
+#define DA732X_MIC_VOL_DB_MIN		0
+#define DA732X_MIC_VOL_DB_INC		50
+#define DA732X_MIC_PRE_VOL_DB_MIN	0
+#define DA732X_MIC_PRE_VOL_DB_INC	600
+#define DA732X_AUX_VOL_DB_MIN		-6000
+#define DA732X_AUX_VOL_DB_INC		150
+#define DA732X_HP_VOL_DB_MIN		-2250
+#define DA732X_HP_VOL_DB_INC		150
+#define	DA732X_LIN2_VOL_DB_MIN		-1650
+#define	DA732X_LIN2_VOL_DB_INC		150
+#define	DA732X_LIN3_VOL_DB_MIN		-1650
+#define DA732X_LIN3_VOL_DB_INC		150
+#define	DA732X_LIN4_VOL_DB_MIN		-2250
+#define DA732X_LIN4_VOL_DB_INC		150
+#define	DA732X_EQ_BAND_VOL_DB_MIN	-1050
+#define	DA732X_EQ_BAND_VOL_DB_INC	150
+#define DA732X_DAC_VOL_DB_MIN		-7725
+#define DA732X_DAC_VOL_DB_INC		75
+#define DA732X_ADC_VOL_DB_MIN		0
+#define DA732X_ADC_VOL_DB_INC		-1
+#define	DA732X_EQ_OVERALL_VOL_DB_MIN	-1800
+#define	DA732X_EQ_OVERALL_VOL_DB_INC	600
+
+#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \
+	{.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext}
+
+enum da732x_sysctl {
+	DA732X_SR_8KHZ		= 0x1,
+	DA732X_SR_11_025KHZ	= 0x2,
+	DA732X_SR_12KHZ		= 0x3,
+	DA732X_SR_16KHZ		= 0x5,
+	DA732X_SR_22_05KHZ	= 0x6,
+	DA732X_SR_24KHZ		= 0x7,
+	DA732X_SR_32KHZ		= 0x9,
+	DA732X_SR_44_1KHZ	= 0xA,
+	DA732X_SR_48KHZ		= 0xB,
+	DA732X_SR_88_1KHZ	= 0xE,
+	DA732X_SR_96KHZ		= 0xF,
+};
+
+#endif /* __DA732X_H_ */
diff --git a/sound/soc/codecs/da732x_reg.h b/sound/soc/codecs/da732x_reg.h
new file mode 100644
index 000000000000..bdd03ca4b2de
--- /dev/null
+++ b/sound/soc/codecs/da732x_reg.h
@@ -0,0 +1,654 @@
+/*
+ * da732x_reg.h --- Dialog DA732X ALSA SoC Audio Registers Header File
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __DA732X_REG_H_
+#define __DA732X_REG_H_
+
+/* DA732X registers */
+#define	DA732X_REG_STATUS_EXT		0x00
+#define DA732X_REG_STATUS		0x01
+#define DA732X_REG_REF1			0x02
+#define DA732X_REG_BIAS_EN		0x03
+#define DA732X_REG_BIAS1		0x04
+#define DA732X_REG_BIAS2		0x05
+#define DA732X_REG_BIAS3		0x06
+#define DA732X_REG_BIAS4		0x07
+#define DA732X_REG_MICBIAS2		0x0F
+#define DA732X_REG_MICBIAS1		0x10
+#define DA732X_REG_MICDET		0x11
+#define DA732X_REG_MIC1_PRE		0x12
+#define DA732X_REG_MIC1			0x13
+#define DA732X_REG_MIC2_PRE		0x14
+#define DA732X_REG_MIC2			0x15
+#define DA732X_REG_AUX1L		0x16
+#define DA732X_REG_AUX1R		0x17
+#define DA732X_REG_MIC3_PRE		0x18
+#define DA732X_REG_MIC3			0x19
+#define DA732X_REG_INP_PINBIAS		0x1A
+#define DA732X_REG_INP_ZC_EN		0x1B
+#define DA732X_REG_INP_MUX		0x1D
+#define DA732X_REG_HP_DET		0x20
+#define DA732X_REG_HPL_DAC_OFFSET	0x21
+#define DA732X_REG_HPL_DAC_OFF_CNTL	0x22
+#define DA732X_REG_HPL_OUT_OFFSET	0x23
+#define DA732X_REG_HPL			0x24
+#define DA732X_REG_HPL_VOL		0x25
+#define DA732X_REG_HPR_DAC_OFFSET	0x26
+#define DA732X_REG_HPR_DAC_OFF_CNTL	0x27
+#define DA732X_REG_HPR_OUT_OFFSET	0x28
+#define DA732X_REG_HPR			0x29
+#define DA732X_REG_HPR_VOL		0x2A
+#define DA732X_REG_LIN2			0x2B
+#define DA732X_REG_LIN3			0x2C
+#define DA732X_REG_LIN4			0x2D
+#define DA732X_REG_OUT_ZC_EN		0x2E
+#define DA732X_REG_HP_LIN1_GNDSEL	0x37
+#define DA732X_REG_CP_HP1		0x3A
+#define DA732X_REG_CP_HP2		0x3B
+#define DA732X_REG_CP_CTRL1		0x40
+#define DA732X_REG_CP_CTRL2		0x41
+#define DA732X_REG_CP_CTRL3		0x42
+#define DA732X_REG_CP_LEVEL_MASK	0x43
+#define DA732X_REG_CP_DET		0x44
+#define DA732X_REG_CP_STATUS		0x45
+#define DA732X_REG_CP_THRESH1		0x46
+#define DA732X_REG_CP_THRESH2		0x47
+#define DA732X_REG_CP_THRESH3		0x48
+#define DA732X_REG_CP_THRESH4		0x49
+#define DA732X_REG_CP_THRESH5		0x4A
+#define DA732X_REG_CP_THRESH6		0x4B
+#define DA732X_REG_CP_THRESH7		0x4C
+#define DA732X_REG_CP_THRESH8		0x4D
+#define DA732X_REG_PLL_DIV_LO		0x50
+#define DA732X_REG_PLL_DIV_MID		0x51
+#define DA732X_REG_PLL_DIV_HI		0x52
+#define DA732X_REG_PLL_CTRL		0x53
+#define DA732X_REG_CLK_CTRL		0x54
+#define DA732X_REG_CLK_DSP		0x5A
+#define DA732X_REG_CLK_EN1		0x5B
+#define DA732X_REG_CLK_EN2		0x5C
+#define DA732X_REG_CLK_EN3		0x5D
+#define DA732X_REG_CLK_EN4		0x5E
+#define DA732X_REG_CLK_EN5		0x5F
+#define DA732X_REG_AIF_MCLK		0x60
+#define DA732X_REG_AIFA1		0x61
+#define DA732X_REG_AIFA2		0x62
+#define DA732X_REG_AIFA3		0x63
+#define DA732X_REG_AIFB1		0x64
+#define DA732X_REG_AIFB2		0x65
+#define DA732X_REG_AIFB3		0x66
+#define DA732X_REG_PC_CTRL		0x6A
+#define DA732X_REG_DATA_ROUTE		0x70
+#define DA732X_REG_DSP_CTRL		0x71
+#define DA732X_REG_CIF_CTRL2		0x74
+#define DA732X_REG_HANDSHAKE		0x75
+#define DA732X_REG_MBOX0		0x76
+#define DA732X_REG_MBOX1		0x77
+#define DA732X_REG_MBOX2		0x78
+#define DA732X_REG_MBOX_STATUS		0x79
+#define DA732X_REG_SPARE1_OUT		0x7D
+#define DA732X_REG_SPARE2_OUT		0x7E
+#define DA732X_REG_SPARE1_IN		0x7F
+#define DA732X_REG_ID			0x81
+#define DA732X_REG_ADC1_PD		0x90
+#define DA732X_REG_ADC1_HPF		0x93
+#define DA732X_REG_ADC1_SEL		0x94
+#define DA732X_REG_ADC1_EQ12		0x95
+#define DA732X_REG_ADC1_EQ34		0x96
+#define DA732X_REG_ADC1_EQ5		0x97
+#define DA732X_REG_ADC2_PD		0x98
+#define DA732X_REG_ADC2_HPF		0x9B
+#define DA732X_REG_ADC2_SEL		0x9C
+#define DA732X_REG_ADC2_EQ12		0x9D
+#define DA732X_REG_ADC2_EQ34		0x9E
+#define DA732X_REG_ADC2_EQ5		0x9F
+#define DA732X_REG_DAC1_HPF		0xA0
+#define DA732X_REG_DAC1_L_VOL		0xA1
+#define DA732X_REG_DAC1_R_VOL		0xA2
+#define DA732X_REG_DAC1_SEL		0xA3
+#define DA732X_REG_DAC1_SOFTMUTE	0xA4
+#define DA732X_REG_DAC1_EQ12		0xA5
+#define DA732X_REG_DAC1_EQ34		0xA6
+#define DA732X_REG_DAC1_EQ5		0xA7
+#define DA732X_REG_DAC2_HPF		0xB0
+#define DA732X_REG_DAC2_L_VOL		0xB1
+#define DA732X_REG_DAC2_R_VOL		0xB2
+#define DA732X_REG_DAC2_SEL		0xB3
+#define DA732X_REG_DAC2_SOFTMUTE	0xB4
+#define DA732X_REG_DAC2_EQ12		0xB5
+#define DA732X_REG_DAC2_EQ34		0xB6
+#define DA732X_REG_DAC2_EQ5		0xB7
+#define DA732X_REG_DAC3_HPF		0xC0
+#define DA732X_REG_DAC3_VOL		0xC1
+#define DA732X_REG_DAC3_SEL		0xC3
+#define DA732X_REG_DAC3_SOFTMUTE	0xC4
+#define DA732X_REG_DAC3_EQ12		0xC5
+#define DA732X_REG_DAC3_EQ34		0xC6
+#define DA732X_REG_DAC3_EQ5		0xC7
+#define DA732X_REG_BIQ_BYP		0xD2
+#define DA732X_REG_DMA_CMD		0xD3
+#define DA732X_REG_DMA_ADDR0		0xD4
+#define DA732X_REG_DMA_ADDR1		0xD5
+#define DA732X_REG_DMA_DATA0		0xD6
+#define DA732X_REG_DMA_DATA1		0xD7
+#define DA732X_REG_DMA_DATA2		0xD8
+#define DA732X_REG_DMA_DATA3		0xD9
+#define DA732X_REG_DMA_STATUS		0xDA
+#define DA732X_REG_BROWNOUT		0xDF
+#define DA732X_REG_UNLOCK		0xE0
+
+#define	DA732X_MAX_REG			DA732X_REG_UNLOCK
+/*
+ * Bits
+ */
+
+/* DA732X_REG_STATUS_EXT (addr=0x00) */
+#define	DA732X_STATUS_EXT_DSP			(1 << 4)
+#define	DA732X_STATUS_EXT_CLEAR			(0 << 0)
+
+/* DA732X_REG_STATUS	(addr=0x01) */
+#define DA732X_STATUS_PLL_LOCK			(1 << 0)
+#define DA732X_STATUS_PLL_MCLK_DET		(1 << 1)
+#define DA732X_STATUS_HPDET_OUT			(1 << 2)
+#define DA732X_STATUS_INP_MIXDET_1		(1 << 3)
+#define DA732X_STATUS_INP_MIXDET_2		(1 << 4)
+#define DA732X_STATUS_BO_STATUS			(1 << 5)
+
+/* DA732X_REG_REF1	(addr=0x02) */
+#define DA732X_VMID_FASTCHG			(1 << 1)
+#define DA732X_VMID_FASTDISCHG			(1 << 2)
+#define DA732X_REFBUFX2_EN			(1 << 6)
+#define DA732X_REFBUFX2_DIS			(0 << 6)
+
+/* DA732X_REG_BIAS_EN	(addr=0x03) */
+#define DA732X_BIAS_BOOST_MASK			(3 << 0)
+#define DA732X_BIAS_BOOST_100PC			(0 << 0)
+#define DA732X_BIAS_BOOST_133PC			(1 << 0)
+#define DA732X_BIAS_BOOST_88PC			(2 << 0)
+#define DA732X_BIAS_BOOST_50PC			(3 << 0)
+#define DA732X_BIAS_EN				(1 << 7)
+#define DA732X_BIAS_DIS				(0 << 7)
+
+/* DA732X_REG_BIAS1	(addr=0x04) */
+#define DA732X_BIAS1_HP_DAC_BIAS_MASK		(3 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_100PC		(0 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_150PC		(1 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_50PC		(2 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_75PC		(3 << 0)
+#define DA732X_BIAS1_HP_OUT_BIAS_MASK		(7 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_100PC		(0 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_125PC		(1 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_150PC		(2 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_175PC		(3 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_200PC		(4 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_250PC		(5 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_300PC		(6 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_350PC		(7 << 4)
+
+/* DA732X_REG_BIAS2	(addr=0x05) */
+#define DA732X_BIAS2_LINE2_DAC_BIAS_MASK	(3 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_100PC	(0 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_150PC	(1 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_50PC	(2 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_75PC	(3 << 0)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_MASK	(7 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_100PC	(0 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_125PC	(1 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_150PC	(2 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_175PC	(3 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_200PC	(4 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_250PC	(5 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_300PC	(6 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_350PC	(7 << 4)
+
+/* DA732X_REG_BIAS3	(addr=0x06) */
+#define DA732X_BIAS3_LINE3_DAC_BIAS_MASK	(3 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_100PC	(0 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_150PC	(1 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_50PC	(2 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_75PC	(3 << 0)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_MASK	(7 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_100PC	(0 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_125PC	(1 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_150PC	(2 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_175PC	(3 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_200PC	(4 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_250PC	(5 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_300PC	(6 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_350PC	(7 << 4)
+
+/* DA732X_REG_BIAS4	(addr=0x07) */
+#define DA732X_BIAS4_LINE4_DAC_BIAS_MASK	(3 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_100PC	(0 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_150PC	(1 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_50PC	(2 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_75PC	(3 << 0)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_MASK	(7 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_100PC	(0 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_125PC	(1 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_150PC	(2 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_175PC	(3 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_200PC	(4 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_250PC	(5 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_300PC	(6 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_350PC	(7 << 4)
+
+/* DA732X_REG_SIF_VDD_SEL	(addr=0x08) */
+#define DA732X_SIF_VDD_SEL_AIFA_VDD2		(1 << 0)
+#define DA732X_SIF_VDD_SEL_AIFB_VDD2		(1 << 1)
+#define DA732X_SIF_VDD_SEL_CIFA_VDD2		(1 << 4)
+
+/* DA732X_REG_MICBIAS2/1	(addr=0x0F/0x10) */
+#define DA732X_MICBIAS_VOLTAGE_MASK		(0x0F << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V		(0x00 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V05		(0x01 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V1		(0x02 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V15		(0x03 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V2		(0x04 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V25		(0x05 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V3		(0x06 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V35		(0x07 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V4		(0x08 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V45		(0x09 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V5		(0x0A << 0)
+#define DA732X_MICBIAS_EN			(1 << 7)
+#define DA732X_MICBIAS_EN_SHIFT			7
+#define DA732X_MICBIAS_VOLTAGE_SHIFT		0
+#define	DA732X_MICBIAS_VOLTAGE_MAX		0x0B
+
+/* DA732X_REG_MICDET	(addr=0x11) */
+#define DA732X_MICDET_INP_MICRES		(1 << 0)
+#define DA732X_MICDET_INP_MICHOOK		(1 << 1)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_8MS	(0 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_16MS	(1 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_32MS	(2 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_64MS	(3 << 0)
+#define DA732X_MICDET_INP_MICDET_EN		(1 << 7)
+
+/* DA732X_REG_MIC1/2/3_PRE (addr=0x11/0x14/0x18) */
+#define	DA732X_MICBOOST_MASK			0x7
+#define	DA732X_MICBOOST_SHIFT			0
+#define	DA732X_MICBOOST_MIN			0x1
+#define	DA732X_MICBOOST_MAX			DA732X_MICBOOST_MASK
+
+/* DA732X_REG_MIC1/2/3	(addr=0x13/0x15/0x19) */
+#define	DA732X_MIC_VOL_SHIFT			0
+#define	DA732X_MIC_VOL_VAL_MASK			0x1F
+#define DA732X_MIC_MUTE_SHIFT			6
+#define DA732X_MIC_EN_SHIFT			7
+#define DA732X_MIC_VOL_VAL_MIN			0x7
+#define	DA732X_MIC_VOL_VAL_MAX			DA732X_MIC_VOL_VAL_MASK
+
+/* DA732X_REG_AUX1L/R	(addr=0x16/0x17) */
+#define	DA732X_AUX_VOL_SHIFT			0
+#define	DA732X_AUX_VOL_MASK			0x7
+#define DA732X_AUX_MUTE_SHIFT			6
+#define DA732X_AUX_EN_SHIFT			7
+#define	DA732X_AUX_VOL_VAL_MAX			DA732X_AUX_VOL_MASK
+
+/* DA732X_REG_INP_PINBIAS	(addr=0x1A) */
+#define DA732X_INP_MICL_PINBIAS_EN		(1 << 0)
+#define DA732X_INP_MICR_PINBIAS_EN		(1 << 1)
+#define DA732X_INP_AUX1L_PINBIAS_EN		(1 << 2)
+#define DA732X_INP_AUX1R_PINBIAS_EN		(1 << 3)
+#define DA732X_INP_AUX2_PINBIAS_EN		(1 << 4)
+
+/* DA732X_REG_INP_ZC_EN	(addr=0x1B) */
+#define	DA732X_MIC1_PRE_ZC_EN			(1 << 0)
+#define	DA732X_MIC1_ZC_EN			(1 << 1)
+#define	DA732X_MIC2_PRE_ZC_EN			(1 << 2)
+#define	DA732X_MIC2_ZC_EN			(1 << 3)
+#define	DA732X_AUXL_ZC_EN			(1 << 4)
+#define	DA732X_AUXR_ZC_EN			(1 << 5)
+#define	DA732X_MIC3_PRE_ZC_EN			(1 << 6)
+#define	DA732X_MIC3_ZC_EN			(1 << 7)
+
+/* DA732X_REG_INP_MUX	(addr=0x1D) */
+#define DA732X_INP_ADC1L_MUX_SEL_AUX1L		(0 << 0)
+#define DA732X_INP_ADC1L_MUX_SEL_MIC1		(1 << 0)
+#define DA732X_INP_ADC1R_MUX_SEL_MASK		(3 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_AUX1R		(0 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_MIC2		(1 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_MIC3		(2 << 2)
+#define DA732X_INP_ADC2L_MUX_SEL_AUX1L		(0 << 4)
+#define DA732X_INP_ADC2L_MUX_SEL_MICL		(1 << 4)
+#define DA732X_INP_ADC2R_MUX_SEL_MASK		(3 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_AUX1R		(0 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_MICR		(1 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_AUX2		(2 << 6)
+#define	DA732X_ADC1L_MUX_SEL_SHIFT		0
+#define	DA732X_ADC1R_MUX_SEL_SHIFT		2
+#define	DA732X_ADC2L_MUX_SEL_SHIFT		4
+#define	DA732X_ADC2R_MUX_SEL_SHIFT		6
+
+/* DA732X_REG_HP_DET		(addr=0x20) */
+#define DA732X_HP_DET_AZ			(1 << 0)
+#define DA732X_HP_DET_SEL1			(1 << 1)
+#define DA732X_HP_DET_IS_MASK			(3 << 2)
+#define DA732X_HP_DET_IS_0_5UA			(0 << 2)
+#define DA732X_HP_DET_IS_1UA			(1 << 2)
+#define DA732X_HP_DET_IS_2UA			(2 << 2)
+#define DA732X_HP_DET_IS_4UA			(3 << 2)
+#define DA732X_HP_DET_RS_MASK			(3 << 4)
+#define DA732X_HP_DET_RS_INFINITE		(0 << 4)
+#define DA732X_HP_DET_RS_100KOHM		(1 << 4)
+#define DA732X_HP_DET_RS_10KOHM			(2 << 4)
+#define DA732X_HP_DET_RS_1KOHM			(3 << 4)
+#define DA732X_HP_DET_EN			(1 << 7)
+
+/* DA732X_REG_HPL_DAC_OFFSET	(addr=0x21/0x26) */
+#define DA732X_HP_DAC_OFFSET_TRIM_MASK		(0x3F << 0)
+#define DA732X_HP_DAC_OFFSET_DAC_SIGN		(1 << 6)
+
+/* DA732X_REG_HPL_DAC_OFF_CNTL	(addr=0x22/0x27) */
+#define DA732X_HP_DAC_OFF_CNTL_CONT_MASK	(7 << 0)
+#define DA732X_HP_DAC_OFF_CNTL_COMPO		(1 << 3)
+#define	DA732X_HP_DAC_OFF_CALIBRATION		(1 << 0)
+#define	DA732X_HP_DAC_OFF_SCALE_STEPS		(1 << 1)
+#define	DA732X_HP_DAC_OFF_MASK			0x7F
+#define DA732X_HP_DAC_COMPO_SHIFT		3
+
+/* DA732X_REG_HPL_OUT_OFFSET	(addr=0x23/0x28) */
+#define DA732X_HP_OUT_OFFSET_MASK		(0xFF << 0)
+#define	DA732X_HP_DAC_OFFSET_TRIM_VAL		0x7F
+
+/* DA732X_REG_HPL/R	(addr=0x24/0x29) */
+#define DA732X_HP_OUT_SIGN			(1 << 0)
+#define DA732X_HP_OUT_COMP			(1 << 1)
+#define DA732X_HP_OUT_RESERVED			(1 << 2)
+#define DA732X_HP_OUT_COMPO			(1 << 3)
+#define DA732X_HP_OUT_DAC_EN			(1 << 4)
+#define DA732X_HP_OUT_HIZ_EN			(1 << 5)
+#define	DA732X_HP_OUT_HIZ_DIS			(0 << 5)
+#define DA732X_HP_OUT_MUTE			(1 << 6)
+#define DA732X_HP_OUT_EN			(1 << 7)
+#define	DA732X_HP_OUT_COMPO_SHIFT		3
+#define	DA732X_HP_OUT_DAC_EN_SHIFT		4
+#define	DA732X_HP_HIZ_SHIFT			5
+#define	DA732X_HP_MUTE_SHIFT			6
+#define DA732X_HP_OUT_EN_SHIFT			7
+
+#define DA732X_OUT_HIZ_EN			(1 << 5)
+#define	DA732X_OUT_HIZ_DIS			(0 << 5)
+
+/* DA732X_REG_HPL/R_VOL	(addr=0x25/0x2A) */
+#define	DA732X_HP_VOL_VAL_MASK			0xF
+#define	DA732X_HP_VOL_SHIFT			0
+#define	DA732X_HP_VOL_VAL_MAX			DA732X_HP_VOL_VAL_MASK
+
+/* DA732X_REG_LIN2/3/4	(addr=0x2B/0x2C/0x2D) */
+#define DA732X_LOUT_VOL_SHIFT			0
+#define DA732X_LOUT_VOL_MASK			0x0F
+#define DA732X_LOUT_DAC_OFF			(0 << 4)
+#define DA732X_LOUT_DAC_EN			(1 << 4)
+#define DA732X_LOUT_HIZ_N_DIS			(0 << 5)
+#define DA732X_LOUT_HIZ_N_EN			(1 << 5)
+#define DA732X_LOUT_UNMUTED			(0 << 6)
+#define DA732X_LOUT_MUTED			(1 << 6)
+#define DA732X_LOUT_EN				(0 << 7)
+#define DA732X_LOUT_DIS				(1 << 7)
+#define DA732X_LOUT_DAC_EN_SHIFT		4
+#define	DA732X_LOUT_MUTE_SHIFT			6
+#define DA732X_LIN_OUT_EN_SHIFT			7
+#define DA732X_LOUT_VOL_VAL_MAX			DA732X_LOUT_VOL_MASK
+
+/* DA732X_REG_OUT_ZC_EN		(addr=0x2E) */
+#define	DA732X_HPL_ZC_EN_SHIFT			0
+#define DA732X_HPR_ZC_EN_SHIFT			1
+#define DA732X_HPL_ZC_EN			(1 << 0)
+#define DA732X_HPL_ZC_DIS			(0 << 0)
+#define DA732X_HPR_ZC_EN			(1 << 1)
+#define DA732X_HPR_ZC_DIS			(0 << 1)
+#define DA732X_LIN2_ZC_EN			(1 << 2)
+#define DA732X_LIN2_ZC_DIS			(0 << 2)
+#define DA732X_LIN3_ZC_EN			(1 << 3)
+#define DA732X_LIN3_ZC_DIS			(0 << 3)
+#define DA732X_LIN4_ZC_EN			(1 << 4)
+#define DA732X_LIN4_ZC_DIS			(0 << 4)
+
+/* DA732X_REG_HP_LIN1_GNDSEL (addr=0x37) */
+#define	DA732X_HP_OUT_GNDSEL			(1 << 0)
+
+/* DA732X_REG_CP_HP2 (addr=0x3a) */
+#define	DA732X_HP_CP_PULSESKIP			(1 << 0)
+#define	DA732X_HP_CP_REG			(1 << 1)
+#define DA732X_HP_CP_EN				(1 << 3)
+#define DA732X_HP_CP_DIS			(0 << 3)
+
+/* DA732X_REG_CP_CTRL1 (addr=0x40) */
+#define	DA732X_CP_MODE_MASK			(7 << 1)
+#define	DA732X_CP_CTRL_STANDBY			(0 << 1)
+#define	DA732X_CP_CTRL_CPVDD6			(2 << 1)
+#define	DA732X_CP_CTRL_CPVDD5			(3 << 1)
+#define	DA732X_CP_CTRL_CPVDD4			(4 << 1)
+#define	DA732X_CP_CTRL_CPVDD3			(5 << 1)
+#define	DA732X_CP_CTRL_CPVDD2			(6 << 1)
+#define	DA732X_CP_CTRL_CPVDD1			(7 << 1)
+#define	DA723X_CP_DIS				(0 << 7)
+#define	DA732X_CP_EN				(1 << 7)
+
+/* DA732X_REG_CP_CTRL2 (addr=0x41) */
+#define	DA732X_CP_BOOST				(1 << 0)
+#define	DA732X_CP_MANAGE_MAGNITUDE		(2 << 2)
+
+/* DA732X_REG_CP_CTRL3 (addr=0x42) */
+#define	DA732X_CP_1MHZ				(0 << 0)
+#define	DA732X_CP_500KHZ			(1 << 0)
+#define	DA732X_CP_250KHZ			(2 << 0)
+#define	DA732X_CP_125KHZ			(3 << 0)
+#define	DA732X_CP_63KHZ				(4 << 0)
+#define	DA732X_CP_0KHZ				(5 << 0)
+
+/* DA732X_REG_PLL_CTRL (addr=0x53) */
+#define	DA732X_PLL_INDIV_MASK			(3 << 0)
+#define	DA732X_PLL_SRM_EN			(1 << 2)
+#define	DA732X_PLL_EN				(1 << 7)
+#define	DA732X_PLL_BYPASS			(0 << 0)
+
+/* DA732X_REG_CLK_CTRL (addr=0x54) */
+#define	DA732X_SR1_MASK				(0xF)
+#define	DA732X_SR2_MASK				(0xF0)
+
+/* DA732X_REG_CLK_DSP (addr=0x5A) */
+#define	DA732X_DSP_FREQ_MASK			(7 << 0)
+#define	DA732X_DSP_FREQ_12MHZ			(0 << 0)
+#define	DA732X_DSP_FREQ_24MHZ			(1 << 0)
+#define	DA732X_DSP_FREQ_36MHZ			(2 << 0)
+#define	DA732X_DSP_FREQ_48MHZ			(3 << 0)
+#define	DA732X_DSP_FREQ_60MHZ			(4 << 0)
+#define	DA732X_DSP_FREQ_72MHZ			(5 << 0)
+#define	DA732X_DSP_FREQ_84MHZ			(6 << 0)
+#define	DA732X_DSP_FREQ_96MHZ			(7 << 0)
+
+/* DA732X_REG_CLK_EN1 (addr=0x5B) */
+#define	DA732X_DSP_CLK_EN			(1 << 0)
+#define	DA732X_SYS3_CLK_EN			(1 << 1)
+#define	DA732X_DSP12_CLK_EN			(1 << 2)
+#define	DA732X_PC_CLK_EN			(1 << 3)
+#define	DA732X_MCLK_SQR_EN			(1 << 7)
+
+/* DA732X_REG_CLK_EN2 (addr=0x5C) */
+#define	DA732X_UART_CLK_EN			(1 << 1)
+#define	DA732X_CP_CLK_EN			(1 << 2)
+#define	DA732X_CP_CLK_DIS			(0 << 2)
+
+/* DA732X_REG_CLK_EN3 (addr=0x5D) */
+#define	DA732X_ADCA_BB_CLK_EN			(1 << 0)
+#define	DA732X_ADCC_BB_CLK_EN			(1 << 4)
+
+/* DA732X_REG_CLK_EN4 (addr=0x5E) */
+#define	DA732X_DACA_BB_CLK_EN			(1 << 0)
+#define	DA732X_DACC_BB_CLK_EN			(1 << 4)
+#define DA732X_DACA_BB_CLK_SHIFT		0
+#define DA732X_DACC_BB_CLK_SHIFT		4
+
+/* DA732X_REG_CLK_EN5 (addr=0x5F) */
+#define	DA732X_DACE_BB_CLK_EN			(1 << 0)
+#define DA732X_DACE_BB_CLK_SHIFT		0
+
+/* DA732X_REG_AIF_MCLK (addr=0x60) */
+#define DA732X_AIFM_FRAME_64			(1 << 2)
+#define	DA732X_AIFM_SRC_SEL_AIFA		(1 << 6)
+#define	DA732X_CLK_GENERATION_AIF_A		(1 << 4)
+#define	DA732X_NO_CLK_GENERATION		0x0
+
+/* DA732X_REG_AIFA1 (addr=0x61) */
+#define	DA732X_AIF_WORD_MASK			(0x3 << 0)
+#define	DA732X_AIF_WORD_16			(0 << 0)
+#define	DA732X_AIF_WORD_20			(1 << 0)
+#define	DA732X_AIF_WORD_24			(2 << 0)
+#define	DA732X_AIF_WORD_32			(3 << 0)
+#define	DA732X_AIF_TDM_MONO_SHIFT		(1 << 6)
+#define	DA732X_AIF1_CLK_MASK			(1 << 7)
+#define	DA732X_AIF_SLAVE			(0 << 7)
+#define DA732X_AIF_CLK_FROM_SRC			(1 << 7)
+
+/* DA732X_REG_AIFA3 (addr=0x63) */
+#define	DA732X_AIF_MODE_SHIFT			0
+#define	DA732X_AIF_MODE_MASK			0x3
+#define	DA732X_AIF_I2S_MODE			(0 << 0)
+#define	DA732X_AIF_LEFT_J_MODE			(1 << 0)
+#define	DA732X_AIF_RIGHT_J_MODE			(2 << 0)
+#define	DA732X_AIF_DSP_MODE			(3 << 0)
+#define DA732X_AIF_WCLK_INV			(1 << 4)
+#define DA732X_AIF_BCLK_INV			(1 << 5)
+#define	DA732X_AIF_EN				(1 << 7)
+#define	DA732X_AIF_EN_SHIFT			7
+
+/* DA732X_REG_PC_CTRL (addr=0x6a) */
+#define	DA732X_PC_PULSE_AIFA			(0 << 0)
+#define	DA732X_PC_PULSE_AIFB			(1 << 0)
+#define	DA732X_PC_RESYNC_AUT			(1 << 6)
+#define	DA732X_PC_RESYNC_NOT_AUT		(0 << 6)
+#define	DA732X_PC_SAME				(1 << 7)
+
+/* DA732X_REG_DATA_ROUTE (addr=0x70) */
+#define DA732X_ADC1_TO_AIFA			(0 << 0)
+#define DA732X_DSP_TO_AIFA			(1 << 0)
+#define DA732X_ADC2_TO_AIFB			(0 << 1)
+#define DA732X_DSP_TO_AIFB			(1 << 1)
+#define DA732X_AIFA_TO_DAC1L			(0 << 2)
+#define DA732X_DSP_TO_DAC1L			(1 << 2)
+#define DA732X_AIFA_TO_DAC1R			(0 << 3)
+#define DA732X_DSP_TO_DAC1R			(1 << 3)
+#define DA732X_AIFB_TO_DAC2L			(0 << 4)
+#define DA732X_DSP_TO_DAC2L			(1 << 4)
+#define DA732X_AIFB_TO_DAC2R			(0 << 5)
+#define DA732X_DSP_TO_DAC2R			(1 << 5)
+#define DA732X_AIFB_TO_DAC3			(0 << 6)
+#define DA732X_DSP_TO_DAC3			(1 << 6)
+#define	DA732X_BYPASS_DSP			(0 << 0)
+#define	DA732X_ALL_TO_DSP			(0x7F << 0)
+
+/* DA732X_REG_DSP_CTRL (addr=0x71) */
+#define	DA732X_DIGITAL_EN			(1 << 0)
+#define	DA732X_DIGITAL_RESET			(0 << 0)
+#define	DA732X_DSP_CORE_EN			(1 << 1)
+#define	DA732X_DSP_CORE_RESET			(0 << 1)
+
+/* DA732X_REG_SPARE1_OUT (addr=0x7D)*/
+#define	DA732X_HP_DRIVER_EN			(1 << 0)
+#define	DA732X_HP_GATE_LOW			(1 << 2)
+#define DA732X_HP_LOOP_GAIN_CTRL		(1 << 3)
+
+/* DA732X_REG_ID (addr=0x81)*/
+#define DA732X_ID_MINOR_MASK			(0xF << 0)
+#define DA732X_ID_MAJOR_MASK			(0xF << 4)
+
+/* DA732X_REG_ADC1/2_PD (addr=0x90/0x98) */
+#define	DA732X_ADC_RST_MASK			(0x3 << 0)
+#define	DA732X_ADC_PD_MASK			(0x3 << 2)
+#define	DA732X_ADC_SET_ACT			(0x3 << 0)
+#define	DA732X_ADC_SET_RST			(0x0 << 0)
+#define	DA732X_ADC_ON				(0x3 << 2)
+#define	DA732X_ADC_OFF				(0x0 << 2)
+
+/* DA732X_REG_ADC1/2_SEL (addr=0x94/0x9C) */
+#define	DA732X_ADC_VOL_VAL_MASK			0x7
+#define	DA732X_ADCL_VOL_SHIFT			0
+#define	DA732X_ADCR_VOL_SHIFT			4
+#define DA732X_ADCL_EN_SHIFT			2
+#define DA732X_ADCR_EN_SHIFT			3
+#define	DA732X_ADCL_EN				(1 << 2)
+#define	DA732X_ADCR_EN				(1 << 3)
+#define	DA732X_ADC_VOL_VAL_MAX			DA732X_ADC_VOL_VAL_MASK
+
+/*
+ * DA732X_REG_ADC1/2_HPF (addr=0x93/0x9b)
+ * DA732x_REG_DAC1/2/3_HPG	(addr=0xA5/0xB5/0xC5)
+ */
+#define	DA732X_HPF_MUSIC_EN			(1 << 3)
+#define	DA732X_HPF_VOICE_EN			((1 << 3) | (1 << 7))
+#define	DA732X_HPF_MASK				((1 << 3) | (1 << 7))
+#define DA732X_HPF_DIS				((0 << 3) | (0 << 7))
+
+/* DA732X_REG_DAC1/2/3_VOL */
+#define DA732X_DAC_VOL_VAL_MASK			0x7F
+#define DA732X_DAC_VOL_SHIFT			0
+#define DA732X_DAC_VOL_VAL_MAX			DA732X_DAC_VOL_VAL_MASK
+
+/* DA732X_REG_DAC1/2/3_SEL (addr=0xA3/0xB3/0xC3) */
+#define DA732X_DACL_EN_SHIFT			3
+#define	DA732X_DACR_EN_SHIFT			7
+#define DA732X_DACL_MUTE_SHIFT			2
+#define	DA732X_DACR_MUTE_SHIFT			6
+#define DA732X_DACL_EN				(1 << 3)
+#define	DA732X_DACR_EN				(1 << 7)
+#define	DA732X_DACL_SDM				(1 << 0)
+#define	DA732X_DACR_SDM				(1 << 4)
+#define	DA732X_DACL_MUTE			(1 << 2)
+#define	DA732X_DACR_MUTE			(1 << 6)
+
+/* DA732X_REG_DAC_SOFTMUTE (addr=0xA4/0xB4/0xC4) */
+#define	DA732X_SOFTMUTE_EN			(1 << 7)
+#define	DA732X_GAIN_RAMPED			(1 << 6)
+#define	DA732X_16_SAMPLES			(4 << 0)
+#define	DA732X_SOFTMUTE_MASK			(1 << 7)
+#define	DA732X_SOFTMUTE_SHIFT			7
+
+/*
+ * DA732x_REG_ADC1/2_EQ12	(addr=0x95/0x9D)
+ * DA732x_REG_ADC1/2_EQ34	(addr=0x96/0x9E)
+ * DA732x_REG_ADC1/2_EQ5	(addr=0x97/0x9F)
+ * DA732x_REG_DAC1/2/3_EQ12	(addr=0xA5/0xB5/0xC5)
+ * DA732x_REG_DAC1/2/3_EQ34	(addr=0xA6/0xB6/0xC6)
+ * DA732x_REG_DAC1/2/3_EQ5	(addr=0xA7/0xB7/0xB7)
+ */
+#define	DA732X_EQ_VOL_VAL_MASK			0xF
+#define	DA732X_EQ_BAND1_SHIFT			0
+#define	DA732X_EQ_BAND2_SHIFT			4
+#define	DA732X_EQ_BAND3_SHIFT			0
+#define	DA732X_EQ_BAND4_SHIFT			4
+#define	DA732X_EQ_BAND5_SHIFT			0
+#define	DA732X_EQ_OVERALL_SHIFT			4
+#define	DA732X_EQ_OVERALL_VOL_VAL_MASK		0x3
+#define	DA732X_EQ_DIS				(0 << 7)
+#define	DA732X_EQ_EN				(1 << 7)
+#define	DA732X_EQ_EN_SHIFT			7
+#define	DA732X_EQ_VOL_VAL_MAX			DA732X_EQ_VOL_VAL_MASK
+#define	DA732X_EQ_OVERALL_VOL_VAL_MAX		DA732X_EQ_OVERALL_VOL_VAL_MASK
+
+/* DA732X_REG_DMA_CMD (addr=0xD3) */
+#define	DA732X_SEL_DSP_DMA_MASK			(3 << 0)
+#define	DA732X_SEL_DSP_DMA_DIS			(0 << 0)
+#define	DA732X_SEL_DSP_DMA_PMEM			(1 << 0)
+#define	DA732X_SEL_DSP_DMA_XMEM			(2 << 0)
+#define	DA732X_SEL_DSP_DMA_YMEM			(3 << 0)
+#define	DA732X_DSP_RW_MASK			(1 << 4)
+#define	DA732X_DSP_DMA_WRITE			(0 << 4)
+#define	DA732X_DSP_DMA_READ			(1 << 4)
+
+/* DA732X_REG_DMA_STATUS (addr=0xDA) */
+#define	DA732X_DSP_DMA_FREE			(0 << 0)
+#define	DA732X_DSP_DMA_BUSY			(1 << 0)
+
+#endif /* __DA732X_REG_H_ */
-- 
cgit v1.2.3


From 20c5fd399482ef5b87a41ab064b3255f1faaaee4 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 10:03:20 +0800
Subject: ASoC: wm8903: Move pin configuration into I2C probe() function

Ensure that the device pins are configured as soon as possible by moving
the pin configration (including MICBIAS) into the I2C probe() function.
This had been done in the CODEC probe() function when we were relying on
the ASoC register I/O code.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
---
 sound/soc/codecs/wm8903.c | 92 ++++++++++++++++++++++++-----------------------
 1 file changed, 47 insertions(+), 45 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 3abd450842ee..64ca9042bad3 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1,7 +1,7 @@
 /*
  * wm8903.c  --  WM8903 ALSA SoC Audio driver
  *
- * Copyright 2008-11 Wolfson Microelectronics
+ * Copyright 2008-12 Wolfson Microelectronics
  * Copyright 2011-2012 NVIDIA, Inc.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
@@ -1880,10 +1880,9 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 {
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	struct wm8903_platform_data *pdata = wm8903->pdata;
-	int ret, i;
+	int ret;
 	int trigger, irq_pol;
 	u16 val;
-	bool mic_gpio = false;
 
 	wm8903->codec = codec;
 	codec->control_data = wm8903->regmap;
@@ -1894,47 +1893,6 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	/* Set up GPIOs, detect if any are MIC detect outputs */
-	for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
-		if ((!pdata->gpio_cfg[i]) ||
-		    (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
-			continue;
-
-		snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
-				pdata->gpio_cfg[i] & 0x7fff);
-
-		val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
-			>> WM8903_GP1_FN_SHIFT;
-
-		switch (val) {
-		case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
-		case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
-			mic_gpio = true;
-			break;
-		default:
-			break;
-		}
-	}
-
-	/* Set up microphone detection */
-	snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
-			pdata->micdet_cfg);
-
-	/* Microphone detection needs the WSEQ clock */
-	if (pdata->micdet_cfg)
-		snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
-				    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
-
-	/* If microphone detection is enabled by pdata but
-	    * detected via IRQ then interrupts can be lost before
-	    * the machine driver has set up microphone detection
-	    * IRQs as the IRQs are clear on read.  The detection
-	    * will be enabled when the machine driver configures.
-	    */
-	WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
-
-	wm8903->mic_delay = pdata->micdet_delay;
-
 	if (wm8903->irq) {
 		if (pdata->irq_active_low) {
 			trigger = IRQF_TRIGGER_LOW;
@@ -2115,8 +2073,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 {
 	struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8903_priv *wm8903;
+	bool mic_gpio = false;
 	unsigned int val;
-	int ret;
+	int ret, i;
 
 	wm8903 = devm_kzalloc(&i2c->dev,  sizeof(struct wm8903_priv),
 			      GFP_KERNEL);
@@ -2160,6 +2119,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 		}
 	}
 
+	pdata = wm8903->pdata;
+
 	ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret);
@@ -2184,6 +2145,47 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 	wm8903_init_gpio(wm8903);
 
+	/* Set up GPIO pin state, detect if any are MIC detect outputs */
+	for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
+		if ((!pdata->gpio_cfg[i]) ||
+		    (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
+			continue;
+
+		regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i,
+				pdata->gpio_cfg[i] & 0x7fff);
+
+		val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
+			>> WM8903_GP1_FN_SHIFT;
+
+		switch (val) {
+		case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
+		case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
+			mic_gpio = true;
+			break;
+		default:
+			break;
+		}
+	}
+
+	/* Set up microphone detection */
+	regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0,
+		     pdata->micdet_cfg);
+
+	/* Microphone detection needs the WSEQ clock */
+	if (pdata->micdet_cfg)
+		regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0,
+				   WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+
+	/* If microphone detection is enabled by pdata but
+	 * detected via IRQ then interrupts can be lost before
+	 * the machine driver has set up microphone detection
+	 * IRQs as the IRQs are clear on read.  The detection
+	 * will be enabled when the machine driver configures.
+	 */
+	WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
+
+	wm8903->mic_delay = pdata->micdet_delay;
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8903, &wm8903_dai, 1);
 	if (ret != 0)
-- 
cgit v1.2.3


From e373cbfb2f7d194e48d528794b3b99274d4c1a97 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 10:06:11 +0800
Subject: ASoC: wm8903: Make interrupt handler use regmap directly

There's no urgent need for the interrupt handler to use the ASoC I/O
functions and it'll support a further move in where we request the
interrupt so call the regmap APIs directly.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
---
 sound/soc/codecs/wm8903.c | 48 +++++++++++++++++++++++++++++++----------------
 1 file changed, 32 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 64ca9042bad3..f5d47c8e5402 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1636,17 +1636,27 @@ EXPORT_SYMBOL_GPL(wm8903_mic_detect);
 
 static irqreturn_t wm8903_irq(int irq, void *data)
 {
-	struct snd_soc_codec *codec = data;
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	int mic_report;
-	int int_pol;
-	int int_val = 0;
-	int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK);
+	struct wm8903_priv *wm8903 = data;
+	int mic_report, ret;
+	unsigned int int_val, mask, int_pol;
+
+	ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK,
+			  &mask);
+	if (ret != 0) {
+		dev_err(wm8903->dev, "Failed to read IRQ mask: %d\n", ret);
+		return IRQ_NONE;
+	}
+
+	ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1, &int_val);
+	if (ret != 0) {
+		dev_err(wm8903->dev, "Failed to read IRQ status: %d\n", ret);
+		return IRQ_NONE;
+	}
 
-	int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
+	int_val &= ~mask;
 
 	if (int_val & WM8903_WSEQ_BUSY_EINT) {
-		dev_warn(codec->dev, "Write sequencer done\n");
+		dev_warn(wm8903->dev, "Write sequencer done\n");
 	}
 
 	/*
@@ -1657,22 +1667,28 @@ static irqreturn_t wm8903_irq(int irq, void *data)
 	 * the polarity register.
 	 */
 	mic_report = wm8903->mic_last_report;
-	int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1);
+	ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1,
+			  &int_pol);
+	if (ret != 0) {
+		dev_err(wm8903->dev, "Failed to read interrupt polarity: %d\n",
+			ret);
+		return IRQ_HANDLED;
+	}
 
 #ifndef CONFIG_SND_SOC_WM8903_MODULE
 	if (int_val & (WM8903_MICSHRT_EINT | WM8903_MICDET_EINT))
-		trace_snd_soc_jack_irq(dev_name(codec->dev));
+		trace_snd_soc_jack_irq(dev_name(wm8903->dev));
 #endif
 
 	if (int_val & WM8903_MICSHRT_EINT) {
-		dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol);
+		dev_dbg(wm8903->dev, "Microphone short (pol=%x)\n", int_pol);
 
 		mic_report ^= wm8903->mic_short;
 		int_pol ^= WM8903_MICSHRT_INV;
 	}
 
 	if (int_val & WM8903_MICDET_EINT) {
-		dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol);
+		dev_dbg(wm8903->dev, "Microphone detect (pol=%x)\n", int_pol);
 
 		mic_report ^= wm8903->mic_det;
 		int_pol ^= WM8903_MICDET_INV;
@@ -1680,8 +1696,8 @@ static irqreturn_t wm8903_irq(int irq, void *data)
 		msleep(wm8903->mic_delay);
 	}
 
-	snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1,
-			    WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
+	regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1,
+			   WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
 
 	snd_soc_jack_report(wm8903->mic_jack, mic_report,
 			    wm8903->mic_short | wm8903->mic_det);
@@ -1907,7 +1923,7 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 		
 		ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq,
 					   trigger | IRQF_ONESHOT,
-					   "wm8903", codec);
+					   "wm8903", wm8903);
 		if (ret != 0) {
 			dev_err(codec->dev, "Failed to request IRQ: %d\n",
 				ret);
@@ -1963,7 +1979,7 @@ static int wm8903_remove(struct snd_soc_codec *codec)
 
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	if (wm8903->irq)
-		free_irq(wm8903->irq, codec);
+		free_irq(wm8903->irq, wm8903);
 
 	return 0;
 }
-- 
cgit v1.2.3


From b7c95d9146c8201740e2ce9dca7fb1eb8b7b0053 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 10:15:10 +0800
Subject: ASoC: wm8903: Move interrupt request to I2C probe

There's no reason to defer requesting of the interrupt until the CODEC
probe and doing so results in more work if we hit an error as we'll have
registered the CODEC with the core. It's neater to acquire as many of the
resources we'll need as we can in the bus probe function.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
---
 sound/soc/codecs/wm8903.c | 65 ++++++++++++++++++++++-------------------------
 1 file changed, 31 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index f5d47c8e5402..7261a68aac6f 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1895,9 +1895,7 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 static int wm8903_probe(struct snd_soc_codec *codec)
 {
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	struct wm8903_platform_data *pdata = wm8903->pdata;
 	int ret;
-	int trigger, irq_pol;
 	u16 val;
 
 	wm8903->codec = codec;
@@ -1909,32 +1907,6 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	if (wm8903->irq) {
-		if (pdata->irq_active_low) {
-			trigger = IRQF_TRIGGER_LOW;
-			irq_pol = WM8903_IRQ_POL;
-		} else {
-			trigger = IRQF_TRIGGER_HIGH;
-			irq_pol = 0;
-		}
-
-		snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL,
-				    WM8903_IRQ_POL, irq_pol);
-		
-		ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq,
-					   trigger | IRQF_ONESHOT,
-					   "wm8903", wm8903);
-		if (ret != 0) {
-			dev_err(codec->dev, "Failed to request IRQ: %d\n",
-				ret);
-			return ret;
-		}
-
-		/* Enable write sequencer interrupts */
-		snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK,
-				    WM8903_IM_WSEQ_BUSY_EINT, 0);
-	}
-
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1975,11 +1947,7 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 /* power down chip */
 static int wm8903_remove(struct snd_soc_codec *codec)
 {
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	if (wm8903->irq)
-		free_irq(wm8903->irq, wm8903);
 
 	return 0;
 }
@@ -2089,8 +2057,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 {
 	struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8903_priv *wm8903;
+	int trigger;
 	bool mic_gpio = false;
-	unsigned int val;
+	unsigned int val, irq_pol;
 	int ret, i;
 
 	wm8903 = devm_kzalloc(&i2c->dev,  sizeof(struct wm8903_priv),
@@ -2108,7 +2077,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 	}
 
 	i2c_set_clientdata(i2c, wm8903);
-	wm8903->irq = i2c->irq;
 
 	/* If no platform data was supplied, create storage for defaults */
 	if (pdata) {
@@ -2202,6 +2170,33 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 
 	wm8903->mic_delay = pdata->micdet_delay;
 
+	if (i2c->irq) {
+		if (pdata->irq_active_low) {
+			trigger = IRQF_TRIGGER_LOW;
+			irq_pol = WM8903_IRQ_POL;
+		} else {
+			trigger = IRQF_TRIGGER_HIGH;
+			irq_pol = 0;
+		}
+
+		regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_CONTROL,
+				   WM8903_IRQ_POL, irq_pol);
+
+		ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq,
+					   trigger | IRQF_ONESHOT,
+					   "wm8903", wm8903);
+		if (ret != 0) {
+			dev_err(wm8903->dev, "Failed to request IRQ: %d\n",
+				ret);
+			return ret;
+		}
+
+		/* Enable write sequencer interrupts */
+		regmap_update_bits(wm8903->regmap,
+				   WM8903_INTERRUPT_STATUS_1_MASK,
+				   WM8903_IM_WSEQ_BUSY_EINT, 0);
+	}
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8903, &wm8903_dai, 1);
 	if (ret != 0)
@@ -2216,6 +2211,8 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct wm8903_priv *wm8903 = i2c_get_clientdata(client);
 
+	if (client->irq)
+		free_irq(client->irq, wm8903);
 	wm8903_free_gpio(wm8903);
 	snd_soc_unregister_codec(&client->dev);
 
-- 
cgit v1.2.3


From a89c3e956ae78cec8926b92f2d61b7a5b675e787 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 10:30:34 +0800
Subject: ASoC: wm8903: Move register default changes to I2C probe

Also convert to use update_bits() while we're at it. No great need to do
this, it's just a bit neater to do as much as possible in the I2C probe.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
---
 sound/soc/codecs/wm8903.c | 63 +++++++++++++++++++++++------------------------
 1 file changed, 31 insertions(+), 32 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 7261a68aac6f..73f1c8d7bafb 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1896,7 +1896,6 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 {
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	int ret;
-	u16 val;
 
 	wm8903->codec = codec;
 	codec->control_data = wm8903->regmap;
@@ -1910,37 +1909,6 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	/* Latch volume update bits */
-	val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT);
-	val |= WM8903_ADCVU;
-	snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val);
-	snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT);
-	val |= WM8903_DACVU;
-	snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val);
-	snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT);
-	val |= WM8903_HPOUTVU;
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val);
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT);
-	val |= WM8903_LINEOUTVU;
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val);
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT);
-	val |= WM8903_SPKVU;
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val);
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
-
-	/* Enable DAC soft mute by default */
-	snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
-			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
-			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
-
 	return ret;
 }
 
@@ -2197,6 +2165,37 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
 				   WM8903_IM_WSEQ_BUSY_EINT, 0);
 	}
 
+	/* Latch volume update bits */
+	regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_LEFT,
+			   WM8903_ADCVU, WM8903_ADCVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_RIGHT,
+			   WM8903_ADCVU, WM8903_ADCVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_LEFT,
+			   WM8903_DACVU, WM8903_DACVU);
+	regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_RIGHT,
+			   WM8903_DACVU, WM8903_DACVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_LEFT,
+			   WM8903_HPOUTVU, WM8903_HPOUTVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_RIGHT,
+			   WM8903_HPOUTVU, WM8903_HPOUTVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_LEFT,
+			   WM8903_LINEOUTVU, WM8903_LINEOUTVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_RIGHT,
+			   WM8903_LINEOUTVU, WM8903_LINEOUTVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_LEFT,
+			   WM8903_SPKVU, WM8903_SPKVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_RIGHT,
+			   WM8903_SPKVU, WM8903_SPKVU);
+
+	/* Enable DAC soft mute by default */
+	regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_1,
+			   WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
+			   WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8903, &wm8903_dai, 1);
 	if (ret != 0)
-- 
cgit v1.2.3


From 85f243912b99b053ce0624c30609f5d8fd4445d2 Mon Sep 17 00:00:00 2001
From: Ola Lilja <ola.o.lilja@stericsson.com>
Date: Wed, 13 Jun 2012 10:09:51 +0200
Subject: ASoC: Ux500: Correct license strings

GPLv2 -> GPL v2

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ab8500-codec.c | 2 +-
 sound/soc/ux500/ux500_msp_dai.c | 2 +-
 sound/soc/ux500/ux500_msp_i2s.c | 2 +-
 sound/soc/ux500/ux500_pcm.c     | 2 +-
 4 files changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 95dc7d5bb076..389dd660b511 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2518,4 +2518,4 @@ static struct platform_driver ab8500_codec_platform_driver = {
 };
 module_platform_driver(ab8500_codec_platform_driver);
 
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 93c6c40e724c..62ac0285bfaf 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -840,4 +840,4 @@ static struct platform_driver msp_i2s_driver = {
 };
 module_platform_driver(msp_i2s_driver);
 
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 496dec10c96e..ee14d2dac2f5 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -739,4 +739,4 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
 	devm_kfree(&pdev->dev, msp);
 }
 
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 66b080e5de96..97d8e4de29c2 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -315,4 +315,4 @@ static struct platform_driver ux500_pcm_driver = {
 };
 module_platform_driver(ux500_pcm_driver);
 
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
-- 
cgit v1.2.3


From 8994a5e1d2443511e677d62e97d7de3718b71325 Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@gmail.com>
Date: Wed, 13 Jun 2012 14:36:07 +0800
Subject: ASoC: ml26124: Convert to devm_regmap_init_i2c

This fixes a leak if snd_soc_register_codec fails.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ml26124.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 22cb5bf59273..96aa5fa05160 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -638,7 +638,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, priv);
 
-	priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap);
+	priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap);
 	if (IS_ERR(priv->regmap)) {
 		ret = PTR_ERR(priv->regmap);
 		dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret);
@@ -651,10 +651,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
 
 static __devexit int ml26124_i2c_remove(struct i2c_client *client)
 {
-	struct ml26124_priv *priv = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(priv->regmap);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 7a824e214e25a49442fe868dac0af8a904b24f58 Mon Sep 17 00:00:00 2001
From: Zhangfei Gao <zhangfei.gao@marvell.com>
Date: Mon, 11 Jun 2012 18:04:38 +0800
Subject: ASoC: mmp: add audio dma support

mmp-pcm handle audio dma based on soc-dmaengine
Support mmp and pxa910

Signed-off-by: Zhangfei Gao <zhangfei.gao@marvell.com>
Signed-off-by: Leo Yan <leoy@marvell.com>
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig   |   9 ++
 sound/soc/pxa/Makefile  |   2 +
 sound/soc/pxa/mmp-pcm.c | 297 ++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 308 insertions(+)
 create mode 100644 sound/soc/pxa/mmp-pcm.c

(limited to 'sound')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index a0f7d3cfa470..5d76e2971fbe 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -8,6 +8,15 @@ config SND_PXA2XX_SOC
 	  the PXA2xx AC97, I2S or SSP interface. You will also need
 	  to select the audio interfaces to support below.
 
+config SND_MMP_SOC
+	bool "Soc Audio for Marvell MMP chips"
+	depends on ARCH_MMP
+	select SND_SOC_DMAENGINE_PCM
+	select SND_ARM
+	help
+	  Say Y if you want to add support for codecs attached to
+	  the MMP SSPA interface.
+
 config SND_PXA2XX_AC97
 	tristate
 	select SND_AC97_CODEC
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index af357623be9d..f913e9bfce4f 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -3,11 +3,13 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o
 snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
 snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
 snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
 obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
 obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 000000000000..73ac5463c9e4
--- /dev/null
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,297 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/mmp_audio.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/sram.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+	int ssp_id;
+	struct resource *dma_res;
+};
+
+#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP |	\
+		SNDRV_PCM_INFO_MMAP_VALID |	\
+		SNDRV_PCM_INFO_INTERLEAVED |	\
+		SNDRV_PCM_INFO_PAUSE |		\
+		SNDRV_PCM_INFO_RESUME)
+
+#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+			 SNDRV_PCM_FMTBIT_S24_LE | \
+			 SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+	{
+		.info			= MMP_PCM_INFO,
+		.formats		= MMP_PCM_FORMATS,
+		.period_bytes_min	= 1024,
+		.period_bytes_max	= 2048,
+		.periods_min		= 2,
+		.periods_max		= 32,
+		.buffer_bytes_max	= 4096,
+		.fifo_size		= 32,
+	},
+	{
+		.info			= MMP_PCM_INFO,
+		.formats		= MMP_PCM_FORMATS,
+		.period_bytes_min	= 1024,
+		.period_bytes_max	= 2048,
+		.periods_min		= 2,
+		.periods_max		= 32,
+		.buffer_bytes_max	= 4096,
+		.fifo_size		= 32,
+	},
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct pxa2xx_pcm_dma_params *dma_params;
+	struct dma_slave_config slave_config;
+	int ret;
+
+	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dma_params)
+		return 0;
+
+	ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config);
+	if (ret)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		slave_config.dst_addr     = dma_params->dev_addr;
+		slave_config.dst_maxburst = 4;
+	} else {
+		slave_config.src_addr	  = dma_params->dev_addr;
+		slave_config.src_maxburst = 4;
+	}
+
+	ret = dmaengine_slave_config(chan, &slave_config);
+	if (ret)
+		return ret;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+	struct mmp_dma_data *dma_data = param;
+	bool found = false;
+	char *devname;
+
+	devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+		dma_data->ssp_id);
+	if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+		(chan->chan_id == dma_data->dma_res->start)) {
+		found = true;
+	}
+
+	kfree(devname);
+	return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct mmp_dma_data *dma_data;
+	struct resource *r;
+	int ret;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+	if (!r)
+		return -EBUSY;
+
+	snd_soc_set_runtime_hwparams(substream,
+				&mmp_pcm_hardware[substream->stream]);
+	dma_data = devm_kzalloc(&pdev->dev,
+			sizeof(struct mmp_dma_data), GFP_KERNEL);
+	if (dma_data == NULL)
+		return -ENOMEM;
+
+	dma_data->dma_res = r;
+	dma_data->ssp_id = cpu_dai->id;
+
+	ret = snd_dmaengine_pcm_open(substream, filter, dma_data);
+	if (ret) {
+		devm_kfree(&pdev->dev, dma_data);
+		return ret;
+	}
+
+	snd_dmaengine_pcm_set_data(substream, dma_data);
+	return 0;
+}
+
+static int mmp_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+
+	snd_dmaengine_pcm_close(substream);
+	devm_kfree(&pdev->dev, dma_data);
+	return 0;
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+			 struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned long off = vma->vm_pgoff;
+
+	vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+	return remap_pfn_range(vma, vma->vm_start,
+		__phys_to_pfn(runtime->dma_addr) + off,
+		vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops mmp_pcm_ops = {
+	.open		= mmp_pcm_open,
+	.close		= mmp_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= mmp_pcm_hw_params,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+	struct gen_pool *gpool;
+
+	gpool = sram_get_gpool("asram");
+	if (!gpool)
+		return;
+
+	for (stream = 0; stream < 2; stream++) {
+		size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+		gen_pool_free(gpool, (unsigned long)buf->area, size);
+		buf->area = NULL;
+	}
+
+	return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+								int stream)
+{
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+	struct gen_pool *gpool;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = substream->pcm->card->dev;
+	buf->private_data = NULL;
+
+	gpool = sram_get_gpool("asram");
+	if (!gpool)
+		return -ENOMEM;
+
+	buf->area = (unsigned char *)gen_pool_alloc(gpool, size);
+	if (!buf->area)
+		return -ENOMEM;
+	buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area);
+	buf->bytes = size;
+	return 0;
+}
+
+int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret = 0, stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+
+		ret = mmp_pcm_preallocate_dma_buffer(substream,	stream);
+		if (ret)
+			goto err;
+	}
+
+	return 0;
+
+err:
+	mmp_pcm_free_dma_buffers(pcm);
+	return ret;
+}
+
+struct snd_soc_platform_driver mmp_soc_platform = {
+	.ops		= &mmp_pcm_ops,
+	.pcm_new	= mmp_pcm_new,
+	.pcm_free	= mmp_pcm_free_dma_buffers,
+};
+
+static __devinit int mmp_pcm_probe(struct platform_device *pdev)
+{
+	struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+	if (pdata) {
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+						pdata->buffer_max_playback;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+						pdata->period_max_playback;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+						pdata->buffer_max_capture;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+						pdata->period_max_capture;
+	}
+	return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int __devexit mmp_pcm_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+	.driver = {
+		.name = "mmp-pcm-audio",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = mmp_pcm_probe,
+	.remove = __devexit_p(mmp_pcm_remove),
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From fa375d42f0e531b7ca4316ea9fd5444e01d585e8 Mon Sep 17 00:00:00 2001
From: Zhangfei Gao <zhangfei.gao@marvell.com>
Date: Mon, 11 Jun 2012 18:04:39 +0800
Subject: ASoC: mmp: add sspa support

The SSPA is a configurable multi-channel audio serial (TDM) interface.
It's configurable at runtime to support up to 128 channels and the
number of bits per sample: 8, 12, 16, 20, 24 and 32 bits. It also
support stereo format: I2S, left-justified or right-justified.

Signed-off-by: Zhangfei Gao <zhangfei.gao@marvell.com>
Signed-off-by: Leo Yan <leoy@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig    |   3 +
 sound/soc/pxa/Makefile   |   2 +
 sound/soc/pxa/mmp-sspa.c | 480 +++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/pxa/mmp-sspa.h |  92 +++++++++
 4 files changed, 577 insertions(+)
 create mode 100644 sound/soc/pxa/mmp-sspa.c
 create mode 100644 sound/soc/pxa/mmp-sspa.h

(limited to 'sound')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 5d76e2971fbe..6c3d00b8ea0b 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -35,6 +35,9 @@ config SND_PXA_SOC_SSP
 	tristate
 	select PXA_SSP
 
+config SND_MMP_SOC_SSPA
+	tristate
+
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
 	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index f913e9bfce4f..07b841746fc3 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -4,12 +4,14 @@ snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
 snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
 snd-soc-pxa-ssp-objs := pxa-ssp.o
 snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
 obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
 obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 000000000000..4d6cb8a30fc8
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,480 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+	struct ssp_device *sspa;
+	struct pxa2xx_pcm_dma_params *dma_params;
+	struct clk *audio_clk;
+	struct clk *sysclk;
+	int dai_fmt;
+	int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+	__raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+	return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+	sspa_sp |= SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+	sspa_sp &= ~SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+	sspa_sp |= SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+	sspa_sp &= ~SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_enable(priv->sysclk);
+	clk_enable(priv->sspa->clk);
+
+	return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable(priv->sspa->clk);
+	clk_disable(priv->sysclk);
+
+	return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+				    int clk_id, unsigned int freq, int dir)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (clk_id) {
+	case MMP_SSPA_CLK_AUDIO:
+		ret = clk_set_rate(priv->audio_clk, freq);
+		if (ret)
+			return ret;
+		break;
+	case MMP_SSPA_CLK_PLL:
+	case MMP_SSPA_CLK_VCXO:
+		/* not support yet */
+		return -EINVAL;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+				 int source, unsigned int freq_in,
+				 unsigned int freq_out)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (pll_id) {
+	case MMP_SYSCLK:
+		ret = clk_set_rate(priv->sysclk, freq_out);
+		if (ret)
+			return ret;
+		break;
+	case MMP_SSPA_CLK:
+		ret = clk_set_rate(priv->sspa->clk, freq_out);
+		if (ret)
+			return ret;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+				 unsigned int fmt)
+{
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	u32 sspa_sp, sspa_ctrl;
+
+	/* check if we need to change anything at all */
+	if (sspa_priv->dai_fmt == fmt)
+		return 0;
+
+	/* we can only change the settings if the port is not in use */
+	if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+	    (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+		dev_err(&sspa->pdev->dev,
+			"can't change hardware dai format: stream is in use\n");
+		return -EINVAL;
+	}
+
+	/* reset port settings */
+	sspa_sp   = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+	sspa_ctrl = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		sspa_sp |= SSPA_SP_MSL;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		sspa_sp |= SSPA_SP_FSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sspa_sp |= SSPA_TXSP_FPER(63);
+		sspa_sp |= SSPA_SP_FWID(31);
+		sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+	sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+	/*
+	 * FIXME: hw issue, for the tx serial port,
+	 * can not config the master/slave mode;
+	 * so must clean this bit.
+	 * The master/slave mode has been set in the
+	 * rx port.
+	 */
+	sspa_sp &= ~SSPA_SP_MSL;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+	mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+	mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	sspa_priv->dai_fmt = fmt;
+	return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_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_dai *cpu_dai = rtd->cpu_dai;
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	struct pxa2xx_pcm_dma_params *dma_params;
+	u32 sspa_ctrl;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+	else
+		sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+	sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+	sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+	sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+	sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+	sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+		mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+	} else {
+		mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+		mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+	}
+
+	dma_params = &sspa_priv->dma_params[substream->stream];
+	dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				(sspa->phys_base + SSPA_TXD) :
+				(sspa->phys_base + SSPA_RXD);
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+	return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+			     struct snd_soc_dai *dai)
+{
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		/*
+		 * whatever playback or capture, must enable rx.
+		 * this is a hw issue, so need check if rx has been
+		 * enabled or not; if has been enabled by another
+		 * stream, do not enable again.
+		 */
+		if (!sspa_priv->running_cnt)
+			mmp_sspa_rx_enable(sspa);
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mmp_sspa_tx_enable(sspa);
+
+		sspa_priv->running_cnt++;
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		sspa_priv->running_cnt--;
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mmp_sspa_tx_disable(sspa);
+
+		/* have no capture stream, disable rx port */
+		if (!sspa_priv->running_cnt)
+			mmp_sspa_rx_disable(sspa);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+	snd_soc_dai_set_drvdata(dai, priv);
+	return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+		SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+	.startup	= mmp_sspa_startup,
+	.shutdown	= mmp_sspa_shutdown,
+	.trigger	= mmp_sspa_trigger,
+	.hw_params	= mmp_sspa_hw_params,
+	.set_sysclk	= mmp_sspa_set_dai_sysclk,
+	.set_pll	= mmp_sspa_set_dai_pll,
+	.set_fmt	= mmp_sspa_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver mmp_sspa_dai = {
+	.probe = mmp_sspa_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 128,
+		.rates = MMP_SSPA_RATES,
+		.formats = MMP_SSPA_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = MMP_SSPA_RATES,
+		.formats = MMP_SSPA_FORMATS,
+	},
+	.ops = &mmp_sspa_dai_ops,
+};
+
+static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+	struct sspa_priv *priv;
+	struct resource *res;
+
+	priv = devm_kzalloc(&pdev->dev,
+				sizeof(struct sspa_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->sspa = devm_kzalloc(&pdev->dev,
+				sizeof(struct ssp_device), GFP_KERNEL);
+	if (priv->sspa == NULL)
+		return -ENOMEM;
+
+	priv->dma_params = devm_kzalloc(&pdev->dev,
+			2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL);
+	if (priv->dma_params == NULL)
+		return -ENOMEM;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (res == NULL)
+		return -ENOMEM;
+
+	priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res);
+	if (priv->sspa->mmio_base == NULL)
+		return -ENODEV;
+
+	priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+	if (IS_ERR(priv->sspa->clk))
+		return PTR_ERR(priv->sspa->clk);
+
+	priv->audio_clk = clk_get(NULL, "mmp-audio");
+	if (IS_ERR(priv->audio_clk))
+		return PTR_ERR(priv->audio_clk);
+
+	priv->sysclk = clk_get(NULL, "mmp-sysclk");
+	if (IS_ERR(priv->sysclk)) {
+		clk_put(priv->audio_clk);
+		return PTR_ERR(priv->sysclk);
+	}
+	clk_enable(priv->audio_clk);
+	priv->dai_fmt = (unsigned int) -1;
+	platform_set_drvdata(pdev, priv);
+
+	return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai);
+}
+
+static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+	struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+	clk_disable(priv->audio_clk);
+	clk_put(priv->audio_clk);
+	clk_put(priv->sysclk);
+	snd_soc_unregister_dai(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+	.driver = {
+		.name = "mmp-sspa-dai",
+		.owner = THIS_MODULE,
+	},
+	.probe = asoc_mmp_sspa_probe,
+	.remove = __devexit_p(asoc_mmp_sspa_remove),
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 000000000000..ea365cb9e784
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#ifndef _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD		(0x00)
+#define SSPA_RXID		(0x04)
+#define SSPA_RXCTL		(0x08)
+#define SSPA_RXSP		(0x0c)
+#define SSPA_RXFIFO_UL		(0x10)
+#define SSPA_RXINT_MASK		(0x14)
+#define SSPA_RXC		(0x18)
+#define SSPA_RXFIFO_NOFS	(0x1c)
+#define SSPA_RXFIFO_SIZE	(0x20)
+
+#define SSPA_TXD		(0x80)
+#define SSPA_TXID		(0x84)
+#define SSPA_TXCTL		(0x88)
+#define SSPA_TXSP		(0x8c)
+#define SSPA_TXFIFO_LL		(0x90)
+#define SSPA_TXINT_MASK		(0x94)
+#define SSPA_TXC		(0x98)
+#define SSPA_TXFIFO_NOFS	(0x9c)
+#define SSPA_TXFIFO_SIZE	(0xa0)
+
+/* SSPA Control Register */
+#define	SSPA_CTL_XPH		(1 << 31)	/* Read Phase */
+#define	SSPA_CTL_XFIG		(1 << 15)	/* Transmit Zeros when FIFO Empty */
+#define	SSPA_CTL_JST		(1 << 3)	/* Audio Sample Justification */
+#define	SSPA_CTL_XFRLEN2_MASK	(7 << 24)
+#define	SSPA_CTL_XFRLEN2(x)	((x) << 24)	/* Transmit Frame Length in Phase 2 */
+#define	SSPA_CTL_XWDLEN2_MASK	(7 << 21)
+#define	SSPA_CTL_XWDLEN2(x)	((x) << 21)	/* Transmit Word Length in Phase 2 */
+#define	SSPA_CTL_XDATDLY(x)	((x) << 19)	/* Tansmit Data Delay */
+#define	SSPA_CTL_XSSZ2_MASK	(7 << 16)
+#define	SSPA_CTL_XSSZ2(x)	((x) << 16)	/* Transmit Sample Audio Size */
+#define	SSPA_CTL_XFRLEN1_MASK	(7 << 8)
+#define	SSPA_CTL_XFRLEN1(x)	((x) << 8)	/* Transmit Frame Length in Phase 1 */
+#define	SSPA_CTL_XWDLEN1_MASK	(7 << 5)
+#define	SSPA_CTL_XWDLEN1(x)	((x) << 5)	/* Transmit Word Length in Phase 1 */
+#define	SSPA_CTL_XSSZ1_MASK	(7 << 0)
+#define	SSPA_CTL_XSSZ1(x)	((x) << 0)	/* XSSZ1 */
+
+#define SSPA_CTL_8_BITS		(0x0)		/* Sample Size */
+#define SSPA_CTL_12_BITS	(0x1)
+#define SSPA_CTL_16_BITS	(0x2)
+#define SSPA_CTL_20_BITS	(0x3)
+#define SSPA_CTL_24_BITS	(0x4)
+#define SSPA_CTL_32_BITS	(0x5)
+
+/* SSPA Serial Port Register */
+#define	SSPA_SP_WEN		(1 << 31)	/* Write Configuration Enable */
+#define	SSPA_SP_MSL		(1 << 18)	/* Master Slave Configuration */
+#define	SSPA_SP_CLKP		(1 << 17)	/* CLKP Polarity Clock Edge Select */
+#define	SSPA_SP_FSP		(1 << 16)	/* FSP Polarity Clock Edge Select */
+#define	SSPA_SP_FFLUSH		(1 << 2)	/* FIFO Flush */
+#define	SSPA_SP_S_RST		(1 << 1)	/* Active High Reset Signal */
+#define	SSPA_SP_S_EN		(1 << 0)	/* Serial Clock Domain Enable */
+#define	SSPA_SP_FWID(x)		((x) << 20)	/* Frame-Sync Width */
+#define	SSPA_TXSP_FPER(x)	((x) << 4)	/* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL	0
+#define MMP_SSPA_CLK_VCXO	1
+#define MMP_SSPA_CLK_AUDIO	3
+
+/* sspa pll id */
+#define MMP_SYSCLK		0
+#define MMP_SSPA_CLK		1
+
+#endif /* _MMP_SSPA_H */
-- 
cgit v1.2.3


From 5ebf20ae286a7d2b02551757166247a901d705e5 Mon Sep 17 00:00:00 2001
From: Zhangfei Gao <zhangfei.gao@marvell.com>
Date: Mon, 11 Jun 2012 18:04:40 +0800
Subject: ASoC: add mmp brownstone support

Adds Alsa audio platform driver for mmp brownstone machine

Signed-off-by: Zhangfei Gao <zhangfei.gao@marvell.com>
Signed-off-by: Leo Yan <leoy@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig      |  10 +++
 sound/soc/pxa/Makefile     |   2 +
 sound/soc/pxa/brownstone.c | 174 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 186 insertions(+)
 create mode 100644 sound/soc/pxa/brownstone.c

(limited to 'sound')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6c3d00b8ea0b..d389fd574efe 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -206,3 +206,13 @@ config SND_PXA2XX_SOC_IMOTE2
        help
          Say Y if you want to add support for SoC audio on the
 	 IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+	tristate "SoC Audio support for Marvell Brownstone"
+	depends on SND_MMP_SOC && MACH_BROWNSTONE
+	select SND_MMP_SOC_SSPA
+	select MFD_WM8994
+	select SND_SOC_WM8994
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 07b841746fc3..c12aa2a9bf74 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -32,6 +32,7 @@ snd-soc-mioa701-objs := mioa701_wm9713.o
 snd-soc-z2-objs := z2.o
 snd-soc-imote2-objs := imote2.o
 snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -51,3 +52,4 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
new file mode 100644
index 000000000000..5e666e03d333
--- /dev/null
+++ b/sound/soc/pxa/brownstone.c
@@ -0,0 +1,174 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+	{"Ext Spk", NULL, "SPKOUTLP"},
+	{"Ext Spk", NULL, "SPKOUTLN"},
+	{"Ext Spk", NULL, "SPKOUTRP"},
+	{"Ext Spk", NULL, "SPKOUTRN"},
+
+	{"Headset Stereophone", NULL, "HPOUT1L"},
+	{"Headset Stereophone", NULL, "HPOUT1R"},
+
+	{"IN1RN", NULL, "Headset Mic"},
+
+	{"DMIC1DAT", NULL, "MICBIAS1"},
+	{"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Main Mic");
+
+	/* set endpoints to not connected */
+	snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "IN1LN");
+	snd_soc_dapm_nc_pin(dapm, "IN1LP");
+	snd_soc_dapm_nc_pin(dapm, "IN1RP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
+				       struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int freq_out, sspa_mclk, sysclk;
+	int sspa_div;
+
+	if (params_rate(params) > 11025) {
+		freq_out  = params_rate(params) * 512;
+		sysclk    = params_rate(params) * 256;
+		sspa_mclk = params_rate(params) * 64;
+	} else {
+		freq_out  = params_rate(params) * 1024;
+		sysclk    = params_rate(params) * 512;
+		sspa_mclk = params_rate(params) * 64;
+	}
+	sspa_div = freq_out;
+	do_div(sspa_div, sspa_mclk);
+
+	snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+	snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+	snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+	/* set wm8994 sysclk */
+	snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+	return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+	.hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+	.name		= "WM8994",
+	.stream_name	= "WM8994 HiFi",
+	.cpu_dai_name	= "mmp-sspa-dai.0",
+	.codec_dai_name	= "wm8994-aif1",
+	.platform_name	= "mmp-pcm-audio",
+	.codec_name	= "wm8994-codec",
+	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				SND_SOC_DAIFMT_CBS_CFS,
+	.ops		= &brownstone_ops,
+	.init		= brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+	.name         = "brownstone",
+	.dai_link     = brownstone_wm8994_dai,
+	.num_links    = ARRAY_SIZE(brownstone_wm8994_dai),
+
+	.controls = brownstone_dapm_control,
+	.num_controls = ARRAY_SIZE(brownstone_dapm_control),
+	.dapm_widgets = brownstone_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+	.dapm_routes = brownstone_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int __devinit brownstone_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	brownstone.dev = &pdev->dev;
+	ret = snd_soc_register_card(&brownstone);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+				ret);
+	return ret;
+}
+
+static int __devexit brownstone_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&brownstone);
+	return 0;
+}
+
+static struct platform_driver mmp_driver = {
+	.driver		= {
+		.name	= "brownstone-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= brownstone_probe,
+	.remove		= __devexit_p(brownstone_remove),
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From b883f363495f3d2e237170f6b8814869a3dd16fe Mon Sep 17 00:00:00 2001
From: Qiao Zhou <zhouqiao@marvell.com>
Date: Mon, 11 Jun 2012 18:04:41 +0800
Subject: ASoC: add ttc-dkb machine support

add ttc-dkb machine support for pxa910. It uses 88pm8607 as codec
dai, mmp-pcm as platform and pxa-ssp as cpu dai.

Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/Kconfig   |  20 ++++++
 sound/soc/pxa/Makefile  |   2 +
 sound/soc/pxa/ttc-dkb.c | 173 ++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 195 insertions(+)
 create mode 100644 sound/soc/pxa/ttc-dkb.c

(limited to 'sound')

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index d389fd574efe..4d2e46fae77c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -150,6 +150,26 @@ config SND_SOC_TAVOREVB3
 	  Say Y if you want to add support for SoC audio on the
 	  Marvell Saarb reference platform.
 
+config SND_PXA910_SOC
+	tristate "SoC Audio for Marvell PXA910 chip"
+	depends on ARCH_MMP && SND
+	select SND_PCM
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell PXA910 reference platform.
+
+config SND_SOC_TTC_DKB
+	bool "SoC Audio support for TTC DKB"
+	depends on SND_PXA910_SOC && MACH_TTC_DKB
+	select PXA_SSP
+	select SND_PXA_SOC_SSP
+	select SND_MMP_SOC
+	select MFD_88PM860X
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on TTC DKB
+
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index c12aa2a9bf74..d8a265d2d5d7 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -33,6 +33,7 @@ snd-soc-z2-objs := z2.o
 snd-soc-imote2-objs := imote2.o
 snd-soc-raumfeld-objs := raumfeld.o
 snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -53,3 +54,4 @@ obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
 obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 000000000000..935491a8a770
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,173 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+	return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+	 .name = "88pm860x i2s",
+	 .stream_name = "audio playback",
+	 .codec_name = "88pm860x-codec",
+	 .platform_name = "mmp-pcm-audio",
+	 .cpu_dai_name = "pxa-ssp-dai.1",
+	 .codec_dai_name = "88pm860x-i2s",
+	 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
+	 .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+	.name = "ttc-dkb-hifi",
+	.dai_link = ttc_pm860x_hifi_dai,
+	.num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+	.dapm_widgets = ttc_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+	.dapm_routes = ttc_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int __devinit ttc_dkb_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &ttc_dkb_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit ttc_dkb_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+	.driver		= {
+		.name	= "ttc-dkb-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= ttc_dkb_probe,
+	.remove		= __devexit_p(ttc_dkb_remove),
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");
-- 
cgit v1.2.3


From a9db7dbee0436f0c741c6dfb39ab0241d4131539 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Fri, 8 Jun 2012 12:34:20 -0600
Subject: ASoC: when initializing CPU DAI, don't duplicate any CODEC init

If the CPU-side of a DAI link is a CODEC rather than a standalone DAI,
the codec initialization will call try_module_get() and create the DAI
widgets. Ensure that this isn't duplicated when the CPU DAI itself is
probed, if the CPU DAI is part of a CODEC.

Note that this is not a complete fix on its own, since there's no
guarantee that the CODEC itself will be initialized - currently that only
happens if the CODEC is also used as the CODEC-side of a DAI link, and
that initialization may happen before or after the DAIs within the CODEC
are initialized. However, such a scenario doesn't necessarily currently
work, and I don't think this change alone makes it any worse. This is
fixed in a couple patches time.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 14 +++++++++-----
 1 file changed, 9 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3d803f3cd272..448d4a7c09dd 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -983,7 +983,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 		}
 		cpu_dai->probed = 0;
 		list_del(&cpu_dai->card_list);
-		module_put(cpu_dai->dev->driver->owner);
+
+		if (!cpu_dai->codec)
+			module_put(cpu_dai->dev->driver->owner);
 	}
 }
 
@@ -1257,11 +1259,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 	/* probe the cpu_dai */
 	if (!cpu_dai->probed &&
 			cpu_dai->driver->probe_order == order) {
-		cpu_dai->dapm.card = card;
-		if (!try_module_get(cpu_dai->dev->driver->owner))
-			return -ENODEV;
+		if (!cpu_dai->codec) {
+			cpu_dai->dapm.card = card;
+			if (!try_module_get(cpu_dai->dev->driver->owner))
+				return -ENODEV;
 
-		snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+			snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+		}
 
 		if (cpu_dai->driver->probe) {
 			ret = cpu_dai->driver->probe(cpu_dai);
-- 
cgit v1.2.3


From 18d756440e8ed292b772682c4440a9d33643225b Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Fri, 8 Jun 2012 12:34:21 -0600
Subject: ASoC: when removing a CPU DAI, clean up its DAPM context

When a standalone CPU DAI (one not part of a CODEC) is probed, widgets
are created for it. Add a call to snd_soc_dapm_free() in order to clean
these up when the CPU DAI is removed.

In order for snd_soc_dapm_free() to work, the CPU DAI's DAPM context's
list member must be initialized, since snd_soc_dapm_free() removes that
from the list it's part of. Add it to the card's list of DAPM contexts.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 448d4a7c09dd..621c5bdea483 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -984,8 +984,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 		cpu_dai->probed = 0;
 		list_del(&cpu_dai->card_list);
 
-		if (!cpu_dai->codec)
+		if (!cpu_dai->codec) {
+			snd_soc_dapm_free(&cpu_dai->dapm);
 			module_put(cpu_dai->dev->driver->owner);
+		}
 	}
 }
 
@@ -1264,6 +1266,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 			if (!try_module_get(cpu_dai->dev->driver->owner))
 				return -ENODEV;
 
+			list_add(&cpu_dai->dapm.list, &card->dapm_list);
 			snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
 		}
 
-- 
cgit v1.2.3


From d12cd198cba7949c70f596296297b772063175c0 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Fri, 8 Jun 2012 12:34:22 -0600
Subject: ASoC: factor out soc_remove_platform()

This change simply factors out part of soc_remove_dai_link() into a
standalone function. This makes platform and CODEC removal much more
similar at the call-sites.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 40 ++++++++++++++++++++++++----------------
 1 file changed, 24 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 621c5bdea483..a539ade477af 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -898,6 +898,28 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	return 0;
 }
 
+static int soc_remove_platform(struct snd_soc_platform *platform)
+{
+	int ret;
+
+	if (platform->driver->remove) {
+		ret = platform->driver->remove(platform);
+		if (ret < 0)
+			pr_err("asoc: failed to remove %s: %d\n",
+				platform->name, ret);
+	}
+
+	/* Make sure all DAPM widgets are freed */
+	snd_soc_dapm_free(&platform->dapm);
+
+	soc_cleanup_platform_debugfs(platform);
+	platform->probed = 0;
+	list_del(&platform->card_list);
+	module_put(platform->dev->driver->owner);
+
+	return 0;
+}
+
 static void soc_remove_codec(struct snd_soc_codec *codec)
 {
 	int err;
@@ -950,22 +972,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 
 	/* remove the platform */
 	if (platform && platform->probed &&
-			platform->driver->remove_order == order) {
-		if (platform->driver->remove) {
-			err = platform->driver->remove(platform);
-			if (err < 0)
-				pr_err("asoc: failed to remove %s: %d\n",
-							platform->name, err);
-		}
-
-		/* Make sure all DAPM widgets are freed */
-		snd_soc_dapm_free(&platform->dapm);
-
-		soc_cleanup_platform_debugfs(platform);
-		platform->probed = 0;
-		list_del(&platform->card_list);
-		module_put(platform->dev->driver->owner);
-	}
+			platform->driver->remove_order == order)
+		soc_remove_platform(platform);
 
 	/* remove the CODEC */
 	if (codec && codec->probed &&
-- 
cgit v1.2.3


From 62ae68fa5d6d6f93d8ca8d00e21ad7ac410f9d58 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Fri, 8 Jun 2012 12:34:23 -0600
Subject: ASoC: probe CODECs and platforms before DAIs and links

soc_probe_dai_link() currently inter-mixes the probing of CODECs,
platforms, and DAIs. This can lead to problems such as a CODEC's DAI
being probed before the CODEC, if that DAI is used as the CPU-side of
a DAI link without any other of the CODEC's DAIs having been used as
the CODEC-side of any DAI link that was probed earlier.

To solve this, split soc_probe_dai_link() into soc_probe_link_components()
and soc_probe_link_dais(). The former is used to probe all CODECs and
platforms used by a card first, and then the latter is used to probe all
the DAIs and links later.

A similar change is made to soc_remove_dai_links().

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 129 +++++++++++++++++++++++++++++++++++++--------------
 1 file changed, 95 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a539ade477af..fe16135250f8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -941,11 +941,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec)
 	module_put(codec->dev->driver->owner);
 }
 
-static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
+static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
 	int err;
 
@@ -970,16 +968,6 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 		list_del(&codec_dai->card_list);
 	}
 
-	/* remove the platform */
-	if (platform && platform->probed &&
-			platform->driver->remove_order == order)
-		soc_remove_platform(platform);
-
-	/* remove the CODEC */
-	if (codec && codec->probed &&
-			codec->driver->remove_order == order)
-		soc_remove_codec(codec);
-
 	/* remove the cpu_dai */
 	if (cpu_dai && cpu_dai->probed &&
 			cpu_dai->driver->remove_order == order) {
@@ -999,6 +987,38 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 	}
 }
 
+static void soc_remove_link_components(struct snd_soc_card *card, int num,
+				       int order)
+{
+	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_codec *codec;
+
+	/* remove the platform */
+	if (platform && platform->probed &&
+	    platform->driver->remove_order == order) {
+		soc_remove_platform(platform);
+	}
+
+	/* remove the CODEC-side CODEC */
+	if (codec_dai) {
+		codec = codec_dai->codec;
+		if (codec && codec->probed &&
+		    codec->driver->remove_order == order)
+			soc_remove_codec(codec);
+	}
+
+	/* remove any CPU-side CODEC */
+	if (cpu_dai) {
+		codec = cpu_dai->codec;
+		if (codec && codec->probed &&
+		    codec->driver->remove_order == order)
+			soc_remove_codec(codec);
+	}
+}
+
 static void soc_remove_dai_links(struct snd_soc_card *card)
 {
 	int dai, order;
@@ -1006,8 +1026,15 @@ static void soc_remove_dai_links(struct snd_soc_card *card)
 	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
 			order++) {
 		for (dai = 0; dai < card->num_rtd; dai++)
-			soc_remove_dai_link(card, dai, order);
+			soc_remove_link_dais(card, dai, order);
+	}
+
+	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+			order++) {
+		for (dai = 0; dai < card->num_rtd; dai++)
+			soc_remove_link_components(card, dai, order);
 	}
+
 	card->num_rtd = 0;
 }
 
@@ -1244,7 +1271,44 @@ out:
 	return 0;
 }
 
-static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
+static int soc_probe_link_components(struct snd_soc_card *card, int num,
+				     int order)
+{
+	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_platform *platform = rtd->platform;
+	int ret;
+
+	/* probe the CPU-side component, if it is a CODEC */
+	if (cpu_dai->codec &&
+	    !cpu_dai->codec->probed &&
+	    cpu_dai->codec->driver->probe_order == order) {
+		ret = soc_probe_codec(card, cpu_dai->codec);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* probe the CODEC-side component */
+	if (!codec_dai->codec->probed &&
+	    codec_dai->codec->driver->probe_order == order) {
+		ret = soc_probe_codec(card, codec_dai->codec);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* probe the platform */
+	if (!platform->probed &&
+	    platform->driver->probe_order == order) {
+		ret = soc_probe_platform(card, platform);
+		if (ret < 0)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_dai_link *dai_link = &card->dai_link[num];
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1292,22 +1356,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 		list_add(&cpu_dai->card_list, &card->dai_dev_list);
 	}
 
-	/* probe the CODEC */
-	if (!codec->probed &&
-			codec->driver->probe_order == order) {
-		ret = soc_probe_codec(card, codec);
-		if (ret < 0)
-			return ret;
-	}
-
-	/* probe the platform */
-	if (!platform->probed &&
-			platform->driver->probe_order == order) {
-		ret = soc_probe_platform(card, platform);
-		if (ret < 0)
-			return ret;
-	}
-
 	/* probe the CODEC DAI */
 	if (!codec_dai->probed && codec_dai->driver->probe_order == order) {
 		if (codec_dai->driver->probe) {
@@ -1582,14 +1630,27 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 			goto card_probe_error;
 	}
 
-	/* early DAI link probe */
+	/* probe all components used by DAI links on this card */
 	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
 			order++) {
 		for (i = 0; i < card->num_links; i++) {
-			ret = soc_probe_dai_link(card, i, order);
+			ret = soc_probe_link_components(card, i, order);
 			if (ret < 0) {
 				pr_err("asoc: failed to instantiate card %s: %d\n",
-			       card->name, ret);
+				       card->name, ret);
+				goto probe_dai_err;
+			}
+		}
+	}
+
+	/* probe all DAI links on this card */
+	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+			order++) {
+		for (i = 0; i < card->num_links; i++) {
+			ret = soc_probe_link_dais(card, i, order);
+			if (ret < 0) {
+				pr_err("asoc: failed to instantiate card %s: %d\n",
+				       card->name, ret);
 				goto probe_dai_err;
 			}
 		}
-- 
cgit v1.2.3


From 80c8bfbe76869bfd6bdf3d260d316e7a32f318c3 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Mon, 4 Jun 2012 09:33:51 +0200
Subject: ALSA: HDA: Create phantom jacks for fixed inputs and outputs

PulseAudio sometimes have difficulties knowing that there is a
"Speaker" or "Internal Mic", if they have no individual volume
controls or selectors. As a result, only e g "Headphone" might
be created for a laptop, but no "Speaker".
To help out, create phantom jacks (that are always present,
at least for now) for "Speaker", "Internal Mic" etc, in case we
detect them.
The naming convention is e g "Speaker Phantom Jack".

In order not to pollute the /dev/input namespace with even more
devices, these are added to the kcontrols only, not the input devices.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_jack.c | 59 +++++++++++++++++++++++++++++++++---------------
 sound/pci/hda/hda_jack.h |  1 +
 2 files changed, 42 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 2dd1c113a4c1..60c976f06280 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -127,10 +127,15 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec)
 static void jack_detect_update(struct hda_codec *codec,
 			       struct hda_jack_tbl *jack)
 {
-	if (jack->jack_dirty || !jack->jack_detect) {
+	if (!jack->jack_dirty)
+		return;
+
+	if (jack->phantom_jack)
+		jack->pin_sense = AC_PINSENSE_PRESENCE;
+	else
 		jack->pin_sense = read_pin_sense(codec, jack->nid);
-		jack->jack_dirty = 0;
-	}
+
+	jack->jack_dirty = 0;
 }
 
 /**
@@ -264,8 +269,8 @@ static void hda_free_jack_priv(struct snd_jack *jack)
  * This assigns a jack-detection kctl to the given pin.  The kcontrol
  * will have the given name and index.
  */
-int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
-			  const char *name, int idx)
+static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
+			  const char *name, int idx, bool phantom_jack)
 {
 	struct hda_jack_tbl *jack;
 	struct snd_kcontrol *kctl;
@@ -283,19 +288,30 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 	if (err < 0)
 		return err;
 	jack->kctl = kctl;
+	jack->phantom_jack = !!phantom_jack;
+
 	state = snd_hda_jack_detect(codec, nid);
 	snd_kctl_jack_report(codec->bus->card, kctl, state);
 #ifdef CONFIG_SND_HDA_INPUT_JACK
-	jack->type = get_input_jack_type(codec, nid);
-	err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack);
-	if (err < 0)
-		return err;
-	jack->jack->private_data = jack;
-	jack->jack->private_free = hda_free_jack_priv;
-	snd_jack_report(jack->jack, state ? jack->type : 0);
+	if (!phantom_jack) {
+		jack->type = get_input_jack_type(codec, nid);
+		err = snd_jack_new(codec->bus->card, name, jack->type,
+				   &jack->jack);
+		if (err < 0)
+			return err;
+		jack->jack->private_data = jack;
+		jack->jack->private_free = hda_free_jack_priv;
+		snd_jack_report(jack->jack, state ? jack->type : 0);
+	}
 #endif
 	return 0;
 }
+
+int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
+			  const char *name, int idx)
+{
+	return __snd_hda_jack_add_kctl(codec, nid, name, idx, false);
+}
 EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
 
 static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
@@ -305,25 +321,32 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
 	unsigned int def_conf, conn;
 	char name[44];
 	int idx, err;
+	bool phantom_jack;
 
 	if (!nid)
 		return 0;
-	if (!is_jack_detectable(codec, nid))
-		return 0;
 	def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	conn = get_defcfg_connect(def_conf);
-	if (conn != AC_JACK_PORT_COMPLEX)
+	if (conn == AC_JACK_PORT_NONE)
 		return 0;
+	phantom_jack = (conn != AC_JACK_PORT_COMPLEX) ||
+		       !is_jack_detectable(codec, nid);
 
 	snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
+	if (phantom_jack)
+		/* Example final name: "Internal Mic Phantom Jack" */
+		strncat(name, " Phantom", sizeof(name) - strlen(name) - 1);
 	if (!strcmp(name, lastname) && idx == *lastidx)
 		idx++;
-	strncpy(lastname, name, 44);
+	strncpy(lastname, name, sizeof(name));
 	*lastidx = idx;
-	err = snd_hda_jack_add_kctl(codec, nid, name, idx);
+	err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack);
 	if (err < 0)
 		return err;
-	return snd_hda_jack_detect_enable(codec, nid, 0);
+
+	if (!phantom_jack)
+		return snd_hda_jack_detect_enable(codec, nid, 0);
+	return 0;
 }
 
 /**
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 8ae52465ec5d..a9803da633c0 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -23,6 +23,7 @@ struct hda_jack_tbl {
 	unsigned int pin_sense;		/* cached pin-sense value */
 	unsigned int jack_detect:1;	/* capable of jack-detection? */
 	unsigned int jack_dirty:1;	/* needs to update? */
+	unsigned int phantom_jack:1;    /* a fixed, always present port? */
 	struct snd_kcontrol *kctl;	/* assigned kctl for jack-detection */
 #ifdef CONFIG_SND_HDA_INPUT_JACK
 	int type;
-- 
cgit v1.2.3


From c20c5a841cbe47f5b7812b57bd25397497e5fbc0 Mon Sep 17 00:00:00 2001
From: Seth Heasley <seth.heasley@intel.com>
Date: Thu, 14 Jun 2012 14:23:53 -0700
Subject: ALSA: hda_intel: activate COMBO mode for Intel client chipsets

This patch activates the COMBO position_fix for recent Intel client chipsets.
COMBO mode is the recommended setting for Intel chipsets and eliminates HD
audio warnings in dmesg.  This patch has been tested on Lynx Point, Panther
Point, and Cougar Pont.

Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 11 ++++++++---
 1 file changed, 8 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d49926e4d19f..1a07d2188dd7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -535,6 +535,7 @@ enum {
 #define AZX_DCAPS_BUFSIZE	(1 << 21)	/* no buffer size alignment */
 #define AZX_DCAPS_ALIGN_BUFSIZE	(1 << 22)	/* buffer size alignment */
 #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23)	/* BDLE in 4k boundary */
+#define AZX_DCAPS_POSFIX_COMBO  (1 << 24)	/* Use COMBO as default */
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
 		snd_printd(SFX "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
 	}
+	if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) {
+		snd_printd(SFX "Using COMBO position fix\n");
+		return POS_FIX_COMBO;
+	}
 	return POS_FIX_AUTO;
 }
 
@@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* CPT */
 	{ PCI_DEVICE(0x8086, 0x1c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
-	  AZX_DCAPS_BUFSIZE },
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* PBG */
 	{ PCI_DEVICE(0x8086, 0x1d20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
@@ -3248,11 +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	/* Panther Point */
 	{ PCI_DEVICE(0x8086, 0x1e20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
-	  AZX_DCAPS_BUFSIZE},
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* Lynx Point */
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
-	  AZX_DCAPS_BUFSIZE},
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
-- 
cgit v1.2.3


From 4b6ace9e7176d93f819cec9df47faadaaceead4b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 15 Jun 2012 11:53:32 +0200
Subject: ALSA: hda - Add the support for VIA HDMI pin detection

This patch adds the hotplug unsol event handling to simple_hdmi*().
It works on VIA VX900.  If AMD or Nvidia chips support the
pin-detection similarly, it can be added easily, too.

Reported-by: Annie Liu <annieliu@viatech.com.cn>
Tested-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 20 ++++++++++++++++++++
 1 file changed, 20 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 696681826b01..8e7333b07b58 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1377,6 +1377,19 @@ static int simple_playback_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+/* unsolicited event for jack sensing */
+static void simple_hdmi_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	snd_hda_jack_get_action(codec, res >> AC_UNSOL_RES_TAG_SHIFT);
+	snd_hda_jack_report_sync(codec);
+}
+
+/* generic_hdmi_build_jack can be used for simple_hdmi, too,
+ * as long as spec->pins[] is set correctly
+ */
+#define simple_hdmi_build_jack	generic_hdmi_build_jack
+
 static int simple_playback_build_controls(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -1389,6 +1402,11 @@ static int simple_playback_build_controls(struct hda_codec *codec)
 						    spec->cvts[i].cvt_nid);
 		if (err < 0)
 			return err;
+		if (codec->patch_ops.unsol_event) {
+			err = simple_hdmi_build_jack(codec, i);
+			if (err < 0)
+				return err;
+		}
 	}
 
 	return 0;
@@ -1876,6 +1894,7 @@ static struct hda_verb viahdmi_basic_init[] = {
 static int via_hdmi_init(struct hda_codec *codec)
 {
 	snd_hda_sequence_write(codec, viahdmi_basic_init);
+	snd_hda_jack_report_sync(codec);
 	return 0;
 }
 
@@ -1884,6 +1903,7 @@ static const struct hda_codec_ops via_hdmi_patch_ops = {
 	.build_pcms = simple_playback_build_pcms,
 	.init = via_hdmi_init,
 	.free = simple_playback_free,
+	.unsol_event = simple_hdmi_unsol_event,
 };
 
 static struct hda_pcm_stream via_hdmi_digital_playback = {
-- 
cgit v1.2.3


From 4f0110ced1b5d61e6df4871f6f800a9d3678bf26 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 15 Jun 2012 12:45:43 +0200
Subject: ALSA: hda - Merge ATI/VIA HDMI simple init functions

Just a minor code cleanup to use the same function for both AMD and VIA
simple_hdmi*().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 56 ++++++++++++++++++----------------------------
 1 file changed, 22 insertions(+), 34 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 8e7333b07b58..c9d0c98bbe86 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1412,6 +1412,24 @@ static int simple_playback_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
+static int simple_playback_init(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_pins; i++) {
+		snd_hda_codec_write(codec, spec->pins[i].pin_nid, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		/* some codecs require to unmute the pin */
+		if (get_wcaps(codec, spec->pins[i].pin_nid) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_write(codec, spec->pins[i].pin_nid, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_OUT_UNMUTE);
+	}
+	snd_hda_jack_report_sync(codec);
+	return 0;
+}
+
 static void simple_playback_free(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -1831,29 +1849,10 @@ static const struct hda_pcm_stream atihdmi_pcm_digital_playback = {
 	},
 };
 
-static const struct hda_verb atihdmi_basic_init[] = {
-	/* enable digital output on pin widget */
-	{ 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{} /* terminator */
-};
-
-static int atihdmi_init(struct hda_codec *codec)
-{
-	struct hdmi_spec *spec = codec->spec;
-
-	snd_hda_sequence_write(codec, atihdmi_basic_init);
-	/* SI codec requires to unmute the pin */
-	if (get_wcaps(codec, spec->pins[0].pin_nid) & AC_WCAP_OUT_AMP)
-		snd_hda_codec_write(codec, spec->pins[0].pin_nid, 0,
-				    AC_VERB_SET_AMP_GAIN_MUTE,
-				    AMP_OUT_UNMUTE);
-	return 0;
-}
-
 static const struct hda_codec_ops atihdmi_patch_ops = {
 	.build_controls = simple_playback_build_controls,
 	.build_pcms = simple_playback_build_pcms,
-	.init = atihdmi_init,
+	.init = simple_playback_init,
 	.free = simple_playback_free,
 };
 
@@ -1872,6 +1871,7 @@ static int patch_atihdmi(struct hda_codec *codec)
 	spec->multiout.max_channels = 2;
 	spec->multiout.dig_out_nid = ATIHDMI_CVT_NID;
 	spec->num_cvts = 1;
+	spec->num_pins = 1;
 	spec->cvts[0].cvt_nid = ATIHDMI_CVT_NID;
 	spec->pins[0].pin_nid = ATIHDMI_PIN_NID;
 	spec->pcm_playback = &atihdmi_pcm_digital_playback;
@@ -1885,23 +1885,10 @@ static int patch_atihdmi(struct hda_codec *codec)
 #define VIAHDMI_CVT_NID	0x02	/* audio converter1 */
 #define VIAHDMI_PIN_NID	0x03	/* HDMI output pin1 */
 
-static struct hda_verb viahdmi_basic_init[] = {
-	/* enable digital output on pin widget */
-	{ VIAHDMI_PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{} /* terminator */
-};
-
-static int via_hdmi_init(struct hda_codec *codec)
-{
-	snd_hda_sequence_write(codec, viahdmi_basic_init);
-	snd_hda_jack_report_sync(codec);
-	return 0;
-}
-
 static const struct hda_codec_ops via_hdmi_patch_ops = {
 	.build_controls = simple_playback_build_controls,
 	.build_pcms = simple_playback_build_pcms,
-	.init = via_hdmi_init,
+	.init = simple_playback_init,
 	.free = simple_playback_free,
 	.unsol_event = simple_hdmi_unsol_event,
 };
@@ -1930,6 +1917,7 @@ static int patch_via_hdmi(struct hda_codec *codec)
 	spec->multiout.max_channels = 2;
 	spec->multiout.dig_out_nid = VIAHDMI_CVT_NID; /* pure-digital case */
 	spec->num_cvts = 1;
+	spec->num_pins = 1;
 	spec->cvts[0].cvt_nid = VIAHDMI_CVT_NID;
 	spec->pins[0].pin_nid = VIAHDMI_PIN_NID;
 	spec->pcm_playback = &via_hdmi_digital_playback;
-- 
cgit v1.2.3


From d0b1252dd11549103a97a13aca25737b084c5618 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 15 Jun 2012 14:34:42 +0200
Subject: ALSA: hda - Use common codes for ATI, Nvidia and VIA simple codecs

The code refactoring using the same helper functions for sharing the
codes among ATI, Nvidia and VIA simple_hdmi* stuff.  Except for that
spec->pcm_playback is no longer pointer, the functionality doesn't
change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 191 +++++++++++++++------------------------------
 1 file changed, 65 insertions(+), 126 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index c9d0c98bbe86..6bf784fc8d6d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -85,7 +85,7 @@ struct hdmi_spec {
 	 * Non-generic ATI/NVIDIA specific
 	 */
 	struct hda_multi_out multiout;
-	const struct hda_pcm_stream *pcm_playback;
+	struct hda_pcm_stream pcm_playback;
 };
 
 
@@ -1367,8 +1367,7 @@ static int simple_playback_build_pcms(struct hda_codec *codec)
 		info->name = get_hdmi_pcm_name(i);
 		info->pcm_type = HDA_PCM_TYPE_HDMI;
 		pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
-		snd_BUG_ON(!spec->pcm_playback);
-		*pstr = *spec->pcm_playback;
+		*pstr = spec->pcm_playback;
 		pstr->nid = spec->cvts[i].cvt_nid;
 		if (pstr->channels_max <= 2 && chans && chans <= 16)
 			pstr->channels_max = chans;
@@ -1560,6 +1559,49 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     stream_tag, format, substream);
 }
 
+static const struct hda_pcm_stream simple_pcm_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.open = simple_playback_pcm_open,
+		.close = simple_playback_pcm_close,
+		.prepare = simple_playback_pcm_prepare
+	},
+};
+
+static const struct hda_codec_ops simple_hdmi_patch_ops = {
+	.build_controls = simple_playback_build_controls,
+	.build_pcms = simple_playback_build_pcms,
+	.init = simple_playback_init,
+	.free = simple_playback_free,
+};
+
+static int patch_simple_hdmi(struct hda_codec *codec,
+			     hda_nid_t cvt_nid, hda_nid_t pin_nid)
+{
+	struct hdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	spec->multiout.num_dacs = 0;  /* no analog */
+	spec->multiout.max_channels = 2;
+	spec->multiout.dig_out_nid = cvt_nid;
+	spec->num_cvts = 1;
+	spec->num_pins = 1;
+	spec->cvts[0].cvt_nid = cvt_nid;
+	spec->cvts[0].cvt_nid = pin_nid;
+	spec->pcm_playback = simple_pcm_playback;
+
+	codec->patch_ops = simple_hdmi_patch_ops;
+
+	return 0;
+}
+
 static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec,
 						    int channels)
 {
@@ -1732,54 +1774,20 @@ static const struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = {
 	},
 };
 
-static const struct hda_pcm_stream nvhdmi_pcm_playback_2ch = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = nvhdmi_master_con_nid_7x,
-	.rates = SUPPORTED_RATES,
-	.maxbps = SUPPORTED_MAXBPS,
-	.formats = SUPPORTED_FORMATS,
-	.ops = {
-		.open = simple_playback_pcm_open,
-		.close = simple_playback_pcm_close,
-		.prepare = simple_playback_pcm_prepare
-	},
-};
-
-static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = nvhdmi_7x_init,
-	.free = simple_playback_free,
-};
-
-static const struct hda_codec_ops nvhdmi_patch_ops_2ch = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = nvhdmi_7x_init,
-	.free = simple_playback_free,
-};
-
 static int patch_nvhdmi_2ch(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
+	int err = patch_simple_hdmi(codec, nvhdmi_master_con_nid_7x,
+				    nvhdmi_master_pin_nid_7x);
+	if (err < 0)
+		return err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
-	spec->multiout.num_dacs = 0;  /* no analog */
-	spec->multiout.max_channels = 2;
-	spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x;
-	spec->num_cvts = 1;
-	spec->cvts[0].cvt_nid = nvhdmi_master_con_nid_7x;
-	spec->pcm_playback = &nvhdmi_pcm_playback_2ch;
-
-	codec->patch_ops = nvhdmi_patch_ops_2ch;
-
+	codec->patch_ops.init = nvhdmi_7x_init;
+	/* override the PCM rates, etc, as the codec doesn't give full list */
+	spec = codec->spec;
+	spec->pcm_playback.rates = SUPPORTED_RATES;
+	spec->pcm_playback.maxbps = SUPPORTED_MAXBPS;
+	spec->pcm_playback.formats = SUPPORTED_FORMATS;
 	return 0;
 }
 
@@ -1787,13 +1795,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
 	int err = patch_nvhdmi_2ch(codec);
-
 	if (err < 0)
 		return err;
 	spec = codec->spec;
 	spec->multiout.max_channels = 8;
-	spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x;
-	codec->patch_ops = nvhdmi_patch_ops_8ch_7x;
+	spec->pcm_playback = nvhdmi_pcm_playback_8ch_7x;
 
 	/* Initialize the audio infoframe channel mask and checksum to something
 	 * valid */
@@ -1837,47 +1843,14 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
-static const struct hda_pcm_stream atihdmi_pcm_digital_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = ATIHDMI_CVT_NID,
-	.ops = {
-		.open = simple_playback_pcm_open,
-		.close = simple_playback_pcm_close,
-		.prepare = atihdmi_playback_pcm_prepare
-	},
-};
-
-static const struct hda_codec_ops atihdmi_patch_ops = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = simple_playback_init,
-	.free = simple_playback_free,
-};
-
-
 static int patch_atihdmi(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
-
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
-	spec->multiout.num_dacs = 0;	  /* no analog */
-	spec->multiout.max_channels = 2;
-	spec->multiout.dig_out_nid = ATIHDMI_CVT_NID;
-	spec->num_cvts = 1;
-	spec->num_pins = 1;
-	spec->cvts[0].cvt_nid = ATIHDMI_CVT_NID;
-	spec->pins[0].pin_nid = ATIHDMI_PIN_NID;
-	spec->pcm_playback = &atihdmi_pcm_digital_playback;
-
-	codec->patch_ops = atihdmi_patch_ops;
-
+	int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID);
+	if (err < 0)
+		return err;
+	spec = codec->spec;
+	spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare;
 	return 0;
 }
 
@@ -1885,46 +1858,12 @@ static int patch_atihdmi(struct hda_codec *codec)
 #define VIAHDMI_CVT_NID	0x02	/* audio converter1 */
 #define VIAHDMI_PIN_NID	0x03	/* HDMI output pin1 */
 
-static const struct hda_codec_ops via_hdmi_patch_ops = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = simple_playback_init,
-	.free = simple_playback_free,
-	.unsol_event = simple_hdmi_unsol_event,
-};
-
-static struct hda_pcm_stream via_hdmi_digital_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = VIAHDMI_CVT_NID, /* NID to query formats and rates*/
-	.ops = {
-		.open = simple_playback_pcm_open,
-		.close = simple_playback_pcm_close,
-		.prepare = simple_playback_pcm_prepare
-	},
-};
-
 static int patch_via_hdmi(struct hda_codec *codec)
 {
-	struct hdmi_spec *spec;
-
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	spec->multiout.num_dacs = 0;	  /* no analog */
-	spec->multiout.max_channels = 2;
-	spec->multiout.dig_out_nid = VIAHDMI_CVT_NID; /* pure-digital case */
-	spec->num_cvts = 1;
-	spec->num_pins = 1;
-	spec->cvts[0].cvt_nid = VIAHDMI_CVT_NID;
-	spec->pins[0].pin_nid = VIAHDMI_PIN_NID;
-	spec->pcm_playback = &via_hdmi_digital_playback;
-
-	codec->spec = spec;
-	codec->patch_ops = via_hdmi_patch_ops;
-
+	int err = patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID);
+	if (err < 0)
+		return err;
+	codec->patch_ops.unsol_event = simple_hdmi_unsol_event;
 	return 0;
 }
 
-- 
cgit v1.2.3


From ceaa86ba2ed90780617be76526de975521374595 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 15 Jun 2012 14:38:31 +0200
Subject: ALSA: hda - Remove invalid init verbs for Nvidia 2ch codecs

Nvidia 2ch codecs have no NIDs greather than 0x05.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 23 +++++++++++++++++++----
 1 file changed, 19 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 6bf784fc8d6d..f51a0b5bfcb2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1453,7 +1453,15 @@ static const hda_nid_t nvhdmi_con_nids_7x[4] = {
 	0x6, 0x8, 0xa, 0xc,
 };
 
-static const struct hda_verb nvhdmi_basic_init_7x[] = {
+static const struct hda_verb nvhdmi_basic_init_7x_2ch[] = {
+	/* set audio protect on */
+	{ 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1},
+	/* enable digital output on pin widget */
+	{ 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
+	{} /* terminator */
+};
+
+static const struct hda_verb nvhdmi_basic_init_7x_8ch[] = {
 	/* set audio protect on */
 	{ 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1},
 	/* enable digital output on pin widget */
@@ -1481,9 +1489,15 @@ static const struct hda_verb nvhdmi_basic_init_7x[] = {
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 #endif
 
-static int nvhdmi_7x_init(struct hda_codec *codec)
+static int nvhdmi_7x_init_2ch(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_2ch);
+	return 0;
+}
+
+static int nvhdmi_7x_init_8ch(struct hda_codec *codec)
 {
-	snd_hda_sequence_write(codec, nvhdmi_basic_init_7x);
+	snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_8ch);
 	return 0;
 }
 
@@ -1782,7 +1796,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	codec->patch_ops.init = nvhdmi_7x_init;
+	codec->patch_ops.init = nvhdmi_7x_init_2ch;
 	/* override the PCM rates, etc, as the codec doesn't give full list */
 	spec = codec->spec;
 	spec->pcm_playback.rates = SUPPORTED_RATES;
@@ -1800,6 +1814,7 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec)
 	spec = codec->spec;
 	spec->multiout.max_channels = 8;
 	spec->pcm_playback = nvhdmi_pcm_playback_8ch_7x;
+	codec->patch_ops.init = nvhdmi_7x_init_8ch;
 
 	/* Initialize the audio infoframe channel mask and checksum to something
 	 * valid */
-- 
cgit v1.2.3


From 250e41ac9f31216db1b592bfd77c6a097f10503d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 15 Jun 2012 14:40:21 +0200
Subject: ALSA: hda - Enable unsol event for ATI and Nvidia HDMI codecs too

ATI and Nvidia HDMI codecs have also the pin-detection capability,
so let's enable the jack-detecion for them, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 15 +++++----------
 1 file changed, 5 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index f51a0b5bfcb2..8891fa658382 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1401,11 +1401,9 @@ static int simple_playback_build_controls(struct hda_codec *codec)
 						    spec->cvts[i].cvt_nid);
 		if (err < 0)
 			return err;
-		if (codec->patch_ops.unsol_event) {
-			err = simple_hdmi_build_jack(codec, i);
-			if (err < 0)
-				return err;
-		}
+		err = simple_hdmi_build_jack(codec, i);
+		if (err < 0)
+			return err;
 	}
 
 	return 0;
@@ -1589,6 +1587,7 @@ static const struct hda_codec_ops simple_hdmi_patch_ops = {
 	.build_pcms = simple_playback_build_pcms,
 	.init = simple_playback_init,
 	.free = simple_playback_free,
+	.unsol_event = simple_hdmi_unsol_event,
 };
 
 static int patch_simple_hdmi(struct hda_codec *codec,
@@ -1875,11 +1874,7 @@ static int patch_atihdmi(struct hda_codec *codec)
 
 static int patch_via_hdmi(struct hda_codec *codec)
 {
-	int err = patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID);
-	if (err < 0)
-		return err;
-	codec->patch_ops.unsol_event = simple_hdmi_unsol_event;
-	return 0;
+	return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID);
 }
 
 /*
-- 
cgit v1.2.3


From e0690385a86cac5403a62d91dc146f2508416ded Mon Sep 17 00:00:00 2001
From: Ola Lilja <ola.o.lilja@stericsson.com>
Date: Tue, 12 Jun 2012 08:50:08 +0200
Subject: ASoC: Ux500: Add machine-driver

Add machine-driver for ST-Ericsson U8500 platform, including
support for the AB8500-codec.

Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/ux500/Kconfig         |  11 +
 sound/soc/ux500/Makefile        |   3 +
 sound/soc/ux500/mop500.c        | 113 +++++++++++
 sound/soc/ux500/mop500_ab8500.c | 431 ++++++++++++++++++++++++++++++++++++++++
 sound/soc/ux500/mop500_ab8500.h |  22 ++
 5 files changed, 580 insertions(+)
 create mode 100644 sound/soc/ux500/mop500.c
 create mode 100644 sound/soc/ux500/mop500_ab8500.c
 create mode 100644 sound/soc/ux500/mop500_ab8500.h

(limited to 'sound')

diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
index 1d385150064f..069330d82be5 100644
--- a/sound/soc/ux500/Kconfig
+++ b/sound/soc/ux500/Kconfig
@@ -19,3 +19,14 @@ config SND_SOC_UX500_PLAT_DMA
 	select SND_SOC_DMAENGINE_PCM
 	help
 		Say Y if you want to enable the Ux500 platform-driver.
+
++config SND_SOC_UX500_MACH_MOP500
++	tristate "Machine - MOP500 (Ux500 + AB8500)"
+	depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500
+	select SND_SOC_AB8500_CODEC
+	select SND_SOC_UX500_PLAT_MSP_I2S
+	select SND_SOC_UX500_PLAT_DMA
+	help
+		Select this to enable the MOP500 machine-driver.
+		This will enable platform-drivers for: Ux500
+		This will enable codec-drivers for: AB8500
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
index 4634bf015f62..cce0c11a4d86 100644
--- a/sound/soc/ux500/Makefile
+++ b/sound/soc/ux500/Makefile
@@ -5,3 +5,6 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o
 
 snd-soc-ux500-plat-dma-objs := ux500_pcm.o
 obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o
+
+snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o
+obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
new file mode 100644
index 000000000000..31c4d26d0359
--- /dev/null
+++ b/sound/soc/ux500/mop500.c
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja (ola.o.lilja@stericsson.com)
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <asm/mach-types.h>
+
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/spi/spi.h>
+
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#include "ux500_pcm.h"
+#include "ux500_msp_dai.h"
+
+#include <mop500_ab8500.h>
+
+/* Define the whole MOP500 soundcard, linking platform to the codec-drivers  */
+struct snd_soc_dai_link mop500_dai_links[] = {
+	{
+		.name = "ab8500_0",
+		.stream_name = "ab8500_0",
+		.cpu_dai_name = "ux500-msp-i2s.1",
+		.codec_dai_name = "ab8500-codec-dai.0",
+		.platform_name = "ux500-pcm.0",
+		.codec_name = "ab8500-codec.0",
+		.init = mop500_ab8500_machine_init,
+		.ops = mop500_ab8500_ops,
+	},
+	{
+		.name = "ab8500_1",
+		.stream_name = "ab8500_1",
+		.cpu_dai_name = "ux500-msp-i2s.3",
+		.codec_dai_name = "ab8500-codec-dai.1",
+		.platform_name = "ux500-pcm.0",
+		.codec_name = "ab8500-codec.0",
+		.init = NULL,
+		.ops = mop500_ab8500_ops,
+	},
+};
+
+static struct snd_soc_card mop500_card = {
+	.name = "MOP500-card",
+	.probe = NULL,
+	.dai_link = mop500_dai_links,
+	.num_links = ARRAY_SIZE(mop500_dai_links),
+};
+
+static int __devinit mop500_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	pr_debug("%s: Enter.\n", __func__);
+
+	dev_dbg(&pdev->dev, "%s: Enter.\n", __func__);
+
+	mop500_card.dev = &pdev->dev;
+
+	dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n",
+		__func__, mop500_card.name);
+	platform_set_drvdata(pdev, &mop500_card);
+
+	snd_soc_card_set_drvdata(&mop500_card, NULL);
+
+	dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n",
+		__func__, mop500_card.name, mop500_card.num_links);
+	dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n",
+		__func__, mop500_card.name, mop500_card.dai_link[0].name);
+	dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n",
+		__func__, mop500_card.name,
+		mop500_card.dai_link[0].stream_name);
+
+	ret = snd_soc_register_card(&mop500_card);
+	if (ret)
+		dev_err(&pdev->dev,
+			"Error: snd_soc_register_card failed (%d)!\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit mop500_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *mop500_card = platform_get_drvdata(pdev);
+
+	pr_debug("%s: Enter.\n", __func__);
+
+	snd_soc_unregister_card(mop500_card);
+	mop500_ab8500_remove(mop500_card);
+	
+	return 0;
+}
+
+static struct platform_driver snd_soc_mop500_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "snd-soc-mop500",
+	},
+	.probe = mop500_probe,
+	.remove = __devexit_p(mop500_remove),
+};
+
+module_platform_driver(snd_soc_mop500_driver);
diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c
new file mode 100644
index 000000000000..78cce236693e
--- /dev/null
+++ b/sound/soc/ux500/mop500_ab8500.c
@@ -0,0 +1,431 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <mach/hardware.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "ux500_pcm.h"
+#include "ux500_msp_dai.h"
+#include "../codecs/ab8500-codec.h"
+
+#define TX_SLOT_MONO	0x0008
+#define TX_SLOT_STEREO	0x000a
+#define RX_SLOT_MONO	0x0001
+#define RX_SLOT_STEREO	0x0003
+#define TX_SLOT_8CH	0x00FF
+#define RX_SLOT_8CH	0x00FF
+
+#define DEF_TX_SLOTS	TX_SLOT_STEREO
+#define DEF_RX_SLOTS	RX_SLOT_MONO
+
+#define DRIVERMODE_NORMAL	0
+#define DRIVERMODE_CODEC_ONLY	1
+
+/* Slot configuration */
+static unsigned int tx_slots = DEF_TX_SLOTS;
+static unsigned int rx_slots = DEF_RX_SLOTS;
+
+/* Clocks */
+static const char * const enum_mclk[] = {
+	"SYSCLK",
+	"ULPCLK"
+};
+enum mclk {
+	MCLK_SYSCLK,
+	MCLK_ULPCLK,
+};
+
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk);
+
+/* Private data for machine-part MOP500<->AB8500 */
+struct mop500_ab8500_drvdata {
+	/* Clocks */
+	enum mclk mclk_sel;
+	struct clk *clk_ptr_intclk;
+	struct clk *clk_ptr_sysclk;
+	struct clk *clk_ptr_ulpclk;
+};
+
+static inline const char *get_mclk_str(enum mclk mclk_sel)
+{
+	switch (mclk_sel) {
+	case MCLK_SYSCLK:
+		return "SYSCLK";
+	case MCLK_ULPCLK:
+		return "ULPCLK";
+	default:
+		return "Unknown";
+	}
+}
+
+static int mop500_ab8500_set_mclk(struct device *dev,
+				struct mop500_ab8500_drvdata *drvdata)
+{
+	int status;
+	struct clk *clk_ptr;
+
+	if (IS_ERR(drvdata->clk_ptr_intclk)) {
+		dev_err(dev,
+			"%s: ERROR: intclk not initialized!\n", __func__);
+		return -EIO;
+	}
+
+	switch (drvdata->mclk_sel) {
+	case MCLK_SYSCLK:
+		clk_ptr = drvdata->clk_ptr_sysclk;
+		break;
+	case MCLK_ULPCLK:
+		clk_ptr = drvdata->clk_ptr_ulpclk;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (IS_ERR(clk_ptr)) {
+		dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__,
+			get_mclk_str(drvdata->mclk_sel));
+		return -EIO;
+	}
+
+	status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr);
+	if (status)
+		dev_err(dev,
+			"%s: ERROR: Setting intclk parent to %s failed (ret = %d)!",
+			__func__, get_mclk_str(drvdata->mclk_sel), status);
+	else
+		dev_dbg(dev,
+			"%s: intclk parent changed to %s.\n",
+			__func__, get_mclk_str(drvdata->mclk_sel));
+
+	return status;
+}
+
+/*
+ * Control-events
+ */
+
+static int mclk_input_control_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct mop500_ab8500_drvdata *drvdata =
+				snd_soc_card_get_drvdata(codec->card);
+
+	ucontrol->value.enumerated.item[0] = drvdata->mclk_sel;
+
+	return 0;
+}
+
+static int mclk_input_control_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct mop500_ab8500_drvdata *drvdata =
+				snd_soc_card_get_drvdata(codec->card);
+	unsigned int val = ucontrol->value.enumerated.item[0];
+
+	if (val > (unsigned int)MCLK_ULPCLK)
+		return -EINVAL;
+	if (drvdata->mclk_sel == val)
+		return 0;
+
+	drvdata->mclk_sel = val;
+
+	return 1;
+}
+
+/*
+ * Controls
+ */
+
+static struct snd_kcontrol_new mop500_ab8500_ctrls[] = {
+	SOC_ENUM_EXT("Master Clock Select",
+		soc_enum_mclk,
+		mclk_input_control_get, mclk_input_control_put),
+	/* Digital interface - Clocks */
+	SOC_SINGLE("Digital Interface Master Generator Switch",
+		AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN,
+		1, 0),
+	SOC_SINGLE("Digital Interface 0 Bit-clock Switch",
+		AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0,
+		1, 0),
+	SOC_SINGLE("Digital Interface 1 Bit-clock Switch",
+		AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1,
+		1, 0),
+	SOC_DAPM_PIN_SWITCH("Headset Left"),
+	SOC_DAPM_PIN_SWITCH("Headset Right"),
+	SOC_DAPM_PIN_SWITCH("Earpiece"),
+	SOC_DAPM_PIN_SWITCH("Speaker Left"),
+	SOC_DAPM_PIN_SWITCH("Speaker Right"),
+	SOC_DAPM_PIN_SWITCH("LineOut Left"),
+	SOC_DAPM_PIN_SWITCH("LineOut Right"),
+	SOC_DAPM_PIN_SWITCH("Vibra 1"),
+	SOC_DAPM_PIN_SWITCH("Vibra 2"),
+	SOC_DAPM_PIN_SWITCH("Mic 1"),
+	SOC_DAPM_PIN_SWITCH("Mic 2"),
+	SOC_DAPM_PIN_SWITCH("LineIn Left"),
+	SOC_DAPM_PIN_SWITCH("LineIn Right"),
+	SOC_DAPM_PIN_SWITCH("DMic 1"),
+	SOC_DAPM_PIN_SWITCH("DMic 2"),
+	SOC_DAPM_PIN_SWITCH("DMic 3"),
+	SOC_DAPM_PIN_SWITCH("DMic 4"),
+	SOC_DAPM_PIN_SWITCH("DMic 5"),
+	SOC_DAPM_PIN_SWITCH("DMic 6"),
+};
+
+/* ASoC */
+
+int mop500_ab8500_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* Set audio-clock source */
+	return mop500_ab8500_set_mclk(rtd->card->dev,
+				snd_soc_card_get_drvdata(rtd->card));
+}
+
+void mop500_ab8500_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->card->dev;
+
+	dev_dbg(dev, "%s: Enter\n", __func__);
+
+	/* Reset slots configuration to default(s) */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		tx_slots = DEF_TX_SLOTS;
+	else
+		rx_slots = DEF_RX_SLOTS;
+}
+
+int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct device *dev = rtd->card->dev;
+	unsigned int fmt;
+	int channels, ret = 0, driver_mode, slots;
+	unsigned int sw_codec, sw_cpu;
+	bool is_playback;
+
+	dev_dbg(dev, "%s: Enter\n", __func__);
+
+	dev_dbg(dev, "%s: substream->pcm->name = %s\n"
+		"substream->pcm->id = %s.\n"
+		"substream->name = %s.\n"
+		"substream->number = %d.\n",
+		__func__,
+		substream->pcm->name,
+		substream->pcm->id,
+		substream->name,
+		substream->number);
+
+	channels = params_channels(params);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sw_cpu = 32;
+		break;
+
+	case SNDRV_PCM_FORMAT_S16_LE:
+		sw_cpu = 16;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	/* Setup codec depending on driver-mode */
+	if (channels == 8)
+		driver_mode = DRIVERMODE_CODEC_ONLY;
+	else
+		driver_mode = DRIVERMODE_NORMAL;
+	dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__,
+		(driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY");
+
+	/* Setup format */
+
+	if (driver_mode == DRIVERMODE_NORMAL) {
+		fmt = SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_CBM_CFM |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CONT;
+	} else {
+		fmt = SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_CBM_CFM |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_GATED;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		dev_err(dev,
+			"%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		dev_err(dev,
+			"%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	/* Setup TDM-slots */
+
+	is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	switch (channels) {
+	case 1:
+		slots = 16;
+		tx_slots = (is_playback) ? TX_SLOT_MONO : 0;
+		rx_slots = (is_playback) ? 0 : RX_SLOT_MONO;
+		break;
+	case 2:
+		slots = 16;
+		tx_slots = (is_playback) ? TX_SLOT_STEREO : 0;
+		rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO;
+		break;
+	case 8:
+		slots = 16;
+		tx_slots = (is_playback) ? TX_SLOT_8CH : 0;
+		rx_slots = (is_playback) ? 0 : RX_SLOT_8CH;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (driver_mode == DRIVERMODE_NORMAL)
+		sw_codec = sw_cpu;
+	else
+		sw_codec = 20;
+
+	dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__,
+		tx_slots, rx_slots);
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots,
+				sw_cpu);
+	if (ret)
+		return ret;
+
+	dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__,
+		tx_slots, rx_slots);
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots,
+				sw_codec);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+struct snd_soc_ops mop500_ab8500_ops[] = {
+	{
+		.hw_params = mop500_ab8500_hw_params,
+		.startup = mop500_ab8500_startup,
+		.shutdown = mop500_ab8500_shutdown,
+	}
+};
+
+int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct device *dev = rtd->card->dev;
+	struct mop500_ab8500_drvdata *drvdata;
+	int ret;
+
+	dev_dbg(dev, "%s Enter.\n", __func__);
+
+	/* Create driver private-data struct */
+	drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata),
+			GFP_KERNEL);
+	snd_soc_card_set_drvdata(rtd->card, drvdata);
+
+	/* Setup clocks */
+
+	drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk");
+	if (IS_ERR(drvdata->clk_ptr_sysclk))
+		dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n",
+			__func__);
+	drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk");
+	if (IS_ERR(drvdata->clk_ptr_ulpclk))
+		dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n",
+			__func__);
+	drvdata->clk_ptr_intclk = clk_get(dev, "intclk");
+	if (IS_ERR(drvdata->clk_ptr_intclk))
+		dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n",
+			__func__);
+
+	/* Set intclk default parent to ulpclk */
+	drvdata->mclk_sel = MCLK_ULPCLK;
+	ret = mop500_ab8500_set_mclk(dev, drvdata);
+	if (ret < 0)
+		dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n",
+			__func__);
+
+	drvdata->mclk_sel = MCLK_ULPCLK;
+
+	/* Add controls */
+	ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls,
+			ARRAY_SIZE(mop500_ab8500_ctrls));
+	if (ret < 0) {
+		pr_err("%s: Failed to add machine-controls (%d)!\n",
+				__func__, ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6");
+
+	return ret;
+}
+
+void mop500_ab8500_remove(struct snd_soc_card *card)
+{
+	struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card);
+
+	if (drvdata->clk_ptr_sysclk != NULL)
+		clk_put(drvdata->clk_ptr_sysclk);
+	if (drvdata->clk_ptr_ulpclk != NULL)
+		clk_put(drvdata->clk_ptr_ulpclk);
+	if (drvdata->clk_ptr_intclk != NULL)
+		clk_put(drvdata->clk_ptr_intclk);
+
+	snd_soc_card_set_drvdata(card, drvdata);
+}
diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h
new file mode 100644
index 000000000000..cca5b33964b6
--- /dev/null
+++ b/sound/soc/ux500/mop500_ab8500.h
@@ -0,0 +1,22 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef MOP500_AB8500_H
+#define MOP500_AB8500_H
+
+extern struct snd_soc_ops mop500_ab8500_ops[];
+
+int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime);
+void mop500_ab8500_remove(struct snd_soc_card *card);
+
+#endif
-- 
cgit v1.2.3


From 9f0ed7a7c547efbce2c15b5017744809e9bba23a Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Sat, 16 Jun 2012 16:19:27 +0300
Subject: ASoC: Ux500: unlock on an error path

There is a missing mutex_unlock() here.  The cleanup path also has more
debug output.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/ab8500-codec.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 389dd660b511..3c795921c5f6 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1235,7 +1235,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 		req != ANC_APPLY_IIR) {
 		dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
 			__func__, enum_anc_state[req]);
-		return -EINVAL;
+		status = -EINVAL;
+		goto cleanup;
 	}
 	apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
 	apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
-- 
cgit v1.2.3


From 4be77a530be1ea62574f31c20dd9848e7e2ab0f6 Mon Sep 17 00:00:00 2001
From: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Date: Fri, 15 Jun 2012 16:35:28 +0100
Subject: ALSA: pcm: Add snd_pcm_rate_bit_to_rate()

This is essentially the reverse of snd_pcm_rate_to_rate_bit().

This is generally useful as the Compress API uses the rate bit
directly and it helps to be able to map back to the actual sample
rate.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_misc.c | 18 ++++++++++++++++++
 1 file changed, 18 insertions(+)

(limited to 'sound')

diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 9c9eff9afbac..d4fc1bfbe457 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -488,3 +488,21 @@ unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate)
 	return SNDRV_PCM_RATE_KNOT;
 }
 EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);
+
+/**
+ * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate
+ * @rate_bit: the rate bit to convert
+ *
+ * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag
+ * or 0 for an unknown rate bit
+ */
+unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit)
+{
+	unsigned int i;
+
+	for (i = 0; i < snd_pcm_known_rates.count; i++)
+		if ((1u << i) == rate_bit)
+			return snd_pcm_known_rates.list[i];
+	return 0;
+}
+EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate);
-- 
cgit v1.2.3


From 10a3061accd897c9e4e3821cbd501660ac482497 Mon Sep 17 00:00:00 2001
From: Ondrej Zary <linux@rainbow-software.org>
Date: Fri, 15 Jun 2012 20:55:54 +0200
Subject: ALSA: snd-opti9xx: fixes for MED3931 card (opti931)

MED3931 card did not work (failed with "OPTI chip not found") because
snd-opti9xx gets mc_indir_index from pnp by adding 2 to the pnp-reported port.
It probably works for some cards but not for this one. Datasheet says that
the port is always at 0xe?e so just force the lowest nibble to be 0xe.

Also this card powers up with (ugly) 3D sound enabled. As there's no mixer
control for this, just disable it.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/opti9xx/opti92x-ad1848.c | 17 ++++++++---------
 1 file changed, 8 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index d7ccf28bd66a..ecc68dfe7b54 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -135,7 +135,6 @@ struct snd_opti9xx {
 	unsigned long mc_base_size;
 #ifdef OPTi93X
 	unsigned long mc_indir_index;
-	unsigned long mc_indir_size;
 	struct resource *res_mc_indir;
 	struct snd_wss *codec;
 #endif	/* OPTi93X */
@@ -245,10 +244,8 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip,
 	case OPTi9XX_HW_82C931:
 	case OPTi9XX_HW_82C933:
 		chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d;
-		if (!chip->mc_indir_index) {
+		if (!chip->mc_indir_index)
 			chip->mc_indir_index = 0xe0e;
-			chip->mc_indir_size = 2;
-		}
 		chip->password = 0xe4;
 		chip->pwd_reg = 0;
 		break;
@@ -403,7 +400,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip,
 
 #else	/* OPTi93X */
 	case OPTi9XX_HW_82C931:
-	case OPTi9XX_HW_82C933:
+		/* disable 3D sound (set GPIO1 as output, low) */
+		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
+	case OPTi9XX_HW_82C933: /* FALL THROUGH */
 		/*
 		 * The BTC 1817DW has QS1000 wavetable which is connected
 		 * to the serial digital input of the OPTI931.
@@ -696,8 +695,7 @@ static int __devinit snd_opti9xx_read_check(struct snd_opti9xx *chip)
 		if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)))
 			return 0;
 #else	/* OPTi93X */
-	chip->res_mc_indir = request_region(chip->mc_indir_index,
-					    chip->mc_indir_size,
+	chip->res_mc_indir = request_region(chip->mc_indir_index, 2,
 					    "OPTi93x MC");
 	if (chip->res_mc_indir == NULL)
 		return -EBUSY;
@@ -770,8 +768,9 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip,
 #ifdef OPTi93X
 	port = pnp_port_start(pdev, 0) - 4;
 	fm_port = pnp_port_start(pdev, 1) + 8;
-	chip->mc_indir_index = pnp_port_start(pdev, 3) + 2;
-	chip->mc_indir_size = pnp_port_len(pdev, 3) - 2;
+	/* adjust mc_indir_index - some cards report it at 0xe?d,
+	   other at 0xe?c but it really is always at 0xe?e */
+	chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe;
 #else
 	devmc = pnp_request_card_device(card, pid->devs[2].id, NULL);
 	if (devmc == NULL)
-- 
cgit v1.2.3


From e0815f35ccd276a903298ad4c3d8d10b65933b86 Mon Sep 17 00:00:00 2001
From: Ezequiel Garcia <elezegarcia@gmail.com>
Date: Mon, 11 Jun 2012 16:58:49 -0300
Subject: ALSA: maestro3: Remove unused AC97 register definitions

Signed-off-by: Ezequiel Garcia <elezegarcia@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/maestro3.c | 68 ----------------------------------------------------
 1 file changed, 68 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index deef21399586..adb3b4c7917e 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -361,74 +361,6 @@ MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)");
 #define DSP2HOST_REQ_I2SRATE    0x02
 #define DSP2HOST_REQ_TIMER      0x04
 
-/* AC97 registers */
-/* XXX fix this crap up */
-/*#define AC97_RESET              0x00*/
-
-#define AC97_VOL_MUTE_B         0x8000
-#define AC97_VOL_M              0x1F
-#define AC97_LEFT_VOL_S         8
-
-#define AC97_MASTER_VOL         0x02
-#define AC97_LINE_LEVEL_VOL     0x04
-#define AC97_MASTER_MONO_VOL    0x06
-#define AC97_PC_BEEP_VOL        0x0A
-#define AC97_PC_BEEP_VOL_M      0x0F
-#define AC97_SROUND_MASTER_VOL  0x38
-#define AC97_PC_BEEP_VOL_S      1
-
-/*#define AC97_PHONE_VOL          0x0C
-#define AC97_MIC_VOL            0x0E*/
-#define AC97_MIC_20DB_ENABLE    0x40
-
-/*#define AC97_LINEIN_VOL         0x10
-#define AC97_CD_VOL             0x12
-#define AC97_VIDEO_VOL          0x14
-#define AC97_AUX_VOL            0x16*/
-#define AC97_PCM_OUT_VOL        0x18
-/*#define AC97_RECORD_SELECT      0x1A*/
-#define AC97_RECORD_MIC         0x00
-#define AC97_RECORD_CD          0x01
-#define AC97_RECORD_VIDEO       0x02
-#define AC97_RECORD_AUX         0x03
-#define AC97_RECORD_MONO_MUX    0x02
-#define AC97_RECORD_DIGITAL     0x03
-#define AC97_RECORD_LINE        0x04
-#define AC97_RECORD_STEREO      0x05
-#define AC97_RECORD_MONO        0x06
-#define AC97_RECORD_PHONE       0x07
-
-/*#define AC97_RECORD_GAIN        0x1C*/
-#define AC97_RECORD_VOL_M       0x0F
-
-/*#define AC97_GENERAL_PURPOSE    0x20*/
-#define AC97_POWER_DOWN_CTRL    0x26
-#define AC97_ADC_READY          0x0001
-#define AC97_DAC_READY          0x0002
-#define AC97_ANALOG_READY       0x0004
-#define AC97_VREF_ON            0x0008
-#define AC97_PR0                0x0100
-#define AC97_PR1                0x0200
-#define AC97_PR2                0x0400
-#define AC97_PR3                0x0800
-#define AC97_PR4                0x1000
-
-#define AC97_RESERVED1          0x28
-
-#define AC97_VENDOR_TEST        0x5A
-
-#define AC97_CLOCK_DELAY        0x5C
-#define AC97_LINEOUT_MUX_SEL    0x0001
-#define AC97_MONO_MUX_SEL       0x0002
-#define AC97_CLOCK_DELAY_SEL    0x1F
-#define AC97_DAC_CDS_SHIFT      6
-#define AC97_ADC_CDS_SHIFT      11
-
-#define AC97_MULTI_CHANNEL_SEL  0x74
-
-/*#define AC97_VENDOR_ID1         0x7C
-#define AC97_VENDOR_ID2         0x7E*/
-
 /*
  * ASSP control regs
  */
-- 
cgit v1.2.3


From 629b15b95d5b12a47791147b7559eacbad04d507 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 18 Jun 2012 21:09:04 +0100
Subject: ASoC: wm5100: Remove stubs of ASoC-level register map code

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5100.c | 9 ---------
 1 file changed, 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 3823af362912..f4817292ef45 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2378,13 +2378,6 @@ static int wm5100_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int wm5100_soc_volatile(struct snd_soc_codec *codec,
-			       unsigned int reg)
-{
-	return true;
-}
-
-
 static struct snd_soc_codec_driver soc_codec_dev_wm5100 = {
 	.probe =	wm5100_probe,
 	.remove =	wm5100_remove,
@@ -2392,8 +2385,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = {
 	.set_sysclk = wm5100_set_sysclk,
 	.set_pll = wm5100_set_fll,
 	.idle_bias_off = 1,
-	.reg_cache_size = WM5100_MAX_REGISTER,
-	.volatile_register = wm5100_soc_volatile,
 
 	.seq_notifier = wm5100_seq_notifier,
 	.controls = wm5100_snd_controls,
-- 
cgit v1.2.3


From d7dc9e32ae64b5db777017344da61a285c2347f8 Mon Sep 17 00:00:00 2001
From: Markus Bollinger <bollinger@digigram.com>
Date: Wed, 20 Jun 2012 08:32:48 +0200
Subject: ALSA: pcxhr: Fix a counter wrap

fix a counter wrap to avoid resynchronization of stream positions every several
minutes. The resynchronization may create stream position jitter

Signed-off-by: Markus Bollinger <bollinger@digigram.com>

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr_core.c | 23 +++++++++++++++--------
 1 file changed, 15 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 304411c1fe4b..841703b5c52a 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -1133,13 +1133,12 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr,
 	hw_sample_count = ((u_int64_t)rmh.stat[0]) << 24;
 	hw_sample_count += (u_int64_t)rmh.stat[1];
 
-	snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n",
+	snd_printdd("stream %c%d : abs samples real(%llu) timer(%llu)\n",
 		    stream->pipe->is_capture ? 'C' : 'P',
 		    stream->substream->number,
-		    (long unsigned int)hw_sample_count,
-		    (long unsigned int)(stream->timer_abs_periods +
-					stream->timer_period_frag +
-					mgr->granularity));
+		    hw_sample_count,
+		    stream->timer_abs_periods + stream->timer_period_frag +
+						mgr->granularity);
 	return hw_sample_count;
 }
 
@@ -1243,10 +1242,18 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
 
 		if ((dsp_time_diff < 0) &&
 		    (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) {
-			snd_printdd("ERROR DSP TIME old(%d) new(%d) -> "
-				    "resynchronize all streams\n",
+			/* handle dsp counter wraparound without resync */
+			int tmp_diff = dsp_time_diff + PCXHR_DSP_TIME_MASK + 1;
+			snd_printdd("WARNING DSP timestamp old(%d) new(%d)",
 				    mgr->dsp_time_last, dsp_time_new);
-			mgr->dsp_time_err++;
+			if (tmp_diff > 0 && tmp_diff <= (2*mgr->granularity)) {
+				snd_printdd("-> timestamp wraparound OK: "
+					    "diff=%d\n", tmp_diff);
+				dsp_time_diff = tmp_diff;
+			} else {
+				snd_printdd("-> resynchronize all streams\n");
+				mgr->dsp_time_err++;
+			}
 		}
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (dsp_time_diff == 0)
-- 
cgit v1.2.3


From fdfbaf69e0b9d6843c0da6501caf4ecacec0f775 Mon Sep 17 00:00:00 2001
From: Markus Bollinger <bollinger@digigram.com>
Date: Wed, 20 Jun 2012 08:34:40 +0200
Subject: ALSA: pcxhr: Add LTC support

add LTC (linear timecode) read function via proc interface to the pcxhr driver

Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.c       | 63 +++++++++++++++++++++++++++++++++++++++++++
 sound/pci/pcxhr/pcxhr.h       |  1 +
 sound/pci/pcxhr/pcxhr_core.c  |  4 +++
 sound/pci/pcxhr/pcxhr_core.h  |  4 ++-
 sound/pci/pcxhr/pcxhr_mix22.c | 11 ++++++++
 sound/pci/pcxhr/pcxhr_mix22.h |  1 +
 6 files changed, 83 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 0435f45e9513..e3ac1f768ff6 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1368,6 +1368,67 @@ static void pcxhr_proc_gpo_write(struct snd_info_entry *entry,
 	}
 }
 
+/* Access to the results of the CMD_GET_TIME_CODE RMH */
+#define TIME_CODE_VALID_MASK	0x00800000
+#define TIME_CODE_NEW_MASK	0x00400000
+#define TIME_CODE_BACK_MASK	0x00200000
+#define TIME_CODE_WAIT_MASK	0x00100000
+
+/* Values for the CMD_MANAGE_SIGNAL RMH */
+#define MANAGE_SIGNAL_TIME_CODE	0x01
+#define MANAGE_SIGNAL_MIDI	0x02
+
+/* linear time code read proc*/
+static void pcxhr_proc_ltc(struct snd_info_entry *entry,
+			   struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	struct pcxhr_rmh rmh;
+	unsigned int ltcHrs, ltcMin, ltcSec, ltcFrm;
+	int err;
+	/* commands available when embedded DSP is running */
+	if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) {
+		snd_iprintf(buffer, "no firmware loaded\n");
+		return;
+	}
+	if (!mgr->capture_ltc) {
+		pcxhr_init_rmh(&rmh, CMD_MANAGE_SIGNAL);
+		rmh.cmd[0] |= MANAGE_SIGNAL_TIME_CODE;
+		err = pcxhr_send_msg(mgr, &rmh);
+		if (err) {
+			snd_iprintf(buffer, "ltc not activated (%d)\n", err);
+			return;
+		}
+		if (mgr->is_hr_stereo)
+			hr222_manage_timecode(mgr, 1);
+		else
+			pcxhr_write_io_num_reg_cont(mgr, REG_CONT_VALSMPTE,
+						    REG_CONT_VALSMPTE, NULL);
+		mgr->capture_ltc = 1;
+	}
+	pcxhr_init_rmh(&rmh, CMD_GET_TIME_CODE);
+	err = pcxhr_send_msg(mgr, &rmh);
+	if (err) {
+		snd_iprintf(buffer, "ltc read error (err=%d)\n", err);
+		return ;
+	}
+	ltcHrs = 10*((rmh.stat[0] >> 8) & 0x3) + (rmh.stat[0] & 0xf);
+	ltcMin = 10*((rmh.stat[1] >> 16) & 0x7) + ((rmh.stat[1] >> 8) & 0xf);
+	ltcSec = 10*(rmh.stat[1] & 0x7) + ((rmh.stat[2] >> 16) & 0xf);
+	ltcFrm = 10*((rmh.stat[2] >> 8) & 0x3) + (rmh.stat[2] & 0xf);
+
+	snd_iprintf(buffer, "timecode: %02u:%02u:%02u-%02u\n",
+			    ltcHrs, ltcMin, ltcSec, ltcFrm);
+	snd_iprintf(buffer, "raw: 0x%04x%06x%06x\n", rmh.stat[0] & 0x00ffff,
+			    rmh.stat[1] & 0xffffff, rmh.stat[2] & 0xffffff);
+	/*snd_iprintf(buffer, "dsp ref time: 0x%06x%06x\n",
+			    rmh.stat[3] & 0xffffff, rmh.stat[4] & 0xffffff);*/
+	if (!(rmh.stat[0] & TIME_CODE_VALID_MASK)) {
+		snd_iprintf(buffer, "warning: linear timecode not valid\n");
+	}
+}
+
 static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 {
 	struct snd_info_entry *entry;
@@ -1383,6 +1444,8 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 		entry->c.text.write = pcxhr_proc_gpo_write;
 		entry->mode |= S_IWUSR;
 	}
+	if (!snd_card_proc_new(chip->card, "ltc", &entry))
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc);
 }
 /* end of proc interface */
 
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index bda776c49884..a4c602c45173 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -103,6 +103,7 @@ struct pcxhr_mgr {
 	unsigned int board_has_mic:1; /* if 1 the board has microphone input */
 	unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
 	unsigned int mono_capture:1; /* if 1 the board does mono capture */
+	unsigned int capture_ltc:1; /* if 1 the board captures LTC input */
 
 	struct snd_dma_buffer hostport;
 
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 841703b5c52a..b33db1e006e7 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -504,6 +504,8 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = {
 [CMD_FORMAT_STREAM_IN] =		{ 0x870000, 0, RMH_SSIZE_FIXED },
 [CMD_STREAM_SAMPLE_COUNT] =		{ 0x902000, 2, RMH_SSIZE_FIXED },
 [CMD_AUDIO_LEVEL_ADJUST] =		{ 0xc22000, 0, RMH_SSIZE_FIXED },
+[CMD_GET_TIME_CODE] =			{ 0x060000, 5, RMH_SSIZE_FIXED },
+[CMD_MANAGE_SIGNAL] =			{ 0x0f0000, 0, RMH_SSIZE_FIXED },
 };
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
@@ -533,6 +535,8 @@ static char* cmd_names[] = {
 [CMD_FORMAT_STREAM_IN] =		"CMD_FORMAT_STREAM_IN",
 [CMD_STREAM_SAMPLE_COUNT] =		"CMD_STREAM_SAMPLE_COUNT",
 [CMD_AUDIO_LEVEL_ADJUST] =		"CMD_AUDIO_LEVEL_ADJUST",
+[CMD_GET_TIME_CODE] =			"CMD_GET_TIME_CODE",
+[CMD_MANAGE_SIGNAL] =			"CMD_MANAGE_SIGNAL",
 };
 #endif
 
diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
index be0173796cdb..a81ab6b811e7 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -79,6 +79,8 @@ enum {
 	CMD_FORMAT_STREAM_IN,		/* cmd_len >= 4	stat_len = 0 */
 	CMD_STREAM_SAMPLE_COUNT,	/* cmd_len = 2	stat_len = (2 * nb_stream) */
 	CMD_AUDIO_LEVEL_ADJUST,		/* cmd_len = 3	stat_len = 0 */
+	CMD_GET_TIME_CODE,		/* cmd_len = 1  stat_len = 5 */
+	CMD_MANAGE_SIGNAL,		/* cmd_len = 1  stat_len = 0 */
 	CMD_LAST_INDEX
 };
 
@@ -116,7 +118,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh);
 #define IO_NUM_REG_OUT_ANA_LEVEL	20
 #define IO_NUM_REG_IN_ANA_LEVEL		21
 
-
+#define REG_CONT_VALSMPTE		0x000800
 #define REG_CONT_UNMUTE_INPUTS		0x020000
 
 /* parameters used with register IO_NUM_REG_STATUS */
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
index 1cb82c0a9cb3..84fe57626eba 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.c
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -53,6 +53,7 @@
 #define PCXHR_DSP_RESET_DSP	0x01
 #define PCXHR_DSP_RESET_MUTE	0x02
 #define PCXHR_DSP_RESET_CODEC	0x08
+#define PCXHR_DSP_RESET_SMPTE	0x10
 #define PCXHR_DSP_RESET_GPO_OFFSET	5
 #define PCXHR_DSP_RESET_GPO_MASK	0x60
 
@@ -527,6 +528,16 @@ int hr222_write_gpo(struct pcxhr_mgr *mgr, int value)
 	return 0;
 }
 
+int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable)
+{
+	if (enable)
+		mgr->dsp_reset |= PCXHR_DSP_RESET_SMPTE;
+	else
+		mgr->dsp_reset &= ~PCXHR_DSP_RESET_SMPTE;
+
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset);
+	return 0;
+}
 
 int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
 				    int is_capture, int channel)
diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h
index 5a37a0007e8f..5971b9933f41 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.h
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -34,6 +34,7 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr,
 
 int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value);
 int hr222_write_gpo(struct pcxhr_mgr *mgr, int value);
+int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable);
 
 #define HR222_LINE_PLAYBACK_LEVEL_MIN		0	/* -25.5 dB */
 #define HR222_LINE_PLAYBACK_ZERO_LEVEL		51	/* 0.0 dB */
-- 
cgit v1.2.3


From 24fe5ddc5f189606bde68136ac808da37e4dfe04 Mon Sep 17 00:00:00 2001
From: Ezequiel Garcia <elezegarcia@gmail.com>
Date: Tue, 19 Jun 2012 16:20:23 -0300
Subject: sound: swarm_cs4297: Provide definitions for AC97 registers

This patch removes the last usage of linux/ac97_codec.h by
re-defining used AC97 registers.
We can't use sound/ac97_codec.h here, since it is an OSS driver.

Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Ezequiel Garcia <elezegarcia@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/swarm_cs4297a.c | 17 ++++++++++++++++-
 1 file changed, 16 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 09d46484bc1a..7d8803a00b79 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -69,7 +69,6 @@
 #include <linux/sound.h>
 #include <linux/slab.h>
 #include <linux/soundcard.h>
-#include <linux/ac97_codec.h>
 #include <linux/pci.h>
 #include <linux/bitops.h>
 #include <linux/interrupt.h>
@@ -199,6 +198,22 @@ static const char invalid_magic[] =
         }                                          \
 })
 
+/* AC97 registers */
+#define AC97_MASTER_VOL_STEREO   0x0002      /* Line Out		*/
+#define AC97_PCBEEP_VOL          0x000a      /* none			*/
+#define AC97_PHONE_VOL           0x000c      /* TAD Input (mono)	*/
+#define AC97_MIC_VOL             0x000e      /* MIC Input (mono)	*/
+#define AC97_LINEIN_VOL          0x0010      /* Line Input (stereo)	*/
+#define AC97_CD_VOL              0x0012      /* CD Input (stereo)	*/
+#define AC97_AUX_VOL             0x0016      /* Aux Input (stereo)	*/
+#define AC97_PCMOUT_VOL          0x0018      /* Wave Output (stereo)	*/
+#define AC97_RECORD_SELECT       0x001a      /*			*/
+#define AC97_RECORD_GAIN         0x001c
+#define AC97_GENERAL_PURPOSE     0x0020
+#define AC97_3D_CONTROL          0x0022
+#define AC97_POWER_CONTROL       0x0026
+#define AC97_VENDOR_ID1           0x007c
+
 struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs };
 
 typedef struct serdma_descr_s {
-- 
cgit v1.2.3


From 21cd683d318041c63876b4acbebb3f6d9d80597b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 20 Jun 2012 09:19:32 +0200
Subject: ALSA: hda - Fix the pin nid assignment in patch_hdmi.c

This fixes the regression introduced by the commit d0b1252d for
refactoring simple_hdmi*().  The pin NID wasn't assigned correctly.

Reported-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 8891fa658382..72a3b26736ae 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1607,7 +1607,7 @@ static int patch_simple_hdmi(struct hda_codec *codec,
 	spec->num_cvts = 1;
 	spec->num_pins = 1;
 	spec->cvts[0].cvt_nid = cvt_nid;
-	spec->cvts[0].cvt_nid = pin_nid;
+	spec->pins[0].pin_nid = pin_nid;
 	spec->pcm_playback = simple_pcm_playback;
 
 	codec->patch_ops = simple_hdmi_patch_ops;
-- 
cgit v1.2.3


From ccfcf7d151c01969133b5555eed635537c41c944 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 20 Jun 2012 09:30:42 +0200
Subject: ALSA: hda - Add missing snd_hda_jack_detect_enable() for
 simple_hdmi*()

Reported-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 72a3b26736ae..db8f6928f839 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1415,13 +1415,15 @@ static int simple_playback_init(struct hda_codec *codec)
 	int i;
 
 	for (i = 0; i < spec->num_pins; i++) {
-		snd_hda_codec_write(codec, spec->pins[i].pin_nid, 0,
+		hda_nid_t pin = spec->pins[i].pin_nid;
+		snd_hda_codec_write(codec, pin, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
 		/* some codecs require to unmute the pin */
-		if (get_wcaps(codec, spec->pins[i].pin_nid) & AC_WCAP_OUT_AMP)
-			snd_hda_codec_write(codec, spec->pins[i].pin_nid, 0,
+		if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_write(codec, pin, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_UNMUTE);
+		snd_hda_jack_detect_enable(codec, pin, pin);
 	}
 	snd_hda_jack_report_sync(codec);
 	return 0;
-- 
cgit v1.2.3


From d2aae47f804830da904d2454d73959eda4ebb0fd Mon Sep 17 00:00:00 2001
From: Scott Jiang <scott.jiang.linux@gmail.com>
Date: Wed, 20 Jun 2012 17:00:30 -0400
Subject: ASoC: add sport driver for bf6xx soc

The SPORT(Serial Port) module on bf6xx soc has a totally different ip
comparing to bf5xx soc. An individual SPORT module consists of two
independently configurable SPORT halves with identical functionality.
Each SPORT half can be configured for either transmitter or receiver.

Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/Kconfig       |  10 +-
 sound/soc/blackfin/Makefile      |   2 +
 sound/soc/blackfin/bf6xx-sport.c | 422 +++++++++++++++++++++++++++++++++++++++
 sound/soc/blackfin/bf6xx-sport.h |  82 ++++++++
 4 files changed, 513 insertions(+), 3 deletions(-)
 create mode 100644 sound/soc/blackfin/bf6xx-sport.c
 create mode 100644 sound/soc/blackfin/bf6xx-sport.h

(limited to 'sound')

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 9f6bc55fc399..0374a3965095 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,7 +1,8 @@
 config SND_BF5XX_I2S
-	tristate "SoC I2S Audio for the ADI BF5xx chip"
+	tristate "SoC I2S Audio for the ADI Blackfin chip"
 	depends on BLACKFIN
-	select SND_BF5XX_SOC_SPORT
+	select SND_BF5XX_SOC_SPORT if !BF60x
+	select SND_BF6XX_SOC_SPORT if BF60x
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -162,6 +163,9 @@ config SND_BF5XX_SOC_AD1980
 config SND_BF5XX_SOC_SPORT
 	tristate
 
+config SND_BF6XX_SOC_SPORT
+	tristate
+
 config SND_BF5XX_SOC_I2S
 	tristate
 
@@ -173,7 +177,7 @@ config SND_BF5XX_SOC_AC97
 
 config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
-	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
+	depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT)
 	range 0 3 if BF54x
 	range 0 1 if !BF54x
 	default 0
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 1bf86ccaa8de..13b092239a82 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -3,6 +3,7 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o
 snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o
 snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o
 snd-soc-bf5xx-sport-objs := bf5xx-sport.o
+snd-soc-bf6xx-sport-objs := bf6xx-sport.o
 snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o
 snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o
 snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o
@@ -11,6 +12,7 @@ obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o
 obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o
 obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o
 obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o
+obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o
 obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o
 obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
 obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
new file mode 100644
index 000000000000..f19a72b8e0c2
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -0,0 +1,422 @@
+/*
+ * bf6xx_sport.c Analog Devices BF6XX SPORT driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <asm/blackfin.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "bf6xx-sport.h"
+
+int sport_set_tx_params(struct sport_device *sport,
+			struct sport_params *params)
+{
+	if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI)
+		return -EBUSY;
+	sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN;
+	sport->tx_regs->div = params->div;
+	SSYNC();
+	return 0;
+}
+EXPORT_SYMBOL(sport_set_tx_params);
+
+int sport_set_rx_params(struct sport_device *sport,
+			struct sport_params *params)
+{
+	if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI)
+		return -EBUSY;
+	sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN;
+	sport->rx_regs->div = params->div;
+	SSYNC();
+	return 0;
+}
+EXPORT_SYMBOL(sport_set_rx_params);
+
+static int compute_wdsize(size_t wdsize)
+{
+	switch (wdsize) {
+	case 1:
+		return WDSIZE_8 | PSIZE_8;
+	case 2:
+		return WDSIZE_16 | PSIZE_16;
+	default:
+		return WDSIZE_32 | PSIZE_32;
+	}
+}
+
+void sport_tx_start(struct sport_device *sport)
+{
+	set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc);
+	set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN
+			| compute_wdsize(sport->wdsize) | NDSIZE_6);
+	enable_dma(sport->tx_dma_chan);
+	sport->tx_regs->spctl |= SPORT_CTL_SPENPRI;
+	SSYNC();
+}
+EXPORT_SYMBOL(sport_tx_start);
+
+void sport_rx_start(struct sport_device *sport)
+{
+	set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc);
+	set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR
+			| compute_wdsize(sport->wdsize) | NDSIZE_6);
+	enable_dma(sport->rx_dma_chan);
+	sport->rx_regs->spctl |= SPORT_CTL_SPENPRI;
+	SSYNC();
+}
+EXPORT_SYMBOL(sport_rx_start);
+
+void sport_tx_stop(struct sport_device *sport)
+{
+	sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI;
+	SSYNC();
+	disable_dma(sport->tx_dma_chan);
+}
+EXPORT_SYMBOL(sport_tx_stop);
+
+void sport_rx_stop(struct sport_device *sport)
+{
+	sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI;
+	SSYNC();
+	disable_dma(sport->rx_dma_chan);
+}
+EXPORT_SYMBOL(sport_rx_stop);
+
+void sport_set_tx_callback(struct sport_device *sport,
+		void (*tx_callback)(void *), void *tx_data)
+{
+	sport->tx_callback = tx_callback;
+	sport->tx_data = tx_data;
+}
+EXPORT_SYMBOL(sport_set_tx_callback);
+
+void sport_set_rx_callback(struct sport_device *sport,
+		void (*rx_callback)(void *), void *rx_data)
+{
+	sport->rx_callback = rx_callback;
+	sport->rx_data = rx_data;
+}
+EXPORT_SYMBOL(sport_set_rx_callback);
+
+static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
+		size_t fragsize, unsigned int cfg,
+		unsigned int count, size_t wdsize)
+{
+
+	int i;
+
+	for (i = 0; i < fragcount; ++i) {
+		desc[i].next_desc_addr  = &(desc[i + 1]);
+		desc[i].start_addr = (unsigned long)buf + i*fragsize;
+		desc[i].cfg = cfg;
+		desc[i].x_count = count;
+		desc[i].x_modify = wdsize;
+		desc[i].y_count = 0;
+		desc[i].y_modify = 0;
+	}
+
+	/* make circular */
+	desc[fragcount-1].next_desc_addr = desc;
+}
+
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+		int fragcount, size_t fragsize)
+{
+	unsigned int count;
+	unsigned int cfg;
+	dma_addr_t addr;
+
+	count = fragsize/sport->wdsize;
+
+	if (sport->tx_desc)
+		dma_free_coherent(NULL, sport->tx_desc_size,
+				sport->tx_desc, 0);
+
+	sport->tx_desc = dma_alloc_coherent(NULL,
+			fragcount * sizeof(struct dmasg), &addr, 0);
+	sport->tx_desc_size = fragcount * sizeof(struct dmasg);
+	if (!sport->tx_desc)
+		return -ENOMEM;
+
+	sport->tx_buf = buf;
+	sport->tx_fragsize = fragsize;
+	sport->tx_frags = fragcount;
+	cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6;
+
+	setup_desc(sport->tx_desc, buf, fragcount, fragsize,
+			cfg|DMAEN, count, sport->wdsize);
+
+	return 0;
+}
+EXPORT_SYMBOL(sport_config_tx_dma);
+
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+		int fragcount, size_t fragsize)
+{
+	unsigned int count;
+	unsigned int cfg;
+	dma_addr_t addr;
+
+	count = fragsize/sport->wdsize;
+
+	if (sport->rx_desc)
+		dma_free_coherent(NULL, sport->rx_desc_size,
+				sport->rx_desc, 0);
+
+	sport->rx_desc = dma_alloc_coherent(NULL,
+			fragcount * sizeof(struct dmasg), &addr, 0);
+	sport->rx_desc_size = fragcount * sizeof(struct dmasg);
+	if (!sport->rx_desc)
+		return -ENOMEM;
+
+	sport->rx_buf = buf;
+	sport->rx_fragsize = fragsize;
+	sport->rx_frags = fragcount;
+	cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize)
+		| WNR | NDSIZE_6;
+
+	setup_desc(sport->rx_desc, buf, fragcount, fragsize,
+			cfg|DMAEN, count, sport->wdsize);
+
+	return 0;
+}
+EXPORT_SYMBOL(sport_config_rx_dma);
+
+unsigned long sport_curr_offset_tx(struct sport_device *sport)
+{
+	unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan);
+
+	return (unsigned char *)curr - sport->tx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_tx);
+
+unsigned long sport_curr_offset_rx(struct sport_device *sport)
+{
+	unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan);
+
+	return (unsigned char *)curr - sport->rx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_rx);
+
+static irqreturn_t sport_tx_irq(int irq, void *dev_id)
+{
+	struct sport_device *sport = dev_id;
+	static unsigned long status;
+
+	status = get_dma_curr_irqstat(sport->tx_dma_chan);
+	if (status & (DMA_DONE|DMA_ERR)) {
+		clear_dma_irqstat(sport->tx_dma_chan);
+		SSYNC();
+	}
+	if (sport->tx_callback)
+		sport->tx_callback(sport->tx_data);
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t sport_rx_irq(int irq, void *dev_id)
+{
+	struct sport_device *sport = dev_id;
+	unsigned long status;
+
+	status = get_dma_curr_irqstat(sport->rx_dma_chan);
+	if (status & (DMA_DONE|DMA_ERR)) {
+		clear_dma_irqstat(sport->rx_dma_chan);
+		SSYNC();
+	}
+	if (sport->rx_callback)
+		sport->rx_callback(sport->rx_data);
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t sport_err_irq(int irq, void *dev_id)
+{
+	struct sport_device *sport = dev_id;
+	struct device *dev = &sport->pdev->dev;
+
+	if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI)
+		dev_dbg(dev, "sport error: TUVF\n");
+	if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI)
+		dev_dbg(dev, "sport error: ROVF\n");
+
+	return IRQ_HANDLED;
+}
+
+static int sport_get_resource(struct sport_device *sport)
+{
+	struct platform_device *pdev = sport->pdev;
+	struct device *dev = &pdev->dev;
+	struct bfin_snd_platform_data *pdata = dev->platform_data;
+	struct resource *res;
+
+	if (!pdata) {
+		dev_err(dev, "No platform data\n");
+		return -ENODEV;
+	}
+	sport->pin_req = pdata->pin_req;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(dev, "No tx MEM resource\n");
+		return -ENODEV;
+	}
+	sport->tx_regs = (struct sport_register *)res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+	if (!res) {
+		dev_err(dev, "No rx MEM resource\n");
+		return -ENODEV;
+	}
+	sport->rx_regs = (struct sport_register *)res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!res) {
+		dev_err(dev, "No tx DMA resource\n");
+		return -ENODEV;
+	}
+	sport->tx_dma_chan = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!res) {
+		dev_err(dev, "No rx DMA resource\n");
+		return -ENODEV;
+	}
+	sport->rx_dma_chan = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	if (!res) {
+		dev_err(dev, "No tx error irq resource\n");
+		return -ENODEV;
+	}
+	sport->tx_err_irq = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_IRQ, 1);
+	if (!res) {
+		dev_err(dev, "No rx error irq resource\n");
+		return -ENODEV;
+	}
+	sport->rx_err_irq = res->start;
+
+	return 0;
+}
+
+static int sport_request_resource(struct sport_device *sport)
+{
+	struct device *dev = &sport->pdev->dev;
+	int ret;
+
+	ret = peripheral_request_list(sport->pin_req, "soc-audio");
+	if (ret) {
+		dev_err(dev, "Unable to request sport pin\n");
+		return ret;
+	}
+
+	ret = request_dma(sport->tx_dma_chan, "SPORT TX Data");
+	if (ret) {
+		dev_err(dev, "Unable to allocate DMA channel for sport tx\n");
+		goto err_tx_dma;
+	}
+	set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport);
+
+	ret = request_dma(sport->rx_dma_chan, "SPORT RX Data");
+	if (ret) {
+		dev_err(dev, "Unable to allocate DMA channel for sport rx\n");
+		goto err_rx_dma;
+	}
+	set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport);
+
+	ret = request_irq(sport->tx_err_irq, sport_err_irq,
+			0, "SPORT TX ERROR", sport);
+	if (ret) {
+		dev_err(dev, "Unable to allocate tx error IRQ for sport\n");
+		goto err_tx_irq;
+	}
+
+	ret = request_irq(sport->rx_err_irq, sport_err_irq,
+			0, "SPORT RX ERROR", sport);
+	if (ret) {
+		dev_err(dev, "Unable to allocate rx error IRQ for sport\n");
+		goto err_rx_irq;
+	}
+
+	return 0;
+err_rx_irq:
+	free_irq(sport->tx_err_irq, sport);
+err_tx_irq:
+	free_dma(sport->rx_dma_chan);
+err_rx_dma:
+	free_dma(sport->tx_dma_chan);
+err_tx_dma:
+	peripheral_free_list(sport->pin_req);
+	return ret;
+}
+
+static void sport_free_resource(struct sport_device *sport)
+{
+	free_irq(sport->rx_err_irq, sport);
+	free_irq(sport->tx_err_irq, sport);
+	free_dma(sport->rx_dma_chan);
+	free_dma(sport->tx_dma_chan);
+	peripheral_free_list(sport->pin_req);
+}
+
+struct sport_device *sport_create(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct sport_device *sport;
+	int ret;
+
+	sport = kzalloc(sizeof(*sport), GFP_KERNEL);
+	if (!sport) {
+		dev_err(dev, "Unable to allocate memory for sport device\n");
+		return NULL;
+	}
+	sport->pdev = pdev;
+
+	ret = sport_get_resource(sport);
+	if (ret) {
+		kfree(sport);
+		return NULL;
+	}
+
+	ret = sport_request_resource(sport);
+	if (ret) {
+		kfree(sport);
+		return NULL;
+	}
+
+	dev_dbg(dev, "SPORT create success\n");
+	return sport;
+}
+EXPORT_SYMBOL(sport_create);
+
+void sport_delete(struct sport_device *sport)
+{
+	sport_free_resource(sport);
+}
+EXPORT_SYMBOL(sport_delete);
+
+MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver");
+MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h
new file mode 100644
index 000000000000..307d193cfcef
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-sport.h
@@ -0,0 +1,82 @@
+/*
+ * bf6xx_sport - Analog Devices BF6XX SPORT driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef _BF6XX_SPORT_H_
+#define _BF6XX_SPORT_H_
+
+#include <linux/platform_device.h>
+#include <asm/bfin_sport3.h>
+
+struct sport_device {
+	struct platform_device *pdev;
+	const unsigned short *pin_req;
+	struct sport_register *tx_regs;
+	struct sport_register *rx_regs;
+	int tx_dma_chan;
+	int rx_dma_chan;
+	int tx_err_irq;
+	int rx_err_irq;
+
+	void (*tx_callback)(void *data);
+	void *tx_data;
+	void (*rx_callback)(void *data);
+	void *rx_data;
+
+	struct dmasg *tx_desc;
+	struct dmasg *rx_desc;
+	unsigned int tx_desc_size;
+	unsigned int rx_desc_size;
+	unsigned char *tx_buf;
+	unsigned char *rx_buf;
+	unsigned int tx_fragsize;
+	unsigned int rx_fragsize;
+	unsigned int tx_frags;
+	unsigned int rx_frags;
+	unsigned int wdsize;
+};
+
+struct sport_params {
+	u32 spctl;
+	u32 div;
+};
+
+struct sport_device *sport_create(struct platform_device *pdev);
+void sport_delete(struct sport_device *sport);
+int sport_set_tx_params(struct sport_device *sport,
+		struct sport_params *params);
+int sport_set_rx_params(struct sport_device *sport,
+		struct sport_params *params);
+void sport_tx_start(struct sport_device *sport);
+void sport_rx_start(struct sport_device *sport);
+void sport_tx_stop(struct sport_device *sport);
+void sport_rx_stop(struct sport_device *sport);
+void sport_set_tx_callback(struct sport_device *sport,
+	void (*tx_callback)(void *), void *tx_data);
+void sport_set_rx_callback(struct sport_device *sport,
+	void (*rx_callback)(void *), void *rx_data);
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+	int fragcount, size_t fragsize);
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+	int fragcount, size_t fragsize);
+unsigned long sport_curr_offset_tx(struct sport_device *sport);
+unsigned long sport_curr_offset_rx(struct sport_device *sport);
+
+
+
+#endif
-- 
cgit v1.2.3


From f62ae7bda434ac5d2bcd6feb4f5bdb5885633177 Mon Sep 17 00:00:00 2001
From: Scott Jiang <scott.jiang.linux@gmail.com>
Date: Wed, 20 Jun 2012 17:00:31 -0400
Subject: ASoC: add i2s dai driver for bf6xx soc

This driver enables i2s mode support on blackfin bf6xx platform.
We reuse bf5xx-i2s-pcm.c as its i2s pcm driver because it's the same
for both bf5xx and bf6xx soc.

Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/Kconfig     |  11 +-
 sound/soc/blackfin/Makefile    |   2 +
 sound/soc/blackfin/bf6xx-i2s.c | 234 +++++++++++++++++++++++++++++++++++++++++
 3 files changed, 244 insertions(+), 3 deletions(-)
 create mode 100644 sound/soc/blackfin/bf6xx-i2s.c

(limited to 'sound')

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 0374a3965095..16b88f5c26e2 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -10,12 +10,14 @@ config SND_BF5XX_I2S
 	  You will also need to select the audio interfaces to support below.
 
 config SND_BF5XX_SOC_SSM2602
-	tristate "SoC SSM2602 Audio support for BF52x ezkit"
+	tristate "SoC SSM2602 Audio Codec Add-On Card support"
 	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
-	select SND_BF5XX_SOC_I2S
+	select SND_BF5XX_SOC_I2S if !BF60x
+	select SND_BF6XX_SOC_I2S if BF60x
 	select SND_SOC_SSM2602
 	help
-	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
+	  Say Y if you want to add support for the Analog Devices
+	  SSM2602 Audio Codec Add-On Card.
 
 config SND_SOC_BFIN_EVAL_ADAU1701
 	tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
@@ -169,6 +171,9 @@ config SND_BF6XX_SOC_SPORT
 config SND_BF5XX_SOC_I2S
 	tristate
 
+config SND_BF6XX_SOC_I2S
+	tristate
+
 config SND_BF5XX_SOC_TDM
 	tristate
 
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 13b092239a82..6fea1f4cbee2 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -6,6 +6,7 @@ snd-soc-bf5xx-sport-objs := bf5xx-sport.o
 snd-soc-bf6xx-sport-objs := bf6xx-sport.o
 snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o
 snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o
+snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o
 snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o
 
 obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o
@@ -15,6 +16,7 @@ obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o
 obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o
 obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o
 obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
+obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o
 obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o
 
 # Blackfin Machine Support
diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
new file mode 100644
index 000000000000..c3c2466d3a42
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -0,0 +1,234 @@
+/*
+ * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "bf6xx-sport.h"
+
+struct sport_params param;
+
+static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai);
+	struct device *dev = &sport->pdev->dev;
+	int ret = 0;
+
+	param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR
+			| SPORT_CTL_LFS | SPORT_CTL_LAFS);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE
+			| SPORT_CTL_LFS;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		param.spctl |= SPORT_CTL_FSR;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS
+			| SPORT_CTL_LAFS;
+		break;
+	default:
+		dev_err(dev, "%s: Unknown DAI format type\n", __func__);
+		ret = -EINVAL;
+		break;
+	}
+
+	param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS);
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		ret = -EINVAL;
+		break;
+	default:
+		dev_err(dev, "%s: Unknown DAI master type\n", __func__);
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int bfin_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+	struct device *dev = &sport->pdev->dev;
+	int ret = 0;
+
+	param.spctl &= ~SPORT_CTL_SLEN;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		param.spctl |= 0x70;
+		sport->wdsize = 1;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		param.spctl |= 0xf0;
+		sport->wdsize = 2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		param.spctl |= 0x170;
+		sport->wdsize = 3;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		param.spctl |= 0x1f0;
+		sport->wdsize = 4;
+		break;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = sport_set_tx_params(sport, &param);
+		if (ret) {
+			dev_err(dev, "SPORT tx is busy!\n");
+			return ret;
+		}
+	} else {
+		ret = sport_set_rx_params(sport, &param);
+		if (ret) {
+			dev_err(dev, "SPORT rx is busy!\n");
+			return ret;
+		}
+	}
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int bfin_i2s_suspend(struct snd_soc_dai *dai)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+
+	if (dai->capture_active)
+		sport_rx_stop(sport);
+	if (dai->playback_active)
+		sport_tx_stop(sport);
+	return 0;
+}
+
+static int bfin_i2s_resume(struct snd_soc_dai *dai)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+	struct device *dev = &sport->pdev->dev;
+	int ret;
+
+	ret = sport_set_tx_params(sport, &param);
+	if (ret) {
+		dev_err(dev, "SPORT tx is busy!\n");
+		return ret;
+	}
+	ret = sport_set_rx_params(sport, &param);
+	if (ret) {
+		dev_err(dev, "SPORT rx is busy!\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+#else
+#define bfin_i2s_suspend NULL
+#define bfin_i2s_resume NULL
+#endif
+
+#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+		SNDRV_PCM_RATE_96000)
+
+#define BFIN_I2S_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 bfin_i2s_dai_ops = {
+	.hw_params	= bfin_i2s_hw_params,
+	.set_fmt	= bfin_i2s_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver bfin_i2s_dai = {
+	.suspend = bfin_i2s_suspend,
+	.resume = bfin_i2s_resume,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = BFIN_I2S_RATES,
+		.formats = BFIN_I2S_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = BFIN_I2S_RATES,
+		.formats = BFIN_I2S_FORMATS,
+	},
+	.ops = &bfin_i2s_dai_ops,
+};
+
+static int __devinit bfin_i2s_probe(struct platform_device *pdev)
+{
+	struct sport_device *sport;
+	struct device *dev = &pdev->dev;
+	int ret;
+
+	sport = sport_create(pdev);
+	if (!sport)
+		return -ENODEV;
+
+	/* register with the ASoC layers */
+	ret = snd_soc_register_dai(dev, &bfin_i2s_dai);
+	if (ret) {
+		dev_err(dev, "Failed to register DAI: %d\n", ret);
+		sport_delete(sport);
+		return ret;
+	}
+	platform_set_drvdata(pdev, sport);
+
+	return 0;
+}
+
+static int __devexit bfin_i2s_remove(struct platform_device *pdev)
+{
+	struct sport_device *sport = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	sport_delete(sport);
+
+	return 0;
+}
+
+static struct platform_driver bfin_i2s_driver = {
+	.probe  = bfin_i2s_probe,
+	.remove = __devexit_p(bfin_i2s_remove),
+	.driver = {
+		.name = "bfin-i2s",
+		.owner = THIS_MODULE,
+	},
+};
+
+module_platform_driver(bfin_i2s_driver);
+
+MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver");
+MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>");
+MODULE_LICENSE("GPL v2");
-- 
cgit v1.2.3


From 8b8d654b55648561287bd8baca0f75f964a17038 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 20 Jun 2012 16:32:22 +0200
Subject: ALSA: hda - Move one-time init codes from generic_hdmi_init()

The codes to initialize work struct or create a proc interface should
be called only once and never although it's called many times through
the init callback.  Move that stuff into patch_generic_hdmi() so that
it's called only once.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++-----
 1 file changed, 17 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index db8f6928f839..64f1fedfd535 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1277,23 +1277,34 @@ static int generic_hdmi_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
-static int generic_hdmi_init(struct hda_codec *codec)
+static int generic_hdmi_init_per_pins(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx;
 
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
-		hda_nid_t pin_nid = per_pin->pin_nid;
 		struct hdmi_eld *eld = &per_pin->sink_eld;
 
-		hdmi_init_pin(codec, pin_nid);
-		snd_hda_jack_detect_enable(codec, pin_nid, pin_nid);
-
 		per_pin->codec = codec;
 		INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld);
 		snd_hda_eld_proc_new(codec, eld, pin_idx);
 	}
+	return 0;
+}
+
+static int generic_hdmi_init(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+	int pin_idx;
+
+	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+		struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+		hda_nid_t pin_nid = per_pin->pin_nid;
+
+		hdmi_init_pin(codec, pin_nid);
+		snd_hda_jack_detect_enable(codec, pin_nid, pin_nid);
+	}
 	snd_hda_jack_report_sync(codec);
 	return 0;
 }
@@ -1338,6 +1349,7 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
+	generic_hdmi_init_per_pins(codec);
 
 	init_channel_allocations();
 
-- 
cgit v1.2.3


From 9883ab229d61b884323f9186b1bd4a41373a491b Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 Jun 2012 20:11:41 +0200
Subject: ASoC: dmaengine-pcm: Rename and deprecate snd_dmaengine_pcm_pointer

Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch renames the current implementation
and documents its shortcomings and that it should not be used anymore in new
drivers.

The next patch will introduce a new snd_dmaengine_pcm_pointer which will be
implemented based on querying the current stream position from the dma device.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/ep93xx/ep93xx-pcm.c |  2 +-
 sound/soc/fsl/imx-pcm-dma.c   |  2 +-
 sound/soc/mxs/mxs-pcm.c       |  2 +-
 sound/soc/soc-dmaengine-pcm.c | 10 +++++-----
 sound/soc/ux500/ux500_pcm.c   |  2 +-
 5 files changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 162dbb74f4cc..4eea98b42bc8 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -136,7 +136,7 @@ static struct snd_pcm_ops ep93xx_pcm_ops = {
 	.hw_params	= ep93xx_pcm_hw_params,
 	.hw_free	= ep93xx_pcm_hw_free,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= ep93xx_pcm_mmap,
 };
 
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index f3c0a5ef35c8..48f9d886f020 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -141,7 +141,7 @@ static struct snd_pcm_ops imx_pcm_ops = {
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= snd_imx_pcm_hw_params,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= snd_imx_pcm_mmap,
 };
 
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 373dec90579f..f82d766cbf9e 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -141,7 +141,7 @@ static struct snd_pcm_ops mxs_pcm_ops = {
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= snd_mxs_pcm_hw_params,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= snd_mxs_pcm_mmap,
 };
 
diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c
index 475695234b3d..7c0877e3731c 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/soc/soc-dmaengine-pcm.c
@@ -200,18 +200,18 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger);
 
 /**
- * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation
+ * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation
  * @substream: PCM substream
  *
- * This function can be used as the PCM pointer callback for dmaengine based PCM
- * driver implementations.
+ * This function is deprecated and should not be used by new drivers, as its
+ * results may be unreliable.
  */
-snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream)
 {
 	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
 	return bytes_to_frames(substream->runtime, prtd->pos);
 }
-EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue);
 
 static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd,
 	dma_filter_fn filter_fn, void *filter_data)
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
index 97d8e4de29c2..1a04e248453c 100644
--- a/sound/soc/ux500/ux500_pcm.c
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -261,7 +261,7 @@ static struct snd_pcm_ops ux500_pcm_ops = {
 	.hw_params	= ux500_pcm_hw_params,
 	.hw_free	= ux500_pcm_hw_free,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= ux500_pcm_mmap
 };
 
-- 
cgit v1.2.3


From 3528f27a5d4ac299e2d8cbe7297c1e9edd601ee6 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 11 Jun 2012 20:11:42 +0200
Subject: ASoC: dmaengine-pcm: Add support for querying stream position from
 DMA driver

Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch addresses the issue by
implementing support for querying the current stream position directly from the
dmaengine driver. Since not all dmaengine drivers support reporting the stream
position yet the old period counting implementation is kept for now.

Furthermore the new mechanism allows to report the stream position with a
sub-period granularity, given that the dmaengine driver supports this.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dmaengine-pcm.c | 29 ++++++++++++++++++++++++++++-
 1 file changed, 28 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c
index 7c0877e3731c..2995334d8000 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/soc/soc-dmaengine-pcm.c
@@ -30,6 +30,7 @@
 
 struct dmaengine_pcm_runtime_data {
 	struct dma_chan *dma_chan;
+	dma_cookie_t cookie;
 
 	unsigned int pos;
 
@@ -153,7 +154,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream)
 
 	desc->callback = dmaengine_pcm_dma_complete;
 	desc->callback_param = substream;
-	dmaengine_submit(desc);
+	prtd->cookie = dmaengine_submit(desc);
 
 	return 0;
 }
@@ -213,6 +214,32 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue);
 
+/**
+ * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation
+ * @substream: PCM substream
+ *
+ * This function can be used as the PCM pointer callback for dmaengine based PCM
+ * driver implementations.
+ */
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+	struct dma_tx_state state;
+	enum dma_status status;
+	unsigned int buf_size;
+	unsigned int pos = 0;
+
+	status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state);
+	if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) {
+		buf_size = snd_pcm_lib_buffer_bytes(substream);
+		if (state.residue > 0 && state.residue <= buf_size)
+			pos = buf_size - state.residue;
+	}
+
+	return bytes_to_frames(substream->runtime, pos);
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
+
 static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd,
 	dma_filter_fn filter_fn, void *filter_data)
 {
-- 
cgit v1.2.3


From 8ceb332df46863ac8f74114a2b1805719cf49dcc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 21 Jun 2012 08:23:27 +0200
Subject: ALSA: hda - Remove loop from simple_hdmi*()

The simple_hdmi stuff is designed only for a single pin and a single
converter (thus a single PCM stream), and no need for loops.
Let's flatten the code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 69 ++++++++++++++++++----------------------------
 1 file changed, 27 insertions(+), 42 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 64f1fedfd535..0a87a1f2988e 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1364,26 +1364,22 @@ static int simple_playback_build_pcms(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct hda_pcm *info = spec->pcm_rec;
-	int i;
+	unsigned int chans;
+	struct hda_pcm_stream *pstr;
 
-	codec->num_pcms = spec->num_cvts;
+	codec->num_pcms = 1;
 	codec->pcm_info = info;
 
-	for (i = 0; i < codec->num_pcms; i++, info++) {
-		unsigned int chans;
-		struct hda_pcm_stream *pstr;
+	chans = get_wcaps(codec, spec->cvts[0].cvt_nid);
+	chans = get_wcaps_channels(chans);
 
-		chans = get_wcaps(codec, spec->cvts[i].cvt_nid);
-		chans = get_wcaps_channels(chans);
-
-		info->name = get_hdmi_pcm_name(i);
-		info->pcm_type = HDA_PCM_TYPE_HDMI;
-		pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
-		*pstr = spec->pcm_playback;
-		pstr->nid = spec->cvts[i].cvt_nid;
-		if (pstr->channels_max <= 2 && chans && chans <= 16)
-			pstr->channels_max = chans;
-	}
+	info->name = get_hdmi_pcm_name(0);
+	info->pcm_type = HDA_PCM_TYPE_HDMI;
+	pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
+	*pstr = spec->pcm_playback;
+	pstr->nid = spec->cvts[0].cvt_nid;
+	if (pstr->channels_max <= 2 && chans && chans <= 16)
+		pstr->channels_max = chans;
 
 	return 0;
 }
@@ -1405,38 +1401,27 @@ static int simple_playback_build_controls(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
 	int err;
-	int i;
-
-	for (i = 0; i < codec->num_pcms; i++) {
-		err = snd_hda_create_spdif_out_ctls(codec,
-						    spec->cvts[i].cvt_nid,
-						    spec->cvts[i].cvt_nid);
-		if (err < 0)
-			return err;
-		err = simple_hdmi_build_jack(codec, i);
-		if (err < 0)
-			return err;
-	}
 
-	return 0;
+	err = snd_hda_create_spdif_out_ctls(codec,
+					    spec->cvts[0].cvt_nid,
+					    spec->cvts[0].cvt_nid);
+	if (err < 0)
+		return err;
+	return simple_hdmi_build_jack(codec, 0);
 }
 
 static int simple_playback_init(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_pins; i++) {
-		hda_nid_t pin = spec->pins[i].pin_nid;
-		snd_hda_codec_write(codec, pin, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
-		/* some codecs require to unmute the pin */
-		if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
-			snd_hda_codec_write(codec, pin, 0,
-					    AC_VERB_SET_AMP_GAIN_MUTE,
-					    AMP_OUT_UNMUTE);
-		snd_hda_jack_detect_enable(codec, pin, pin);
-	}
+	hda_nid_t pin = spec->pins[0].pin_nid;
+
+	snd_hda_codec_write(codec, pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	/* some codecs require to unmute the pin */
+	if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+		snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_OUT_UNMUTE);
+	snd_hda_jack_detect_enable(codec, pin, pin);
 	snd_hda_jack_report_sync(codec);
 	return 0;
 }
-- 
cgit v1.2.3


From 9dd8cf125d27742a25219bfdf82026e7efed27d9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 21 Jun 2012 10:43:15 +0200
Subject: ALSA: hda - Don't rely on event tag for simple_hdmi

VIA codecs seem not returning the event tag in the unsolicited events,
thus the current code relying on the tag value doesn't work.
Since simple_hdmi stuff has only a single pin, we can use simply
snd_hda_jack_set_dirty_all() to activate the pin-detection
independently from the tag value.

Tested-by: Annie Liu <AnnieLiu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 0a87a1f2988e..75ad1a10646b 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1388,7 +1388,7 @@ static int simple_playback_build_pcms(struct hda_codec *codec)
 static void simple_hdmi_unsol_event(struct hda_codec *codec,
 				    unsigned int res)
 {
-	snd_hda_jack_get_action(codec, res >> AC_UNSOL_RES_TAG_SHIFT);
+	snd_hda_jack_set_dirty_all(codec);
 	snd_hda_jack_report_sync(codec);
 }
 
-- 
cgit v1.2.3


From e9ea8e8f229f4963bf01658e79c1c01780de25dd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 21 Jun 2012 11:41:05 +0200
Subject: ALSA: hda - Correct info print in HDMI non-intrinsic unsol event

In the recent code, the value shown there is a tag number, and it's no
longer same as the pin nid.  Correct the message to avoid confusion.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 75ad1a10646b..d6a8260a6f74 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -787,7 +787,7 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	int cp_ready = !!(res & AC_UNSOL_RES_CP_READY);
 
 	printk(KERN_INFO
-		"HDMI CP event: CODEC=%d PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+		"HDMI CP event: CODEC=%d TAG=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
 		codec->addr,
 		tag,
 		subtag,
-- 
cgit v1.2.3


From f4a6348391fa029c0e230230adfafb7f33d4683e Mon Sep 17 00:00:00 2001
From: Scott Jiang <scott.jiang.linux@gmail.com>
Date: Thu, 21 Jun 2012 13:51:58 -0400
Subject: ASoC: bfin: use dev_err to print error log instead of dev_dbg

Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/bf6xx-sport.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index f19a72b8e0c2..318c5ba5360f 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -256,9 +256,9 @@ static irqreturn_t sport_err_irq(int irq, void *dev_id)
 	struct device *dev = &sport->pdev->dev;
 
 	if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI)
-		dev_dbg(dev, "sport error: TUVF\n");
+		dev_err(dev, "sport error: TUVF\n");
 	if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI)
-		dev_dbg(dev, "sport error: ROVF\n");
+		dev_err(dev, "sport error: ROVF\n");
 
 	return IRQ_HANDLED;
 }
-- 
cgit v1.2.3


From b43d224767e426cf1a8b6622d1d172f2b2b0e857 Mon Sep 17 00:00:00 2001
From: Dylan Reid <dgreid@chromium.org>
Date: Thu, 21 Jun 2012 21:51:22 -0700
Subject: ALSA: hda - Don't power up when not powered down.

After cancel_delayed_work_sync returns, the power down work either never
started (power_on == 1) or finished (power_on == 0).  In the former case
there is no need to power up again.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 604699cf85f5..045b5e7b8245 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4444,6 +4444,13 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 	cancel_delayed_work_sync(&codec->power_work);
 
 	spin_lock(&codec->power_lock);
+	/* If the power down delayed work was cancelled above before starting,
+	 * then there is no need to go through power up here.
+	 */
+	if (codec->power_on) {
+		spin_unlock(&codec->power_lock);
+		return;
+	}
 	trace_hda_power_up(codec);
 	snd_hda_update_power_acct(codec);
 	codec->power_on = 1;
-- 
cgit v1.2.3


From 3a9cf8efd7b64f26f1e0f02afb70382f90cc11ca Mon Sep 17 00:00:00 2001
From: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Date: Thu, 21 Jun 2012 15:54:51 +0530
Subject: ASoC: Add support for synopsys i2s controller as per ASoC framework.

This patch add support for synopsys I2S controller as per the ASoC
framework.

Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/Kconfig              |   1 +
 sound/soc/Makefile             |   1 +
 sound/soc/dwc/Kconfig          |   8 +
 sound/soc/dwc/Makefile         |   3 +
 sound/soc/dwc/designware_i2s.c | 454 +++++++++++++++++++++++++++++++++++++++++
 5 files changed, 467 insertions(+)
 create mode 100644 sound/soc/dwc/Kconfig
 create mode 100644 sound/soc/dwc/Makefile
 create mode 100644 sound/soc/dwc/designware_i2s.c

(limited to 'sound')

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 40b2ad1bb1cd..c5de0a84566f 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -33,6 +33,7 @@ source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
 source "sound/soc/blackfin/Kconfig"
 source "sound/soc/davinci/Kconfig"
+source "sound/soc/dwc/Kconfig"
 source "sound/soc/ep93xx/Kconfig"
 source "sound/soc/fsl/Kconfig"
 source "sound/soc/jz4740/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 70990f4017f4..00a555a743b6 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -11,6 +11,7 @@ obj-$(CONFIG_SND_SOC)	+= atmel/
 obj-$(CONFIG_SND_SOC)	+= au1x/
 obj-$(CONFIG_SND_SOC)	+= blackfin/
 obj-$(CONFIG_SND_SOC)	+= davinci/
+obj-$(CONFIG_SND_SOC)	+= dwc/
 obj-$(CONFIG_SND_SOC)	+= ep93xx/
 obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig
new file mode 100644
index 000000000000..93e9fc33560c
--- /dev/null
+++ b/sound/soc/dwc/Kconfig
@@ -0,0 +1,8 @@
+config SND_DESIGNWARE_I2S
+	tristate "Synopsys I2S Device Driver"
+	help
+	 Say Y or M if you want to add support for I2S driver for
+	 Synopsys desigwnware I2S device. The device supports upto
+	 maximum of 8 channels each for play and record.
+
+
diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile
new file mode 100644
index 000000000000..319371f690f4
--- /dev/null
+++ b/sound/soc/dwc/Makefile
@@ -0,0 +1,3 @@
+# SYNOPSYS Platform Support
+obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o
+
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
new file mode 100644
index 000000000000..e667e2b45e67
--- /dev/null
+++ b/sound/soc/dwc/designware_i2s.c
@@ -0,0 +1,454 @@
+/*
+ * ALSA SoC Synopsys I2S Audio Layer
+ *
+ * sound/soc/spear/designware_i2s.c
+ *
+ * Copyright (C) 2010 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <sound/designware_i2s.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+/* common register for all channel */
+#define IER		0x000
+#define IRER		0x004
+#define ITER		0x008
+#define CER		0x00C
+#define CCR		0x010
+#define RXFFR		0x014
+#define TXFFR		0x018
+
+/* I2STxRxRegisters for all channels */
+#define LRBR_LTHR(x)	(0x40 * x + 0x020)
+#define RRBR_RTHR(x)	(0x40 * x + 0x024)
+#define RER(x)		(0x40 * x + 0x028)
+#define TER(x)		(0x40 * x + 0x02C)
+#define RCR(x)		(0x40 * x + 0x030)
+#define TCR(x)		(0x40 * x + 0x034)
+#define ISR(x)		(0x40 * x + 0x038)
+#define IMR(x)		(0x40 * x + 0x03C)
+#define ROR(x)		(0x40 * x + 0x040)
+#define TOR(x)		(0x40 * x + 0x044)
+#define RFCR(x)		(0x40 * x + 0x048)
+#define TFCR(x)		(0x40 * x + 0x04C)
+#define RFF(x)		(0x40 * x + 0x050)
+#define TFF(x)		(0x40 * x + 0x054)
+
+/* I2SCOMPRegisters */
+#define I2S_COMP_PARAM_2	0x01F0
+#define I2S_COMP_PARAM_1	0x01F4
+#define I2S_COMP_VERSION	0x01F8
+#define I2S_COMP_TYPE		0x01FC
+
+#define MAX_CHANNEL_NUM		8
+#define MIN_CHANNEL_NUM		2
+
+struct dw_i2s_dev {
+	void __iomem *i2s_base;
+	struct clk *clk;
+	int active;
+	unsigned int capability;
+	struct device *dev;
+
+	/* data related to DMA transfers b/w i2s and DMAC */
+	struct i2s_dma_data play_dma_data;
+	struct i2s_dma_data capture_dma_data;
+	struct i2s_clk_config_data config;
+	int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
+};
+
+static inline void i2s_write_reg(void *io_base, int reg, u32 val)
+{
+	writel(val, io_base + reg);
+}
+
+static inline u32 i2s_read_reg(void *io_base, int reg)
+{
+	return readl(io_base + reg);
+}
+
+static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream)
+{
+	u32 i = 0;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, TER(i), 0);
+	} else {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, RER(i), 0);
+	}
+}
+
+static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
+{
+	u32 i = 0;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, TOR(i), 0);
+	} else {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, ROR(i), 0);
+	}
+}
+
+void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream)
+{
+
+	i2s_write_reg(dev->i2s_base, IER, 1);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		i2s_write_reg(dev->i2s_base, ITER, 1);
+	else
+		i2s_write_reg(dev->i2s_base, IRER, 1);
+
+	i2s_write_reg(dev->i2s_base, CER, 1);
+}
+
+static void i2s_stop(struct dw_i2s_dev *dev,
+		struct snd_pcm_substream *substream)
+{
+	u32 i = 0, irq;
+
+	i2s_clear_irqs(dev, substream->stream);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		i2s_write_reg(dev->i2s_base, ITER, 0);
+
+		for (i = 0; i < 4; i++) {
+			irq = i2s_read_reg(dev->i2s_base, IMR(i));
+			i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30);
+		}
+	} else {
+		i2s_write_reg(dev->i2s_base, IRER, 0);
+
+		for (i = 0; i < 4; i++) {
+			irq = i2s_read_reg(dev->i2s_base, IMR(i));
+			i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03);
+		}
+	}
+
+	if (!dev->active) {
+		i2s_write_reg(dev->i2s_base, CER, 0);
+		i2s_write_reg(dev->i2s_base, IER, 0);
+	}
+}
+
+static int dw_i2s_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+	struct i2s_dma_data *dma_data = NULL;
+
+	if (!(dev->capability & DWC_I2S_RECORD) &&
+			(substream->stream == SNDRV_PCM_STREAM_CAPTURE))
+		return -EINVAL;
+
+	if (!(dev->capability & DWC_I2S_PLAY) &&
+			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+		return -EINVAL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_data = &dev->play_dma_data;
+	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		dma_data = &dev->capture_dma_data;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data);
+
+	return 0;
+}
+
+static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	struct i2s_clk_config_data *config = &dev->config;
+	u32 ccr, xfer_resolution, ch_reg, irq;
+	int ret;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		config->data_width = 16;
+		ccr = 0x00;
+		xfer_resolution = 0x02;
+		break;
+
+	case SNDRV_PCM_FORMAT_S24_LE:
+		config->data_width = 24;
+		ccr = 0x08;
+		xfer_resolution = 0x04;
+		break;
+
+	case SNDRV_PCM_FORMAT_S32_LE:
+		config->data_width = 32;
+		ccr = 0x10;
+		xfer_resolution = 0x05;
+		break;
+
+	default:
+		dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt");
+		return -EINVAL;
+	}
+
+	config->chan_nr = params_channels(params);
+
+	switch (config->chan_nr) {
+	case EIGHT_CHANNEL_SUPPORT:
+		ch_reg = 3;
+	case SIX_CHANNEL_SUPPORT:
+		ch_reg = 2;
+	case FOUR_CHANNEL_SUPPORT:
+		ch_reg = 1;
+	case TWO_CHANNEL_SUPPORT:
+		ch_reg = 0;
+		break;
+	default:
+		dev_err(dev->dev, "channel not supported\n");
+	}
+
+	i2s_disable_channels(dev, substream->stream);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution);
+		i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+		irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
+		i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
+		i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
+	} else {
+		i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution);
+		i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+		irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
+		i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
+		i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
+	}
+
+	i2s_write_reg(dev->i2s_base, CCR, ccr);
+
+	config->sample_rate = params_rate(params);
+
+	if (!dev->i2s_clk_cfg)
+		return -EINVAL;
+
+	ret = dev->i2s_clk_cfg(config);
+	if (ret < 0) {
+		dev_err(dev->dev, "runtime audio clk config fail\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static int dw_i2s_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		dev->active++;
+		i2s_start(dev, substream);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		dev->active--;
+		i2s_stop(dev, substream);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
+static struct snd_soc_dai_ops dw_i2s_dai_ops = {
+	.startup	= dw_i2s_startup,
+	.shutdown	= dw_i2s_shutdown,
+	.hw_params	= dw_i2s_hw_params,
+	.trigger	= dw_i2s_trigger,
+};
+
+#ifdef CONFIG_PM
+
+static int dw_i2s_suspend(struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable(dev->clk);
+	return 0;
+}
+
+static int dw_i2s_resume(struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	clk_enable(dev->clk);
+	return 0;
+}
+
+#else
+#define dw_i2s_suspend	NULL
+#define dw_i2s_resume	NULL
+#endif
+
+static int dw_i2s_probe(struct platform_device *pdev)
+{
+	const struct i2s_platform_data *pdata = pdev->dev.platform_data;
+	struct dw_i2s_dev *dev;
+	struct resource *res;
+	int ret;
+	unsigned int cap;
+	struct snd_soc_dai_driver *dw_i2s_dai;
+
+	if (!pdata) {
+		dev_err(&pdev->dev, "Invalid platform data\n");
+		return -EINVAL;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(&pdev->dev, "no i2s resource defined\n");
+		return -ENODEV;
+	}
+
+	if (!devm_request_mem_region(&pdev->dev, res->start,
+				resource_size(res), pdev->name)) {
+		dev_err(&pdev->dev, "i2s region already claimed\n");
+		return -EBUSY;
+	}
+
+	dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
+	if (!dev) {
+		dev_warn(&pdev->dev, "kzalloc fail\n");
+		return -ENOMEM;
+	}
+
+	dev->i2s_base = devm_ioremap(&pdev->dev, res->start,
+			resource_size(res));
+	if (!dev->i2s_base) {
+		dev_err(&pdev->dev, "ioremap fail for i2s_region\n");
+		return -ENOMEM;
+	}
+
+	cap = pdata->cap;
+	dev->capability = cap;
+	dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+
+	/* Set DMA slaves info */
+
+	dev->play_dma_data.data = pdata->play_dma_data;
+	dev->capture_dma_data.data = pdata->capture_dma_data;
+	dev->play_dma_data.addr = res->start + I2S_TXDMA;
+	dev->capture_dma_data.addr = res->start + I2S_RXDMA;
+	dev->play_dma_data.max_burst = 16;
+	dev->capture_dma_data.max_burst = 16;
+	dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	dev->play_dma_data.filter = pdata->filter;
+	dev->capture_dma_data.filter = pdata->filter;
+
+	dev->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(dev->clk))
+		return  PTR_ERR(dev->clk);
+
+	ret = clk_enable(dev->clk);
+	if (ret < 0)
+		goto err_clk_put;
+
+	dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
+	if (!dw_i2s_dai) {
+		dev_err(&pdev->dev, "mem allocation failed for dai driver\n");
+		ret = -ENOMEM;
+		goto err_clk_disable;
+	}
+
+	if (cap & DWC_I2S_PLAY) {
+		dev_dbg(&pdev->dev, " SPEAr: play supported\n");
+		dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
+		dw_i2s_dai->playback.channels_max = pdata->channel;
+		dw_i2s_dai->playback.formats = pdata->snd_fmts;
+		dw_i2s_dai->playback.rates = pdata->snd_rates;
+	}
+
+	if (cap & DWC_I2S_RECORD) {
+		dev_dbg(&pdev->dev, "SPEAr: record supported\n");
+		dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM;
+		dw_i2s_dai->capture.channels_max = pdata->channel;
+		dw_i2s_dai->capture.formats = pdata->snd_fmts;
+		dw_i2s_dai->capture.rates = pdata->snd_rates;
+	}
+
+	dw_i2s_dai->ops = &dw_i2s_dai_ops;
+	dw_i2s_dai->suspend = dw_i2s_suspend;
+	dw_i2s_dai->resume = dw_i2s_resume;
+
+	dev->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, dev);
+	ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "not able to register dai\n");
+		goto err_set_drvdata;
+	}
+
+	return 0;
+
+err_set_drvdata:
+	dev_set_drvdata(&pdev->dev, NULL);
+err_clk_disable:
+	clk_disable(dev->clk);
+err_clk_put:
+	clk_put(dev->clk);
+	return ret;
+}
+
+static int dw_i2s_remove(struct platform_device *pdev)
+{
+	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	clk_put(dev->clk);
+
+	return 0;
+}
+
+static struct platform_driver dw_i2s_driver = {
+	.probe		= dw_i2s_probe,
+	.remove		= dw_i2s_remove,
+	.driver		= {
+		.name	= "designware-i2s",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver(dw_i2s_driver);
+
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:designware_i2s");
-- 
cgit v1.2.3


From 241b446f30de171b627524c107ce03e5ecee0124 Mon Sep 17 00:00:00 2001
From: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Date: Thu, 21 Jun 2012 15:54:52 +0530
Subject: ASoC: Add support for SPEAr ASoC pcm layer.

This patch add support for the SPEAr ASoC pcm layer in ASoC
framework. The pcm layer uses common snd_dmaengine framework.

Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/spear/spear_pcm.c | 214 ++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 214 insertions(+)
 create mode 100644 sound/soc/spear/spear_pcm.c

(limited to 'sound')

diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
new file mode 100644
index 000000000000..97c2cac8e92c
--- /dev/null
+++ b/sound/soc/spear/spear_pcm.c
@@ -0,0 +1,214 @@
+/*
+ * ALSA PCM interface for ST SPEAr Processors
+ *
+ * sound/soc/spear/spear_pcm.c
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar<rajeev-dlh.kumar@st.com>
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/scatterlist.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+
+struct snd_pcm_hardware spear_pcm_hardware = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+	.buffer_bytes_max = 16 * 1024, /* max buffer size */
+	.period_bytes_min = 2 * 1024, /* 1 msec data minimum period size */
+	.period_bytes_max = 2 * 1024, /* maximum period size */
+	.periods_min = 1, /* min # periods */
+	.periods_max = 8, /* max # of periods */
+	.fifo_size = 0, /* fifo size in bytes */
+};
+
+static int spear_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static int spear_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+
+	return 0;
+}
+
+static int spear_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	struct spear_dma_data *dma_data = (struct spear_dma_data *)
+		snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	int ret;
+
+	ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware);
+	if (ret)
+		return ret;
+
+	ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data);
+	if (ret)
+		return ret;
+
+	snd_dmaengine_pcm_set_data(substream, dma_data);
+
+	return 0;
+}
+
+static int spear_pcm_close(struct snd_pcm_substream *substream)
+{
+
+	snd_dmaengine_pcm_close(substream);
+
+	return 0;
+}
+
+static int spear_pcm_mmap(struct snd_pcm_substream *substream,
+		struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+			runtime->dma_area, runtime->dma_addr,
+			runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops spear_pcm_ops = {
+	.open		= spear_pcm_open,
+	.close		= spear_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= spear_pcm_hw_params,
+	.hw_free	= spear_pcm_hw_free,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= spear_pcm_mmap,
+};
+
+static int
+spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+		size_t size)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+			&buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+
+	dev_info(buf->dev.dev,
+			" preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+			(void *)buf->area, (void *)buf->addr, size);
+
+	buf->bytes = size;
+	return 0;
+}
+
+static void spear_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf && !buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int spear_pcm_new(struct snd_card *card,
+		struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	int ret;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &spear_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+	if (dai->driver->playback.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(pcm,
+				SNDRV_PCM_STREAM_PLAYBACK,
+				spear_pcm_hardware.buffer_bytes_max);
+		if (ret)
+			return ret;
+	}
+
+	if (dai->driver->capture.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(pcm,
+				SNDRV_PCM_STREAM_CAPTURE,
+				spear_pcm_hardware.buffer_bytes_max);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+struct snd_soc_platform_driver spear_soc_platform = {
+	.ops		=	&spear_pcm_ops,
+	.pcm_new	=	spear_pcm_new,
+	.pcm_free	=	spear_pcm_free,
+};
+
+static int __devinit spear_soc_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev, &spear_soc_platform);
+}
+
+static int __devexit spear_soc_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver spear_pcm_driver = {
+	.driver = {
+		.name = "spear-pcm-audio",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = spear_soc_platform_probe,
+	.remove = __devexit_p(spear_soc_platform_remove),
+};
+
+module_platform_driver(spear_pcm_driver);
+
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr PCM DMA module");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spear-pcm-audio");
-- 
cgit v1.2.3


From ace36d85809f6005b559802f194d44c6aa61af88 Mon Sep 17 00:00:00 2001
From: Vipin Kumar <vipin.kumar@st.com>
Date: Thu, 21 Jun 2012 15:54:53 +0530
Subject: ASoC: SPEAr spdif_in: Add spdif IN support

This patch implements the spdif IN driver for ST peripheral

Signed-off-by: Vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/spear/spdif_in.c      | 297 ++++++++++++++++++++++++++++++++++++++++
 sound/soc/spear/spdif_in_regs.h |  60 ++++++++
 2 files changed, 357 insertions(+)
 create mode 100644 sound/soc/spear/spdif_in.c
 create mode 100644 sound/soc/spear/spdif_in_regs.h

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
new file mode 100644
index 000000000000..c7c4b20395bb
--- /dev/null
+++ b/sound/soc/spear/spdif_in.c
@@ -0,0 +1,297 @@
+/*
+ * ALSA SoC SPDIF In Audio Layer for spear processors
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Vipin Kumar <vipin.kumar@st.com>
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/ioport.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+#include <sound/spear_spdif.h>
+#include "spdif_in_regs.h"
+
+struct spdif_in_params {
+	u32 format;
+};
+
+struct spdif_in_dev {
+	struct clk *clk;
+	struct spear_dma_data dma_params;
+	struct spdif_in_params saved_params;
+	void *io_base;
+	struct device *dev;
+	void (*reset_perip)(void);
+	int irq;
+};
+
+static void spdif_in_configure(struct spdif_in_dev *host)
+{
+	u32 ctrl = SPDIF_IN_PRTYEN | SPDIF_IN_STATEN | SPDIF_IN_USREN |
+		SPDIF_IN_VALEN | SPDIF_IN_BLKEN;
+	ctrl |= SPDIF_MODE_16BIT | SPDIF_FIFO_THRES_16;
+
+	writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+	writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK);
+}
+
+static int spdif_in_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return -EINVAL;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params);
+	return 0;
+}
+
+static void spdif_in_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return;
+
+	writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK);
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static void spdif_in_format(struct spdif_in_dev *host, u32 format)
+{
+	u32 ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		ctrl |= SPDIF_XTRACT_16BIT;
+		break;
+
+	case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE:
+		ctrl &= ~SPDIF_XTRACT_16BIT;
+		break;
+	}
+
+	writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+}
+
+static int spdif_in_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 format;
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return -EINVAL;
+
+	format = params_format(params);
+	host->saved_params.format = format;
+
+	return 0;
+}
+
+static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 ctrl;
+	int ret = 0;
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return -EINVAL;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		clk_enable(host->clk);
+		spdif_in_configure(host);
+		spdif_in_format(host, host->saved_params.format);
+
+		ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+		ctrl |= SPDIF_IN_SAMPLE | SPDIF_IN_ENB;
+		writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+		writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+		ctrl &= ~(SPDIF_IN_SAMPLE | SPDIF_IN_ENB);
+		writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+		writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK);
+
+		if (host->reset_perip)
+			host->reset_perip();
+		clk_disable(host->clk);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
+static struct snd_soc_dai_ops spdif_in_dai_ops = {
+	.startup	= spdif_in_startup,
+	.shutdown	= spdif_in_shutdown,
+	.trigger	= spdif_in_trigger,
+	.hw_params	= spdif_in_hw_params,
+};
+
+struct snd_soc_dai_driver spdif_in_dai = {
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+				 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+				 SNDRV_PCM_RATE_192000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | \
+			   SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE,
+	},
+	.ops = &spdif_in_dai_ops,
+};
+
+static irqreturn_t spdif_in_irq(int irq, void *arg)
+{
+	struct spdif_in_dev *host = (struct spdif_in_dev *)arg;
+
+	u32 irq_status = readl(host->io_base + SPDIF_IN_IRQ);
+
+	if (!irq_status)
+		return IRQ_NONE;
+
+	if (irq_status & SPDIF_IRQ_FIFOWRITE)
+		dev_err(host->dev, "spdif in: fifo write error");
+	if (irq_status & SPDIF_IRQ_EMPTYFIFOREAD)
+		dev_err(host->dev, "spdif in: empty fifo read error");
+	if (irq_status & SPDIF_IRQ_FIFOFULL)
+		dev_err(host->dev, "spdif in: fifo full error");
+	if (irq_status & SPDIF_IRQ_OUTOFRANGE)
+		dev_err(host->dev, "spdif in: out of range error");
+
+	writel(0, host->io_base + SPDIF_IN_IRQ);
+
+	return IRQ_HANDLED;
+}
+
+static int spdif_in_probe(struct platform_device *pdev)
+{
+	struct spdif_in_dev *host;
+	struct spear_spdif_platform_data *pdata;
+	struct resource *res, *res_fifo;
+	int ret;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res)
+		return -EINVAL;
+
+	res_fifo = platform_get_resource(pdev, IORESOURCE_IO, 0);
+	if (!res_fifo)
+		return -EINVAL;
+
+	if (!devm_request_mem_region(&pdev->dev, res->start,
+				resource_size(res), pdev->name)) {
+		dev_warn(&pdev->dev, "Failed to get memory resourse\n");
+		return -ENOENT;
+	}
+
+	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
+	if (!host) {
+		dev_warn(&pdev->dev, "kzalloc fail\n");
+		return -ENOMEM;
+	}
+
+	host->io_base = devm_ioremap(&pdev->dev, res->start,
+				resource_size(res));
+	if (!host->io_base) {
+		dev_warn(&pdev->dev, "ioremap failed\n");
+		return -ENOMEM;
+	}
+
+	host->irq = platform_get_irq(pdev, 0);
+	if (host->irq < 0)
+		return -EINVAL;
+
+	host->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(host->clk))
+		return PTR_ERR(host->clk);
+
+	pdata = dev_get_platdata(&pdev->dev);
+
+	if (!pdata)
+		return -EINVAL;
+
+	host->dma_params.data = pdata->dma_params;
+	host->dma_params.addr = res_fifo->start;
+	host->dma_params.max_burst = 16;
+	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	host->dma_params.filter = pdata->filter;
+	host->reset_perip = pdata->reset_perip;
+
+	host->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, host);
+
+	ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0,
+			"spdif-in", host);
+	if (ret) {
+		clk_put(host->clk);
+		dev_warn(&pdev->dev, "request_irq failed\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai);
+	if (ret != 0) {
+		clk_put(host->clk);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int spdif_in_remove(struct platform_device *pdev)
+{
+	struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	clk_put(host->clk);
+
+	return 0;
+}
+
+
+static struct platform_driver spdif_in_driver = {
+	.probe		= spdif_in_probe,
+	.remove		= spdif_in_remove,
+	.driver		= {
+		.name	= "spdif-in",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver(spdif_in_driver);
+
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr SPDIF IN SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spdif_in");
diff --git a/sound/soc/spear/spdif_in_regs.h b/sound/soc/spear/spdif_in_regs.h
new file mode 100644
index 000000000000..37af7bc66b7f
--- /dev/null
+++ b/sound/soc/spear/spdif_in_regs.h
@@ -0,0 +1,60 @@
+/*
+ * SPEAr SPDIF IN controller header file
+ *
+ * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef SPDIF_IN_REGS_H
+#define SPDIF_IN_REGS_H
+
+#define SPDIF_IN_CTRL		0x00
+	#define SPDIF_IN_PRTYEN		(1 << 20)
+	#define SPDIF_IN_STATEN		(1 << 19)
+	#define SPDIF_IN_USREN		(1 << 18)
+	#define SPDIF_IN_VALEN		(1 << 17)
+	#define SPDIF_IN_BLKEN		(1 << 16)
+
+	#define SPDIF_MODE_24BIT	(8 << 12)
+	#define SPDIF_MODE_23BIT	(7 << 12)
+	#define SPDIF_MODE_22BIT	(6 << 12)
+	#define SPDIF_MODE_21BIT	(5 << 12)
+	#define SPDIF_MODE_20BIT	(4 << 12)
+	#define SPDIF_MODE_19BIT	(3 << 12)
+	#define SPDIF_MODE_18BIT	(2 << 12)
+	#define SPDIF_MODE_17BIT	(1 << 12)
+	#define SPDIF_MODE_16BIT	(0 << 12)
+	#define SPDIF_MODE_MASK		(0x0F << 12)
+
+	#define SPDIF_IN_VALID		(1 << 11)
+	#define SPDIF_IN_SAMPLE		(1 << 10)
+	#define SPDIF_DATA_SWAP		(1 << 9)
+	#define SPDIF_IN_ENB		(1 << 8)
+	#define SPDIF_DATA_REVERT	(1 << 7)
+	#define SPDIF_XTRACT_16BIT	(1 << 6)
+	#define SPDIF_FIFO_THRES_16	(16 << 0)
+
+#define SPDIF_IN_IRQ_MASK	0x04
+#define SPDIF_IN_IRQ		0x08
+	#define SPDIF_IRQ_FIFOWRITE	(1 << 0)
+	#define SPDIF_IRQ_EMPTYFIFOREAD	(1 << 1)
+	#define SPDIF_IRQ_FIFOFULL	(1 << 2)
+	#define SPDIF_IRQ_OUTOFRANGE	(1 << 3)
+
+#define SPDIF_IN_STA		0x0C
+	#define SPDIF_IN_LOCK		(0x1 << 0)
+
+#endif /* SPDIF_IN_REGS_H */
-- 
cgit v1.2.3


From b2a4ec3d48fb53c99cb2e332f6e58eef770f1ed9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 22 Jun 2012 11:34:49 +0100
Subject: ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/da732x.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 04af369f228c..01be2a320e21 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1543,7 +1543,7 @@ static int da732x_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-struct snd_soc_codec_driver soc_codec_dev_da732x = {
+static struct snd_soc_codec_driver soc_codec_dev_da732x = {
 	.probe			= da732x_probe,
 	.remove			= da732x_remove,
 	.set_bias_level		= da732x_set_bias_level,
-- 
cgit v1.2.3


From bb1591b3de7c9c6b28f337e214100a394a126ab2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 22 Jun 2012 11:33:38 +0100
Subject: ASoC: isabelle: Staticise non-exported isabelle_dai

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

(limited to 'sound')

diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 0d62f3b0f474..5d8f39e32978 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -1036,7 +1036,7 @@ static struct snd_soc_dai_ops isabelle_ul_dai_ops = {
 };
 
 /* ISABELLE dai structure */
-struct snd_soc_dai_driver isabelle_dai[] = {
+static struct snd_soc_dai_driver isabelle_dai[] = {
 	{
 		.name = "isabelle-dl1",
 		.playback = {
-- 
cgit v1.2.3


From 62d4a4b99dfd647ef88b8434334eaa7497602857 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 22 Jun 2012 12:21:49 +0100
Subject: ASoC: dapm: Try to add all routes even if one fails

We may as well print as many errors as we can in one go rather than
requiring developers to iterate through all their typos.

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

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7365fed1ba74..32fbf10127f1 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2276,15 +2276,15 @@ err:
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num)
 {
-	int i, ret = 0;
+	int i, r, ret = 0;
 
 	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
-		ret = snd_soc_dapm_add_route(dapm, route);
-		if (ret < 0) {
+		r = snd_soc_dapm_add_route(dapm, route);
+		if (r < 0) {
 			dev_err(dapm->dev, "Failed to add route %s->%s\n",
 				route->source, route->sink);
-			break;
+			ret = r;
 		}
 		route++;
 	}
-- 
cgit v1.2.3


From 9dfdd5abcf2b350d4fdb207c0dff3194e2fd73db Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 22 Jun 2012 12:40:52 +0100
Subject: ASoC: io: Don't dereference regmap if we failed to get one

Avoids a crash in invalid configurations.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
 sound/soc/soc-io.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 44d0174b4d97..29183ef2b93d 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -145,10 +145,13 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 		if (!codec->control_data)
 			codec->control_data = dev_get_regmap(codec->dev, NULL);
 
-		ret = regmap_get_val_bytes(codec->control_data);
-		/* Errors are legitimate for non-integer byte multiples */
-		if (ret > 0)
-			codec->val_bytes = ret;
+		if (codec->control_data) {
+			ret = regmap_get_val_bytes(codec->control_data);
+			/* Errors are legitimate for non-integer byte
+			 * multiples */
+			if (ret > 0)
+				codec->val_bytes = ret;
+		}
 		break;
 
 	default:
-- 
cgit v1.2.3


From 07ed873e4c975a26c327a6bd306693678ef63351 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 18 Jun 2012 21:08:44 +0100
Subject: ASoC: Add shared code for Wolfson Arizona class devices

The Wolfson Arizona series of audio hub CODECs can share a large amount
of their driver code as the result of a common register map. This patch
adds some of this core support, providing a basis for the initial WM5102
audio driver.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig   |   5 +
 sound/soc/codecs/Makefile  |   2 +
 sound/soc/codecs/arizona.c | 781 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/arizona.h | 149 +++++++++
 4 files changed, 937 insertions(+)
 create mode 100644 sound/soc/codecs/arizona.c
 create mode 100644 sound/soc/codecs/arizona.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 43f5240e6942..2ae8082f23b5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -129,6 +129,11 @@ config SND_SOC_ALL_CODECS
 config SND_SOC_88PM860X
 	tristate
 
+config SND_SOC_ARIZONA
+	tristate
+	default y if SND_SOC_WM5102=y
+	default m if SND_SOC_WM5102=m
+
 config SND_SOC_WM_HUBS
 	tristate
 	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 3d30654f6fcc..3005ea6c1fd8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
+snd-soc-arizona-objs := arizona.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs42l52-objs := cs42l52.o
@@ -128,6 +129,7 @@ obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
+obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
new file mode 100644
index 000000000000..3b5730b90686
--- /dev/null
+++ b/sound/soc/codecs/arizona.c
@@ -0,0 +1,781 @@
+/*
+ * arizona.c - Wolfson Arizona class device shared support
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/gcd.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+
+#define ARIZONA_AIF_BCLK_CTRL                   0x00
+#define ARIZONA_AIF_TX_PIN_CTRL                 0x01
+#define ARIZONA_AIF_RX_PIN_CTRL                 0x02
+#define ARIZONA_AIF_RATE_CTRL                   0x03
+#define ARIZONA_AIF_FORMAT                      0x04
+#define ARIZONA_AIF_TX_BCLK_RATE                0x05
+#define ARIZONA_AIF_RX_BCLK_RATE                0x06
+#define ARIZONA_AIF_FRAME_CTRL_1                0x07
+#define ARIZONA_AIF_FRAME_CTRL_2                0x08
+#define ARIZONA_AIF_FRAME_CTRL_3                0x09
+#define ARIZONA_AIF_FRAME_CTRL_4                0x0A
+#define ARIZONA_AIF_FRAME_CTRL_5                0x0B
+#define ARIZONA_AIF_FRAME_CTRL_6                0x0C
+#define ARIZONA_AIF_FRAME_CTRL_7                0x0D
+#define ARIZONA_AIF_FRAME_CTRL_8                0x0E
+#define ARIZONA_AIF_FRAME_CTRL_9                0x0F
+#define ARIZONA_AIF_FRAME_CTRL_10               0x10
+#define ARIZONA_AIF_FRAME_CTRL_11               0x11
+#define ARIZONA_AIF_FRAME_CTRL_12               0x12
+#define ARIZONA_AIF_FRAME_CTRL_13               0x13
+#define ARIZONA_AIF_FRAME_CTRL_14               0x14
+#define ARIZONA_AIF_FRAME_CTRL_15               0x15
+#define ARIZONA_AIF_FRAME_CTRL_16               0x16
+#define ARIZONA_AIF_FRAME_CTRL_17               0x17
+#define ARIZONA_AIF_FRAME_CTRL_18               0x18
+#define ARIZONA_AIF_TX_ENABLES                  0x19
+#define ARIZONA_AIF_RX_ENABLES                  0x1A
+#define ARIZONA_AIF_FORCE_WRITE                 0x1B
+
+#define arizona_fll_err(_fll, fmt, ...) \
+	dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+#define arizona_fll_warn(_fll, fmt, ...) \
+	dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+#define arizona_fll_dbg(_fll, fmt, ...) \
+	dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+
+#define arizona_aif_err(_dai, fmt, ...) \
+	dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+#define arizona_aif_warn(_dai, fmt, ...) \
+	dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+#define arizona_aif_dbg(_dai, fmt, ...) \
+	dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+
+const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
+	"None",
+	"Tone Generator 1",
+	"Tone Generator 2",
+	"Haptics",
+	"AEC",
+	"Mic Mute Mixer",
+	"Noise Generator",
+	"IN1L",
+	"IN1R",
+	"IN2L",
+	"IN2R",
+	"IN3L",
+	"IN3R",
+	"AIF1RX1",
+	"AIF1RX2",
+	"AIF1RX3",
+	"AIF1RX4",
+	"AIF1RX5",
+	"AIF1RX6",
+	"AIF1RX7",
+	"AIF1RX8",
+	"AIF2RX1",
+	"AIF2RX2",
+	"AIF3RX1",
+	"AIF3RX2",
+	"SLIMRX1",
+	"SLIMRX2",
+	"SLIMRX3",
+	"SLIMRX4",
+	"SLIMRX5",
+	"SLIMRX6",
+	"SLIMRX7",
+	"SLIMRX8",
+	"EQ1",
+	"EQ2",
+	"EQ3",
+	"EQ4",
+	"DRC1L",
+	"DRC1R",
+	"DRC2L",
+	"DRC2R",
+	"LHPF1",
+	"LHPF2",
+	"LHPF3",
+	"LHPF4",
+	"DSP1.1",
+	"DSP1.2",
+	"DSP1.3",
+	"DSP1.4",
+	"DSP1.5",
+	"DSP1.6",
+	"ASRC1L",
+	"ASRC1R",
+	"ASRC2L",
+	"ASRC2R",
+};
+EXPORT_SYMBOL_GPL(arizona_mixer_texts);
+
+int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
+	0x00,  /* None */
+	0x04,  /* Tone */
+	0x05,
+	0x06,  /* Haptics */
+	0x08,  /* AEC */
+	0x0c,  /* Noise mixer */
+	0x0d,  /* Comfort noise */
+	0x10,  /* IN1L */
+	0x11,
+	0x12,
+	0x13,
+	0x14,
+	0x15,
+	0x20,  /* AIF1RX1 */
+	0x21,
+	0x22,
+	0x23,
+	0x24,
+	0x25,
+	0x26,
+	0x27,
+	0x28,  /* AIF2RX1 */
+	0x29,
+	0x30,  /* AIF3RX1 */
+	0x31,
+	0x38,  /* SLIMRX1 */
+	0x39,
+	0x3a,
+	0x3b,
+	0x3c,
+	0x3d,
+	0x3e,
+	0x3f,
+	0x50,  /* EQ1 */
+	0x51,
+	0x52,
+	0x53,
+	0x58,  /* DRC1L */
+	0x59,
+	0x5a,
+	0x5b,
+	0x60,  /* LHPF1 */
+	0x61,
+	0x62,
+	0x63,
+	0x68,  /* DSP1.1 */
+	0x69,
+	0x6a,
+	0x6b,
+	0x6c,
+	0x6d,
+	0x90,  /* ASRC1L */
+	0x91,
+	0x92,
+	0x93,
+};
+EXPORT_SYMBOL_GPL(arizona_mixer_values);
+
+const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
+EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
+
+static const char *arizona_lhpf_mode_text[] = {
+	"Low-pass", "High-pass"
+};
+
+const struct soc_enum arizona_lhpf1_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf1_mode);
+
+const struct soc_enum arizona_lhpf2_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf2_mode);
+
+const struct soc_enum arizona_lhpf3_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf3_mode);
+
+const struct soc_enum arizona_lhpf4_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
+
+int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
+		  int event)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_in_ev);
+
+int arizona_out_ev(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol,
+		   int event)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_out_ev);
+
+int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+		       int source, unsigned int freq, int dir)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	char *name;
+	unsigned int reg;
+	unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK;
+	unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT;
+	unsigned int *clk;
+
+	switch (clk_id) {
+	case ARIZONA_CLK_SYSCLK:
+		name = "SYSCLK";
+		reg = ARIZONA_SYSTEM_CLOCK_1;
+		clk = &priv->sysclk;
+		mask |= ARIZONA_SYSCLK_FRAC;
+		break;
+	case ARIZONA_CLK_ASYNCCLK:
+		name = "ASYNCCLK";
+		reg = ARIZONA_ASYNC_CLOCK_1;
+		clk = &priv->asyncclk;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (freq) {
+	case  5644800:
+	case  6144000:
+		break;
+	case 11289600:
+	case 12288000:
+		val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		break;
+	case 22579200:
+	case 24576000:
+		val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		break;
+	case 45158400:
+	case 49152000:
+		val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	*clk = freq;
+
+	if (freq % 6144000)
+		val |= ARIZONA_SYSCLK_FRAC;
+
+	dev_dbg(arizona->dev, "%s set to %uHz", name, freq);
+
+	return regmap_update_bits(arizona->regmap, reg, mask, val);
+}
+EXPORT_SYMBOL_GPL(arizona_set_sysclk);
+
+static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int lrclk, bclk, mode, base;
+
+	base = dai->driver->base;
+
+	lrclk = 0;
+	bclk = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		mode = 0;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		mode = 1;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		mode = 2;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode = 3;
+		break;
+	default:
+		arizona_aif_err(dai, "Unsupported DAI format %d\n",
+				fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		bclk |= ARIZONA_AIF1_BCLK_MSTR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		bclk |= ARIZONA_AIF1_BCLK_MSTR;
+		lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR;
+		break;
+	default:
+		arizona_aif_err(dai, "Unsupported master mode %d\n",
+				fmt & SND_SOC_DAIFMT_MASTER_MASK);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		bclk |= ARIZONA_AIF1_BCLK_INV;
+		lrclk |= ARIZONA_AIF1TX_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		bclk |= ARIZONA_AIF1_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		lrclk |= ARIZONA_AIF1TX_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
+			    ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR,
+			    bclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL,
+			    ARIZONA_AIF1TX_LRCLK_INV |
+			    ARIZONA_AIF1TX_LRCLK_MSTR, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL,
+			    ARIZONA_AIF1RX_LRCLK_INV |
+			    ARIZONA_AIF1RX_LRCLK_MSTR, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT,
+			    ARIZONA_AIF1_FMT_MASK, mode);
+
+	return 0;
+}
+
+static const int arizona_48k_rates[] = {
+	-1,
+	48000,
+	64000,
+	96000,
+	128000,
+	192000,
+	256000,
+	384000,
+	512000,
+	768000,
+	1024000,
+	1536000,
+	2048000,
+	3072000,
+	4096000,
+	6144000,
+	8192000,
+	12288000,
+	24576000,
+};
+
+static const int arizona_44k1_rates[] = {
+	-1,
+	44100,
+	58800,
+	88200,
+	117600,
+	177640,
+	235200,
+	352800,
+	470400,
+	705600,
+	940800,
+	1411200,
+	1881600,
+	2882400,
+	3763200,
+	5644800,
+	7526400,
+	11289600,
+	22579200,
+};
+
+static int arizona_sr_vals[] = {
+	0,
+	12000,
+	24000,
+	48000,
+	96000,
+	192000,
+	384000,
+	768000,
+	0,
+	11025,
+	22050,
+	44100,
+	88200,
+	176400,
+	352800,
+	705600,
+	4000,
+	8000,
+	16000,
+	32000,
+	64000,
+	128000,
+	256000,
+	512000,
+};
+
+static int arizona_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int base = dai->driver->base;
+	const int *rates;
+	int i;
+	int bclk, lrclk, wl, frame, sr_val;
+
+	if (params_rate(params) % 8000)
+		rates = &arizona_44k1_rates[0];
+	else
+		rates = &arizona_48k_rates[0];
+
+	for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) {
+		if (rates[i] == snd_soc_params_to_bclk(params)) {
+			bclk = i;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(arizona_44k1_rates)) {
+		arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+				params_rate(params));
+		return -EINVAL;
+	}
+
+	/*
+	 * We will need to be more flexible than this in future,
+	 * currently we use a single sample rate for the chip.
+	 */
+	for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++)
+		if (arizona_sr_vals[i] == params_rate(params))
+			break;
+	if (i == ARRAY_SIZE(arizona_sr_vals)) {
+		arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+				params_rate(params));
+		return -EINVAL;
+	}
+	sr_val = i;
+
+	lrclk = snd_soc_params_to_bclk(params) / params_rate(params);
+
+	arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
+			rates[bclk], rates[bclk] / lrclk);
+
+	wl = snd_pcm_format_width(params_format(params));
+	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+
+	snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
+			    ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
+			    ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
+			    ARIZONA_AIF1TX_BCPF_MASK, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE,
+			    ARIZONA_AIF1RX_BCPF_MASK, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1,
+			    ARIZONA_AIF1TX_WL_MASK |
+			    ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2,
+			    ARIZONA_AIF1RX_WL_MASK |
+			    ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
+
+	return 0;
+}
+
+const struct snd_soc_dai_ops arizona_dai_ops = {
+	.set_fmt = arizona_set_fmt,
+	.hw_params = arizona_hw_params,
+};
+
+static irqreturn_t arizona_fll_lock(int irq, void *data)
+{
+	struct arizona_fll *fll = data;
+
+	arizona_fll_dbg(fll, "Locked\n");
+
+	complete(&fll->lock);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t arizona_fll_clock_ok(int irq, void *data)
+{
+	struct arizona_fll *fll = data;
+
+	arizona_fll_dbg(fll, "clock OK\n");
+
+	complete(&fll->ok);
+
+	return IRQ_HANDLED;
+}
+
+static struct {
+	unsigned int min;
+	unsigned int max;
+	u16 fratio;
+	int ratio;
+} fll_fratios[] = {
+	{       0,    64000, 4, 16 },
+	{   64000,   128000, 3,  8 },
+	{  128000,   256000, 2,  4 },
+	{  256000,  1000000, 1,  2 },
+	{ 1000000, 13500000, 0,  1 },
+};
+
+struct arizona_fll_cfg {
+	int n;
+	int theta;
+	int lambda;
+	int refdiv;
+	int outdiv;
+	int fratio;
+};
+
+static int arizona_calc_fll(struct arizona_fll *fll,
+			    struct arizona_fll_cfg *cfg,
+			    unsigned int Fref,
+			    unsigned int Fout)
+{
+	unsigned int target, div, gcd_fll;
+	int i, ratio;
+
+	arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout);
+
+	/* Fref must be <=13.5MHz */
+	div = 1;
+	cfg->refdiv = 0;
+	while ((Fref / div) > 13500000) {
+		div *= 2;
+		cfg->refdiv++;
+
+		if (div > 8) {
+			arizona_fll_err(fll,
+					"Can't scale %dMHz in to <=13.5MHz\n",
+					Fref);
+			return -EINVAL;
+		}
+	}
+
+	/* Apply the division for our remaining calculations */
+	Fref /= div;
+
+	/* Fvco should be 90-100MHz; don't check the upper bound */
+	div = 1;
+	while (Fout * div < 90000000) {
+		div++;
+		if (div > 7) {
+			arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
+					Fout);
+			return -EINVAL;
+		}
+	}
+	target = Fout * div;
+	cfg->outdiv = div;
+
+	arizona_fll_dbg(fll, "Fvco=%dHz\n", target);
+
+	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+			cfg->fratio = fll_fratios[i].fratio;
+			ratio = fll_fratios[i].ratio;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(fll_fratios)) {
+		arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
+				Fref);
+		return -EINVAL;
+	}
+
+	cfg->n = target / (ratio * Fref);
+
+	if (target % Fref) {
+		gcd_fll = gcd(target, ratio * Fref);
+		arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll);
+
+		cfg->theta = (target - (cfg->n * ratio * Fref))
+			/ gcd_fll;
+		cfg->lambda = (ratio * Fref) / gcd_fll;
+	} else {
+		cfg->theta = 0;
+		cfg->lambda = 0;
+	}
+
+	arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
+			cfg->n, cfg->theta, cfg->lambda);
+	arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
+			cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
+
+	return 0;
+
+}
+
+static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
+			      struct arizona_fll_cfg *cfg, int source)
+{
+	regmap_update_bits(arizona->regmap, base + 3,
+			   ARIZONA_FLL1_THETA_MASK, cfg->theta);
+	regmap_update_bits(arizona->regmap, base + 4,
+			   ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda);
+	regmap_update_bits(arizona->regmap, base + 5,
+			   ARIZONA_FLL1_FRATIO_MASK,
+			   cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT);
+	regmap_update_bits(arizona->regmap, base + 6,
+			   ARIZONA_FLL1_CLK_REF_DIV_MASK |
+			   ARIZONA_FLL1_CLK_REF_SRC_MASK,
+			   cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
+			   source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
+
+	regmap_update_bits(arizona->regmap, base + 2,
+			   ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
+			   ARIZONA_FLL1_CTRL_UPD | cfg->n);
+}
+
+int arizona_set_fll(struct arizona_fll *fll, int source,
+		    unsigned int Fref, unsigned int Fout)
+{
+	struct arizona *arizona = fll->arizona;
+	struct arizona_fll_cfg cfg, sync;
+	unsigned int reg, val;
+	int syncsrc;
+	bool ena;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, fll->base + 1, &reg);
+	if (ret != 0) {
+		arizona_fll_err(fll, "Failed to read current state: %d\n",
+				ret);
+		return ret;
+	}
+	ena = reg & ARIZONA_FLL1_ENA;
+
+	if (Fout) {
+		/* Do we have a 32kHz reference? */
+		regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
+		switch (val & ARIZONA_CLK_32K_SRC_MASK) {
+		case ARIZONA_CLK_SRC_MCLK1:
+		case ARIZONA_CLK_SRC_MCLK2:
+			syncsrc = val & ARIZONA_CLK_32K_SRC_MASK;
+			break;
+		default:
+			syncsrc = -1;
+		}
+
+		if (source == syncsrc)
+			syncsrc = -1;
+
+		if (syncsrc >= 0) {
+			ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+			if (ret != 0)
+				return ret;
+
+			ret = arizona_calc_fll(fll, &cfg, 32768, Fout);
+			if (ret != 0)
+				return ret;
+		} else {
+			ret = arizona_calc_fll(fll, &cfg, Fref, Fout);
+			if (ret != 0)
+				return ret;
+		}
+	} else {
+		regmap_update_bits(arizona->regmap, fll->base + 1,
+				   ARIZONA_FLL1_ENA, 0);
+		regmap_update_bits(arizona->regmap, fll->base + 0x11,
+				   ARIZONA_FLL1_SYNC_ENA, 0);
+
+		if (ena)
+			pm_runtime_put_autosuspend(arizona->dev);
+
+		return 0;
+	}
+
+	regmap_update_bits(arizona->regmap, fll->base + 5,
+			   ARIZONA_FLL1_OUTDIV_MASK,
+			   cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+	if (syncsrc >= 0) {
+		arizona_apply_fll(arizona, fll->base, &cfg, syncsrc);
+		arizona_apply_fll(arizona, fll->base + 0x10, &sync, source);
+	} else {
+		arizona_apply_fll(arizona, fll->base, &cfg, source);
+	}
+
+	if (!ena)
+		pm_runtime_get(arizona->dev);
+
+	/* Clear any pending completions */
+	try_wait_for_completion(&fll->ok);
+
+	regmap_update_bits(arizona->regmap, fll->base + 1,
+			   ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
+	if (syncsrc >= 0)
+		regmap_update_bits(arizona->regmap, fll->base + 0x11,
+				   ARIZONA_FLL1_SYNC_ENA,
+				   ARIZONA_FLL1_SYNC_ENA);
+
+	ret = wait_for_completion_timeout(&fll->ok,
+					  msecs_to_jiffies(25));
+	if (ret == 0)
+		arizona_fll_warn(fll, "Timed out waiting for lock\n");
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_set_fll);
+
+int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
+		     int ok_irq, struct arizona_fll *fll)
+{
+	int ret;
+
+	init_completion(&fll->lock);
+	init_completion(&fll->ok);
+
+	fll->id = id;
+	fll->base = base;
+	fll->arizona = arizona;
+
+	snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id);
+	snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
+		 "FLL%d clock OK", id);
+
+	ret = arizona_request_irq(arizona, lock_irq, fll->lock_name,
+				  arizona_fll_lock, fll);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n",
+			id, ret);
+	}
+
+	ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name,
+				  arizona_fll_clock_ok, fll);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n",
+			id, ret);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_fll);
+
+MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
new file mode 100644
index 000000000000..8c2ca1d9dbae
--- /dev/null
+++ b/sound/soc/codecs/arizona.h
@@ -0,0 +1,149 @@
+/*
+ * arizona.h - Wolfson Arizona class device shared support
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _ASOC_ARIZONA_H
+#define _ASOC_ARIZONA_H
+
+#include <linux/completion.h>
+
+#include <sound/soc.h>
+
+#define ARIZONA_CLK_SYSCLK   1
+#define ARIZONA_CLK_ASYNCCLK 2
+
+#define ARIZONA_CLK_SRC_MCLK1    0x0
+#define ARIZONA_CLK_SRC_MCLK2    0x1
+#define ARIZONA_CLK_SRC_FLL1     0x4
+#define ARIZONA_CLK_SRC_FLL2     0x5
+#define ARIZONA_CLK_SRC_AIF1BCLK 0x8
+#define ARIZONA_CLK_SRC_AIF2BCLK 0x9
+#define ARIZONA_CLK_SRC_AIF3BCLK 0xa
+
+#define ARIZONA_FLL_SRC_MCLK1      0
+#define ARIZONA_FLL_SRC_MCLK2      1
+#define ARIZONA_FLL_SRC_SLIMCLK    2
+#define ARIZONA_FLL_SRC_FLL1       3
+#define ARIZONA_FLL_SRC_FLL2       4
+#define ARIZONA_FLL_SRC_AIF1BCLK   5
+#define ARIZONA_FLL_SRC_AIF2BCLK   6
+#define ARIZONA_FLL_SRC_AIF3BCLK   7
+#define ARIZONA_FLL_SRC_AIF1LRCLK  8
+#define ARIZONA_FLL_SRC_AIF2LRCLK  9
+#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+
+#define ARIZONA_MIXER_VOL_MASK             0x00FE
+#define ARIZONA_MIXER_VOL_SHIFT                 1
+#define ARIZONA_MIXER_VOL_WIDTH                 7
+
+struct arizona;
+
+struct arizona_priv {
+	struct arizona *arizona;
+	int sysclk;
+	int asyncclk;
+};
+
+#define ARIZONA_NUM_MIXER_INPUTS 55
+
+extern const unsigned int arizona_mixer_tlv[];
+extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
+extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
+
+#define ARIZONA_MIXER_CONTROLS(name, base) \
+	SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv),			\
+	SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv),			\
+	SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv),			\
+	SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv)
+
+#define ARIZONA_MUX_ENUM_DECL(name, reg) \
+	SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff,			\
+				   arizona_mixer_texts, arizona_mixer_values)
+
+#define ARIZONA_MUX_CTL_DECL(name) \
+	const struct snd_kcontrol_new name##_mux =	\
+		SOC_DAPM_VALUE_ENUM("Route", name##_enum)
+
+#define ARIZONA_MIXER_ENUMS(name, base_reg) \
+	static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg);      \
+	static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2);  \
+	static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4);  \
+	static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6);  \
+	static ARIZONA_MUX_CTL_DECL(name##_in1); \
+	static ARIZONA_MUX_CTL_DECL(name##_in2); \
+	static ARIZONA_MUX_CTL_DECL(name##_in3); \
+	static ARIZONA_MUX_CTL_DECL(name##_in4)
+
+#define ARIZONA_MUX(name, ctrl) \
+	SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+#define ARIZONA_MIXER_WIDGETS(name, name_str)	\
+	ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \
+	ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \
+	ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \
+	ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \
+	SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0)
+
+#define ARIZONA_MIXER_ROUTES(widget, name) \
+	{ widget, NULL, name " Mixer" },         \
+	{ name " Mixer", NULL, name " Input 1" }, \
+	{ name " Mixer", NULL, name " Input 2" }, \
+	{ name " Mixer", NULL, name " Input 3" }, \
+	{ name " Mixer", NULL, name " Input 4" }, \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 4")
+
+extern const struct soc_enum arizona_lhpf1_mode;
+extern const struct soc_enum arizona_lhpf2_mode;
+extern const struct soc_enum arizona_lhpf3_mode;
+extern const struct soc_enum arizona_lhpf4_mode;
+
+extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol,
+			 int event);
+extern int arizona_out_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event);
+
+extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+			      int source, unsigned int freq, int dir);
+
+extern const struct snd_soc_dai_ops arizona_dai_ops;
+
+#define ARIZONA_FLL_NAME_LEN 20
+
+struct arizona_fll {
+	struct arizona *arizona;
+	int id;
+	unsigned int base;
+	struct completion lock;
+	struct completion ok;
+
+	char lock_name[ARIZONA_FLL_NAME_LEN];
+	char clock_ok_name[ARIZONA_FLL_NAME_LEN];
+};
+
+extern int arizona_init_fll(struct arizona *arizona, int id, int base,
+			    int lock_irq, int ok_irq, struct arizona_fll *fll);
+extern int arizona_set_fll(struct arizona_fll *fll, int source,
+			   unsigned int Fref, unsigned int Fout);
+
+#endif
-- 
cgit v1.2.3


From 93e8791dd34ca0c3371d65c4488249d41de02776 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 19 Jun 2012 16:38:15 +0100
Subject: ASoC: wm5102: Initial driver

The WM5102 is a highly-integrated low-power audio system for smartphones,
tablets and other portable audio devices based on the Arizona platform.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   4 +
 sound/soc/codecs/Makefile |   2 +
 sound/soc/codecs/wm5102.c | 870 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm5102.h |  21 ++
 4 files changed, 897 insertions(+)
 create mode 100644 sound/soc/codecs/wm5102.c
 create mode 100644 sound/soc/codecs/wm5102.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 2ae8082f23b5..1de24ccfe1c3 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -73,6 +73,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM2000 if I2C
 	select SND_SOC_WM2200 if I2C
 	select SND_SOC_WM5100 if I2C
+	select SND_SOC_WM5102 if MFD_WM5102
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
@@ -330,6 +331,9 @@ config SND_SOC_WM2200
 config SND_SOC_WM5100
 	tristate
 
+config SND_SOC_WM5102
+	tristate
+
 config SND_SOC_WM8350
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 3005ea6c1fd8..d35ba7f06fcf 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -63,6 +63,7 @@ snd-soc-wm1250-ev1-objs := wm1250-ev1.o
 snd-soc-wm2000-objs := wm2000.o
 snd-soc-wm2200-objs := wm2200.o
 snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
+snd-soc-wm5102-objs := wm5102.o
 snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
@@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
 obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o
 obj-$(CONFIG_SND_SOC_WM2200)	+= snd-soc-wm2200.o
 obj-$(CONFIG_SND_SOC_WM5100)	+= snd-soc-wm5100.o
+obj-$(CONFIG_SND_SOC_WM5102)	+= snd-soc-wm5102.o
 obj-$(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
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
new file mode 100644
index 000000000000..9b9ea7fd0d7d
--- /dev/null
+++ b/sound/soc/codecs/wm5102.c
@@ -0,0 +1,870 @@
+/*
+ * wm5102.c  --  WM5102 ALSA SoC Audio driver
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/wm5102.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm5102.h"
+
+struct wm5102_priv {
+	struct arizona_priv core;
+	struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new wm5102_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
+	   ARIZONA_IN3_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL,
+		       ARIZONA_IN1R_CONTROL,
+		       ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+		       ARIZONA_IN2R_CONTROL,
+		       ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL,
+		       ARIZONA_IN3R_CONTROL,
+		       ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_PGA_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_PGA_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_PGA_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
+		   ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+	       ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+	   ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	       ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
+		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
+		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+		     ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+		    0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+		    ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+		 ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+		 ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "IN1L", "IN1L PGA" }, \
+	{ name, "IN1R", "IN1R PGA" }, \
+	{ name, "IN2L", "IN2L PGA" }, \
+	{ name, "IN2R", "IN2R PGA" }, \
+	{ name, "IN3L", "IN3L PGA" }, \
+	{ name, "IN3R", "IN3R PGA" }, \
+	{ name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF1RX7", "AIF1RX7" }, \
+	{ name, "AIF1RX8", "AIF1RX8" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "DRC2L", "DRC2L" }, \
+	{ name, "DRC2R", "DRC2R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }
+
+static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
+	{ "AIF2 Capture", NULL, "DBVDD2" },
+	{ "AIF2 Playback", NULL, "DBVDD2" },
+
+	{ "AIF3 Capture", NULL, "DBVDD3" },
+	{ "AIF3 Playback", NULL, "DBVDD3" },
+
+	{ "OUT1L", NULL, "CPVDD" },
+	{ "OUT1R", NULL, "CPVDD" },
+	{ "OUT2L", NULL, "CPVDD" },
+	{ "OUT2R", NULL, "CPVDD" },
+	{ "OUT3L", NULL, "CPVDD" },
+
+	{ "OUT4L", NULL, "SPKVDDL" },
+	{ "OUT4R", NULL, "SPKVDDR" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT2L", NULL, "SYSCLK" },
+	{ "OUT2R", NULL, "SYSCLK" },
+	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+	{ "OUT4R", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+	{ "MICBIAS3", NULL, "MICVDD" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "Mic Mute Mixer", NULL, "Noise Mixer" },
+	{ "Mic Mute Mixer", NULL, "Mic Mixer" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+	{ "AIF1 Capture", NULL, "AIF1TX7" },
+	{ "AIF1 Capture", NULL, "AIF1TX8" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+	{ "AIF1RX7", NULL, "AIF1 Playback" },
+	{ "AIF1RX8", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+
+	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+	ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+	ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+	ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+	ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+	ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+	ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+	ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+	ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+	ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+	ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+	ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+	ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+	ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
+	ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	{ "HPOUT1L", NULL, "OUT1L" },
+	{ "HPOUT1R", NULL, "OUT1R" },
+
+	{ "HPOUT2L", NULL, "OUT2L" },
+	{ "HPOUT2R", NULL, "OUT2R" },
+
+	{ "EPOUTN", NULL, "OUT3L" },
+	{ "EPOUTP", NULL, "OUT3L" },
+
+	{ "SPKOUTLN", NULL, "OUT4L" },
+	{ "SPKOUTLP", NULL, "OUT4L" },
+
+	{ "SPKOUTRN", NULL, "OUT4R" },
+	{ "SPKOUTRP", NULL, "OUT4R" },
+
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+};
+
+static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fll_id) {
+	case WM5102_FLL1:
+		return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout);
+	case WM5102_FLL2:
+		return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+#define WM5102_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM5102_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_driver wm5102_dai[] = {
+	{
+		.name = "wm5102-aif1",
+		.id = 1,
+		.base = ARIZONA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 8,
+			 .rates = WM5102_RATES,
+			 .formats = WM5102_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5102-aif2",
+		.id = 2,
+		.base = ARIZONA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5102_RATES,
+			 .formats = WM5102_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5102-aif3",
+		.id = 3,
+		.base = ARIZONA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF3 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5102_RATES,
+			 .formats = WM5102_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static int wm5102_codec_probe(struct snd_soc_codec *codec)
+{
+	struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	codec->control_data = priv->core.arizona->regmap;
+	return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+}
+
+#define WM5102_DIG_VU 0x0200
+
+static unsigned int wm5102_digital_vu[] = {
+	ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	ARIZONA_ADC_DIGITAL_VOLUME_1R,
+	ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	ARIZONA_ADC_DIGITAL_VOLUME_2R,
+	ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	ARIZONA_ADC_DIGITAL_VOLUME_3R,
+
+	ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	ARIZONA_DAC_DIGITAL_VOLUME_1R,
+	ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	ARIZONA_DAC_DIGITAL_VOLUME_2R,
+	ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	ARIZONA_DAC_DIGITAL_VOLUME_3R,
+	ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4R,
+	ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm5102 = {
+	.probe = wm5102_codec_probe,
+
+	.idle_bias_off = true,
+
+	.set_sysclk = arizona_set_sysclk,
+	.set_pll = wm5102_set_fll,
+
+	.controls = wm5102_snd_controls,
+	.num_controls = ARRAY_SIZE(wm5102_snd_controls),
+	.dapm_widgets = wm5102_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm5102_dapm_widgets),
+	.dapm_routes = wm5102_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes),
+};
+
+static int __devinit wm5102_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct wm5102_priv *wm5102;
+	int i;
+
+	wm5102 = devm_kzalloc(&pdev->dev, sizeof(struct wm5102_priv),
+			      GFP_KERNEL);
+	if (wm5102 == NULL)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, wm5102);
+
+	wm5102->core.arizona = arizona;
+
+	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+			 &wm5102->fll[0]);
+	arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+			 &wm5102->fll[1]);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++)
+		regmap_update_bits(arizona->regmap, wm5102_digital_vu[i],
+				   WM5102_DIG_VU, WM5102_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102,
+				      wm5102_dai, ARRAY_SIZE(wm5102_dai));
+}
+
+static int __devexit wm5102_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver wm5102_codec_driver = {
+	.driver = {
+		.name = "wm5102-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm5102_probe,
+	.remove = __devexit_p(wm5102_remove),
+};
+
+module_platform_driver(wm5102_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM5102 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm5102-codec");
diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h
new file mode 100644
index 000000000000..d30477f3070c
--- /dev/null
+++ b/sound/soc/codecs/wm5102.h
@@ -0,0 +1,21 @@
+/*
+ * wm5102.h  --  WM5102 ALSA SoC Audio driver
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM5102_H
+#define _WM5102_H
+
+#include "arizona.h"
+
+#define WM5102_FLL1 1
+#define WM5102_FLL2 2
+
+#endif
-- 
cgit v1.2.3


From 125821ae539ab60f432b5e10dadfd7bbf069ca7a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 22 Jun 2012 14:30:29 +0200
Subject: ALSA: hda - Add the inverted digital mic workaround to Realtek codecs

Some laptops are equipped with ForteMedia digital mics that give the
differential input.  With such devices, summing stereo streams into a
mono (like PulseAudio does) results in almost silence.

This patch provides a workaround for this bug by adding a new mixer
switch to turn on/off the right channel of digital mic, just like a
similar fix for Conexant codecs.

When the new switch "Inverted Internal Mic Capture Switch" is off and
the current input source is the digital mic, the right channel of the
recording stream is muted.  When another input source is selected, the
right channel is restored.

Tested-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 127 +++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 125 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f8f4906e498d..a0a3cf956503 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -170,6 +170,7 @@ struct alc_spec {
 	hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS];
 	unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS];
 	int int_mic_idx, ext_mic_idx, dock_mic_idx; /* for auto-mic */
+	hda_nid_t inv_dmic_pin;
 
 	/* hooks */
 	void (*init_hook)(struct hda_codec *codec);
@@ -201,6 +202,8 @@ struct alc_spec {
 	unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
 	unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
 	unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
+	unsigned int inv_dmic_fixup:1; /* has inverted digital-mic workaround */
+	unsigned int inv_dmic_muted:1; /* R-ch of inv d-mic is muted? */
 
 	/* auto-mute control */
 	int automute_mode;
@@ -298,6 +301,7 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx)
 }
 
 static void call_update_outputs(struct hda_codec *codec);
+static void alc_inv_dmic_sync(struct hda_codec *codec, bool force);
 
 /* select the given imux item; either unmute exclusively or select the route */
 static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
@@ -368,6 +372,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 					  AC_VERB_SET_CONNECT_SEL,
 					  imux->items[idx].index);
 	}
+	alc_inv_dmic_sync(codec, true);
 	return 1;
 }
 
@@ -1556,14 +1561,14 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
 
 static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol,
-				 getput_call_t func, bool check_adc_switch)
+				 getput_call_t func, bool is_put)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
 	int i, err = 0;
 
 	mutex_lock(&codec->control_mutex);
-	if (check_adc_switch && spec->dyn_adc_switch) {
+	if (is_put && spec->dyn_adc_switch) {
 		for (i = 0; i < spec->num_adc_nids; i++) {
 			kcontrol->private_value =
 				HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
@@ -1584,6 +1589,8 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
 						    3, 0, HDA_INPUT);
 		err = func(kcontrol, ucontrol);
 	}
+	if (err >= 0 && is_put)
+		alc_inv_dmic_sync(codec, false);
  error:
 	mutex_unlock(&codec->control_mutex);
 	return err;
@@ -1675,6 +1682,108 @@ DEFINE_CAPMIX_NOSRC(1);
 DEFINE_CAPMIX_NOSRC(2);
 DEFINE_CAPMIX_NOSRC(3);
 
+/*
+ * Inverted digital-mic handling
+ *
+ * First off, it's a bit tricky.  The "Inverted Internal Mic Capture Switch"
+ * gives the additional mute only to the right channel of the digital mic
+ * capture stream.  This is a workaround for avoiding the almost silence
+ * by summing the stereo stream from some (known to be ForteMedia)
+ * digital mic unit.
+ *
+ * The logic is to call alc_inv_dmic_sync() after each action (possibly)
+ * modifying ADC amp.  When the mute flag is set, it mutes the R-channel
+ * without caching so that the cache can still keep the original value.
+ * The cached value is then restored when the flag is set off or any other
+ * than d-mic is used as the current input source.
+ */
+static void alc_inv_dmic_sync(struct hda_codec *codec, bool force)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	if (!spec->inv_dmic_fixup)
+		return;
+	if (!spec->inv_dmic_muted && !force)
+		return;
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		int src = spec->dyn_adc_switch ? 0 : i;
+		bool dmic_fixup = false;
+		hda_nid_t nid;
+		int parm, dir, v;
+
+		if (spec->inv_dmic_muted &&
+		    spec->imux_pins[spec->cur_mux[src]] == spec->inv_dmic_pin)
+			dmic_fixup = true;
+		if (!dmic_fixup && !force)
+			continue;
+		if (spec->vol_in_capsrc) {
+			nid = spec->capsrc_nids[i];
+			parm = AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT;
+			dir = HDA_OUTPUT;
+		} else {
+			nid = spec->adc_nids[i];
+			parm = AC_AMP_SET_RIGHT | AC_AMP_SET_INPUT;
+			dir = HDA_INPUT;
+		}
+		/* we care only right channel */
+		v = snd_hda_codec_amp_read(codec, nid, 1, dir, 0);
+		if (v & 0x80) /* if already muted, we don't need to touch */
+			continue;
+		if (dmic_fixup) /* add mute for d-mic */
+			v |= 0x80;
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    parm | v);
+	}
+}
+
+static int alc_inv_dmic_sw_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+
+	ucontrol->value.integer.value[0] = !spec->inv_dmic_muted;
+	return 0;
+}
+
+static int alc_inv_dmic_sw_put(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	unsigned int val = !ucontrol->value.integer.value[0];
+
+	if (val == spec->inv_dmic_muted)
+		return 0;
+	spec->inv_dmic_muted = val;
+	alc_inv_dmic_sync(codec, true);
+	return 0;
+}
+
+static const struct snd_kcontrol_new alc_inv_dmic_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.info = snd_ctl_boolean_mono_info,
+	.get = alc_inv_dmic_sw_get,
+	.put = alc_inv_dmic_sw_put,
+};
+
+static int alc_add_inv_dmic_mixer(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct alc_spec *spec = codec->spec;
+	struct snd_kcontrol_new *knew = alc_kcontrol_new(spec);
+	if (!knew)
+		return -ENOMEM;
+	*knew = alc_inv_dmic_sw;
+	knew->name = kstrdup("Inverted Internal Mic Capture Switch", GFP_KERNEL);
+	if (!knew->name)
+		return -ENOMEM;
+	spec->inv_dmic_fixup = 1;
+	spec->inv_dmic_muted = 0;
+	spec->inv_dmic_pin = nid;
+	return 0;
+}
+
 /*
  * virtual master controls
  */
@@ -2316,6 +2425,7 @@ static int alc_resume(struct hda_codec *codec)
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
+	alc_inv_dmic_sync(codec, true);
 	hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
@@ -6424,6 +6534,13 @@ static void alc272_fixup_mario(struct hda_codec *codec,
 		       "hda_codec: failed to override amp caps for NID 0x2\n");
 }
 
+static void alc662_fixup_inv_dmic(struct hda_codec *codec,
+				  const struct alc_fixup *fix, int action)
+{
+	if (action == ALC_FIXUP_ACT_PROBE)
+		alc_add_inv_dmic_mixer(codec, 0x12);
+}
+
 enum {
 	ALC662_FIXUP_ASPIRE,
 	ALC662_FIXUP_IDEAPAD,
@@ -6441,6 +6558,7 @@ enum {
 	ALC662_FIXUP_ASUS_MODE8,
 	ALC662_FIXUP_NO_JACK_DETECT,
 	ALC662_FIXUP_ZOTAC_Z68,
+	ALC662_FIXUP_INV_DMIC,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -6597,12 +6715,17 @@ static const struct alc_fixup alc662_fixups[] = {
 			{ }
 		}
 	},
+	[ALC662_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc662_fixup_inv_dmic,
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
-- 
cgit v1.2.3


From 693b613dc4657e3f9af2625e0097b1870c78bf8c Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 22 Jun 2012 19:12:10 +0200
Subject: ALSA: hda - Add inverted mic quirks for Asus U41SV, Acer 1810TZ and
 AOD260

These machines have inverted phase on right channel for their
internal mics.

BugLink: https://bugs.launchpad.net/bugs/997227
BugLink: https://bugs.launchpad.net/bugs/996611
BugLink: https://bugs.launchpad.net/bugs/1006089
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++------
 1 file changed, 17 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a0a3cf956503..d11fd0160748 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5918,6 +5918,14 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec,
 	}
 }
 
+static void alc269_fixup_inv_dmic(struct hda_codec *codec,
+				  const struct alc_fixup *fix, int action)
+{
+	if (action == ALC_FIXUP_ACT_PROBE)
+		alc_add_inv_dmic_mixer(codec, 0x12);
+}
+
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -5936,6 +5944,7 @@ enum {
 	ALC269VB_FIXUP_AMIC,
 	ALC269VB_FIXUP_DMIC,
 	ALC269_FIXUP_MIC2_MUTE_LED,
+	ALC269_FIXUP_INV_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -6060,12 +6069,19 @@ static const struct alc_fixup alc269_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_mic2_mute,
 	},
+	[ALC269_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_inv_dmic,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+	SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -6534,12 +6550,7 @@ static void alc272_fixup_mario(struct hda_codec *codec,
 		       "hda_codec: failed to override amp caps for NID 0x2\n");
 }
 
-static void alc662_fixup_inv_dmic(struct hda_codec *codec,
-				  const struct alc_fixup *fix, int action)
-{
-	if (action == ALC_FIXUP_ACT_PROBE)
-		alc_add_inv_dmic_mixer(codec, 0x12);
-}
+#define alc662_fixup_inv_dmic alc269_fixup_inv_dmic
 
 enum {
 	ALC662_FIXUP_ASPIRE,
-- 
cgit v1.2.3


From 6e72aa5f511daa2ffbd333ea99635c551b86013b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 25 Jun 2012 10:52:25 +0200
Subject: ALSA: hda - Add inv-dmic model to possible Realtek codecs

For convenience, add "inv-dmic" model string for enabling the inverted
internal mic workaround to possible Realtek codecs, so far,
ALC882-variants, ALC262, ALC268, ALC269-variants, and ALC662-variants.

Also, the model strings for hardware inv-dmic workarounds,
"alc269-dmic" and "alc271-dmic", are added for ALC269(VA) and ALC271
codecs as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 64 +++++++++++++++++++++++++++++++++++--------
 1 file changed, 52 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d11fd0160748..3e698e239dd8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1784,6 +1784,14 @@ static int alc_add_inv_dmic_mixer(struct hda_codec *codec, hda_nid_t nid)
 	return 0;
 }
 
+/* typically the digital mic is put at node 0x12 */
+static void alc_fixup_inv_dmic_0x12(struct hda_codec *codec,
+				    const struct alc_fixup *fix, int action)
+{
+	if (action == ALC_FIXUP_ACT_PROBE)
+		alc_add_inv_dmic_mixer(codec, 0x12);
+}
+
 /*
  * virtual master controls
  */
@@ -5017,6 +5025,7 @@ enum {
 	ALC889_FIXUP_DAC_ROUTE,
 	ALC889_FIXUP_MBP_VREF,
 	ALC889_FIXUP_IMAC91_VREF,
+	ALC882_FIXUP_INV_DMIC,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -5320,6 +5329,10 @@ static const struct alc_fixup alc882_fixups[] = {
 		.chained = true,
 		.chain_id = ALC882_FIXUP_GPIO1,
 	},
+	[ALC882_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -5394,6 +5407,7 @@ static const struct alc_model_fixup alc882_fixup_models[] = {
 	{.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"},
 	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
 	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
+	{.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{}
 };
 
@@ -5481,6 +5495,7 @@ enum {
 	ALC262_FIXUP_LENOVO_3000,
 	ALC262_FIXUP_BENQ,
 	ALC262_FIXUP_BENQ_T31,
+	ALC262_FIXUP_INV_DMIC,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -5532,6 +5547,10 @@ static const struct alc_fixup alc262_fixups[] = {
 			{}
 		}
 	},
+	[ALC262_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
@@ -5546,6 +5565,10 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
 	{}
 };
 
+static const struct alc_model_fixup alc262_fixup_models[] = {
+	{.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{}
+};
 
 /*
  */
@@ -5574,7 +5597,8 @@ static int patch_alc262(struct hda_codec *codec)
 #endif
 	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
 
-	alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
+	alc_pick_fixup(codec, alc262_fixup_models, alc262_fixup_tbl,
+		       alc262_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
@@ -5630,6 +5654,22 @@ static const struct hda_verb alc268_beep_init_verbs[] = {
 	{ }
 };
 
+enum {
+	ALC268_FIXUP_INV_DMIC,
+};
+
+static const struct alc_fixup alc268_fixups[] = {
+	[ALC268_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
+};
+
+static const struct alc_model_fixup alc268_fixup_models[] = {
+	{.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{}
+};
+
 /*
  * BIOS auto configuration
  */
@@ -5661,6 +5701,9 @@ static int patch_alc268(struct hda_codec *codec)
 
 	spec = codec->spec;
 
+	alc_pick_fixup(codec, alc268_fixup_models, NULL, alc268_fixups);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
 	/* automatic parse from the BIOS config */
 	err = alc268_parse_auto_config(codec);
 	if (err < 0)
@@ -5690,6 +5733,8 @@ static int patch_alc268(struct hda_codec *codec)
 	codec->patch_ops = alc_patch_ops;
 	spec->shutup = alc_eapd_shutup;
 
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
@@ -5918,13 +5963,6 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec,
 	}
 }
 
-static void alc269_fixup_inv_dmic(struct hda_codec *codec,
-				  const struct alc_fixup *fix, int action)
-{
-	if (action == ALC_FIXUP_ACT_PROBE)
-		alc_add_inv_dmic_mixer(codec, 0x12);
-}
-
 
 enum {
 	ALC269_FIXUP_SONY_VAIO,
@@ -6071,7 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = {
 	},
 	[ALC269_FIXUP_INV_DMIC] = {
 		.type = ALC_FIXUP_FUNC,
-		.v.func = alc269_fixup_inv_dmic,
+		.v.func = alc_fixup_inv_dmic_0x12,
 	},
 };
 
@@ -6157,6 +6195,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 static const struct alc_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"},
 	{.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"},
+	{.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"},
+	{.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"},
+	{.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{}
 };
 
@@ -6550,8 +6591,6 @@ static void alc272_fixup_mario(struct hda_codec *codec,
 		       "hda_codec: failed to override amp caps for NID 0x2\n");
 }
 
-#define alc662_fixup_inv_dmic alc269_fixup_inv_dmic
-
 enum {
 	ALC662_FIXUP_ASPIRE,
 	ALC662_FIXUP_IDEAPAD,
@@ -6728,7 +6767,7 @@ static const struct alc_fixup alc662_fixups[] = {
 	},
 	[ALC662_FIXUP_INV_DMIC] = {
 		.type = ALC_FIXUP_FUNC,
-		.v.func = alc662_fixup_inv_dmic,
+		.v.func = alc_fixup_inv_dmic_0x12,
 	},
 };
 
@@ -6817,6 +6856,7 @@ static const struct alc_model_fixup alc662_fixup_models[] = {
 	{.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"},
 	{.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
 	{.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
+	{.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{}
 };
 
-- 
cgit v1.2.3


From 1573ee81cb9ef24fa5acee6b7442e215e63ede2f Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= <u.kleine-koenig@pengutronix.de>
Date: Mon, 25 Jun 2012 09:28:46 +0200
Subject: ASoC: dmaengine_pcm: fix typo in comment
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dmaengine-pcm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c
index 2995334d8000..5df529eda251 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/soc/soc-dmaengine-pcm.c
@@ -270,7 +270,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd
  * Note that this function will use private_data field of the substream's
  * runtime. So it is not availabe to your pcm driver implementation. If you need
  * to keep additional data attached to a substream use
- * snd_dmaeinge_pcm_{set,get}_data.
+ * snd_dmaengine_pcm_{set,get}_data.
  */
 int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
 	dma_filter_fn filter_fn, void *filter_data)
-- 
cgit v1.2.3


From fd88759a42dc10f8230b3933a1ceb40bd88fccea Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 11:30:43 +0800
Subject: ASoC: wm8904: Rely entirely on the core for bias level management

Even though the WM8904 is able to use idle_bias_off during both probe
and resume we were needlessly leaving the device in standby mode. Instead
power the device down as soon as we've confirmed that we can talk to it
and don't manage the bias level at all over suspend and resume, the core
will take us down to our minimum power level.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8904.c | 30 +++---------------------------
 1 file changed, 3 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 5417b1183acb..ecab871573b1 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1945,25 +1945,6 @@ static struct snd_soc_dai_driver wm8904_dai = {
 	.symmetric_rates = 1,
 };
 
-#ifdef CONFIG_PM
-static int wm8904_suspend(struct snd_soc_codec *codec)
-{
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int wm8904_resume(struct snd_soc_codec *codec)
-{
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-#else
-#define wm8904_suspend NULL
-#define wm8904_resume NULL
-#endif
-
 static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
@@ -2143,7 +2124,10 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 		goto err_enable;
 	}
 
+	/* Can leave the device powered off until we need it */
 	regcache_cache_only(wm8904->regmap, true);
+	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+
 	/* Change some default settings - latch VU and enable ZC */
 	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
 			    WM8904_ADC_VU, WM8904_ADC_VU);
@@ -2198,11 +2182,6 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
 			    WM8904_POBCTRL, 0);
 
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	/* Bias level configuration will have done an extra enable */
-	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-
 	wm8904_handle_pdata(codec);
 
 	wm8904_add_widgets(codec);
@@ -2220,7 +2199,6 @@ static int wm8904_remove(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 	kfree(wm8904->retune_mobile_texts);
 	kfree(wm8904->drc_texts);
@@ -2231,8 +2209,6 @@ static int wm8904_remove(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_dev_wm8904 = {
 	.probe =	wm8904_probe,
 	.remove =	wm8904_remove,
-	.suspend =	wm8904_suspend,
-	.resume =	wm8904_resume,
 	.set_bias_level = wm8904_set_bias_level,
 	.idle_bias_off = true,
 };
-- 
cgit v1.2.3


From 03862cf62ea36d6cf3d94eee84b89578cbcf0213 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 11:41:58 +0800
Subject: ASoC: wm8904: Move regulator acquisition and device identification to
 I2C

It's more idiomatic to have the resource allocation at this level.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8904.c | 112 +++++++++++++++++++++-------------------------
 1 file changed, 51 insertions(+), 61 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index ecab871573b1..b178232c990c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -314,11 +314,6 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg)
 	}
 }
 
-static int wm8904_reset(struct snd_soc_codec *codec)
-{
-	return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0);
-}
-
 static int wm8904_configure_clocking(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
@@ -2082,52 +2077,6 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
-		wm8904->supplies[i].supply = wm8904_supply_names[i];
-
-	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies),
-				 wm8904->supplies);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
-		return ret;
-	}
-
-	ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
-				    wm8904->supplies);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
-		goto err_get;
-	}
-
-	ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read ID register\n");
-		goto err_enable;
-	}
-	if (ret != 0x8904) {
-		dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret);
-		ret = -EINVAL;
-		goto err_enable;
-	}
-
-	ret = snd_soc_read(codec, WM8904_REVISION);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read device revision: %d\n",
-			ret);
-		goto err_enable;
-	}
-	dev_info(codec->dev, "revision %c\n", ret + 'A');
-
-	ret = wm8904_reset(codec);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to issue reset\n");
-		goto err_enable;
-	}
-
-	/* Can leave the device powered off until we need it */
-	regcache_cache_only(wm8904->regmap, true);
-	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-
 	/* Change some default settings - latch VU and enable ZC */
 	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
 			    WM8904_ADC_VU, WM8904_ADC_VU);
@@ -2187,19 +2136,12 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 	wm8904_add_widgets(codec);
 
 	return 0;
-
-err_enable:
-	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-err_get:
-	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-	return ret;
 }
 
 static int wm8904_remove(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
-	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 	kfree(wm8904->retune_mobile_texts);
 	kfree(wm8904->drc_texts);
 
@@ -2230,7 +2172,8 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *id)
 {
 	struct wm8904_priv *wm8904;
-	int ret;
+	unsigned int val;
+	int ret, i;
 
 	wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv),
 			      GFP_KERNEL);
@@ -2249,14 +2192,61 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
 	i2c_set_clientdata(i2c, wm8904);
 	wm8904->pdata = i2c->dev.platform_data;
 
+	for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
+		wm8904->supplies[i].supply = wm8904_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies),
+				      wm8904->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
+				    wm8904->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
+		goto err_enable;
+	}
+	if (val != 0x8904) {
+		dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val);
+		ret = -EINVAL;
+		goto err_enable;
+	}
+
+	ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to read device revision: %d\n",
+			ret);
+		goto err_enable;
+	}
+	dev_info(&i2c->dev, "revision %c\n", val + 'A');
+
+	ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+		goto err_enable;
+	}
+
+	/* Can leave the device powered off until we need it */
+	regcache_cache_only(wm8904->regmap, true);
+	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8904, &wm8904_dai, 1);
 	if (ret != 0)
-		goto err;
+		return ret;
 
 	return 0;
 
-err:
+err_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 	return ret;
 }
 
-- 
cgit v1.2.3


From 725e7a7b58fb27d8f97a1c4eae47cb5d37564725 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sat, 9 Jun 2012 11:57:37 +0800
Subject: ASoC: wm8904: Move register default setup into I2C probe()

Get it done as early as possible, it's neater and minimises the time the
pins aren't configured as requested.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8904.c | 111 +++++++++++++++++++++++-----------------------
 1 file changed, 55 insertions(+), 56 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index b178232c990c..0013afe48e66 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2054,8 +2054,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec)
 static int wm8904_probe(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
-	struct wm8904_pdata *pdata = wm8904->pdata;
-	int ret, i;
+	int ret;
 
 	codec->control_data = wm8904->regmap;
 
@@ -2077,60 +2076,6 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	/* Change some default settings - latch VU and enable ZC */
-	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
-			    WM8904_ADC_VU, WM8904_ADC_VU);
-	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
-			    WM8904_ADC_VU, WM8904_ADC_VU);
-	snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT,
-			    WM8904_DAC_VU, WM8904_DAC_VU);
-	snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
-			    WM8904_DAC_VU, WM8904_DAC_VU);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT,
-			    WM8904_HPOUT_VU | WM8904_HPOUTLZC,
-			    WM8904_HPOUT_VU | WM8904_HPOUTLZC);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT,
-			    WM8904_HPOUT_VU | WM8904_HPOUTRZC,
-			    WM8904_HPOUT_VU | WM8904_HPOUTRZC);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
-	snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0,
-			    WM8904_SR_MODE, 0);
-
-	/* Apply configuration from the platform data. */
-	if (wm8904->pdata) {
-		for (i = 0; i < WM8904_GPIO_REGS; i++) {
-			if (!pdata->gpio_cfg[i])
-				continue;
-
-			regmap_update_bits(wm8904->regmap,
-					   WM8904_GPIO_CONTROL_1 + i,
-					   0xffff,
-					   pdata->gpio_cfg[i]);
-		}
-
-		/* Zero is the default value for these anyway */
-		for (i = 0; i < WM8904_MIC_REGS; i++)
-			regmap_update_bits(wm8904->regmap,
-					   WM8904_MIC_BIAS_CONTROL_0 + i,
-					   0xffff,
-					   pdata->mic_cfg[i]);
-	}
-
-	/* Set Class W by default - this will be managed by the Class
-	 * G widget at runtime where bypass paths are available.
-	 */
-	snd_soc_update_bits(codec, WM8904_CLASS_W_0,
-			    WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
-
-	/* Use normal bias source */
-	snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
-			    WM8904_POBCTRL, 0);
-
 	wm8904_handle_pdata(codec);
 
 	wm8904_add_widgets(codec);
@@ -2234,6 +2179,60 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
 		goto err_enable;
 	}
 
+	/* Change some default settings - latch VU and enable ZC */
+	regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT,
+			   WM8904_ADC_VU, WM8904_ADC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
+			   WM8904_ADC_VU, WM8904_ADC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT,
+			   WM8904_DAC_VU, WM8904_DAC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
+			   WM8904_DAC_VU, WM8904_DAC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT,
+			   WM8904_HPOUT_VU | WM8904_HPOUTLZC,
+			   WM8904_HPOUT_VU | WM8904_HPOUTLZC);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT,
+			   WM8904_HPOUT_VU | WM8904_HPOUTRZC,
+			   WM8904_HPOUT_VU | WM8904_HPOUTRZC);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
+	regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0,
+			   WM8904_SR_MODE, 0);
+
+	/* Apply configuration from the platform data. */
+	if (wm8904->pdata) {
+		for (i = 0; i < WM8904_GPIO_REGS; i++) {
+			if (!wm8904->pdata->gpio_cfg[i])
+				continue;
+
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_GPIO_CONTROL_1 + i,
+					   0xffff,
+					   wm8904->pdata->gpio_cfg[i]);
+		}
+
+		/* Zero is the default value for these anyway */
+		for (i = 0; i < WM8904_MIC_REGS; i++)
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_MIC_BIAS_CONTROL_0 + i,
+					   0xffff,
+					   wm8904->pdata->mic_cfg[i]);
+	}
+
+	/* Set Class W by default - this will be managed by the Class
+	 * G widget at runtime where bypass paths are available.
+	 */
+	regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0,
+			    WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
+
+	/* Use normal bias source */
+	regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0,
+			    WM8904_POBCTRL, 0);
+
 	/* Can leave the device powered off until we need it */
 	regcache_cache_only(wm8904->regmap, true);
 	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-- 
cgit v1.2.3


From 625c4888fff33c300779ed38963e1ee7ad878a12 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 11 Jun 2012 18:42:06 +0800
Subject: ASoC: wm8996: Move regulator notifier callbacks into I2C level

Now that we're using regmap the cache is available here.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8996.c | 41 ++++++++++++++++++++---------------------
 1 file changed, 20 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 49e0e8d6663e..e0cf5b0b5203 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2644,21 +2644,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 		goto err;
 	}
 
-	wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
-	wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
-	wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
-
-	/* This should really be moved into the regulator core */
-	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
-		ret = regulator_register_notifier(wm8996->supplies[i].consumer,
-						  &wm8996->disable_nb[i]);
-		if (ret != 0) {
-			dev_err(codec->dev,
-				"Failed to register regulator notifier: %d\n",
-				ret);
-		}
-	}
-
 	/* Apply platform data settings */
 	snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
 			    WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
@@ -2858,9 +2843,7 @@ err:
 
 static int wm8996_remove(struct snd_soc_codec *codec)
 {
-	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
-	int i;
 
 	snd_soc_update_bits(codec, WM8996_INTERRUPT_CONTROL,
 			    WM8996_IM_IRQ, WM8996_IM_IRQ);
@@ -2868,10 +2851,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
 	if (i2c->irq)
 		free_irq(i2c->irq, codec);
 
-	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
-		regulator_unregister_notifier(wm8996->supplies[i].consumer,
-					      &wm8996->disable_nb[i]);
-
 	return 0;
 }
 
@@ -2985,6 +2964,21 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 		goto err_gpio;
 	}
 
+	wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
+	wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
+	wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
+
+	/* This should really be moved into the regulator core */
+	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
+		ret = regulator_register_notifier(wm8996->supplies[i].consumer,
+						  &wm8996->disable_nb[i]);
+		if (ret != 0) {
+			dev_err(&i2c->dev,
+				"Failed to register regulator notifier: %d\n",
+				ret);
+		}
+	}
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies),
 				    wm8996->supplies);
 	if (ret != 0) {
@@ -3062,6 +3056,7 @@ err:
 static __devexit int wm8996_i2c_remove(struct i2c_client *client)
 {
 	struct wm8996_priv *wm8996 = i2c_get_clientdata(client);
+	int i;
 
 	snd_soc_unregister_codec(&client->dev);
 	wm8996_free_gpio(wm8996);
@@ -3069,6 +3064,10 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client)
 		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
 		gpio_free(wm8996->pdata.ldo_ena);
 	}
+	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
+		regulator_unregister_notifier(wm8996->supplies[i].consumer,
+					      &wm8996->disable_nb[i]);
+
 	return 0;
 }
 
-- 
cgit v1.2.3


From d4b3d0fbb7617a65cb919ac86110b0790ae568c5 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 11 Jun 2012 18:41:16 +0800
Subject: ASoC: wm8996: Inline wm8996_reset() and optimise cache-only usage

There is only one caller and this allows us to cleanly leave the CODEC
with the internal LDO powered down which is the default state we're
looking for and means that we can robustly disable the register cache
only when we either disable the LDO or power down the external
regulators.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8996.c | 31 +++++++++++++------------------
 1 file changed, 13 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index e0cf5b0b5203..1579880ac05d 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1528,18 +1528,6 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg)
 	}
 }
 
-static int wm8996_reset(struct wm8996_priv *wm8996)
-{
-	if (wm8996->pdata.ldo_ena > 0) {
-		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
-		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1);
-		return 0;
-	} else {
-		return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET,
-				    0x8915);
-	}
-}
-
 static const int bclk_divs[] = {
 	1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
 };
@@ -1631,8 +1619,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_OFF:
 		regcache_cache_only(codec->control_data, true);
-		if (wm8996->pdata.ldo_ena >= 0)
+		if (wm8996->pdata.ldo_ena >= 0) {
 			gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
+			regcache_cache_only(codec->control_data, true);
+		}
 		regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies),
 				       wm8996->supplies);
 		break;
@@ -3019,13 +3009,18 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 	dev_info(&i2c->dev, "revision %c\n",
 		 (reg & WM8996_CHIP_REV_MASK) + 'A');
 
-	ret = wm8996_reset(wm8996);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "Failed to issue reset\n");
-		goto err_regmap;
+	if (wm8996->pdata.ldo_ena > 0) {
+		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
+		regcache_cache_only(wm8996->regmap, true);
+	} else {
+		ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET,
+				   0x8915);
+		if (ret != 0) {
+			dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+			goto err_regmap;
+		}
 	}
 
-	regcache_cache_only(wm8996->regmap, true);
 	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
 
 	wm8996_init_gpio(wm8996);
-- 
cgit v1.2.3


From ec8ffe1868f45a72eb243f1c9797806be58276fd Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 11 Jun 2012 19:10:50 +0800
Subject: ASoC: wm8996: Move register default configuration to I2C probe

This gets the registers set up as early as possible, mainly useful for the
GPIOs to ensure that they're in the correct mode.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8996.c | 331 +++++++++++++++++++++++++---------------------
 1 file changed, 180 insertions(+), 151 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 1579880ac05d..00f183dfa454 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2619,7 +2619,7 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 	int ret;
 	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
-	int i, irq_flags;
+	int irq_flags;
 
 	wm8996->codec = codec;
 
@@ -2634,162 +2634,12 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 		goto err;
 	}
 
-	/* Apply platform data settings */
-	snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
-			    WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
-			    wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT |
-			    wm8996->pdata.inr_mode);
-
-	for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) {
-		if (!wm8996->pdata.gpio_default[i])
-			continue;
-
-		snd_soc_write(codec, WM8996_GPIO_1 + i,
-			      wm8996->pdata.gpio_default[i] & 0xffff);
-	}
-
-	if (wm8996->pdata.spkmute_seq)
-		snd_soc_update_bits(codec, WM8996_PDM_SPEAKER_MUTE_SEQUENCE,
-				    WM8996_SPK_MUTE_ENDIAN |
-				    WM8996_SPK_MUTE_SEQ1_MASK,
-				    wm8996->pdata.spkmute_seq);
-
-	snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_2,
-			    WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC |
-			    WM8996_MICD_SRC, wm8996->pdata.micdet_def);
-
-	/* Latch volume update bits */
-	snd_soc_update_bits(codec, WM8996_LEFT_LINE_INPUT_VOLUME,
-			    WM8996_IN1_VU, WM8996_IN1_VU);
-	snd_soc_update_bits(codec, WM8996_RIGHT_LINE_INPUT_VOLUME,
-			    WM8996_IN1_VU, WM8996_IN1_VU);
-
-	snd_soc_update_bits(codec, WM8996_DAC1_LEFT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_DAC1_RIGHT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_DAC2_LEFT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-	snd_soc_update_bits(codec, WM8996_DAC2_RIGHT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-
-	snd_soc_update_bits(codec, WM8996_OUTPUT1_LEFT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_OUTPUT1_RIGHT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_OUTPUT2_LEFT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-	snd_soc_update_bits(codec, WM8996_OUTPUT2_RIGHT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-
-	snd_soc_update_bits(codec, WM8996_DSP1_TX_LEFT_VOLUME,
-			    WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP1_TX_RIGHT_VOLUME,
-			    WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_TX_LEFT_VOLUME,
-			    WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_TX_RIGHT_VOLUME,
-			    WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
-
-	snd_soc_update_bits(codec, WM8996_DSP1_RX_LEFT_VOLUME,
-			    WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP1_RX_RIGHT_VOLUME,
-			    WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_RX_LEFT_VOLUME,
-			    WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_RX_RIGHT_VOLUME,
-			    WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
-
-	/* No support currently for the underclocked TDM modes and
-	 * pick a default TDM layout with each channel pair working with
-	 * slots 0 and 1. */
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN1_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_2_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN2_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN2_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_3_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN3_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_4_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN4_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_5_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN5_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1);
-
-	snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF2RX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF2RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF2RX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF2RX_CHAN1_START_SLOT_MASK,
-			    1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1);
-
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_2_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN2_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_3_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN3_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_4_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN4_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_5_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN5_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1);
-
-	snd_soc_update_bits(codec, WM8996_AIF2TX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF2TX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF2TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF2TX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF2TX_CHAN1_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
-
 	if (wm8996->pdata.num_retune_mobile_cfgs)
 		wm8996_retune_mobile_pdata(codec);
 	else
 		snd_soc_add_codec_controls(codec, wm8996_eq_controls,
 				     ARRAY_SIZE(wm8996_eq_controls));
 
-	/* If the TX LRCLK pins are not in LRCLK mode configure the
-	 * AIFs to source their clocks from the RX LRCLKs.
-	 */
-	if ((snd_soc_read(codec, WM8996_GPIO_1)))
-		snd_soc_update_bits(codec, WM8996_AIF1_TX_LRCLK_2,
-				    WM8996_AIF1TX_LRCLK_MODE,
-				    WM8996_AIF1TX_LRCLK_MODE);
-
-	if ((snd_soc_read(codec, WM8996_GPIO_2)))
-		snd_soc_update_bits(codec, WM8996_AIF2_TX_LRCLK_2,
-				    WM8996_AIF2TX_LRCLK_MODE,
-				    WM8996_AIF2TX_LRCLK_MODE);
-
 	if (i2c->irq) {
 		if (wm8996->pdata.irq_flags)
 			irq_flags = wm8996->pdata.irq_flags;
@@ -3023,6 +2873,185 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 
 	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
 
+	/* Apply platform data settings */
+	regmap_update_bits(wm8996->regmap, WM8996_LINE_INPUT_CONTROL,
+			   WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
+			   wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT |
+			   wm8996->pdata.inr_mode);
+
+	for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) {
+		if (!wm8996->pdata.gpio_default[i])
+			continue;
+
+		regmap_write(wm8996->regmap, WM8996_GPIO_1 + i,
+			     wm8996->pdata.gpio_default[i] & 0xffff);
+	}
+
+	if (wm8996->pdata.spkmute_seq)
+		regmap_update_bits(wm8996->regmap,
+				   WM8996_PDM_SPEAKER_MUTE_SEQUENCE,
+				   WM8996_SPK_MUTE_ENDIAN |
+				   WM8996_SPK_MUTE_SEQ1_MASK,
+				   wm8996->pdata.spkmute_seq);
+
+	regmap_update_bits(wm8996->regmap, WM8996_ACCESSORY_DETECT_MODE_2,
+			   WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC |
+			   WM8996_MICD_SRC, wm8996->pdata.micdet_def);
+
+	/* Latch volume update bits */
+	regmap_update_bits(wm8996->regmap, WM8996_LEFT_LINE_INPUT_VOLUME,
+			   WM8996_IN1_VU, WM8996_IN1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_RIGHT_LINE_INPUT_VOLUME,
+			   WM8996_IN1_VU, WM8996_IN1_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_DAC1_LEFT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DAC1_RIGHT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DAC2_LEFT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DAC2_RIGHT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_LEFT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_RIGHT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_LEFT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_RIGHT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_LEFT_VOLUME,
+			   WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_RIGHT_VOLUME,
+			   WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_LEFT_VOLUME,
+			   WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_RIGHT_VOLUME,
+			   WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_LEFT_VOLUME,
+			   WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_RIGHT_VOLUME,
+			   WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_LEFT_VOLUME,
+			   WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_RIGHT_VOLUME,
+			   WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
+
+	/* No support currently for the underclocked TDM modes and
+	 * pick a default TDM layout with each channel pair working with
+	 * slots 0 and 1. */
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN1_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_2_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN2_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN2_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_3_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN3_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_4_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN4_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_5_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN5_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1);
+
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF2RX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF2RX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF2RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF2RX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF2RX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF2RX_CHAN1_START_SLOT_MASK,
+			   1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1);
+
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_2_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN2_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_3_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN3_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_4_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN4_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_5_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN5_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1);
+
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF2TX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF2TX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF2TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF2TX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF2TX_CHAN1_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
+
+	/* If the TX LRCLK pins are not in LRCLK mode configure the
+	 * AIFs to source their clocks from the RX LRCLKs.
+	 */
+	ret = regmap_read(wm8996->regmap, WM8996_GPIO_1, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to read GPIO1: %d\n", ret);
+		goto err_regmap;
+	}
+
+	if (reg & WM8996_GP1_FN_MASK)
+		regmap_update_bits(wm8996->regmap, WM8996_AIF1_TX_LRCLK_2,
+				   WM8996_AIF1TX_LRCLK_MODE,
+				   WM8996_AIF1TX_LRCLK_MODE);
+
+	ret = regmap_read(wm8996->regmap, WM8996_GPIO_2, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to read GPIO2: %d\n", ret);
+		goto err_regmap;
+	}
+
+	if (reg & WM8996_GP2_FN_MASK)
+		regmap_update_bits(wm8996->regmap, WM8996_AIF2_TX_LRCLK_2,
+				   WM8996_AIF2TX_LRCLK_MODE,
+				   WM8996_AIF2TX_LRCLK_MODE);
+
 	wm8996_init_gpio(wm8996);
 
 	ret = snd_soc_register_codec(&i2c->dev,
-- 
cgit v1.2.3


From d69d65226a7910d1cfd9f3841180a0f250eeb2e9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 25 Jun 2012 10:01:27 +0100
Subject: ASoC: dwc: Bodge around continuing absence of clock API stubs

The patches for stubbing out the generic clock API still haven't been
applied so we need to either add ifdefs here or add a dependency until
someone decides to actually apply them.

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/dwc/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig
index 93e9fc33560c..e334900cf0b8 100644
--- a/sound/soc/dwc/Kconfig
+++ b/sound/soc/dwc/Kconfig
@@ -1,5 +1,6 @@
 config SND_DESIGNWARE_I2S
 	tristate "Synopsys I2S Device Driver"
+	depends on CLKDEV_LOOKUP
 	help
 	 Say Y or M if you want to add support for I2S driver for
 	 Synopsys desigwnware I2S device. The device supports upto
-- 
cgit v1.2.3


From e6656369da73f8a4206a72ea6fb0e35247f42364 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 25 Jun 2012 11:03:12 +0200
Subject: ALSA: hda - Remove suprefluous EAPD init verbs for ALC660vd

The EAPD on nodes 0x14 and 0x15 are initialized in alc_auto_setup_eapd().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 -----------
 1 file changed, 11 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3e698e239dd8..4377a9539735 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6494,12 +6494,6 @@ static const struct snd_pci_quirk alc861vd_fixup_tbl[] = {
 	{}
 };
 
-static const struct hda_verb alc660vd_eapd_verbs[] = {
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
 /*
  */
 static int patch_alc861vd(struct hda_codec *codec)
@@ -6521,11 +6515,6 @@ static int patch_alc861vd(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	if (codec->vendor_id == 0x10ec0660) {
-		/* always turn on EAPD */
-		snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs);
-	}
-
 	if (!spec->no_analog) {
 		err = snd_hda_attach_beep_device(codec, 0x23);
 		if (err < 0)
-- 
cgit v1.2.3


From da185443c12f5ef7416af50293833a5654854186 Mon Sep 17 00:00:00 2001
From: Daniel Mack <zonque@gmail.com>
Date: Mon, 18 Jun 2012 21:16:31 +0200
Subject: ALSA: snd-usb-caiaq: initialize card pointer
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fixes the following warning:

  CC [M]  sound/usb/caiaq/device.o
sound/usb/caiaq/device.c: In function ‘snd_probe’:
sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used
uninitialized in this function [-Wmaybe-uninitialized]

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/caiaq/device.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 64aed432ae22..7da0d0aa72cb 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf,
 		     const struct usb_device_id *id)
 {
 	int ret;
-	struct snd_card *card;
+	struct snd_card *card = NULL;
 	struct usb_device *device = interface_to_usbdev(intf);
 
 	ret = create_card(device, intf, &card);
-- 
cgit v1.2.3


From 3dba1c265268950b1ddd22e53ea823c1cb57b684 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 27 Jun 2012 14:59:20 +0100
Subject: ASoC: wm5102: Remove unused platform data header

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5102.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 9b9ea7fd0d7d..e76c41e1f847 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -25,7 +25,6 @@
 #include <sound/jack.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
-#include <sound/wm5102.h>
 
 #include <linux/mfd/arizona/core.h>
 #include <linux/mfd/arizona/registers.h>
-- 
cgit v1.2.3


From 00227f15a0ad8401d2b0b67905da63e75b544895 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 27 Jun 2012 18:45:44 +0200
Subject: ALSA: HDA: Support single 3-pin jack without VREF on the actual pin

Some ASUS device has a single 3-pin jack that can either be a mic or
a headphone, but the pin does not have VREF capabilities. We've been
told by Realtek to instead enable VREF on pin 0x18 in that case.

BugLink: https://bugs.launchpad.net/bugs/1018262
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++++-------------
 1 file changed, 34 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5c81ee95d7f0..40dda2a83774 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -303,6 +303,38 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx)
 static void call_update_outputs(struct hda_codec *codec);
 static void alc_inv_dmic_sync(struct hda_codec *codec, bool force);
 
+/* for shared I/O, change the pin-control accordingly */
+static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int val;
+	hda_nid_t pin = spec->autocfg.inputs[1].pin;
+	/* NOTE: this assumes that there are only two inputs, the
+	 * first is the real internal mic and the second is HP/mic jack.
+	 */
+
+	val = snd_hda_get_default_vref(codec, pin);
+
+	/* This pin does not have vref caps - let's enable vref on pin 0x18
+	   instead, as suggested by Realtek */
+	if (val == AC_PINCTL_VREF_HIZ) {
+		const hda_nid_t vref_pin = 0x18;
+		/* Sanity check pin 0x18 */
+		if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN &&
+		    get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) {
+			unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin);
+			if (vref_val != AC_PINCTL_VREF_HIZ)
+				snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0));
+		}
+	}
+
+	val = set_as_mic ? val | PIN_IN : PIN_HP;
+	snd_hda_set_pin_ctl(codec, pin, val);
+
+	spec->automute_speaker = !set_as_mic;
+	call_update_outputs(codec);
+}
+
 /* select the given imux item; either unmute exclusively or select the route */
 static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 			  unsigned int idx, bool force)
@@ -329,21 +361,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
 		return 0;
 	spec->cur_mux[adc_idx] = idx;
 
-	/* for shared I/O, change the pin-control accordingly */
-	if (spec->shared_mic_hp) {
-		unsigned int val;
-		hda_nid_t pin = spec->autocfg.inputs[1].pin;
-		/* NOTE: this assumes that there are only two inputs, the
-		 * first is the real internal mic and the second is HP jack.
-		 */
-		if (spec->cur_mux[adc_idx])
-			val = snd_hda_get_default_vref(codec, pin) | PIN_IN;
-		else
-			val = PIN_HP;
-		snd_hda_set_pin_ctl(codec, pin, val);
-		spec->automute_speaker = !spec->cur_mux[adc_idx];
-		call_update_outputs(codec);
-	}
+	if (spec->shared_mic_hp)
+		update_shared_mic_hp(codec, spec->cur_mux[adc_idx]);
 
 	if (spec->dyn_adc_switch) {
 		alc_dyn_adc_pcm_resetup(codec, idx);
-- 
cgit v1.2.3


From 5780b627e24113323427c102175296ae43dfb9d7 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 27 Jun 2012 18:45:45 +0200
Subject: ALSA: hda - give 3-pin jack the name "Headphone Mic Jack"

This 3-pin jack was labeled "Headphone Jack", but it could also be
used as a mic jack just by switching "Input Source". Therefore we need
to call the jack something else, to make sure PulseAudio can use the
speaker together with the external mic. (PulseAudio might mute the
speaker if it detects a headphone being plugged in.)

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++--
 1 file changed, 20 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 40dda2a83774..f912d74438a6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2001,13 +2001,31 @@ static int __alc_build_controls(struct hda_codec *codec)
 	return 0;
 }
 
-static int alc_build_controls(struct hda_codec *codec)
+static int alc_build_jacks(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+
+	if (spec->shared_mic_hp) {
+		int err;
+		int nid = spec->autocfg.inputs[1].pin;
+		err = snd_hda_jack_add_kctl(codec, nid, "Headphone Mic", 0);
+		if (err < 0)
+			return err;
+		err = snd_hda_jack_detect_enable(codec, nid, 0);
+		if (err < 0)
+			return err;
+	}
+
+	return snd_hda_jack_add_kctls(codec, &spec->autocfg);
+}
+
+static int alc_build_controls(struct hda_codec *codec)
+{
 	int err = __alc_build_controls(codec);
 	if (err < 0)
 		return err;
-	err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
+
+	err = alc_build_jacks(codec);
 	if (err < 0)
 		return err;
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD);
-- 
cgit v1.2.3


From 6b4a21b64ccc218a00dc0e38676092e64df159dc Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 28 Jun 2012 13:11:47 +0100
Subject: ASoC: dwc: Add missing __iomem annotations

Otherwise sparse gets very upset with us.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/dwc/designware_i2s.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index e667e2b45e67..1bd042b15aef 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -71,12 +71,12 @@ struct dw_i2s_dev {
 	int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
 };
 
-static inline void i2s_write_reg(void *io_base, int reg, u32 val)
+static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val)
 {
 	writel(val, io_base + reg);
 }
 
-static inline u32 i2s_read_reg(void *io_base, int reg)
+static inline u32 i2s_read_reg(void __iomem *io_base, int reg)
 {
 	return readl(io_base + reg);
 }
-- 
cgit v1.2.3


From cdf605255c2b592d7dbc1de19688feae941b5567 Mon Sep 17 00:00:00 2001
From: Vipin Kumar <vipin.kumar@st.com>
Date: Thu, 28 Jun 2012 12:31:37 +0530
Subject: ASoC: spdif_receiver: Add support for spdif in Audio Codec

This patch adds the support for spdif in audio codec.

Signed-off-by: vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Makefile         |  2 +-
 sound/soc/codecs/spdif_receiver.c | 67 +++++++++++++++++++++++++++++++++++++++
 2 files changed, 68 insertions(+), 1 deletion(-)
 create mode 100644 sound/soc/codecs/spdif_receiver.c

(limited to 'sound')

diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index d35ba7f06fcf..62c3d4dd1872 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -45,7 +45,7 @@ snd-soc-alc5623-objs := alc5623.o
 snd-soc-alc5632-objs := alc5632.o
 snd-soc-sigmadsp-objs := sigmadsp.o
 snd-soc-sn95031-objs := sn95031.o
-snd-soc-spdif-objs := spdif_transciever.o
+snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-sta32x-objs := sta32x.o
 snd-soc-stac9766-objs := stac9766.o
diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c
new file mode 100644
index 000000000000..dd8d856053fc
--- /dev/null
+++ b/sound/soc/codecs/spdif_receiver.c
@@ -0,0 +1,67 @@
+/*
+ * ALSA SoC SPDIF DIR (Digital Interface Reciever) driver
+ *
+ * Based on ALSA SoC SPDIF DIT driver
+ *
+ *  This driver is used by controllers which can operate in DIR (SPDI/F) where
+ *  no codec is needed.  This file provides stub codec that can be used
+ *  in these configurations. SPEAr SPDIF IN Audio controller uses this driver.
+ *
+ * Author:      Vipin Kumar,  <vipin.kumar@st.com>
+ * Copyright:   (C) 2012  ST Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_192000
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static struct snd_soc_codec_driver soc_codec_spdif_dir;
+
+static struct snd_soc_dai_driver dir_stub_dai = {
+	.name		= "dir-hifi",
+	.capture	= {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+};
+
+static int spdif_dir_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dir,
+			&dir_stub_dai, 1);
+}
+
+static int spdif_dir_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver spdif_dir_driver = {
+	.probe		= spdif_dir_probe,
+	.remove		= spdif_dir_remove,
+	.driver		= {
+		.name	= "spdif-dir",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver(spdif_dir_driver);
+
+MODULE_DESCRIPTION("ASoC SPDIF DIR driver");
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From adf643aba8ed620f8c8e2533f4ace3a90e5daecf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 29 Jun 2012 02:34:46 +0100
Subject: ASoC: spdif: Build separate RX and TX objects

Otherwise we fail to link when building as modules due to multiple
init/exit functions.

Reported-by: Fengguang Wu <wfg@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Makefile | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 62c3d4dd1872..acf80888790c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -45,7 +45,8 @@ snd-soc-alc5623-objs := alc5623.o
 snd-soc-alc5632-objs := alc5632.o
 snd-soc-sigmadsp-objs := sigmadsp.o
 snd-soc-sn95031-objs := sn95031.o
-snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o
+snd-soc-spdif-tx-objs := spdif_transciever.o
+snd-soc-spdif-rx-objs := spdif_receiver.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-sta32x-objs := sta32x.o
 snd-soc-stac9766-objs := stac9766.o
@@ -159,7 +160,7 @@ obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SIGMADSP)	+= snd-soc-sigmadsp.o
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
-obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
+obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif-rx.o snd-soc-spdif-tx.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
-- 
cgit v1.2.3


From 8a720718b37d00cf8ab311902705ae7c7890bb95 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Beno=C3=AEt=20Th=C3=A9baudeau?=
 <benoit.thebaudeau@advansee.com>
Date: Mon, 18 Jun 2012 22:41:28 +0200
Subject: ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

snd_soc_dapm_put_volsw() sets connect incorrectly in the case max > 1 with
invert. In that case, the raw disconnect value should be max, which corresponds
to the userspace value 0.

This use case currently does not appear upstream, but it could break
SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future.

Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 8 +-------
 1 file changed, 1 insertion(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c2206bc835da..967066873aad 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2515,19 +2515,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	int wi;
 
 	val = (ucontrol->value.integer.value[0] & mask);
+	connect = !!val;
 
 	if (invert)
 		val = max - val;
 	mask = mask << shift;
 	val = val << shift;
 
-	if (val)
-		/* new connection */
-		connect = invert ? 0 : 1;
-	else
-		/* old connection must be powered down */
-		connect = invert ? 1 : 0;
-
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	change = snd_soc_test_bits(widget->codec, reg, mask, val);
-- 
cgit v1.2.3


From 890255e704826a20caec54dcec1926316baf4263 Mon Sep 17 00:00:00 2001
From: Robert Jarzmik <robert.jarzmik@free.fr>
Date: Sat, 30 Jun 2012 19:25:08 +0200
Subject: ASoC: mioa701: convert to snd_soc_register_card API

The mioa701 board code is converted to the snd_soc_register_card()
and snd_soc_unregister_card() APIs.

Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/mioa701_wm9713.c | 33 ++++++++++++---------------------
 1 file changed, 12 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 9c585af59b5f..8687c1c65d29 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = {
 	.num_links = ARRAY_SIZE(mioa701_dai),
 };
 
-static struct platform_device *mioa701_snd_device;
-
-static int mioa701_wm9713_probe(struct platform_device *pdev)
+static int __devinit mioa701_wm9713_probe(struct platform_device *pdev)
 {
-	int ret;
+	int rc;
 
 	if (!machine_is_mioa701())
 		return -ENODEV;
 
-	dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
-		 "lead to overheating and possible destruction of your device."
-		 "Do not use without a good knowledge of mio's board design!\n");
-
-	mioa701_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!mioa701_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(mioa701_snd_device, &mioa701);
-
-	ret = platform_device_add(mioa701_snd_device);
-	if (!ret)
-		return 0;
-
-	platform_device_put(mioa701_snd_device);
-	return ret;
+	mioa701.dev = &pdev->dev;
+	rc =  snd_soc_register_card(&mioa701);
+	if (!rc)
+		dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+			 "lead to overheating and possible destruction of your device."
+			 " Do not use without a good knowledge of mio's board design!\n");
+	return rc;
 }
 
 static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
 {
-	platform_device_unregister(mioa701_snd_device);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 8bf01d8abc55eaf8e19a2d48911c8e49ee6f5bab Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Jul 2012 10:50:24 +0200
Subject: ALSA: Add missing .owner=THIS_MODULE to platform_driver definitions

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/atmel/abdac.c           | 1 +
 sound/atmel/ac97c.c           | 1 +
 sound/drivers/aloop.c         | 3 ++-
 sound/drivers/dummy.c         | 3 ++-
 sound/drivers/mpu401/mpu401.c | 3 ++-
 sound/drivers/mtpav.c         | 3 ++-
 sound/drivers/mts64.c         | 3 ++-
 sound/drivers/portman2x4.c    | 3 ++-
 sound/drivers/serial-u16550.c | 3 ++-
 sound/drivers/virmidi.c       | 3 ++-
 sound/ppc/powermac.c          | 3 ++-
 sound/sh/aica.c               | 4 +++-
 sound/sh/sh_dac_audio.c       | 1 +
 13 files changed, 24 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index f7c2bb08055d..2e866398bffe 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -589,6 +589,7 @@ static struct platform_driver atmel_abdac_driver = {
 	.remove		= __devexit_p(atmel_abdac_remove),
 	.driver		= {
 		.name	= "atmel_abdac",
+		.owner	= THIS_MODULE,
 	},
 	.suspend	= atmel_abdac_suspend,
 	.resume		= atmel_abdac_resume,
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index f5ded640b395..3d0ea82ff068 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1210,6 +1210,7 @@ static struct platform_driver atmel_ac97c_driver = {
 	.remove		= __devexit_p(atmel_ac97c_remove),
 	.driver		= {
 		.name	= "atmel_ac97c",
+		.owner	= THIS_MODULE,
 	},
 	.suspend	= atmel_ac97c_suspend,
 	.resume		= atmel_ac97c_resume,
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 8b5c36f4d303..3484411bd5e6 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1209,7 +1209,8 @@ static struct platform_driver loopback_driver = {
 	.resume		= loopback_resume,
 #endif
 	.driver		= {
-		.name	= SND_LOOPBACK_DRIVER
+		.name	= SND_LOOPBACK_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index ad9434fd6370..bc79c441a8f2 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1094,7 +1094,8 @@ static struct platform_driver snd_dummy_driver = {
 	.resume		= snd_dummy_resume,
 #endif
 	.driver		= {
-		.name	= SND_DUMMY_DRIVER
+		.name	= SND_DUMMY_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 86f5fbc2da72..bc03a2046c9c 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -139,7 +139,8 @@ static struct platform_driver snd_mpu401_driver = {
 	.probe		= snd_mpu401_probe,
 	.remove		= __devexit_p(snd_mpu401_remove),
 	.driver		= {
-		.name	= SND_MPU401_DRIVER
+		.name	= SND_MPU401_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 76930793fb69..cad73af3860c 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -759,7 +759,8 @@ static struct platform_driver snd_mtpav_driver = {
 	.probe		= snd_mtpav_probe,
 	.remove		= __devexit_p(snd_mtpav_remove),
 	.driver		= {
-		.name	= SND_MTPAV_DRIVER
+		.name	= SND_MTPAV_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
index 621e60e2029f..2d5514b0a290 100644
--- a/sound/drivers/mts64.c
+++ b/sound/drivers/mts64.c
@@ -1040,7 +1040,8 @@ static struct platform_driver snd_mts64_driver = {
 	.probe  = snd_mts64_probe,
 	.remove = __devexit_p(snd_mts64_remove),
 	.driver = {
-		.name = PLATFORM_DRIVER
+		.name = PLATFORM_DRIVER,
+		.owner	= THIS_MODULE,
 	}
 };
 
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index 3e32bd3d95d9..8364855ed14f 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -829,7 +829,8 @@ static struct platform_driver snd_portman_driver = {
 	.probe  = snd_portman_probe,
 	.remove = __devexit_p(snd_portman_remove),
 	.driver = {
-		.name = PLATFORM_DRIVER
+		.name = PLATFORM_DRIVER,
+		.owner	= THIS_MODULE,
 	}
 };
 
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index b2d0e8e49bed..86700671d1ac 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -995,7 +995,8 @@ static struct platform_driver snd_serial_driver = {
 	.probe		= snd_serial_probe,
 	.remove		= __devexit_p( snd_serial_remove),
 	.driver		= {
-		.name	= SND_SERIAL_DRIVER
+		.name	= SND_SERIAL_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c
index 9d97478a18b3..d7d514df9058 100644
--- a/sound/drivers/virmidi.c
+++ b/sound/drivers/virmidi.c
@@ -142,7 +142,8 @@ static struct platform_driver snd_virmidi_driver = {
 	.probe		= snd_virmidi_probe,
 	.remove		= __devexit_p(snd_virmidi_remove),
 	.driver		= {
-		.name	= SND_VIRMIDI_DRIVER
+		.name	= SND_VIRMIDI_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index 5a4e263b5b0f..aef54beaf8b7 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -169,7 +169,8 @@ static struct platform_driver snd_pmac_driver = {
 	.resume		= snd_pmac_driver_resume,
 #endif
 	.driver		= {
-		.name	= SND_PMAC_DRIVER
+		.name	= SND_PMAC_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 391a38ca58bc..d48b523207eb 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -654,7 +654,9 @@ static struct platform_driver snd_aica_driver = {
 	.probe = snd_aica_probe,
 	.remove = __devexit_p(snd_aica_remove),
 	.driver = {
-		   .name = SND_AICA_DRIVER},
+		.name = SND_AICA_DRIVER,
+		.owner	= THIS_MODULE,
+	},
 };
 
 static int __init aica_init(void)
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index f8b01c77b298..0a3394751ed2 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -438,6 +438,7 @@ static struct platform_driver sh_dac_driver = {
 	.remove = snd_sh_dac_remove,
 	.driver = {
 		.name = "dac_audio",
+		.owner	= THIS_MODULE,
 	},
 };
 
-- 
cgit v1.2.3


From 284e7ca75f96a18f182cce38ba76ee724fb97e16 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Jul 2012 11:22:40 +0200
Subject: ALSA: convert PM ops of platform_driver to new pm ops

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/arm/pxa2xx-ac97.c   |  9 +++------
 sound/atmel/abdac.c       | 17 +++++++++--------
 sound/atmel/ac97c.c       | 17 +++++++++--------
 sound/drivers/aloop.c     | 19 ++++++++++---------
 sound/drivers/dummy.c     | 18 ++++++++++--------
 sound/drivers/pcsp/pcsp.c | 11 +++++++----
 sound/ppc/powermac.c      | 18 ++++++++++--------
 7 files changed, 58 insertions(+), 51 deletions(-)

(limited to 'sound')

diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index afef72c4f0d3..0d7b25e81643 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -108,7 +108,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
 
 #ifdef CONFIG_PM
 
-static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state)
+static int pxa2xx_ac97_do_suspend(struct snd_card *card)
 {
 	pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data;
 
@@ -144,7 +144,7 @@ static int pxa2xx_ac97_suspend(struct device *dev)
 	int ret = 0;
 
 	if (card)
-		ret = pxa2xx_ac97_do_suspend(card, PMSG_SUSPEND);
+		ret = pxa2xx_ac97_do_suspend(card);
 
 	return ret;
 }
@@ -160,10 +160,7 @@ static int pxa2xx_ac97_resume(struct device *dev)
 	return ret;
 }
 
-static const struct dev_pm_ops pxa2xx_ac97_pm_ops = {
-	.suspend	= pxa2xx_ac97_suspend,
-	.resume		= pxa2xx_ac97_resume,
-};
+static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume);
 #endif
 
 static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 2e866398bffe..eb4ceb71123e 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -535,9 +535,9 @@ out_put_pclk:
 }
 
 #ifdef CONFIG_PM
-static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg)
+static int atmel_abdac_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_abdac *dac = card->private_data;
 
 	dw_dma_cyclic_stop(dac->dma.chan);
@@ -547,9 +547,9 @@ static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg)
 	return 0;
 }
 
-static int atmel_abdac_resume(struct platform_device *pdev)
+static int atmel_abdac_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_abdac *dac = card->private_data;
 
 	clk_enable(dac->pclk);
@@ -559,9 +559,11 @@ static int atmel_abdac_resume(struct platform_device *pdev)
 
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(atmel_abdac_pm, atmel_abdac_suspend, atmel_abdac_resume);
+#define ATMEL_ABDAC_PM_OPS	&atmel_abdac_pm
 #else
-#define atmel_abdac_suspend NULL
-#define atmel_abdac_resume NULL
+#define ATMEL_ABDAC_PM_OPS	NULL
 #endif
 
 static int __devexit atmel_abdac_remove(struct platform_device *pdev)
@@ -590,9 +592,8 @@ static struct platform_driver atmel_abdac_driver = {
 	.driver		= {
 		.name	= "atmel_abdac",
 		.owner	= THIS_MODULE,
+		.pm	= ATMEL_ABDAC_PM_OPS,
 	},
-	.suspend	= atmel_abdac_suspend,
-	.resume		= atmel_abdac_resume,
 };
 
 static int __init atmel_abdac_init(void)
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 3d0ea82ff068..bf47025bdf45 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1135,9 +1135,9 @@ err_snd_card_new:
 }
 
 #ifdef CONFIG_PM
-static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg)
+static int atmel_ac97c_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_ac97c *chip = card->private_data;
 
 	if (cpu_is_at32ap7000()) {
@@ -1151,9 +1151,9 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg)
 	return 0;
 }
 
-static int atmel_ac97c_resume(struct platform_device *pdev)
+static int atmel_ac97c_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_ac97c *chip = card->private_data;
 
 	clk_enable(chip->pclk);
@@ -1165,9 +1165,11 @@ static int atmel_ac97c_resume(struct platform_device *pdev)
 	}
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume);
+#define ATMEL_AC97C_PM_OPS	&atmel_ac97c_pm
 #else
-#define atmel_ac97c_suspend NULL
-#define atmel_ac97c_resume NULL
+#define ATMEL_AC97C_PM_OPS	NULL
 #endif
 
 static int __devexit atmel_ac97c_remove(struct platform_device *pdev)
@@ -1211,9 +1213,8 @@ static struct platform_driver atmel_ac97c_driver = {
 	.driver		= {
 		.name	= "atmel_ac97c",
 		.owner	= THIS_MODULE,
+		.pm	= ATMEL_AC97C_PM_OPS,
 	},
-	.suspend	= atmel_ac97c_suspend,
-	.resume		= atmel_ac97c_resume,
 };
 
 static int __init atmel_ac97c_init(void)
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 3484411bd5e6..1128b35b2b05 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1177,10 +1177,9 @@ static int __devexit loopback_remove(struct platform_device *devptr)
 }
 
 #ifdef CONFIG_PM
-static int loopback_suspend(struct platform_device *pdev,
-				pm_message_t state)
+static int loopback_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct loopback *loopback = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -1190,13 +1189,18 @@ static int loopback_suspend(struct platform_device *pdev,
 	return 0;
 }
 	
-static int loopback_resume(struct platform_device *pdev)
+static int loopback_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(loopback_pm, loopback_suspend, loopback_resume);
+#define LOOPBACK_PM_OPS	&loopback_pm
+#else
+#define LOOPBACK_PM_OPS	NULL
 #endif
 
 #define SND_LOOPBACK_DRIVER	"snd_aloop"
@@ -1204,13 +1208,10 @@ static int loopback_resume(struct platform_device *pdev)
 static struct platform_driver loopback_driver = {
 	.probe		= loopback_probe,
 	.remove		= __devexit_p(loopback_remove),
-#ifdef CONFIG_PM
-	.suspend	= loopback_suspend,
-	.resume		= loopback_resume,
-#endif
 	.driver		= {
 		.name	= SND_LOOPBACK_DRIVER,
 		.owner	= THIS_MODULE,
+		.pm	= LOOPBACK_PM_OPS,
 	},
 };
 
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index bc79c441a8f2..f7d3bfc6bca8 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1065,9 +1065,9 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr)
 }
 
 #ifdef CONFIG_PM
-static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state)
+static int snd_dummy_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct snd_dummy *dummy = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -1075,13 +1075,18 @@ static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state)
 	return 0;
 }
 	
-static int snd_dummy_resume(struct platform_device *pdev)
+static int snd_dummy_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_dummy_pm, snd_dummy_suspend, snd_dummy_resume);
+#define SND_DUMMY_PM_OPS	&snd_dummy_pm
+#else
+#define SND_DUMMY_PM_OPS	NULL
 #endif
 
 #define SND_DUMMY_DRIVER	"snd_dummy"
@@ -1089,13 +1094,10 @@ static int snd_dummy_resume(struct platform_device *pdev)
 static struct platform_driver snd_dummy_driver = {
 	.probe		= snd_dummy_probe,
 	.remove		= __devexit_p(snd_dummy_remove),
-#ifdef CONFIG_PM
-	.suspend	= snd_dummy_suspend,
-	.resume		= snd_dummy_resume,
-#endif
 	.driver		= {
 		.name	= SND_DUMMY_DRIVER,
 		.owner	= THIS_MODULE,
+		.pm	= SND_DUMMY_PM_OPS,
 	},
 };
 
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 99704e6a2e26..6ca59fc6dcb9 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -200,15 +200,18 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
 }
 
 #ifdef CONFIG_PM
-static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
+static int pcsp_suspend(struct device *dev)
 {
-	struct snd_pcsp *chip = platform_get_drvdata(dev);
+	struct snd_pcsp *chip = dev_get_drvdata(dev);
 	pcsp_stop_beep(chip);
 	snd_pcm_suspend_all(chip->pcm);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL);
+#define PCSP_PM_OPS	&pcsp_pm
 #else
-#define pcsp_suspend NULL
+#define PCSP_PM_OPS	NULL
 #endif	/* CONFIG_PM */
 
 static void pcsp_shutdown(struct platform_device *dev)
@@ -221,10 +224,10 @@ static struct platform_driver pcsp_platform_driver = {
 	.driver		= {
 		.name	= "pcspkr",
 		.owner	= THIS_MODULE,
+		.pm	= PCSP_PM_OPS,
 	},
 	.probe		= pcsp_probe,
 	.remove		= __devexit_p(pcsp_remove),
-	.suspend	= pcsp_suspend,
 	.shutdown	= pcsp_shutdown,
 };
 
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index aef54beaf8b7..f5ceb6f282de 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -144,19 +144,24 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr)
 }
 
 #ifdef CONFIG_PM
-static int snd_pmac_driver_suspend(struct platform_device *devptr, pm_message_t state)
+static int snd_pmac_driver_suspend(struct device *dev)
 {
-	struct snd_card *card = platform_get_drvdata(devptr);
+	struct snd_card *card = dev_get_drvdata(dev);
 	snd_pmac_suspend(card->private_data);
 	return 0;
 }
 
-static int snd_pmac_driver_resume(struct platform_device *devptr)
+static int snd_pmac_driver_resume(struct device *dev)
 {
-	struct snd_card *card = platform_get_drvdata(devptr);
+	struct snd_card *card = dev_get_drvdata(dev);
 	snd_pmac_resume(card->private_data);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_pmac_pm, snd_pmac_driver_suspend, snd_pmac_driver_resume);
+#define SND_PMAC_PM_OPS	&snd_pmac_pm
+#else
+#define SND_PMAC_PM_OPS	NULL
 #endif
 
 #define SND_PMAC_DRIVER		"snd_powermac"
@@ -164,13 +169,10 @@ static int snd_pmac_driver_resume(struct platform_device *devptr)
 static struct platform_driver snd_pmac_driver = {
 	.probe		= snd_pmac_probe,
 	.remove		= __devexit_p(snd_pmac_remove),
-#ifdef CONFIG_PM
-	.suspend	= snd_pmac_driver_suspend,
-	.resume		= snd_pmac_driver_resume,
-#endif
 	.driver		= {
 		.name	= SND_PMAC_DRIVER,
 		.owner	= THIS_MODULE,
+		.pm	= SND_PMAC_PM_OPS,
 	},
 };
 
-- 
cgit v1.2.3


From 68cb2b559278858ef9f3a7722f95df88797c7840 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Jul 2012 15:20:37 +0200
Subject: ALSA: Convert to new pm_ops for PCI drivers

Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.

Since we change vx222, vx_core and vxpocket have to be converted,
too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/vx/vx_core.c             |  2 +-
 sound/pci/ali5451/ali5451.c            | 24 +++++++++++++++---------
 sound/pci/als300.c                     | 24 +++++++++++++++---------
 sound/pci/als4000.c                    | 25 +++++++++++++++----------
 sound/pci/atiixp.c                     | 24 +++++++++++++++---------
 sound/pci/atiixp_modem.c               | 25 +++++++++++++++----------
 sound/pci/azt3328.c                    | 25 +++++++++++++++----------
 sound/pci/ca0106/ca0106_main.c         | 24 +++++++++++++++---------
 sound/pci/cmipci.c                     | 24 +++++++++++++++---------
 sound/pci/cs4281.c                     | 24 +++++++++++++++---------
 sound/pci/cs46xx/cs46xx.c              |  5 +++--
 sound/pci/cs46xx/cs46xx_lib.c          | 14 +++++++++-----
 sound/pci/cs5535audio/cs5535audio.c    |  5 +++--
 sound/pci/cs5535audio/cs5535audio.h    |  5 +----
 sound/pci/cs5535audio/cs5535audio_pm.c | 13 ++++++++-----
 sound/pci/ctxfi/ctatc.c                |  4 ++--
 sound/pci/ctxfi/ctatc.h                |  2 +-
 sound/pci/ctxfi/cthardware.h           |  2 +-
 sound/pci/ctxfi/cthw20k1.c             |  4 ++--
 sound/pci/ctxfi/cthw20k2.c             |  4 ++--
 sound/pci/ctxfi/xfi.c                  | 22 +++++++++++++---------
 sound/pci/echoaudio/echoaudio.c        | 22 +++++++++++++---------
 sound/pci/emu10k1/emu10k1.c            | 26 ++++++++++++++++----------
 sound/pci/ens1370.c                    | 25 +++++++++++++++----------
 sound/pci/es1938.c                     | 24 +++++++++++++++---------
 sound/pci/es1968.c                     | 24 +++++++++++++++---------
 sound/pci/fm801.c                      | 26 ++++++++++++++++----------
 sound/pci/hda/hda_codec.c              |  2 +-
 sound/pci/hda/hda_codec.h              |  2 +-
 sound/pci/hda/hda_intel.c              | 29 ++++++++++++++++++-----------
 sound/pci/hda/patch_analog.c           |  2 +-
 sound/pci/hda/patch_cirrus.c           |  2 +-
 sound/pci/hda/patch_conexant.c         |  2 +-
 sound/pci/hda/patch_realtek.c          |  2 +-
 sound/pci/hda/patch_sigmatel.c         |  2 +-
 sound/pci/hda/patch_via.c              |  2 +-
 sound/pci/ice1712/ice1724.c            | 26 ++++++++++++++++----------
 sound/pci/intel8x0.c                   | 24 +++++++++++++++---------
 sound/pci/intel8x0m.c                  | 24 +++++++++++++++---------
 sound/pci/maestro3.c                   | 24 +++++++++++++++---------
 sound/pci/nm256/nm256.c                | 24 +++++++++++++++---------
 sound/pci/oxygen/oxygen.c              |  5 +++--
 sound/pci/oxygen/oxygen.h              |  3 +--
 sound/pci/oxygen/oxygen_lib.c          | 17 ++++++++++-------
 sound/pci/oxygen/virtuoso.c            |  5 +++--
 sound/pci/riptide/riptide.c            | 26 ++++++++++++++++----------
 sound/pci/sis7019.c                    | 25 +++++++++++++++----------
 sound/pci/trident/trident.c            |  5 +++--
 sound/pci/trident/trident_main.c       | 14 +++++++++-----
 sound/pci/via82xx.c                    | 24 +++++++++++++++---------
 sound/pci/via82xx_modem.c              | 24 +++++++++++++++---------
 sound/pci/vx222/vx222.c                | 26 ++++++++++++++++----------
 sound/pci/ymfpci/ymfpci.c              |  5 +++--
 sound/pci/ymfpci/ymfpci_main.c         | 14 +++++++++-----
 sound/pcmcia/vx/vxpocket.c             |  2 +-
 55 files changed, 492 insertions(+), 318 deletions(-)

(limited to 'sound')

diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index b8e515999bc2..de5055a3b0d0 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -725,7 +725,7 @@ EXPORT_SYMBOL(snd_vx_dsp_load);
 /*
  * suspend
  */
-int snd_vx_suspend(struct vx_core *chip, pm_message_t state)
+int snd_vx_suspend(struct vx_core *chip)
 {
 	unsigned int i;
 
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 9dfc27bf6cc6..ee895f3c8605 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1884,9 +1884,10 @@ static int __devinit snd_ali_mixer(struct snd_ali * codec)
 }
 
 #ifdef CONFIG_PM
-static int ali_suspend(struct pci_dev *pci, pm_message_t state)
+static int ali_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ali *chip = card->private_data;
 	struct snd_ali_image *im;
 	int i, j;
@@ -1929,13 +1930,14 @@ static int ali_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int ali_resume(struct pci_dev *pci)
+static int ali_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ali *chip = card->private_data;
 	struct snd_ali_image *im;
 	int i, j;
@@ -1982,6 +1984,11 @@ static int ali_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume);
+#define ALI_PM_OPS	&ali_pm
+#else
+#define ALI_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_ali_free(struct snd_ali * codec)
@@ -2299,10 +2306,9 @@ static struct pci_driver ali5451_driver = {
 	.id_table = snd_ali_ids,
 	.probe = snd_ali_probe,
 	.remove = __devexit_p(snd_ali_remove),
-#ifdef CONFIG_PM
-	.suspend = ali_suspend,
-	.resume = ali_resume,
-#endif
+	.driver = {
+		.pm = ALI_PM_OPS,
+	},
 };                                
 
 module_pci_driver(ali5451_driver);
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 59d65388faf5..68c4469c6d19 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -766,9 +766,10 @@ static int __devinit snd_als300_create(struct snd_card *card,
 }
 
 #ifdef CONFIG_PM
-static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_als300_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_als300 *chip = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -777,13 +778,14 @@ static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_als300_resume(struct pci_dev *pci)
+static int snd_als300_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_als300 *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -802,6 +804,11 @@ static int snd_als300_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_als300_pm, snd_als300_suspend, snd_als300_resume);
+#define SND_ALS300_PM_OPS	&snd_als300_pm
+#else
+#define SND_ALS300_PM_OPS	NULL
 #endif
 
 static int __devinit snd_als300_probe(struct pci_dev *pci,
@@ -857,10 +864,9 @@ static struct pci_driver als300_driver = {
 	.id_table = snd_als300_ids,
 	.probe = snd_als300_probe,
 	.remove = __devexit_p(snd_als300_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_als300_suspend,
-	.resume = snd_als300_resume,
-#endif
+	.driver = {
+		.pm = SND_ALS300_PM_OPS,
+	},
 };
 
 module_pci_driver(als300_driver);
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 7d7f2598c748..0eeca49c5754 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -988,9 +988,10 @@ static void __devexit snd_card_als4000_remove(struct pci_dev *pci)
 }
 
 #ifdef CONFIG_PM
-static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_als4000_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_card_als4000 *acard = card->private_data;
 	struct snd_sb *chip = acard->chip;
 
@@ -1001,13 +1002,14 @@ static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_als4000_resume(struct pci_dev *pci)
+static int snd_als4000_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_card_als4000 *acard = card->private_data;
 	struct snd_sb *chip = acard->chip;
 
@@ -1033,18 +1035,21 @@ static int snd_als4000_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume);
+#define SND_ALS4000_PM_OPS	&snd_als4000_pm
+#else
+#define SND_ALS4000_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver als4000_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_als4000_ids,
 	.probe = snd_card_als4000_probe,
 	.remove = __devexit_p(snd_card_als4000_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_als4000_suspend,
-	.resume = snd_als4000_resume,
-#endif
+	.driver = {
+		.pm = SND_ALS4000_PM_OPS,
+	},
 };
 
 module_pci_driver(als4000_driver);
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 156a94f8a123..31020d2a868b 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1462,9 +1462,10 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock,
 /*
  * power management
  */
-static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_atiixp_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp *chip = card->private_data;
 	int i;
 
@@ -1484,13 +1485,14 @@ static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_atiixp_resume(struct pci_dev *pci)
+static int snd_atiixp_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp *chip = card->private_data;
 	int i;
 
@@ -1526,6 +1528,11 @@ static int snd_atiixp_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume);
+#define SND_ATIIXP_PM_OPS	&snd_atiixp_pm
+#else
+#define SND_ATIIXP_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 
@@ -1705,10 +1712,9 @@ static struct pci_driver atiixp_driver = {
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
 	.remove = __devexit_p(snd_atiixp_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_atiixp_suspend,
-	.resume = snd_atiixp_resume,
-#endif
+	.driver = {
+		.pm = SND_ATIIXP_PM_OPS,
+	},
 };
 
 module_pci_driver(atiixp_driver);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 30a4fd96ce73..79e204ec623f 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1117,9 +1117,10 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock)
 /*
  * power management
  */
-static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_atiixp_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp_modem *chip = card->private_data;
 	int i;
 
@@ -1133,13 +1134,14 @@ static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_atiixp_resume(struct pci_dev *pci)
+static int snd_atiixp_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp_modem *chip = card->private_data;
 	int i;
 
@@ -1162,8 +1164,12 @@ static int snd_atiixp_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume);
+#define SND_ATIIXP_PM_OPS	&snd_atiixp_pm
+#else
+#define SND_ATIIXP_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 #ifdef CONFIG_PROC_FS
 /*
@@ -1336,10 +1342,9 @@ static struct pci_driver atiixp_modem_driver = {
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
 	.remove = __devexit_p(snd_atiixp_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_atiixp_suspend,
-	.resume = snd_atiixp_resume,
-#endif
+	.driver = {
+		.pm = SND_ATIIXP_PM_OPS,
+	},
 };
 
 module_pci_driver(atiixp_modem_driver);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index f0b4d7493af5..4dddd871548b 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2794,9 +2794,10 @@ snd_azf3328_resume_ac97(const struct snd_azf3328 *chip)
 }
 
 static int
-snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
+snd_azf3328_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_azf3328 *chip = card->private_data;
 	u16 *saved_regs_ctrl_u16;
 
@@ -2824,14 +2825,15 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
 static int
-snd_azf3328_resume(struct pci_dev *pci)
+snd_azf3328_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	const struct snd_azf3328 *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -2859,18 +2861,21 @@ snd_azf3328_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume);
+#define SND_AZF3328_PM_OPS	&snd_azf3328_pm
+#else
+#define SND_AZF3328_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver azf3328_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_azf3328_ids,
 	.probe = snd_azf3328_probe,
 	.remove = __devexit_p(snd_azf3328_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_azf3328_suspend,
-	.resume = snd_azf3328_resume,
-#endif
+	.driver = {
+		.pm = SND_AZF3328_PM_OPS,
+	},
 };
 
 module_pci_driver(azf3328_driver);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index e76d68a7081f..83277b747b36 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1872,9 +1872,10 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci)
 }
 
 #ifdef CONFIG_PM
-static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_ca0106_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ca0106 *chip = card->private_data;
 	int i;
 
@@ -1889,13 +1890,14 @@ static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_ca0106_resume(struct pci_dev *pci)
+static int snd_ca0106_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ca0106 *chip = card->private_data;
 	int i;
 
@@ -1922,6 +1924,11 @@ static int snd_ca0106_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume);
+#define SND_CA0106_PM_OPS	&snd_ca0106_pm
+#else
+#define SND_CA0106_PM_OPS	NULL
 #endif
 
 // PCI IDs
@@ -1937,10 +1944,9 @@ static struct pci_driver ca0106_driver = {
 	.id_table = snd_ca0106_ids,
 	.probe = snd_ca0106_probe,
 	.remove = __devexit_p(snd_ca0106_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_ca0106_suspend,
-	.resume = snd_ca0106_resume,
-#endif
+	.driver = {
+		.pm = SND_CA0106_PM_OPS,
+	},
 };
 
 module_pci_driver(ca0106_driver);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 3815bd4c6779..b7d6f2b886ef 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3338,9 +3338,10 @@ static unsigned char saved_mixers[] = {
 	SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT,
 };
 
-static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_cmipci_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cmipci *cm = card->private_data;
 	int i;
 
@@ -3361,13 +3362,14 @@ static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_cmipci_resume(struct pci_dev *pci)
+static int snd_cmipci_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cmipci *cm = card->private_data;
 	int i;
 
@@ -3396,6 +3398,11 @@ static int snd_cmipci_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume);
+#define SND_CMIPCI_PM_OPS	&snd_cmipci_pm
+#else
+#define SND_CMIPCI_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static struct pci_driver cmipci_driver = {
@@ -3403,10 +3410,9 @@ static struct pci_driver cmipci_driver = {
 	.id_table = snd_cmipci_ids,
 	.probe = snd_cmipci_probe,
 	.remove = __devexit_p(snd_cmipci_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_cmipci_suspend,
-	.resume = snd_cmipci_resume,
-#endif
+	.driver = {
+		.pm = SND_CMIPCI_PM_OPS,
+	},
 };
 	
 module_pci_driver(cmipci_driver);
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 33506ee569bd..45a8317085f4 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1997,9 +1997,10 @@ static int saved_regs[SUSPEND_REGISTERS] = {
 
 #define CLKCR1_CKRA                             0x00010000L
 
-static int cs4281_suspend(struct pci_dev *pci, pm_message_t state)
+static int cs4281_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs4281 *chip = card->private_data;
 	u32 ulCLK;
 	unsigned int i;
@@ -2040,13 +2041,14 @@ static int cs4281_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int cs4281_resume(struct pci_dev *pci)
+static int cs4281_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs4281 *chip = card->private_data;
 	unsigned int i;
 	u32 ulCLK;
@@ -2082,6 +2084,11 @@ static int cs4281_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume);
+#define CS4281_PM_OPS	&cs4281_pm
+#else
+#define CS4281_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static struct pci_driver cs4281_driver = {
@@ -2089,10 +2096,9 @@ static struct pci_driver cs4281_driver = {
 	.id_table = snd_cs4281_ids,
 	.probe = snd_cs4281_probe,
 	.remove = __devexit_p(snd_cs4281_remove),
-#ifdef CONFIG_PM
-	.suspend = cs4281_suspend,
-	.resume = cs4281_resume,
-#endif
+	.driver = {
+		.pm = CS4281_PM_OPS,
+	},
 };
 	
 module_pci_driver(cs4281_driver);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 6cc7404e0e8f..00e03bc9a762 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -167,8 +167,9 @@ static struct pci_driver cs46xx_driver = {
 	.probe = snd_card_cs46xx_probe,
 	.remove = __devexit_p(snd_card_cs46xx_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_cs46xx_suspend,
-	.resume = snd_cs46xx_resume,
+	.driver = {
+		.pm = &snd_cs46xx_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 4fa53161b094..28b9747becc9 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -3599,9 +3599,10 @@ static unsigned int saved_regs[] = {
 	BA1_CVOL,
 };
 
-int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_cs46xx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_cs46xx *chip = card->private_data;
 	int i, amp_saved;
 
@@ -3628,13 +3629,14 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_cs46xx_resume(struct pci_dev *pci)
+static int snd_cs46xx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_cs46xx *chip = card->private_data;
 	int amp_saved;
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
@@ -3707,6 +3709,8 @@ int snd_cs46xx_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+SIMPLE_DEV_PM_OPS(snd_cs46xx_pm, snd_cs46xx_suspend, snd_cs46xx_resume);
 #endif /* CONFIG_PM */
 
 
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 2c9697cf0a1a..51f64ba5facf 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -400,8 +400,9 @@ static struct pci_driver cs5535audio_driver = {
 	.probe = snd_cs5535audio_probe,
 	.remove = __devexit_p(snd_cs5535audio_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_cs5535audio_suspend,
-	.resume = snd_cs5535audio_resume,
+	.driver = {
+		.pm = &snd_cs5535audio_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 51966d782a3c..bb3cc641130c 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -94,10 +94,7 @@ struct cs5535audio {
 	struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS];
 };
 
-#ifdef CONFIG_PM
-int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
-int snd_cs5535audio_resume(struct pci_dev *pci);
-#endif
+extern const struct dev_pm_ops snd_cs5535audio_pm;
 
 #ifdef CONFIG_OLPC
 void __devinit olpc_prequirks(struct snd_card *card,
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index 185b00088320..6c34def5986d 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -55,9 +55,10 @@ static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au)
 
 }
 
-int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_cs5535audio_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs5535audio *cs5535au = card->private_data;
 	int i;
 
@@ -77,13 +78,14 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
 		return -EIO;
 	}
 	pci_disable_device(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_cs5535audio_resume(struct pci_dev *pci)
+static int snd_cs5535audio_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs5535audio *cs5535au = card->private_data;
 	u32 tmp;
 	int timeout;
@@ -129,3 +131,4 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
 	return 0;
 }
 
+SIMPLE_DEV_PM_OPS(snd_cs5535audio_pm, snd_cs5535audio_suspend, snd_cs5535audio_resume);
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index d8a4423539ce..8e40262d4117 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1537,7 +1537,7 @@ static void atc_connect_resources(struct ct_atc *atc)
 }
 
 #ifdef CONFIG_PM
-static int atc_suspend(struct ct_atc *atc, pm_message_t state)
+static int atc_suspend(struct ct_atc *atc)
 {
 	int i;
 	struct hw *hw = atc->hw;
@@ -1553,7 +1553,7 @@ static int atc_suspend(struct ct_atc *atc, pm_message_t state)
 
 	atc_release_resources(atc);
 
-	hw->suspend(hw, state);
+	hw->suspend(hw);
 
 	return 0;
 }
diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h
index 3a0def656af0..653e813ad142 100644
--- a/sound/pci/ctxfi/ctatc.h
+++ b/sound/pci/ctxfi/ctatc.h
@@ -144,7 +144,7 @@ struct ct_atc {
 	struct ct_timer *timer;
 
 #ifdef CONFIG_PM
-	int (*suspend)(struct ct_atc *atc, pm_message_t state);
+	int (*suspend)(struct ct_atc *atc);
 	int (*resume)(struct ct_atc *atc);
 #define NUM_PCMS (NUM_CTALSADEVS - 1)
 	struct snd_pcm *pcms[NUM_PCMS];
diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h
index 908315bec3b4..c56fe533b3f3 100644
--- a/sound/pci/ctxfi/cthardware.h
+++ b/sound/pci/ctxfi/cthardware.h
@@ -73,7 +73,7 @@ struct hw {
 	int (*card_stop)(struct hw *hw);
 	int (*pll_init)(struct hw *hw, unsigned int rsr);
 #ifdef CONFIG_PM
-	int (*suspend)(struct hw *hw, pm_message_t state);
+	int (*suspend)(struct hw *hw);
 	int (*resume)(struct hw *hw, struct card_conf *info);
 #endif
 	int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source);
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index a7df19791f5a..dc1969bc67d4 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -2086,7 +2086,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info)
 }
 
 #ifdef CONFIG_PM
-static int hw_suspend(struct hw *hw, pm_message_t state)
+static int hw_suspend(struct hw *hw)
 {
 	struct pci_dev *pci = hw->pci;
 
@@ -2099,7 +2099,7 @@ static int hw_suspend(struct hw *hw, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 
 	return 0;
 }
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index d6c54b524bfa..9d1231dc4ae2 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -2202,7 +2202,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info)
 }
 
 #ifdef CONFIG_PM
-static int hw_suspend(struct hw *hw, pm_message_t state)
+static int hw_suspend(struct hw *hw)
 {
 	struct pci_dev *pci = hw->pci;
 
@@ -2210,7 +2210,7 @@ static int hw_suspend(struct hw *hw, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 
 	return 0;
 }
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index 75aa2c338410..e002183ef8b2 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -126,21 +126,26 @@ static void __devexit ct_card_remove(struct pci_dev *pci)
 }
 
 #ifdef CONFIG_PM
-static int ct_card_suspend(struct pci_dev *pci, pm_message_t state)
+static int ct_card_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ct_atc *atc = card->private_data;
 
-	return atc->suspend(atc, state);
+	return atc->suspend(atc);
 }
 
-static int ct_card_resume(struct pci_dev *pci)
+static int ct_card_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ct_atc *atc = card->private_data;
 
 	return atc->resume(atc);
 }
+
+static SIMPLE_DEV_PM_OPS(ct_card_pm, ct_card_suspend, ct_card_resume);
+#define CT_CARD_PM_OPS	&ct_card_pm
+#else
+#define CT_CARD_PM_OPS	NULL
 #endif
 
 static struct pci_driver ct_driver = {
@@ -148,10 +153,9 @@ static struct pci_driver ct_driver = {
 	.id_table = ct_pci_dev_ids,
 	.probe = ct_card_probe,
 	.remove = __devexit_p(ct_card_remove),
-#ifdef CONFIG_PM
-	.suspend = ct_card_suspend,
-	.resume = ct_card_resume,
-#endif
+	.driver = {
+		.pm = CT_CARD_PM_OPS,
+	},
 };
 
 module_pci_driver(ct_driver);
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 0f8eda1dafdb..0ff754f180d0 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2205,9 +2205,10 @@ ctl_error:
 
 #if defined(CONFIG_PM)
 
-static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_echo_suspend(struct device *dev)
 {
-	struct echoaudio *chip = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct echoaudio *chip = dev_get_drvdata(dev);
 
 	DE_INIT(("suspend start\n"));
 	snd_pcm_suspend_all(chip->analog_pcm);
@@ -2242,9 +2243,10 @@ static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state)
 
 
 
-static int snd_echo_resume(struct pci_dev *pci)
+static int snd_echo_resume(struct device *dev)
 {
-	struct echoaudio *chip = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct echoaudio *chip = dev_get_drvdata(dev);
 	struct comm_page *commpage, *commpage_bak;
 	u32 pipe_alloc_mask;
 	int err;
@@ -2307,10 +2309,13 @@ static int snd_echo_resume(struct pci_dev *pci)
 	return 0;
 }
 
+static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume);
+#define SND_ECHO_PM_OPS	&snd_echo_pm
+#else
+#define SND_ECHO_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 
-
 static void __devexit snd_echo_remove(struct pci_dev *pci)
 {
 	struct echoaudio *chip;
@@ -2333,10 +2338,9 @@ static struct pci_driver echo_driver = {
 	.id_table = snd_echo_ids,
 	.probe = snd_echo_probe,
 	.remove = __devexit_p(snd_echo_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_echo_suspend,
-	.resume = snd_echo_resume,
-#endif /* CONFIG_PM */
+	.driver = {
+		.pm = SND_ECHO_PM_OPS,
+	},
 };
 
 module_pci_driver(echo_driver);
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 7fdbbe4d9965..ddac4e6d660d 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -207,9 +207,10 @@ static void __devexit snd_card_emu10k1_remove(struct pci_dev *pci)
 
 
 #ifdef CONFIG_PM
-static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_emu10k1_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_emu10k1 *emu = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -231,13 +232,14 @@ static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_emu10k1_resume(struct pci_dev *pci)
+static int snd_emu10k1_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_emu10k1 *emu = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -261,17 +263,21 @@ static int snd_emu10k1_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(snd_emu10k1_pm, snd_emu10k1_suspend, snd_emu10k1_resume);
+#define SND_EMU10K1_PM_OPS	&snd_emu10k1_pm
+#else
+#define SND_EMU10K1_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver emu10k1_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_emu10k1_ids,
 	.probe = snd_card_emu10k1_probe,
 	.remove = __devexit_p(snd_card_emu10k1_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_emu10k1_suspend,
-	.resume = snd_emu10k1_resume,
-#endif
+	.driver = {
+		.pm = SND_EMU10K1_PM_OPS,
+	},
 };
 
 module_pci_driver(emu10k1_driver);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 3821c81d1c99..f7e6f73186e1 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2033,9 +2033,10 @@ static void snd_ensoniq_chip_init(struct ensoniq *ensoniq)
 }
 
 #ifdef CONFIG_PM
-static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_ensoniq_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ensoniq *ensoniq = card->private_data;
 	
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -2058,13 +2059,14 @@ static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_ensoniq_resume(struct pci_dev *pci)
+static int snd_ensoniq_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ensoniq *ensoniq = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -2087,8 +2089,12 @@ static int snd_ensoniq_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume);
+#define SND_ENSONIQ_PM_OPS	&snd_ensoniq_pm
+#else
+#define SND_ENSONIQ_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static int __devinit snd_ensoniq_create(struct snd_card *card,
 				     struct pci_dev *pci,
@@ -2493,10 +2499,9 @@ static struct pci_driver ens137x_driver = {
 	.id_table = snd_audiopci_ids,
 	.probe = snd_audiopci_probe,
 	.remove = __devexit_p(snd_audiopci_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_ensoniq_suspend,
-	.resume = snd_ensoniq_resume,
-#endif
+	.driver = {
+		.pm = SND_ENSONIQ_PM_OPS,
+	},
 };
 	
 module_pci_driver(ens137x_driver);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 82c8d8c5c52a..227dff70069f 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1474,9 +1474,10 @@ static unsigned char saved_regs[SAVED_REG_SIZE+1] = {
 };
 
 
-static int es1938_suspend(struct pci_dev *pci, pm_message_t state)
+static int es1938_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1938 *chip = card->private_data;
 	unsigned char *s, *d;
 
@@ -1494,13 +1495,14 @@ static int es1938_suspend(struct pci_dev *pci, pm_message_t state)
 	}
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int es1938_resume(struct pci_dev *pci)
+static int es1938_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1938 *chip = card->private_data;
 	unsigned char *s, *d;
 
@@ -1534,6 +1536,11 @@ static int es1938_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume);
+#define ES1938_PM_OPS	&es1938_pm
+#else
+#define ES1938_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef SUPPORT_JOYSTICK
@@ -1887,10 +1894,9 @@ static struct pci_driver es1938_driver = {
 	.id_table = snd_es1938_ids,
 	.probe = snd_es1938_probe,
 	.remove = __devexit_p(snd_es1938_remove),
-#ifdef CONFIG_PM
-	.suspend = es1938_suspend,
-	.resume = es1938_resume,
-#endif
+	.driver = {
+		.pm = ES1938_PM_OPS,
+	},
 };
 
 module_pci_driver(es1938_driver);
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 52b5c0bf90c1..fb4c90b99c00 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2381,9 +2381,10 @@ static void snd_es1968_start_irq(struct es1968 *chip)
 /*
  * PM support
  */
-static int es1968_suspend(struct pci_dev *pci, pm_message_t state)
+static int es1968_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1968 *chip = card->private_data;
 
 	if (! chip->do_pm)
@@ -2398,13 +2399,14 @@ static int es1968_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int es1968_resume(struct pci_dev *pci)
+static int es1968_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1968 *chip = card->private_data;
 	struct esschan *es;
 
@@ -2454,6 +2456,11 @@ static int es1968_resume(struct pci_dev *pci)
 	chip->in_suspend = 0;
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume);
+#define ES1968_PM_OPS	&es1968_pm
+#else
+#define ES1968_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef SUPPORT_JOYSTICK
@@ -2903,10 +2910,9 @@ static struct pci_driver es1968_driver = {
 	.id_table = snd_es1968_ids,
 	.probe = snd_es1968_probe,
 	.remove = __devexit_p(snd_es1968_remove),
-#ifdef CONFIG_PM
-	.suspend = es1968_suspend,
-	.resume = es1968_resume,
-#endif
+	.driver = {
+		.pm = ES1968_PM_OPS,
+	},
 };
 
 module_pci_driver(es1968_driver);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index b32e8024ea86..522c8706f244 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1369,9 +1369,10 @@ static unsigned char saved_regs[] = {
 	FM801_CODEC_CTRL, FM801_I2S_MODE, FM801_VOLUME, FM801_GEN_CTRL,
 };
 
-static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_fm801_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct fm801 *chip = card->private_data;
 	int i;
 
@@ -1385,13 +1386,14 @@ static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_fm801_resume(struct pci_dev *pci)
+static int snd_fm801_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct fm801 *chip = card->private_data;
 	int i;
 
@@ -1414,17 +1416,21 @@ static int snd_fm801_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume);
+#define SND_FM801_PM_OPS	&snd_fm801_pm
+#else
+#define SND_FM801_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver fm801_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_fm801_ids,
 	.probe = snd_card_fm801_probe,
 	.remove = __devexit_p(snd_card_fm801_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_fm801_suspend,
-	.resume = snd_fm801_resume,
-#endif
+	.driver = {
+		.pm = SND_FM801_PM_OPS,
+	},
 };
 
 module_pci_driver(fm801_driver);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 51cb2a2e4fce..f4c274c5144b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3545,7 +3545,7 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {}
 static void hda_call_codec_suspend(struct hda_codec *codec)
 {
 	if (codec->patch_ops.suspend)
-		codec->patch_ops.suspend(codec, PMSG_SUSPEND);
+		codec->patch_ops.suspend(codec);
 	hda_cleanup_all_streams(codec);
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2fdaadbb4326..5aab408dcb6a 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -703,7 +703,7 @@ struct hda_codec_ops {
 	void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state);
 #ifdef CONFIG_PM
-	int (*suspend)(struct hda_codec *codec, pm_message_t state);
+	int (*suspend)(struct hda_codec *codec);
 	int (*resume)(struct hda_codec *codec);
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7757536b9d5f..a69ec7448714 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2403,9 +2403,10 @@ static void azx_power_notify(struct hda_bus *bus)
  * power management
  */
 
-static int azx_suspend(struct pci_dev *pci, pm_message_t state)
+static int azx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 	struct azx_pcm *p;
 
@@ -2424,13 +2425,14 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 		pci_disable_msi(chip->pci);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int azx_resume(struct pci_dev *pci)
+static int azx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -2455,6 +2457,12 @@ static int azx_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume);
+#define AZX_PM_OPS	&azx_pm
+#else
+#define azx_suspend(dev)
+#define azx_resume(dev)
+#define AZX_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 
@@ -2521,13 +2529,13 @@ static void azx_vs_set_state(struct pci_dev *pci,
 			   disabled ? "Disabling" : "Enabling",
 			   pci_name(chip->pci));
 		if (disabled) {
-			azx_suspend(pci, PMSG_FREEZE);
+			azx_suspend(&pci->dev);
 			chip->disabled = true;
 			snd_hda_lock_devices(chip->bus);
 		} else {
 			snd_hda_unlock_devices(chip->bus);
 			chip->disabled = false;
-			azx_resume(pci);
+			azx_resume(&pci->dev);
 		}
 	}
 }
@@ -3398,10 +3406,9 @@ static struct pci_driver azx_driver = {
 	.id_table = azx_ids,
 	.probe = azx_probe,
 	.remove = __devexit_p(azx_remove),
-#ifdef CONFIG_PM
-	.suspend = azx_suspend,
-	.resume = azx_resume,
-#endif
+	.driver = {
+		.pm = AZX_PM_OPS,
+	},
 };
 
 module_pci_driver(azx_driver);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d8b2d6dee986..0208fa121e5a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -642,7 +642,7 @@ static void ad198x_free(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
+static int ad198x_suspend(struct hda_codec *codec)
 {
 	ad198x_shutup(codec);
 	return 0;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 9647ed4d7929..0c4c1a61b378 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1892,7 +1892,7 @@ static int cs421x_parse_auto_config(struct hda_codec *codec)
 	Manage PDREF, when transitioning to D3hot
 	(DAC,ADC) -> D3, PDREF=1, AFG->D3
 */
-static int cs421x_suspend(struct hda_codec *codec, pm_message_t state)
+static int cs421x_suspend(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	unsigned int coef;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2bf99fc1cbf2..14361184ae1e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -554,7 +554,7 @@ static int conexant_build_controls(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static int conexant_suspend(struct hda_codec *codec, pm_message_t state)
+static int conexant_suspend(struct hda_codec *codec)
 {
 	snd_hda_shutup_pins(codec);
 	return 0;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5ccf10a4d593..ab2c729b88ea 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2300,7 +2300,7 @@ static void alc_power_eapd(struct hda_codec *codec)
 	alc_auto_setup_eapd(codec, false);
 }
 
-static int alc_suspend(struct hda_codec *codec, pm_message_t state)
+static int alc_suspend(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	alc_shutup(codec);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 07675282015a..a1596a3b171c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4997,7 +4997,7 @@ static int stac92xx_resume(struct hda_codec *codec)
 	return 0;
 }
 
-static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
+static int stac92xx_suspend(struct hda_codec *codec)
 {
 	stac92xx_shutup(codec);
 	return 0;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 82b368068e08..90645560ed39 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1748,7 +1748,7 @@ static void via_unsol_event(struct hda_codec *codec,
 }
 
 #ifdef CONFIG_PM
-static int via_suspend(struct hda_codec *codec, pm_message_t state)
+static int via_suspend(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	vt1708_stop_hp_work(spec);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index a01a00d1cf4d..bed9f34f4efe 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2793,9 +2793,10 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci)
 }
 
 #ifdef CONFIG_PM
-static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_vt1724_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ice1712 *ice = card->private_data;
 
 	if (!ice->pm_suspend_enabled)
@@ -2820,13 +2821,14 @@ static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_vt1724_resume(struct pci_dev *pci)
+static int snd_vt1724_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ice1712 *ice = card->private_data;
 
 	if (!ice->pm_suspend_enabled)
@@ -2871,17 +2873,21 @@ static int snd_vt1724_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(snd_vt1724_pm, snd_vt1724_suspend, snd_vt1724_resume);
+#define SND_VT1724_PM_OPS	&snd_vt1724_pm
+#else
+#define SND_VT1724_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver vt1724_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_vt1724_ids,
 	.probe = snd_vt1724_probe,
 	.remove = __devexit_p(snd_vt1724_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_vt1724_suspend,
-	.resume = snd_vt1724_resume,
-#endif
+	.driver = {
+		.pm = SND_VT1724_PM_OPS,
+	},
 };
 
 module_pci_driver(vt1724_driver);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index f4e2dd4da8cf..cd553f592e2d 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2624,9 +2624,10 @@ static int snd_intel8x0_free(struct intel8x0 *chip)
 /*
  * power management
  */
-static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state)
+static int intel8x0_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0 *chip = card->private_data;
 	int i;
 
@@ -2658,13 +2659,14 @@ static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state)
 	/* The call below may disable built-in speaker on some laptops
 	 * after S2RAM.  So, don't touch it.
 	 */
-	/* pci_set_power_state(pci, pci_choose_state(pci, state)); */
+	/* pci_set_power_state(pci, PCI_D3hot); */
 	return 0;
 }
 
-static int intel8x0_resume(struct pci_dev *pci)
+static int intel8x0_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0 *chip = card->private_data;
 	int i;
 
@@ -2734,6 +2736,11 @@ static int intel8x0_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume);
+#define INTEL8X0_PM_OPS	&intel8x0_pm
+#else
+#define INTEL8X0_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #define INTEL8X0_TESTBUF_SIZE	32768	/* enough large for one shot */
@@ -3343,10 +3350,9 @@ static struct pci_driver intel8x0_driver = {
 	.id_table = snd_intel8x0_ids,
 	.probe = snd_intel8x0_probe,
 	.remove = __devexit_p(snd_intel8x0_remove),
-#ifdef CONFIG_PM
-	.suspend = intel8x0_suspend,
-	.resume = intel8x0_resume,
-#endif
+	.driver = {
+		.pm = INTEL8X0_PM_OPS,
+	},
 };
 
 module_pci_driver(intel8x0_driver);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index fc27a6a69e77..da44bb3f8e7a 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1012,9 +1012,10 @@ static int snd_intel8x0m_free(struct intel8x0m *chip)
 /*
  * power management
  */
-static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state)
+static int intel8x0m_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0m *chip = card->private_data;
 	int i;
 
@@ -1028,13 +1029,14 @@ static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state)
 	}
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int intel8x0m_resume(struct pci_dev *pci)
+static int intel8x0m_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0m *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -1060,6 +1062,11 @@ static int intel8x0m_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume);
+#define INTEL8X0M_PM_OPS	&intel8x0m_pm
+#else
+#define INTEL8X0M_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef CONFIG_PROC_FS
@@ -1329,10 +1336,9 @@ static struct pci_driver intel8x0m_driver = {
 	.id_table = snd_intel8x0m_ids,
 	.probe = snd_intel8x0m_probe,
 	.remove = __devexit_p(snd_intel8x0m_remove),
-#ifdef CONFIG_PM
-	.suspend = intel8x0m_suspend,
-	.resume = intel8x0m_resume,
-#endif
+	.driver = {
+		.pm = INTEL8X0M_PM_OPS,
+	},
 };
 
 module_pci_driver(intel8x0m_driver);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index deef21399586..36008a943aa3 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2459,9 +2459,10 @@ static int snd_m3_free(struct snd_m3 *chip)
  * APM support
  */
 #ifdef CONFIG_PM
-static int m3_suspend(struct pci_dev *pci, pm_message_t state)
+static int m3_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_m3 *chip = card->private_data;
 	int i, dsp_index;
 
@@ -2489,13 +2490,14 @@ static int m3_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int m3_resume(struct pci_dev *pci)
+static int m3_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_m3 *chip = card->private_data;
 	int i, dsp_index;
 
@@ -2546,6 +2548,11 @@ static int m3_resume(struct pci_dev *pci)
 	chip->in_suspend = 0;
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume);
+#define M3_PM_OPS	&m3_pm
+#else
+#define M3_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef CONFIG_SND_MAESTRO3_INPUT
@@ -2842,10 +2849,9 @@ static struct pci_driver m3_driver = {
 	.id_table = snd_m3_ids,
 	.probe = snd_m3_probe,
 	.remove = __devexit_p(snd_m3_remove),
-#ifdef CONFIG_PM
-	.suspend = m3_suspend,
-	.resume = m3_resume,
-#endif
+	.driver = {
+		.pm = M3_PM_OPS,
+	},
 };
 	
 module_pci_driver(m3_driver);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 8159b05ee94d..465cff25b146 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1382,9 +1382,10 @@ snd_nm256_peek_for_sig(struct nm256 *chip)
  * APM event handler, so the card is properly reinitialized after a power
  * event.
  */
-static int nm256_suspend(struct pci_dev *pci, pm_message_t state)
+static int nm256_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct nm256 *chip = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -1393,13 +1394,14 @@ static int nm256_suspend(struct pci_dev *pci, pm_message_t state)
 	chip->coeffs_current = 0;
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int nm256_resume(struct pci_dev *pci)
+static int nm256_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct nm256 *chip = card->private_data;
 	int i;
 
@@ -1434,6 +1436,11 @@ static int nm256_resume(struct pci_dev *pci)
 	chip->in_resume = 0;
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume);
+#define NM256_PM_OPS	&nm256_pm
+#else
+#define NM256_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_nm256_free(struct nm256 *chip)
@@ -1747,10 +1754,9 @@ static struct pci_driver nm256_driver = {
 	.id_table = snd_nm256_ids,
 	.probe = snd_nm256_probe,
 	.remove = __devexit_p(snd_nm256_remove),
-#ifdef CONFIG_PM
-	.suspend = nm256_suspend,
-	.resume = nm256_resume,
-#endif
+	.driver = {
+		.pm = NM256_PM_OPS,
+	},
 };
 
 module_pci_driver(nm256_driver);
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 610275bfbaeb..37520a2b4dcf 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -873,8 +873,9 @@ static struct pci_driver oxygen_driver = {
 	.probe = generic_oxygen_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
 #ifdef CONFIG_PM
-	.suspend = oxygen_pci_suspend,
-	.resume = oxygen_pci_resume,
+	.driver = {
+		.pm = &oxygen_pci_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index f53897a708b4..7112a89fb8bd 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -162,8 +162,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 		    );
 void oxygen_pci_remove(struct pci_dev *pci);
 #ifdef CONFIG_PM
-int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
-int oxygen_pci_resume(struct pci_dev *pci);
+extern const struct dev_pm_ops oxygen_pci_pm;
 #endif
 void oxygen_pci_shutdown(struct pci_dev *pci);
 
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 92e2d67f16a1..ab8738e21ad1 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -727,9 +727,10 @@ void oxygen_pci_remove(struct pci_dev *pci)
 EXPORT_SYMBOL(oxygen_pci_remove);
 
 #ifdef CONFIG_PM
-int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
+static int oxygen_pci_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct oxygen *chip = card->private_data;
 	unsigned int i, saved_interrupt_mask;
 
@@ -756,10 +757,9 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
-EXPORT_SYMBOL(oxygen_pci_suspend);
 
 static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = {
 	0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff,
@@ -787,9 +787,10 @@ static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec)
 					  chip->saved_ac97_registers[codec][i]);
 }
 
-int oxygen_pci_resume(struct pci_dev *pci)
+static int oxygen_pci_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct oxygen *chip = card->private_data;
 	unsigned int i;
 
@@ -820,7 +821,9 @@ int oxygen_pci_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-EXPORT_SYMBOL(oxygen_pci_resume);
+
+SIMPLE_DEV_PM_OPS(oxygen_pci_pm, oxygen_pci_suspend, oxygen_pci_resume);
+EXPORT_SYMBOL(oxygen_pci_pm);
 #endif /* CONFIG_PM */
 
 void oxygen_pci_shutdown(struct pci_dev *pci)
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 19962c6d38c3..d3b606b69f3b 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -94,8 +94,9 @@ static struct pci_driver xonar_driver = {
 	.probe = xonar_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
 #ifdef CONFIG_PM
-	.suspend = oxygen_pci_suspend,
-	.resume = oxygen_pci_resume,
+	.driver = {
+		.pm = &oxygen_pci_pm,
+	},
 #endif
 	.shutdown = oxygen_pci_shutdown,
 };
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index cbeb3f77350c..760ee467cd9a 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1151,9 +1151,10 @@ static void riptide_handleirq(unsigned long dev_id)
 }
 
 #ifdef CONFIG_PM
-static int riptide_suspend(struct pci_dev *pci, pm_message_t state)
+static int riptide_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_riptide *chip = card->private_data;
 
 	chip->in_suspend = 1;
@@ -1162,13 +1163,14 @@ static int riptide_suspend(struct pci_dev *pci, pm_message_t state)
 	snd_ac97_suspend(chip->ac97);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int riptide_resume(struct pci_dev *pci)
+static int riptide_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_riptide *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -1186,7 +1188,12 @@ static int riptide_resume(struct pci_dev *pci)
 	chip->in_suspend = 0;
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume);
+#define RIPTIDE_PM_OPS	&riptide_pm
+#else
+#define RIPTIDE_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
 {
@@ -2180,10 +2187,9 @@ static struct pci_driver driver = {
 	.id_table = snd_riptide_ids,
 	.probe = snd_card_riptide_probe,
 	.remove = __devexit_p(snd_card_riptide_remove),
-#ifdef CONFIG_PM
-	.suspend = riptide_suspend,
-	.resume = riptide_resume,
-#endif
+	.driver = {
+		.pm = RIPTIDE_PM_OPS,
+	},
 };
 
 #ifdef SUPPORT_JOYSTICK
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 1552642765d6..512434efcc31 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1209,9 +1209,10 @@ static int sis_chip_init(struct sis7019 *sis)
 }
 
 #ifdef CONFIG_PM
-static int sis_suspend(struct pci_dev *pci, pm_message_t state)
+static int sis_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct sis7019 *sis = card->private_data;
 	void __iomem *ioaddr = sis->ioaddr;
 	int i;
@@ -1241,13 +1242,14 @@ static int sis_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int sis_resume(struct pci_dev *pci)
+static int sis_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct sis7019 *sis = card->private_data;
 	void __iomem *ioaddr = sis->ioaddr;
 	int i;
@@ -1298,6 +1300,11 @@ error:
 	snd_card_disconnect(card);
 	return -EIO;
 }
+
+static SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume);
+#define SIS_PM_OPS	&sis_pm
+#else
+#define SIS_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int sis_alloc_suspend(struct sis7019 *sis)
@@ -1481,11 +1488,9 @@ static struct pci_driver sis7019_driver = {
 	.id_table = snd_sis7019_ids,
 	.probe = snd_sis7019_probe,
 	.remove = __devexit_p(snd_sis7019_remove),
-
-#ifdef CONFIG_PM
-	.suspend = sis_suspend,
-	.resume = sis_resume,
-#endif
+	.driver = {
+		.pm = SIS_PM_OPS,
+	},
 };
 
 module_pci_driver(sis7019_driver);
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 611983ec7321..f61346a555bb 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -178,8 +178,9 @@ static struct pci_driver trident_driver = {
 	.probe = snd_trident_probe,
 	.remove = __devexit_p(snd_trident_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_trident_suspend,
-	.resume = snd_trident_resume,
+	.driver = {
+		.pm = &snd_trident_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 61d3c0e8d4ce..b4430c093bad 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -3920,9 +3920,10 @@ static void snd_trident_clear_voices(struct snd_trident * trident, unsigned shor
 }
 
 #ifdef CONFIG_PM
-int snd_trident_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_trident_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_trident *trident = card->private_data;
 
 	trident->in_suspend = 1;
@@ -3936,13 +3937,14 @@ int snd_trident_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_trident_resume(struct pci_dev *pci)
+static int snd_trident_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_trident *trident = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -3979,4 +3981,6 @@ int snd_trident_resume(struct pci_dev *pci)
 	trident->in_suspend = 0;
 	return 0;
 }
+
+SIMPLE_DEV_PM_OPS(snd_trident_pm, snd_trident_suspend, snd_trident_resume);
 #endif /* CONFIG_PM */
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index b5afab48943e..0eb7245dd362 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2242,9 +2242,10 @@ static int snd_via82xx_chip_init(struct via82xx *chip)
 /*
  * power management
  */
-static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_via82xx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx *chip = card->private_data;
 	int i;
 
@@ -2265,13 +2266,14 @@ static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_via82xx_resume(struct pci_dev *pci)
+static int snd_via82xx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx *chip = card->private_data;
 	int i;
 
@@ -2306,6 +2308,11 @@ static int snd_via82xx_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume);
+#define SND_VIA82XX_PM_OPS	&snd_via82xx_pm
+#else
+#define SND_VIA82XX_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_via82xx_free(struct via82xx *chip)
@@ -2624,10 +2631,9 @@ static struct pci_driver via82xx_driver = {
 	.id_table = snd_via82xx_ids,
 	.probe = snd_via82xx_probe,
 	.remove = __devexit_p(snd_via82xx_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_via82xx_suspend,
-	.resume = snd_via82xx_resume,
-#endif
+	.driver = {
+		.pm = SND_VIA82XX_PM_OPS,
+	},
 };
 
 module_pci_driver(via82xx_driver);
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 59fd47ed0a31..e886bc16999d 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1023,9 +1023,10 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip)
 /*
  * power management
  */
-static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_via82xx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx_modem *chip = card->private_data;
 	int i;
 
@@ -1039,13 +1040,14 @@ static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_via82xx_resume(struct pci_dev *pci)
+static int snd_via82xx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx_modem *chip = card->private_data;
 	int i;
 
@@ -1069,6 +1071,11 @@ static int snd_via82xx_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume);
+#define SND_VIA82XX_PM_OPS	&snd_via82xx_pm
+#else
+#define SND_VIA82XX_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_via82xx_free(struct via82xx_modem *chip)
@@ -1228,10 +1235,9 @@ static struct pci_driver via82xx_modem_driver = {
 	.id_table = snd_via82xx_modem_ids,
 	.probe = snd_via82xx_probe,
 	.remove = __devexit_p(snd_via82xx_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_via82xx_suspend,
-	.resume = snd_via82xx_resume,
-#endif
+	.driver = {
+		.pm = SND_VIA82XX_PM_OPS,
+	},
 };
 
 module_pci_driver(via82xx_modem_driver);
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 1ea1f656a5dc..b89e7a86e9d8 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -258,22 +258,24 @@ static void __devexit snd_vx222_remove(struct pci_dev *pci)
 }
 
 #ifdef CONFIG_PM
-static int snd_vx222_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_vx222_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_vx222 *vx = card->private_data;
 	int err;
 
-	err = snd_vx_suspend(&vx->core, state);
+	err = snd_vx_suspend(&vx->core);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return err;
 }
 
-static int snd_vx222_resume(struct pci_dev *pci)
+static int snd_vx222_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_vx222 *vx = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -287,6 +289,11 @@ static int snd_vx222_resume(struct pci_dev *pci)
 	pci_set_master(pci);
 	return snd_vx_resume(&vx->core);
 }
+
+static SIMPLE_DEV_PM_OPS(snd_vx222_pm, snd_vx222_suspend, snd_vx222_resume);
+#define SND_VX222_PM_OPS	&snd_vx222_pm
+#else
+#define SND_VX222_PM_OPS	NULL
 #endif
 
 static struct pci_driver vx222_driver = {
@@ -294,10 +301,9 @@ static struct pci_driver vx222_driver = {
 	.id_table = snd_vx222_ids,
 	.probe = snd_vx222_probe,
 	.remove = __devexit_p(snd_vx222_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_vx222_suspend,
-	.resume = snd_vx222_resume,
-#endif
+	.driver = {
+		.pm = SND_VX222_PM_OPS,
+	},
 };
 
 module_pci_driver(vx222_driver);
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 9a1d01d653a7..7e20ddb9123a 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -356,8 +356,9 @@ static struct pci_driver ymfpci_driver = {
 	.probe = snd_card_ymfpci_probe,
 	.remove = __devexit_p(snd_card_ymfpci_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_ymfpci_suspend,
-	.resume = snd_ymfpci_resume,
+	.driver = {
+		.pm = &snd_ymfpci_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index a8159b81e9c4..c706901d6ff6 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -2302,9 +2302,10 @@ static int saved_regs_index[] = {
 };
 #define YDSXGR_NUM_SAVED_REGS	ARRAY_SIZE(saved_regs_index)
 
-int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_ymfpci_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ymfpci *chip = card->private_data;
 	unsigned int i;
 	
@@ -2326,13 +2327,14 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
 	snd_ymfpci_disable_dsp(chip);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_ymfpci_resume(struct pci_dev *pci)
+static int snd_ymfpci_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ymfpci *chip = card->private_data;
 	unsigned int i;
 
@@ -2370,6 +2372,8 @@ int snd_ymfpci_resume(struct pci_dev *pci)
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume);
 #endif /* CONFIG_PM */
 
 int __devinit snd_ymfpci_create(struct snd_card *card,
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 512f0b472375..8f9350475c7b 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -260,7 +260,7 @@ static int vxp_suspend(struct pcmcia_device *link)
 	snd_printdd(KERN_DEBUG "SUSPEND\n");
 	if (chip) {
 		snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n");
-		snd_vx_suspend(chip, PMSG_SUSPEND);
+		snd_vx_suspend(chip);
 	}
 
 	return 0;
-- 
cgit v1.2.3


From 2cb1331d9d647643a52be770377ab67ea078fd99 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Jul 2012 15:42:33 +0200
Subject: ALSA: pdaudiocf: Remove superfluous pm_message_t argument from
 suspend

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pcmcia/pdaudiocf/pdaudiocf.c      | 2 +-
 sound/pcmcia/pdaudiocf/pdaudiocf.h      | 2 +-
 sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 830839a874b6..f9b5229b2723 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -251,7 +251,7 @@ static int pdacf_suspend(struct pcmcia_device *link)
 	snd_printdd(KERN_DEBUG "SUSPEND\n");
 	if (chip) {
 		snd_printdd(KERN_DEBUG "snd_pdacf_suspend calling\n");
-		snd_pdacf_suspend(chip, PMSG_SUSPEND);
+		snd_pdacf_suspend(chip);
 	}
 
 	return 0;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index 6ce9ad700290..ea41e57d7179 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -131,7 +131,7 @@ struct snd_pdacf *snd_pdacf_create(struct snd_card *card);
 int snd_pdacf_ak4117_create(struct snd_pdacf *pdacf);
 void snd_pdacf_powerdown(struct snd_pdacf *chip);
 #ifdef CONFIG_PM
-int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state);
+int snd_pdacf_suspend(struct snd_pdacf *chip);
 int snd_pdacf_resume(struct snd_pdacf *chip);
 #endif
 int snd_pdacf_pcm_new(struct snd_pdacf *chip);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
index 9dce0bde5c05..ea0adfb984ad 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
@@ -262,7 +262,7 @@ void snd_pdacf_powerdown(struct snd_pdacf *chip)
 
 #ifdef CONFIG_PM
 
-int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state)
+int snd_pdacf_suspend(struct snd_pdacf *chip)
 {
 	u16 val;
 	
-- 
cgit v1.2.3


From 81fcb170852d58d7ebd8101a8ef970c82056426e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 2 Jul 2012 16:37:05 +0200
Subject: ALSA: Move some headers to local directories from include/sound

This is a bit clean up of public sound header directory.
Some header files in include/sound aren't really necessary to be
located there but can be moved to their local directories gracefully.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/cs46xx/cs46xx.c                |    2 +-
 sound/pci/cs46xx/cs46xx.h                | 1744 ++++++++++++++++++++++++++++++
 sound/pci/cs46xx/cs46xx_dsp_scb_types.h  | 1213 +++++++++++++++++++++
 sound/pci/cs46xx/cs46xx_dsp_spos.h       |  234 ++++
 sound/pci/cs46xx/cs46xx_dsp_task_types.h |  252 +++++
 sound/pci/cs46xx/cs46xx_lib.c            |    2 +-
 sound/pci/cs46xx/dsp_spos.c              |    2 +-
 sound/pci/cs46xx/dsp_spos_scb_lib.c      |    2 +-
 sound/pci/trident/trident.c              |    2 +-
 sound/pci/trident/trident.h              |  444 ++++++++
 sound/pci/trident/trident_main.c         |    2 +-
 sound/pci/trident/trident_memory.c       |    2 +-
 sound/pci/ymfpci/ymfpci.c                |    2 +-
 sound/pci/ymfpci/ymfpci.h                |  389 +++++++
 sound/pci/ymfpci/ymfpci_main.c           |    2 +-
 15 files changed, 4285 insertions(+), 9 deletions(-)
 create mode 100644 sound/pci/cs46xx/cs46xx.h
 create mode 100644 sound/pci/cs46xx/cs46xx_dsp_scb_types.h
 create mode 100644 sound/pci/cs46xx/cs46xx_dsp_spos.h
 create mode 100644 sound/pci/cs46xx/cs46xx_dsp_task_types.h
 create mode 100644 sound/pci/trident/trident.h
 create mode 100644 sound/pci/ymfpci/ymfpci.h

(limited to 'sound')

diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 00e03bc9a762..1e007c736a8b 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -30,7 +30,7 @@
 #include <linux/init.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 #include <sound/initval.h>
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
diff --git a/sound/pci/cs46xx/cs46xx.h b/sound/pci/cs46xx/cs46xx.h
new file mode 100644
index 000000000000..29d8a8da1ba7
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx.h
@@ -0,0 +1,1744 @@
+#ifndef __SOUND_CS46XX_H
+#define __SOUND_CS46XX_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *		     Cirrus Logic, Inc.
+ *  Definitions for Cirrus Logic CS46xx chips
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <sound/pcm.h>
+#include <sound/pcm-indirect.h>
+#include <sound/rawmidi.h>
+#include <sound/ac97_codec.h>
+#include "cs46xx_dsp_spos.h"
+
+/*
+ *  Direct registers
+ */
+
+/*
+ *  The following define the offsets of the registers accessed via base address
+ *  register zero on the CS46xx part.
+ */
+#define BA0_HISR				0x00000000
+#define BA0_HSR0                                0x00000004
+#define BA0_HICR                                0x00000008
+#define BA0_DMSR                                0x00000100
+#define BA0_HSAR                                0x00000110
+#define BA0_HDAR                                0x00000114
+#define BA0_HDMR                                0x00000118
+#define BA0_HDCR                                0x0000011C
+#define BA0_PFMC                                0x00000200
+#define BA0_PFCV1                               0x00000204
+#define BA0_PFCV2                               0x00000208
+#define BA0_PCICFG00                            0x00000300
+#define BA0_PCICFG04                            0x00000304
+#define BA0_PCICFG08                            0x00000308
+#define BA0_PCICFG0C                            0x0000030C
+#define BA0_PCICFG10                            0x00000310
+#define BA0_PCICFG14                            0x00000314
+#define BA0_PCICFG18                            0x00000318
+#define BA0_PCICFG1C                            0x0000031C
+#define BA0_PCICFG20                            0x00000320
+#define BA0_PCICFG24                            0x00000324
+#define BA0_PCICFG28                            0x00000328
+#define BA0_PCICFG2C                            0x0000032C
+#define BA0_PCICFG30                            0x00000330
+#define BA0_PCICFG34                            0x00000334
+#define BA0_PCICFG38                            0x00000338
+#define BA0_PCICFG3C                            0x0000033C
+#define BA0_CLKCR1                              0x00000400
+#define BA0_CLKCR2                              0x00000404
+#define BA0_PLLM                                0x00000408
+#define BA0_PLLCC                               0x0000040C
+#define BA0_FRR                                 0x00000410 
+#define BA0_CFL1                                0x00000414
+#define BA0_CFL2                                0x00000418
+#define BA0_SERMC1                              0x00000420
+#define BA0_SERMC2                              0x00000424
+#define BA0_SERC1                               0x00000428
+#define BA0_SERC2                               0x0000042C
+#define BA0_SERC3                               0x00000430
+#define BA0_SERC4                               0x00000434
+#define BA0_SERC5                               0x00000438
+#define BA0_SERBSP                              0x0000043C
+#define BA0_SERBST                              0x00000440
+#define BA0_SERBCM                              0x00000444
+#define BA0_SERBAD                              0x00000448
+#define BA0_SERBCF                              0x0000044C
+#define BA0_SERBWP                              0x00000450
+#define BA0_SERBRP                              0x00000454
+#ifndef NO_CS4612
+#define BA0_ASER_FADDR                          0x00000458
+#endif
+#define BA0_ACCTL                               0x00000460
+#define BA0_ACSTS                               0x00000464
+#define BA0_ACOSV                               0x00000468
+#define BA0_ACCAD                               0x0000046C
+#define BA0_ACCDA                               0x00000470
+#define BA0_ACISV                               0x00000474
+#define BA0_ACSAD                               0x00000478
+#define BA0_ACSDA                               0x0000047C
+#define BA0_JSPT                                0x00000480
+#define BA0_JSCTL                               0x00000484
+#define BA0_JSC1                                0x00000488
+#define BA0_JSC2                                0x0000048C
+#define BA0_MIDCR                               0x00000490
+#define BA0_MIDSR                               0x00000494
+#define BA0_MIDWP                               0x00000498
+#define BA0_MIDRP                               0x0000049C
+#define BA0_JSIO                                0x000004A0
+#ifndef NO_CS4612
+#define BA0_ASER_MASTER                         0x000004A4
+#endif
+#define BA0_CFGI                                0x000004B0
+#define BA0_SSVID                               0x000004B4
+#define BA0_GPIOR                               0x000004B8
+#ifndef NO_CS4612
+#define BA0_EGPIODR                             0x000004BC
+#define BA0_EGPIOPTR                            0x000004C0
+#define BA0_EGPIOTR                             0x000004C4
+#define BA0_EGPIOWR                             0x000004C8
+#define BA0_EGPIOSR                             0x000004CC
+#define BA0_SERC6                               0x000004D0
+#define BA0_SERC7                               0x000004D4
+#define BA0_SERACC                              0x000004D8
+#define BA0_ACCTL2                              0x000004E0
+#define BA0_ACSTS2                              0x000004E4
+#define BA0_ACOSV2                              0x000004E8
+#define BA0_ACCAD2                              0x000004EC
+#define BA0_ACCDA2                              0x000004F0
+#define BA0_ACISV2                              0x000004F4
+#define BA0_ACSAD2                              0x000004F8
+#define BA0_ACSDA2                              0x000004FC
+#define BA0_IOTAC0                              0x00000500
+#define BA0_IOTAC1                              0x00000504
+#define BA0_IOTAC2                              0x00000508
+#define BA0_IOTAC3                              0x0000050C
+#define BA0_IOTAC4                              0x00000510
+#define BA0_IOTAC5                              0x00000514
+#define BA0_IOTAC6                              0x00000518
+#define BA0_IOTAC7                              0x0000051C
+#define BA0_IOTAC8                              0x00000520
+#define BA0_IOTAC9                              0x00000524
+#define BA0_IOTAC10                             0x00000528
+#define BA0_IOTAC11                             0x0000052C
+#define BA0_IOTFR0                              0x00000540
+#define BA0_IOTFR1                              0x00000544
+#define BA0_IOTFR2                              0x00000548
+#define BA0_IOTFR3                              0x0000054C
+#define BA0_IOTFR4                              0x00000550
+#define BA0_IOTFR5                              0x00000554
+#define BA0_IOTFR6                              0x00000558
+#define BA0_IOTFR7                              0x0000055C
+#define BA0_IOTFIFO                             0x00000580
+#define BA0_IOTRRD                              0x00000584
+#define BA0_IOTFP                               0x00000588
+#define BA0_IOTCR                               0x0000058C
+#define BA0_DPCID                               0x00000590
+#define BA0_DPCIA                               0x00000594
+#define BA0_DPCIC                               0x00000598
+#define BA0_PCPCIR                              0x00000600
+#define BA0_PCPCIG                              0x00000604
+#define BA0_PCPCIEN                             0x00000608
+#define BA0_EPCIPMC                             0x00000610
+#endif
+
+/*
+ *  The following define the offsets of the registers and memories accessed via
+ *  base address register one on the CS46xx part.
+ */
+#define BA1_SP_DMEM0                            0x00000000
+#define BA1_SP_DMEM1                            0x00010000
+#define BA1_SP_PMEM                             0x00020000
+#define BA1_SP_REG				0x00030000
+#define BA1_SPCR                                0x00030000
+#define BA1_DREG                                0x00030004
+#define BA1_DSRWP                               0x00030008
+#define BA1_TWPR                                0x0003000C
+#define BA1_SPWR                                0x00030010
+#define BA1_SPIR                                0x00030014
+#define BA1_FGR1                                0x00030020
+#define BA1_SPCS                                0x00030028
+#define BA1_SDSR                                0x0003002C
+#define BA1_FRMT                                0x00030030
+#define BA1_FRCC                                0x00030034
+#define BA1_FRSC                                0x00030038
+#define BA1_OMNI_MEM                            0x000E0000
+
+
+/*
+ *  The following defines are for the flags in the host interrupt status
+ *  register.
+ */
+#define HISR_VC_MASK                            0x0000FFFF
+#define HISR_VC0                                0x00000001
+#define HISR_VC1                                0x00000002
+#define HISR_VC2                                0x00000004
+#define HISR_VC3                                0x00000008
+#define HISR_VC4                                0x00000010
+#define HISR_VC5                                0x00000020
+#define HISR_VC6                                0x00000040
+#define HISR_VC7                                0x00000080
+#define HISR_VC8                                0x00000100
+#define HISR_VC9                                0x00000200
+#define HISR_VC10                               0x00000400
+#define HISR_VC11                               0x00000800
+#define HISR_VC12                               0x00001000
+#define HISR_VC13                               0x00002000
+#define HISR_VC14                               0x00004000
+#define HISR_VC15                               0x00008000
+#define HISR_INT0                               0x00010000
+#define HISR_INT1                               0x00020000
+#define HISR_DMAI                               0x00040000
+#define HISR_FROVR                              0x00080000
+#define HISR_MIDI                               0x00100000
+#ifdef NO_CS4612
+#define HISR_RESERVED                           0x0FE00000
+#else
+#define HISR_SBINT                              0x00200000
+#define HISR_RESERVED                           0x0FC00000
+#endif
+#define HISR_H0P                                0x40000000
+#define HISR_INTENA                             0x80000000
+
+/*
+ *  The following defines are for the flags in the host signal register 0.
+ */
+#define HSR0_VC_MASK                            0xFFFFFFFF
+#define HSR0_VC16                               0x00000001
+#define HSR0_VC17                               0x00000002
+#define HSR0_VC18                               0x00000004
+#define HSR0_VC19                               0x00000008
+#define HSR0_VC20                               0x00000010
+#define HSR0_VC21                               0x00000020
+#define HSR0_VC22                               0x00000040
+#define HSR0_VC23                               0x00000080
+#define HSR0_VC24                               0x00000100
+#define HSR0_VC25                               0x00000200
+#define HSR0_VC26                               0x00000400
+#define HSR0_VC27                               0x00000800
+#define HSR0_VC28                               0x00001000
+#define HSR0_VC29                               0x00002000
+#define HSR0_VC30                               0x00004000
+#define HSR0_VC31                               0x00008000
+#define HSR0_VC32                               0x00010000
+#define HSR0_VC33                               0x00020000
+#define HSR0_VC34                               0x00040000
+#define HSR0_VC35                               0x00080000
+#define HSR0_VC36                               0x00100000
+#define HSR0_VC37                               0x00200000
+#define HSR0_VC38                               0x00400000
+#define HSR0_VC39                               0x00800000
+#define HSR0_VC40                               0x01000000
+#define HSR0_VC41                               0x02000000
+#define HSR0_VC42                               0x04000000
+#define HSR0_VC43                               0x08000000
+#define HSR0_VC44                               0x10000000
+#define HSR0_VC45                               0x20000000
+#define HSR0_VC46                               0x40000000
+#define HSR0_VC47                               0x80000000
+
+/*
+ *  The following defines are for the flags in the host interrupt control
+ *  register.
+ */
+#define HICR_IEV                                0x00000001
+#define HICR_CHGM                               0x00000002
+
+/*
+ *  The following defines are for the flags in the DMA status register.
+ */
+#define DMSR_HP                                 0x00000001
+#define DMSR_HR                                 0x00000002
+#define DMSR_SP                                 0x00000004
+#define DMSR_SR                                 0x00000008
+
+/*
+ *  The following defines are for the flags in the host DMA source address
+ *  register.
+ */
+#define HSAR_HOST_ADDR_MASK                     0xFFFFFFFF
+#define HSAR_DSP_ADDR_MASK                      0x0000FFFF
+#define HSAR_MEMID_MASK                         0x000F0000
+#define HSAR_MEMID_SP_DMEM0                     0x00000000
+#define HSAR_MEMID_SP_DMEM1                     0x00010000
+#define HSAR_MEMID_SP_PMEM                      0x00020000
+#define HSAR_MEMID_SP_DEBUG                     0x00030000
+#define HSAR_MEMID_OMNI_MEM                     0x000E0000
+#define HSAR_END                                0x40000000
+#define HSAR_ERR                                0x80000000
+
+/*
+ *  The following defines are for the flags in the host DMA destination address
+ *  register.
+ */
+#define HDAR_HOST_ADDR_MASK                     0xFFFFFFFF
+#define HDAR_DSP_ADDR_MASK                      0x0000FFFF
+#define HDAR_MEMID_MASK                         0x000F0000
+#define HDAR_MEMID_SP_DMEM0                     0x00000000
+#define HDAR_MEMID_SP_DMEM1                     0x00010000
+#define HDAR_MEMID_SP_PMEM                      0x00020000
+#define HDAR_MEMID_SP_DEBUG                     0x00030000
+#define HDAR_MEMID_OMNI_MEM                     0x000E0000
+#define HDAR_END                                0x40000000
+#define HDAR_ERR                                0x80000000
+
+/*
+ *  The following defines are for the flags in the host DMA control register.
+ */
+#define HDMR_AC_MASK                            0x0000F000
+#define HDMR_AC_8_16                            0x00001000
+#define HDMR_AC_M_S                             0x00002000
+#define HDMR_AC_B_L                             0x00004000
+#define HDMR_AC_S_U                             0x00008000
+
+/*
+ *  The following defines are for the flags in the host DMA control register.
+ */
+#define HDCR_COUNT_MASK                         0x000003FF
+#define HDCR_DONE                               0x00004000
+#define HDCR_OPT                                0x00008000
+#define HDCR_WBD                                0x00400000
+#define HDCR_WBS                                0x00800000
+#define HDCR_DMS_MASK                           0x07000000
+#define HDCR_DMS_LINEAR                         0x00000000
+#define HDCR_DMS_16_DWORDS                      0x01000000
+#define HDCR_DMS_32_DWORDS                      0x02000000
+#define HDCR_DMS_64_DWORDS                      0x03000000
+#define HDCR_DMS_128_DWORDS                     0x04000000
+#define HDCR_DMS_256_DWORDS                     0x05000000
+#define HDCR_DMS_512_DWORDS                     0x06000000
+#define HDCR_DMS_1024_DWORDS                    0x07000000
+#define HDCR_DH                                 0x08000000
+#define HDCR_SMS_MASK                           0x70000000
+#define HDCR_SMS_LINEAR                         0x00000000
+#define HDCR_SMS_16_DWORDS                      0x10000000
+#define HDCR_SMS_32_DWORDS                      0x20000000
+#define HDCR_SMS_64_DWORDS                      0x30000000
+#define HDCR_SMS_128_DWORDS                     0x40000000
+#define HDCR_SMS_256_DWORDS                     0x50000000
+#define HDCR_SMS_512_DWORDS                     0x60000000
+#define HDCR_SMS_1024_DWORDS                    0x70000000
+#define HDCR_SH                                 0x80000000
+#define HDCR_COUNT_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the performance monitor control
+ *  register.
+ */
+#define PFMC_C1SS_MASK                          0x0000001F
+#define PFMC_C1EV                               0x00000020
+#define PFMC_C1RS                               0x00008000
+#define PFMC_C2SS_MASK                          0x001F0000
+#define PFMC_C2EV                               0x00200000
+#define PFMC_C2RS                               0x80000000
+#define PFMC_C1SS_SHIFT                         0
+#define PFMC_C2SS_SHIFT                         16
+#define PFMC_BUS_GRANT                          0
+#define PFMC_GRANT_AFTER_REQ                    1
+#define PFMC_TRANSACTION                        2
+#define PFMC_DWORD_TRANSFER                     3
+#define PFMC_SLAVE_READ                         4
+#define PFMC_SLAVE_WRITE                        5
+#define PFMC_PREEMPTION                         6
+#define PFMC_DISCONNECT_RETRY                   7
+#define PFMC_INTERRUPT                          8
+#define PFMC_BUS_OWNERSHIP                      9
+#define PFMC_TRANSACTION_LAG                    10
+#define PFMC_PCI_CLOCK                          11
+#define PFMC_SERIAL_CLOCK                       12
+#define PFMC_SP_CLOCK                           13
+
+/*
+ *  The following defines are for the flags in the performance counter value 1
+ *  register.
+ */
+#define PFCV1_PC1V_MASK                         0xFFFFFFFF
+#define PFCV1_PC1V_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the performance counter value 2
+ *  register.
+ */
+#define PFCV2_PC2V_MASK                         0xFFFFFFFF
+#define PFCV2_PC2V_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the clock control register 1.
+ */
+#define CLKCR1_OSCS                             0x00000001
+#define CLKCR1_OSCP                             0x00000002
+#define CLKCR1_PLLSS_MASK                       0x0000000C
+#define CLKCR1_PLLSS_SERIAL                     0x00000000
+#define CLKCR1_PLLSS_CRYSTAL                    0x00000004
+#define CLKCR1_PLLSS_PCI                        0x00000008
+#define CLKCR1_PLLSS_RESERVED                   0x0000000C
+#define CLKCR1_PLLP                             0x00000010
+#define CLKCR1_SWCE                             0x00000020
+#define CLKCR1_PLLOS                            0x00000040
+
+/*
+ *  The following defines are for the flags in the clock control register 2.
+ */
+#define CLKCR2_PDIVS_MASK                       0x0000000F
+#define CLKCR2_PDIVS_1                          0x00000001
+#define CLKCR2_PDIVS_2                          0x00000002
+#define CLKCR2_PDIVS_4                          0x00000004
+#define CLKCR2_PDIVS_7                          0x00000007
+#define CLKCR2_PDIVS_8                          0x00000008
+#define CLKCR2_PDIVS_16                         0x00000000
+
+/*
+ *  The following defines are for the flags in the PLL multiplier register.
+ */
+#define PLLM_MASK                               0x000000FF
+#define PLLM_SHIFT                              0
+
+/*
+ *  The following defines are for the flags in the PLL capacitor coefficient
+ *  register.
+ */
+#define PLLCC_CDR_MASK                          0x00000007
+#ifndef NO_CS4610
+#define PLLCC_CDR_240_350_MHZ                   0x00000000
+#define PLLCC_CDR_184_265_MHZ                   0x00000001
+#define PLLCC_CDR_144_205_MHZ                   0x00000002
+#define PLLCC_CDR_111_160_MHZ                   0x00000003
+#define PLLCC_CDR_87_123_MHZ                    0x00000004
+#define PLLCC_CDR_67_96_MHZ                     0x00000005
+#define PLLCC_CDR_52_74_MHZ                     0x00000006
+#define PLLCC_CDR_45_58_MHZ                     0x00000007
+#endif
+#ifndef NO_CS4612
+#define PLLCC_CDR_271_398_MHZ                   0x00000000
+#define PLLCC_CDR_227_330_MHZ                   0x00000001
+#define PLLCC_CDR_167_239_MHZ                   0x00000002
+#define PLLCC_CDR_150_215_MHZ                   0x00000003
+#define PLLCC_CDR_107_154_MHZ                   0x00000004
+#define PLLCC_CDR_98_140_MHZ                    0x00000005
+#define PLLCC_CDR_73_104_MHZ                    0x00000006
+#define PLLCC_CDR_63_90_MHZ                     0x00000007
+#endif
+#define PLLCC_LPF_MASK                          0x000000F8
+#ifndef NO_CS4610
+#define PLLCC_LPF_23850_60000_KHZ               0x00000000
+#define PLLCC_LPF_7960_26290_KHZ                0x00000008
+#define PLLCC_LPF_4160_10980_KHZ                0x00000018
+#define PLLCC_LPF_1740_4580_KHZ                 0x00000038
+#define PLLCC_LPF_724_1910_KHZ                  0x00000078
+#define PLLCC_LPF_317_798_KHZ                   0x000000F8
+#endif
+#ifndef NO_CS4612
+#define PLLCC_LPF_25580_64530_KHZ               0x00000000
+#define PLLCC_LPF_14360_37270_KHZ               0x00000008
+#define PLLCC_LPF_6100_16020_KHZ                0x00000018
+#define PLLCC_LPF_2540_6690_KHZ                 0x00000038
+#define PLLCC_LPF_1050_2780_KHZ                 0x00000078
+#define PLLCC_LPF_450_1160_KHZ                  0x000000F8
+#endif
+
+/*
+ *  The following defines are for the flags in the feature reporting register.
+ */
+#define FRR_FAB_MASK                            0x00000003
+#define FRR_MASK_MASK                           0x0000001C
+#ifdef NO_CS4612
+#define FRR_CFOP_MASK                           0x000000E0
+#else
+#define FRR_CFOP_MASK                           0x00000FE0
+#endif
+#define FRR_CFOP_NOT_DVD                        0x00000020
+#define FRR_CFOP_A3D                            0x00000040
+#define FRR_CFOP_128_PIN                        0x00000080
+#ifndef NO_CS4612
+#define FRR_CFOP_CS4280                         0x00000800
+#endif
+#define FRR_FAB_SHIFT                           0
+#define FRR_MASK_SHIFT                          2
+#define FRR_CFOP_SHIFT                          5
+
+/*
+ *  The following defines are for the flags in the configuration load 1
+ *  register.
+ */
+#define CFL1_CLOCK_SOURCE_MASK                  0x00000003
+#define CFL1_CLOCK_SOURCE_CS423X                0x00000000
+#define CFL1_CLOCK_SOURCE_AC97                  0x00000001
+#define CFL1_CLOCK_SOURCE_CRYSTAL               0x00000002
+#define CFL1_CLOCK_SOURCE_DUAL_AC97             0x00000003
+#define CFL1_VALID_DATA_MASK                    0x000000FF
+
+/*
+ *  The following defines are for the flags in the configuration load 2
+ *  register.
+ */
+#define CFL2_VALID_DATA_MASK                    0x000000FF
+
+/*
+ *  The following defines are for the flags in the serial port master control
+ *  register 1.
+ */
+#define SERMC1_MSPE                             0x00000001
+#define SERMC1_PTC_MASK                         0x0000000E
+#define SERMC1_PTC_CS423X                       0x00000000
+#define SERMC1_PTC_AC97                         0x00000002
+#define SERMC1_PTC_DAC                          0x00000004
+#define SERMC1_PLB                              0x00000010
+#define SERMC1_XLB                              0x00000020
+
+/*
+ *  The following defines are for the flags in the serial port master control
+ *  register 2.
+ */
+#define SERMC2_LROE                             0x00000001
+#define SERMC2_MCOE                             0x00000002
+#define SERMC2_MCDIV                            0x00000004
+
+/*
+ *  The following defines are for the flags in the serial port 1 configuration
+ *  register.
+ */
+#define SERC1_SO1EN                             0x00000001
+#define SERC1_SO1F_MASK                         0x0000000E
+#define SERC1_SO1F_CS423X                       0x00000000
+#define SERC1_SO1F_AC97                         0x00000002
+#define SERC1_SO1F_DAC                          0x00000004
+#define SERC1_SO1F_SPDIF                        0x00000006
+
+/*
+ *  The following defines are for the flags in the serial port 2 configuration
+ *  register.
+ */
+#define SERC2_SI1EN                             0x00000001
+#define SERC2_SI1F_MASK                         0x0000000E
+#define SERC2_SI1F_CS423X                       0x00000000
+#define SERC2_SI1F_AC97                         0x00000002
+#define SERC2_SI1F_ADC                          0x00000004
+#define SERC2_SI1F_SPDIF                        0x00000006
+
+/*
+ *  The following defines are for the flags in the serial port 3 configuration
+ *  register.
+ */
+#define SERC3_SO2EN                             0x00000001
+#define SERC3_SO2F_MASK                         0x00000006
+#define SERC3_SO2F_DAC                          0x00000000
+#define SERC3_SO2F_SPDIF                        0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port 4 configuration
+ *  register.
+ */
+#define SERC4_SO3EN                             0x00000001
+#define SERC4_SO3F_MASK                         0x00000006
+#define SERC4_SO3F_DAC                          0x00000000
+#define SERC4_SO3F_SPDIF                        0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port 5 configuration
+ *  register.
+ */
+#define SERC5_SI2EN                             0x00000001
+#define SERC5_SI2F_MASK                         0x00000006
+#define SERC5_SI2F_ADC                          0x00000000
+#define SERC5_SI2F_SPDIF                        0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port backdoor sample
+ *  pointer register.
+ */
+#define SERBSP_FSP_MASK                         0x0000000F
+#define SERBSP_FSP_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the serial port backdoor status
+ *  register.
+ */
+#define SERBST_RRDY                             0x00000001
+#define SERBST_WBSY                             0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port backdoor command
+ *  register.
+ */
+#define SERBCM_RDC                              0x00000001
+#define SERBCM_WRC                              0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port backdoor address
+ *  register.
+ */
+#ifdef NO_CS4612
+#define SERBAD_FAD_MASK                         0x000000FF
+#else
+#define SERBAD_FAD_MASK                         0x000001FF
+#endif
+#define SERBAD_FAD_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the serial port backdoor
+ *  configuration register.
+ */
+#define SERBCF_HBP                              0x00000001
+
+/*
+ *  The following defines are for the flags in the serial port backdoor write
+ *  port register.
+ */
+#define SERBWP_FWD_MASK                         0x000FFFFF
+#define SERBWP_FWD_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the serial port backdoor read
+ *  port register.
+ */
+#define SERBRP_FRD_MASK                         0x000FFFFF
+#define SERBRP_FRD_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the async FIFO address register.
+ */
+#ifndef NO_CS4612
+#define ASER_FADDR_A1_MASK                      0x000001FF
+#define ASER_FADDR_EN1                          0x00008000
+#define ASER_FADDR_A2_MASK                      0x01FF0000
+#define ASER_FADDR_EN2                          0x80000000
+#define ASER_FADDR_A1_SHIFT                     0
+#define ASER_FADDR_A2_SHIFT                     16
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 control register.
+ */
+#define ACCTL_RSTN                              0x00000001
+#define ACCTL_ESYN                              0x00000002
+#define ACCTL_VFRM                              0x00000004
+#define ACCTL_DCV                               0x00000008
+#define ACCTL_CRW                               0x00000010
+#define ACCTL_ASYN                              0x00000020
+#ifndef NO_CS4612
+#define ACCTL_TC                                0x00000040
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status register.
+ */
+#define ACSTS_CRDY                              0x00000001
+#define ACSTS_VSTS                              0x00000002
+#ifndef NO_CS4612
+#define ACSTS_WKUP                              0x00000004
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 output slot valid
+ *  register.
+ */
+#define ACOSV_SLV3                              0x00000001
+#define ACOSV_SLV4                              0x00000002
+#define ACOSV_SLV5                              0x00000004
+#define ACOSV_SLV6                              0x00000008
+#define ACOSV_SLV7                              0x00000010
+#define ACOSV_SLV8                              0x00000020
+#define ACOSV_SLV9                              0x00000040
+#define ACOSV_SLV10                             0x00000080
+#define ACOSV_SLV11                             0x00000100
+#define ACOSV_SLV12                             0x00000200
+
+/*
+ *  The following defines are for the flags in the AC97 command address
+ *  register.
+ */
+#define ACCAD_CI_MASK                           0x0000007F
+#define ACCAD_CI_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the AC97 command data register.
+ */
+#define ACCDA_CD_MASK                           0x0000FFFF
+#define ACCDA_CD_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the AC97 input slot valid
+ *  register.
+ */
+#define ACISV_ISV3                              0x00000001
+#define ACISV_ISV4                              0x00000002
+#define ACISV_ISV5                              0x00000004
+#define ACISV_ISV6                              0x00000008
+#define ACISV_ISV7                              0x00000010
+#define ACISV_ISV8                              0x00000020
+#define ACISV_ISV9                              0x00000040
+#define ACISV_ISV10                             0x00000080
+#define ACISV_ISV11                             0x00000100
+#define ACISV_ISV12                             0x00000200
+
+/*
+ *  The following defines are for the flags in the AC97 status address
+ *  register.
+ */
+#define ACSAD_SI_MASK                           0x0000007F
+#define ACSAD_SI_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the AC97 status data register.
+ */
+#define ACSDA_SD_MASK                           0x0000FFFF
+#define ACSDA_SD_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the joystick poll/trigger
+ *  register.
+ */
+#define JSPT_CAX                                0x00000001
+#define JSPT_CAY                                0x00000002
+#define JSPT_CBX                                0x00000004
+#define JSPT_CBY                                0x00000008
+#define JSPT_BA1                                0x00000010
+#define JSPT_BA2                                0x00000020
+#define JSPT_BB1                                0x00000040
+#define JSPT_BB2                                0x00000080
+
+/*
+ *  The following defines are for the flags in the joystick control register.
+ */
+#define JSCTL_SP_MASK                           0x00000003
+#define JSCTL_SP_SLOW                           0x00000000
+#define JSCTL_SP_MEDIUM_SLOW                    0x00000001
+#define JSCTL_SP_MEDIUM_FAST                    0x00000002
+#define JSCTL_SP_FAST                           0x00000003
+#define JSCTL_ARE                               0x00000004
+
+/*
+ *  The following defines are for the flags in the joystick coordinate pair 1
+ *  readback register.
+ */
+#define JSC1_Y1V_MASK                           0x0000FFFF
+#define JSC1_X1V_MASK                           0xFFFF0000
+#define JSC1_Y1V_SHIFT                          0
+#define JSC1_X1V_SHIFT                          16
+
+/*
+ *  The following defines are for the flags in the joystick coordinate pair 2
+ *  readback register.
+ */
+#define JSC2_Y2V_MASK                           0x0000FFFF
+#define JSC2_X2V_MASK                           0xFFFF0000
+#define JSC2_Y2V_SHIFT                          0
+#define JSC2_X2V_SHIFT                          16
+
+/*
+ *  The following defines are for the flags in the MIDI control register.
+ */
+#define MIDCR_TXE                               0x00000001	/* Enable transmitting. */
+#define MIDCR_RXE                               0x00000002	/* Enable receiving. */
+#define MIDCR_RIE                               0x00000004	/* Interrupt upon tx ready. */
+#define MIDCR_TIE                               0x00000008	/* Interrupt upon rx ready. */
+#define MIDCR_MLB                               0x00000010	/* Enable midi loopback. */
+#define MIDCR_MRST                              0x00000020	/* Reset interface. */
+
+/*
+ *  The following defines are for the flags in the MIDI status register.
+ */
+#define MIDSR_TBF                               0x00000001	/* Tx FIFO is full. */
+#define MIDSR_RBE                               0x00000002	/* Rx FIFO is empty. */
+
+/*
+ *  The following defines are for the flags in the MIDI write port register.
+ */
+#define MIDWP_MWD_MASK                          0x000000FF
+#define MIDWP_MWD_SHIFT                         0
+
+/*
+ *  The following defines are for the flags in the MIDI read port register.
+ */
+#define MIDRP_MRD_MASK                          0x000000FF
+#define MIDRP_MRD_SHIFT                         0
+
+/*
+ *  The following defines are for the flags in the joystick GPIO register.
+ */
+#define JSIO_DAX                                0x00000001
+#define JSIO_DAY                                0x00000002
+#define JSIO_DBX                                0x00000004
+#define JSIO_DBY                                0x00000008
+#define JSIO_AXOE                               0x00000010
+#define JSIO_AYOE                               0x00000020
+#define JSIO_BXOE                               0x00000040
+#define JSIO_BYOE                               0x00000080
+
+/*
+ *  The following defines are for the flags in the master async/sync serial
+ *  port enable register.
+ */
+#ifndef NO_CS4612
+#define ASER_MASTER_ME                          0x00000001
+#endif
+
+/*
+ *  The following defines are for the flags in the configuration interface
+ *  register.
+ */
+#define CFGI_CLK                                0x00000001
+#define CFGI_DOUT                               0x00000002
+#define CFGI_DIN_EEN                            0x00000004
+#define CFGI_EELD                               0x00000008
+
+/*
+ *  The following defines are for the flags in the subsystem ID and vendor ID
+ *  register.
+ */
+#define SSVID_VID_MASK                          0x0000FFFF
+#define SSVID_SID_MASK                          0xFFFF0000
+#define SSVID_VID_SHIFT                         0
+#define SSVID_SID_SHIFT                         16
+
+/*
+ *  The following defines are for the flags in the GPIO pin interface register.
+ */
+#define GPIOR_VOLDN                             0x00000001
+#define GPIOR_VOLUP                             0x00000002
+#define GPIOR_SI2D                              0x00000004
+#define GPIOR_SI2OE                             0x00000008
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin direction
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIODR_GPOE0                           0x00000001
+#define EGPIODR_GPOE1                           0x00000002
+#define EGPIODR_GPOE2                           0x00000004
+#define EGPIODR_GPOE3                           0x00000008
+#define EGPIODR_GPOE4                           0x00000010
+#define EGPIODR_GPOE5                           0x00000020
+#define EGPIODR_GPOE6                           0x00000040
+#define EGPIODR_GPOE7                           0x00000080
+#define EGPIODR_GPOE8                           0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin polarity/
+ *  type register.
+ */
+#ifndef NO_CS4612
+#define EGPIOPTR_GPPT0                          0x00000001
+#define EGPIOPTR_GPPT1                          0x00000002
+#define EGPIOPTR_GPPT2                          0x00000004
+#define EGPIOPTR_GPPT3                          0x00000008
+#define EGPIOPTR_GPPT4                          0x00000010
+#define EGPIOPTR_GPPT5                          0x00000020
+#define EGPIOPTR_GPPT6                          0x00000040
+#define EGPIOPTR_GPPT7                          0x00000080
+#define EGPIOPTR_GPPT8                          0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin sticky
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIOTR_GPS0                            0x00000001
+#define EGPIOTR_GPS1                            0x00000002
+#define EGPIOTR_GPS2                            0x00000004
+#define EGPIOTR_GPS3                            0x00000008
+#define EGPIOTR_GPS4                            0x00000010
+#define EGPIOTR_GPS5                            0x00000020
+#define EGPIOTR_GPS6                            0x00000040
+#define EGPIOTR_GPS7                            0x00000080
+#define EGPIOTR_GPS8                            0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO ping wakeup
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIOWR_GPW0                            0x00000001
+#define EGPIOWR_GPW1                            0x00000002
+#define EGPIOWR_GPW2                            0x00000004
+#define EGPIOWR_GPW3                            0x00000008
+#define EGPIOWR_GPW4                            0x00000010
+#define EGPIOWR_GPW5                            0x00000020
+#define EGPIOWR_GPW6                            0x00000040
+#define EGPIOWR_GPW7                            0x00000080
+#define EGPIOWR_GPW8                            0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin status
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIOSR_GPS0                            0x00000001
+#define EGPIOSR_GPS1                            0x00000002
+#define EGPIOSR_GPS2                            0x00000004
+#define EGPIOSR_GPS3                            0x00000008
+#define EGPIOSR_GPS4                            0x00000010
+#define EGPIOSR_GPS5                            0x00000020
+#define EGPIOSR_GPS6                            0x00000040
+#define EGPIOSR_GPS7                            0x00000080
+#define EGPIOSR_GPS8                            0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the serial port 6 configuration
+ *  register.
+ */
+#ifndef NO_CS4612
+#define SERC6_ASDO2EN                           0x00000001
+#endif
+
+/*
+ *  The following defines are for the flags in the serial port 7 configuration
+ *  register.
+ */
+#ifndef NO_CS4612
+#define SERC7_ASDI2EN                           0x00000001
+#define SERC7_POSILB                            0x00000002
+#define SERC7_SIPOLB                            0x00000004
+#define SERC7_SOSILB                            0x00000008
+#define SERC7_SISOLB                            0x00000010
+#endif
+
+/*
+ *  The following defines are for the flags in the serial port AC link
+ *  configuration register.
+ */
+#ifndef NO_CS4612
+#define SERACC_CHIP_TYPE_MASK                  0x00000001
+#define SERACC_CHIP_TYPE_1_03                  0x00000000
+#define SERACC_CHIP_TYPE_2_0                   0x00000001
+#define SERACC_TWO_CODECS                      0x00000002
+#define SERACC_MDM                             0x00000004
+#define SERACC_HSP                             0x00000008
+#define SERACC_ODT                             0x00000010 /* only CS4630 */
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 control register 2.
+ */
+#ifndef NO_CS4612
+#define ACCTL2_RSTN                             0x00000001
+#define ACCTL2_ESYN                             0x00000002
+#define ACCTL2_VFRM                             0x00000004
+#define ACCTL2_DCV                              0x00000008
+#define ACCTL2_CRW                              0x00000010
+#define ACCTL2_ASYN                             0x00000020
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status register 2.
+ */
+#ifndef NO_CS4612
+#define ACSTS2_CRDY                             0x00000001
+#define ACSTS2_VSTS                             0x00000002
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 output slot valid
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACOSV2_SLV3                             0x00000001
+#define ACOSV2_SLV4                             0x00000002
+#define ACOSV2_SLV5                             0x00000004
+#define ACOSV2_SLV6                             0x00000008
+#define ACOSV2_SLV7                             0x00000010
+#define ACOSV2_SLV8                             0x00000020
+#define ACOSV2_SLV9                             0x00000040
+#define ACOSV2_SLV10                            0x00000080
+#define ACOSV2_SLV11                            0x00000100
+#define ACOSV2_SLV12                            0x00000200
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 command address
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACCAD2_CI_MASK                          0x0000007F
+#define ACCAD2_CI_SHIFT                         0
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 command data register
+ *  2.
+ */
+#ifndef NO_CS4612
+#define ACCDA2_CD_MASK                          0x0000FFFF
+#define ACCDA2_CD_SHIFT                         0  
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 input slot valid
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACISV2_ISV3                             0x00000001
+#define ACISV2_ISV4                             0x00000002
+#define ACISV2_ISV5                             0x00000004
+#define ACISV2_ISV6                             0x00000008
+#define ACISV2_ISV7                             0x00000010
+#define ACISV2_ISV8                             0x00000020
+#define ACISV2_ISV9                             0x00000040
+#define ACISV2_ISV10                            0x00000080
+#define ACISV2_ISV11                            0x00000100
+#define ACISV2_ISV12                            0x00000200
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status address
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACSAD2_SI_MASK                          0x0000007F
+#define ACSAD2_SI_SHIFT                         0
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status data register 2.
+ */
+#ifndef NO_CS4612
+#define ACSDA2_SD_MASK                          0x0000FFFF
+#define ACSDA2_SD_SHIFT                         0
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap address and control
+ *  registers (all 12).
+ */
+#ifndef NO_CS4612
+#define IOTAC_SA_MASK                           0x0000FFFF
+#define IOTAC_MSK_MASK                          0x000F0000
+#define IOTAC_IODC_MASK                         0x06000000
+#define IOTAC_IODC_16_BIT                       0x00000000
+#define IOTAC_IODC_10_BIT                       0x02000000
+#define IOTAC_IODC_12_BIT                       0x04000000
+#define IOTAC_WSPI                              0x08000000
+#define IOTAC_RSPI                              0x10000000
+#define IOTAC_WSE                               0x20000000
+#define IOTAC_WE                                0x40000000
+#define IOTAC_RE                                0x80000000
+#define IOTAC_SA_SHIFT                          0
+#define IOTAC_MSK_SHIFT                         16
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap fast read registers
+ *  (all 8).
+ */
+#ifndef NO_CS4612
+#define IOTFR_D_MASK                            0x0000FFFF
+#define IOTFR_A_MASK                            0x000F0000
+#define IOTFR_R_MASK                            0x0F000000
+#define IOTFR_ALL                               0x40000000
+#define IOTFR_VL                                0x80000000
+#define IOTFR_D_SHIFT                           0
+#define IOTFR_A_SHIFT                           16
+#define IOTFR_R_SHIFT                           24
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap FIFO register.
+ */
+#ifndef NO_CS4612
+#define IOTFIFO_BA_MASK                         0x00003FFF
+#define IOTFIFO_S_MASK                          0x00FF0000
+#define IOTFIFO_OF                              0x40000000
+#define IOTFIFO_SPIOF                           0x80000000
+#define IOTFIFO_BA_SHIFT                        0
+#define IOTFIFO_S_SHIFT                         16
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap retry read data
+ *  register.
+ */
+#ifndef NO_CS4612
+#define IOTRRD_D_MASK                           0x0000FFFF
+#define IOTRRD_RDV                              0x80000000
+#define IOTRRD_D_SHIFT                          0
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap FIFO pointer
+ *  register.
+ */
+#ifndef NO_CS4612
+#define IOTFP_CA_MASK                           0x00003FFF
+#define IOTFP_PA_MASK                           0x3FFF0000
+#define IOTFP_CA_SHIFT                          0
+#define IOTFP_PA_SHIFT                          16
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap control register.
+ */
+#ifndef NO_CS4612
+#define IOTCR_ITD                               0x00000001
+#define IOTCR_HRV                               0x00000002
+#define IOTCR_SRV                               0x00000004
+#define IOTCR_DTI                               0x00000008
+#define IOTCR_DFI                               0x00000010
+#define IOTCR_DDP                               0x00000020
+#define IOTCR_JTE                               0x00000040
+#define IOTCR_PPE                               0x00000080
+#endif
+
+/*
+ *  The following defines are for the flags in the direct PCI data register.
+ */
+#ifndef NO_CS4612
+#define DPCID_D_MASK                            0xFFFFFFFF
+#define DPCID_D_SHIFT                           0
+#endif
+
+/*
+ *  The following defines are for the flags in the direct PCI address register.
+ */
+#ifndef NO_CS4612
+#define DPCIA_A_MASK                            0xFFFFFFFF
+#define DPCIA_A_SHIFT                           0
+#endif
+
+/*
+ *  The following defines are for the flags in the direct PCI command register.
+ */
+#ifndef NO_CS4612
+#define DPCIC_C_MASK                            0x0000000F
+#define DPCIC_C_IOREAD                          0x00000002
+#define DPCIC_C_IOWRITE                         0x00000003
+#define DPCIC_BE_MASK                           0x000000F0
+#endif
+
+/*
+ *  The following defines are for the flags in the PC/PCI request register.
+ */
+#ifndef NO_CS4612
+#define PCPCIR_RDC_MASK                         0x00000007
+#define PCPCIR_C_MASK                           0x00007000
+#define PCPCIR_REQ                              0x00008000
+#define PCPCIR_RDC_SHIFT                        0
+#define PCPCIR_C_SHIFT                          12
+#endif
+
+/*
+ *  The following defines are for the flags in the PC/PCI grant register.
+ */ 
+#ifndef NO_CS4612
+#define PCPCIG_GDC_MASK                         0x00000007
+#define PCPCIG_VL                               0x00008000
+#define PCPCIG_GDC_SHIFT                        0
+#endif
+
+/*
+ *  The following defines are for the flags in the PC/PCI master enable
+ *  register.
+ */
+#ifndef NO_CS4612
+#define PCPCIEN_EN                              0x00000001
+#endif
+
+/*
+ *  The following defines are for the flags in the extended PCI power
+ *  management control register.
+ */
+#ifndef NO_CS4612
+#define EPCIPMC_GWU                             0x00000001
+#define EPCIPMC_FSPC                            0x00000002
+#endif 
+
+/*
+ *  The following defines are for the flags in the SP control register.
+ */
+#define SPCR_RUN                                0x00000001
+#define SPCR_STPFR                              0x00000002
+#define SPCR_RUNFR                              0x00000004
+#define SPCR_TICK                               0x00000008
+#define SPCR_DRQEN                              0x00000020
+#define SPCR_RSTSP                              0x00000040
+#define SPCR_OREN                               0x00000080
+#ifndef NO_CS4612
+#define SPCR_PCIINT                             0x00000100
+#define SPCR_OINTD                              0x00000200
+#define SPCR_CRE                                0x00008000
+#endif
+
+/*
+ *  The following defines are for the flags in the debug index register.
+ */
+#define DREG_REGID_MASK                         0x0000007F
+#define DREG_DEBUG                              0x00000080
+#define DREG_RGBK_MASK                          0x00000700
+#define DREG_TRAP                               0x00000800
+#if !defined(NO_CS4612)
+#if !defined(NO_CS4615)
+#define DREG_TRAPX                              0x00001000
+#endif
+#endif
+#define DREG_REGID_SHIFT                        0
+#define DREG_RGBK_SHIFT                         8
+#define DREG_RGBK_REGID_MASK                    0x0000077F
+#define DREG_REGID_R0                           0x00000010
+#define DREG_REGID_R1                           0x00000011
+#define DREG_REGID_R2                           0x00000012
+#define DREG_REGID_R3                           0x00000013
+#define DREG_REGID_R4                           0x00000014
+#define DREG_REGID_R5                           0x00000015
+#define DREG_REGID_R6                           0x00000016
+#define DREG_REGID_R7                           0x00000017
+#define DREG_REGID_R8                           0x00000018
+#define DREG_REGID_R9                           0x00000019
+#define DREG_REGID_RA                           0x0000001A
+#define DREG_REGID_RB                           0x0000001B
+#define DREG_REGID_RC                           0x0000001C
+#define DREG_REGID_RD                           0x0000001D
+#define DREG_REGID_RE                           0x0000001E
+#define DREG_REGID_RF                           0x0000001F
+#define DREG_REGID_RA_BUS_LOW                   0x00000020
+#define DREG_REGID_RA_BUS_HIGH                  0x00000038
+#define DREG_REGID_YBUS_LOW                     0x00000050
+#define DREG_REGID_YBUS_HIGH                    0x00000058
+#define DREG_REGID_TRAP_0                       0x00000100
+#define DREG_REGID_TRAP_1                       0x00000101
+#define DREG_REGID_TRAP_2                       0x00000102
+#define DREG_REGID_TRAP_3                       0x00000103
+#define DREG_REGID_TRAP_4                       0x00000104
+#define DREG_REGID_TRAP_5                       0x00000105
+#define DREG_REGID_TRAP_6                       0x00000106
+#define DREG_REGID_TRAP_7                       0x00000107
+#define DREG_REGID_INDIRECT_ADDRESS             0x0000010E
+#define DREG_REGID_TOP_OF_STACK                 0x0000010F
+#if !defined(NO_CS4612)
+#if !defined(NO_CS4615)
+#define DREG_REGID_TRAP_8                       0x00000110
+#define DREG_REGID_TRAP_9                       0x00000111
+#define DREG_REGID_TRAP_10                      0x00000112
+#define DREG_REGID_TRAP_11                      0x00000113
+#define DREG_REGID_TRAP_12                      0x00000114
+#define DREG_REGID_TRAP_13                      0x00000115
+#define DREG_REGID_TRAP_14                      0x00000116
+#define DREG_REGID_TRAP_15                      0x00000117
+#define DREG_REGID_TRAP_16                      0x00000118
+#define DREG_REGID_TRAP_17                      0x00000119
+#define DREG_REGID_TRAP_18                      0x0000011A
+#define DREG_REGID_TRAP_19                      0x0000011B
+#define DREG_REGID_TRAP_20                      0x0000011C
+#define DREG_REGID_TRAP_21                      0x0000011D
+#define DREG_REGID_TRAP_22                      0x0000011E
+#define DREG_REGID_TRAP_23                      0x0000011F
+#endif
+#endif
+#define DREG_REGID_RSA0_LOW                     0x00000200
+#define DREG_REGID_RSA0_HIGH                    0x00000201
+#define DREG_REGID_RSA1_LOW                     0x00000202
+#define DREG_REGID_RSA1_HIGH                    0x00000203
+#define DREG_REGID_RSA2                         0x00000204
+#define DREG_REGID_RSA3                         0x00000205
+#define DREG_REGID_RSI0_LOW                     0x00000206
+#define DREG_REGID_RSI0_HIGH                    0x00000207
+#define DREG_REGID_RSI1                         0x00000208
+#define DREG_REGID_RSI2                         0x00000209
+#define DREG_REGID_SAGUSTATUS                   0x0000020A
+#define DREG_REGID_RSCONFIG01_LOW               0x0000020B
+#define DREG_REGID_RSCONFIG01_HIGH              0x0000020C
+#define DREG_REGID_RSCONFIG23_LOW               0x0000020D
+#define DREG_REGID_RSCONFIG23_HIGH              0x0000020E
+#define DREG_REGID_RSDMA01E                     0x0000020F
+#define DREG_REGID_RSDMA23E                     0x00000210
+#define DREG_REGID_RSD0_LOW                     0x00000211
+#define DREG_REGID_RSD0_HIGH                    0x00000212
+#define DREG_REGID_RSD1_LOW                     0x00000213
+#define DREG_REGID_RSD1_HIGH                    0x00000214
+#define DREG_REGID_RSD2_LOW                     0x00000215
+#define DREG_REGID_RSD2_HIGH                    0x00000216
+#define DREG_REGID_RSD3_LOW                     0x00000217
+#define DREG_REGID_RSD3_HIGH                    0x00000218
+#define DREG_REGID_SRAR_HIGH                    0x0000021A
+#define DREG_REGID_SRAR_LOW                     0x0000021B
+#define DREG_REGID_DMA_STATE                    0x0000021C
+#define DREG_REGID_CURRENT_DMA_STREAM           0x0000021D
+#define DREG_REGID_NEXT_DMA_STREAM              0x0000021E
+#define DREG_REGID_CPU_STATUS                   0x00000300
+#define DREG_REGID_MAC_MODE                     0x00000301
+#define DREG_REGID_STACK_AND_REPEAT             0x00000302
+#define DREG_REGID_INDEX0                       0x00000304
+#define DREG_REGID_INDEX1                       0x00000305
+#define DREG_REGID_DMA_STATE_0_3                0x00000400
+#define DREG_REGID_DMA_STATE_4_7                0x00000404
+#define DREG_REGID_DMA_STATE_8_11               0x00000408
+#define DREG_REGID_DMA_STATE_12_15              0x0000040C
+#define DREG_REGID_DMA_STATE_16_19              0x00000410
+#define DREG_REGID_DMA_STATE_20_23              0x00000414
+#define DREG_REGID_DMA_STATE_24_27              0x00000418
+#define DREG_REGID_DMA_STATE_28_31              0x0000041C
+#define DREG_REGID_DMA_STATE_32_35              0x00000420
+#define DREG_REGID_DMA_STATE_36_39              0x00000424
+#define DREG_REGID_DMA_STATE_40_43              0x00000428
+#define DREG_REGID_DMA_STATE_44_47              0x0000042C
+#define DREG_REGID_DMA_STATE_48_51              0x00000430
+#define DREG_REGID_DMA_STATE_52_55              0x00000434
+#define DREG_REGID_DMA_STATE_56_59              0x00000438
+#define DREG_REGID_DMA_STATE_60_63              0x0000043C
+#define DREG_REGID_DMA_STATE_64_67              0x00000440
+#define DREG_REGID_DMA_STATE_68_71              0x00000444
+#define DREG_REGID_DMA_STATE_72_75              0x00000448
+#define DREG_REGID_DMA_STATE_76_79              0x0000044C
+#define DREG_REGID_DMA_STATE_80_83              0x00000450
+#define DREG_REGID_DMA_STATE_84_87              0x00000454
+#define DREG_REGID_DMA_STATE_88_91              0x00000458
+#define DREG_REGID_DMA_STATE_92_95              0x0000045C
+#define DREG_REGID_TRAP_SELECT                  0x00000500
+#define DREG_REGID_TRAP_WRITE_0                 0x00000500
+#define DREG_REGID_TRAP_WRITE_1                 0x00000501
+#define DREG_REGID_TRAP_WRITE_2                 0x00000502
+#define DREG_REGID_TRAP_WRITE_3                 0x00000503
+#define DREG_REGID_TRAP_WRITE_4                 0x00000504
+#define DREG_REGID_TRAP_WRITE_5                 0x00000505
+#define DREG_REGID_TRAP_WRITE_6                 0x00000506
+#define DREG_REGID_TRAP_WRITE_7                 0x00000507
+#if !defined(NO_CS4612)
+#if !defined(NO_CS4615)
+#define DREG_REGID_TRAP_WRITE_8                 0x00000510
+#define DREG_REGID_TRAP_WRITE_9                 0x00000511
+#define DREG_REGID_TRAP_WRITE_10                0x00000512
+#define DREG_REGID_TRAP_WRITE_11                0x00000513
+#define DREG_REGID_TRAP_WRITE_12                0x00000514
+#define DREG_REGID_TRAP_WRITE_13                0x00000515
+#define DREG_REGID_TRAP_WRITE_14                0x00000516
+#define DREG_REGID_TRAP_WRITE_15                0x00000517
+#define DREG_REGID_TRAP_WRITE_16                0x00000518
+#define DREG_REGID_TRAP_WRITE_17                0x00000519
+#define DREG_REGID_TRAP_WRITE_18                0x0000051A
+#define DREG_REGID_TRAP_WRITE_19                0x0000051B
+#define DREG_REGID_TRAP_WRITE_20                0x0000051C
+#define DREG_REGID_TRAP_WRITE_21                0x0000051D
+#define DREG_REGID_TRAP_WRITE_22                0x0000051E
+#define DREG_REGID_TRAP_WRITE_23                0x0000051F
+#endif
+#endif
+#define DREG_REGID_MAC0_ACC0_LOW                0x00000600
+#define DREG_REGID_MAC0_ACC1_LOW                0x00000601
+#define DREG_REGID_MAC0_ACC2_LOW                0x00000602
+#define DREG_REGID_MAC0_ACC3_LOW                0x00000603
+#define DREG_REGID_MAC1_ACC0_LOW                0x00000604
+#define DREG_REGID_MAC1_ACC1_LOW                0x00000605
+#define DREG_REGID_MAC1_ACC2_LOW                0x00000606
+#define DREG_REGID_MAC1_ACC3_LOW                0x00000607
+#define DREG_REGID_MAC0_ACC0_MID                0x00000608
+#define DREG_REGID_MAC0_ACC1_MID                0x00000609
+#define DREG_REGID_MAC0_ACC2_MID                0x0000060A
+#define DREG_REGID_MAC0_ACC3_MID                0x0000060B
+#define DREG_REGID_MAC1_ACC0_MID                0x0000060C
+#define DREG_REGID_MAC1_ACC1_MID                0x0000060D
+#define DREG_REGID_MAC1_ACC2_MID                0x0000060E
+#define DREG_REGID_MAC1_ACC3_MID                0x0000060F
+#define DREG_REGID_MAC0_ACC0_HIGH               0x00000610
+#define DREG_REGID_MAC0_ACC1_HIGH               0x00000611
+#define DREG_REGID_MAC0_ACC2_HIGH               0x00000612
+#define DREG_REGID_MAC0_ACC3_HIGH               0x00000613
+#define DREG_REGID_MAC1_ACC0_HIGH               0x00000614
+#define DREG_REGID_MAC1_ACC1_HIGH               0x00000615
+#define DREG_REGID_MAC1_ACC2_HIGH               0x00000616
+#define DREG_REGID_MAC1_ACC3_HIGH               0x00000617
+#define DREG_REGID_RSHOUT_LOW                   0x00000620
+#define DREG_REGID_RSHOUT_MID                   0x00000628
+#define DREG_REGID_RSHOUT_HIGH                  0x00000630
+
+/*
+ *  The following defines are for the flags in the DMA stream requestor write
+ */
+#define DSRWP_DSR_MASK                          0x0000000F
+#define DSRWP_DSR_BG_RQ                         0x00000001
+#define DSRWP_DSR_PRIORITY_MASK                 0x00000006
+#define DSRWP_DSR_PRIORITY_0                    0x00000000
+#define DSRWP_DSR_PRIORITY_1                    0x00000002
+#define DSRWP_DSR_PRIORITY_2                    0x00000004
+#define DSRWP_DSR_PRIORITY_3                    0x00000006
+#define DSRWP_DSR_RQ_PENDING                    0x00000008
+
+/*
+ *  The following defines are for the flags in the trap write port register.
+ */
+#define TWPR_TW_MASK                            0x0000FFFF
+#define TWPR_TW_SHIFT                           0
+
+/*
+ *  The following defines are for the flags in the stack pointer write
+ *  register.
+ */
+#define SPWR_STKP_MASK                          0x0000000F
+#define SPWR_STKP_SHIFT                         0
+
+/*
+ *  The following defines are for the flags in the SP interrupt register.
+ */
+#define SPIR_FRI                                0x00000001
+#define SPIR_DOI                                0x00000002
+#define SPIR_GPI2                               0x00000004
+#define SPIR_GPI3                               0x00000008
+#define SPIR_IP0                                0x00000010
+#define SPIR_IP1                                0x00000020
+#define SPIR_IP2                                0x00000040
+#define SPIR_IP3                                0x00000080
+
+/*
+ *  The following defines are for the flags in the functional group 1 register.
+ */
+#define FGR1_F1S_MASK                           0x0000FFFF
+#define FGR1_F1S_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the SP clock status register.
+ */
+#define SPCS_FRI                                0x00000001
+#define SPCS_DOI                                0x00000002
+#define SPCS_GPI2                               0x00000004
+#define SPCS_GPI3                               0x00000008
+#define SPCS_IP0                                0x00000010
+#define SPCS_IP1                                0x00000020
+#define SPCS_IP2                                0x00000040
+#define SPCS_IP3                                0x00000080
+#define SPCS_SPRUN                              0x00000100
+#define SPCS_SLEEP                              0x00000200
+#define SPCS_FG                                 0x00000400
+#define SPCS_ORUN                               0x00000800
+#define SPCS_IRQ                                0x00001000
+#define SPCS_FGN_MASK                           0x0000E000
+#define SPCS_FGN_SHIFT                          13
+
+/*
+ *  The following defines are for the flags in the SP DMA requestor status
+ *  register.
+ */
+#define SDSR_DCS_MASK                           0x000000FF
+#define SDSR_DCS_SHIFT                          0
+#define SDSR_DCS_NONE                           0x00000007
+
+/*
+ *  The following defines are for the flags in the frame timer register.
+ */
+#define FRMT_FTV_MASK                           0x0000FFFF
+#define FRMT_FTV_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the frame timer current count
+ *  register.
+ */
+#define FRCC_FCC_MASK                           0x0000FFFF
+#define FRCC_FCC_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the frame timer save count
+ *  register.
+ */
+#define FRSC_FCS_MASK                           0x0000FFFF
+#define FRSC_FCS_SHIFT                          0
+
+/*
+ *  The following define the various flags stored in the scatter/gather
+ *  descriptors.
+ */
+#define DMA_SG_NEXT_ENTRY_MASK                  0x00000FF8
+#define DMA_SG_SAMPLE_END_MASK                  0x0FFF0000
+#define DMA_SG_SAMPLE_END_FLAG                  0x10000000
+#define DMA_SG_LOOP_END_FLAG                    0x20000000
+#define DMA_SG_SIGNAL_END_FLAG                  0x40000000
+#define DMA_SG_SIGNAL_PAGE_FLAG                 0x80000000
+#define DMA_SG_NEXT_ENTRY_SHIFT                 3
+#define DMA_SG_SAMPLE_END_SHIFT                 16
+
+/*
+ *  The following define the offsets of the fields within the on-chip generic
+ *  DMA requestor.
+ */
+#define DMA_RQ_CONTROL1                         0x00000000
+#define DMA_RQ_CONTROL2                         0x00000004
+#define DMA_RQ_SOURCE_ADDR                      0x00000008
+#define DMA_RQ_DESTINATION_ADDR                 0x0000000C
+#define DMA_RQ_NEXT_PAGE_ADDR                   0x00000010
+#define DMA_RQ_NEXT_PAGE_SGDESC                 0x00000014
+#define DMA_RQ_LOOP_START_ADDR                  0x00000018
+#define DMA_RQ_POST_LOOP_ADDR                   0x0000001C
+#define DMA_RQ_PAGE_MAP_ADDR                    0x00000020
+
+/*
+ *  The following defines are for the flags in the first control word of the
+ *  on-chip generic DMA requestor.
+ */
+#define DMA_RQ_C1_COUNT_MASK                    0x000003FF
+#define DMA_RQ_C1_DESTINATION_SCATTER           0x00001000
+#define DMA_RQ_C1_SOURCE_GATHER                 0x00002000
+#define DMA_RQ_C1_DONE_FLAG                     0x00004000
+#define DMA_RQ_C1_OPTIMIZE_STATE                0x00008000
+#define DMA_RQ_C1_SAMPLE_END_STATE_MASK         0x00030000
+#define DMA_RQ_C1_FULL_PAGE                     0x00000000
+#define DMA_RQ_C1_BEFORE_SAMPLE_END             0x00010000
+#define DMA_RQ_C1_PAGE_MAP_ERROR                0x00020000
+#define DMA_RQ_C1_AT_SAMPLE_END                 0x00030000
+#define DMA_RQ_C1_LOOP_END_STATE_MASK           0x000C0000
+#define DMA_RQ_C1_NOT_LOOP_END                  0x00000000
+#define DMA_RQ_C1_BEFORE_LOOP_END               0x00040000
+#define DMA_RQ_C1_2PAGE_LOOP_BEGIN              0x00080000
+#define DMA_RQ_C1_LOOP_BEGIN                    0x000C0000
+#define DMA_RQ_C1_PAGE_MAP_MASK                 0x00300000
+#define DMA_RQ_C1_PM_NONE_PENDING               0x00000000
+#define DMA_RQ_C1_PM_NEXT_PENDING               0x00100000
+#define DMA_RQ_C1_PM_RESERVED                   0x00200000
+#define DMA_RQ_C1_PM_LOOP_NEXT_PENDING          0x00300000
+#define DMA_RQ_C1_WRITEBACK_DEST_FLAG           0x00400000
+#define DMA_RQ_C1_WRITEBACK_SRC_FLAG            0x00800000
+#define DMA_RQ_C1_DEST_SIZE_MASK                0x07000000
+#define DMA_RQ_C1_DEST_LINEAR                   0x00000000
+#define DMA_RQ_C1_DEST_MOD16                    0x01000000
+#define DMA_RQ_C1_DEST_MOD32                    0x02000000
+#define DMA_RQ_C1_DEST_MOD64                    0x03000000
+#define DMA_RQ_C1_DEST_MOD128                   0x04000000
+#define DMA_RQ_C1_DEST_MOD256                   0x05000000
+#define DMA_RQ_C1_DEST_MOD512                   0x06000000
+#define DMA_RQ_C1_DEST_MOD1024                  0x07000000
+#define DMA_RQ_C1_DEST_ON_HOST                  0x08000000
+#define DMA_RQ_C1_SOURCE_SIZE_MASK              0x70000000
+#define DMA_RQ_C1_SOURCE_LINEAR                 0x00000000
+#define DMA_RQ_C1_SOURCE_MOD16                  0x10000000
+#define DMA_RQ_C1_SOURCE_MOD32                  0x20000000
+#define DMA_RQ_C1_SOURCE_MOD64                  0x30000000
+#define DMA_RQ_C1_SOURCE_MOD128                 0x40000000
+#define DMA_RQ_C1_SOURCE_MOD256                 0x50000000
+#define DMA_RQ_C1_SOURCE_MOD512                 0x60000000
+#define DMA_RQ_C1_SOURCE_MOD1024                0x70000000
+#define DMA_RQ_C1_SOURCE_ON_HOST                0x80000000
+#define DMA_RQ_C1_COUNT_SHIFT                   0
+
+/*
+ *  The following defines are for the flags in the second control word of the
+ *  on-chip generic DMA requestor.
+ */
+#define DMA_RQ_C2_VIRTUAL_CHANNEL_MASK          0x0000003F
+#define DMA_RQ_C2_VIRTUAL_SIGNAL_MASK           0x00000300
+#define DMA_RQ_C2_NO_VIRTUAL_SIGNAL             0x00000000
+#define DMA_RQ_C2_SIGNAL_EVERY_DMA              0x00000100
+#define DMA_RQ_C2_SIGNAL_SOURCE_PINGPONG        0x00000200
+#define DMA_RQ_C2_SIGNAL_DEST_PINGPONG          0x00000300
+#define DMA_RQ_C2_AUDIO_CONVERT_MASK            0x0000F000
+#define DMA_RQ_C2_AC_NONE                       0x00000000
+#define DMA_RQ_C2_AC_8_TO_16_BIT                0x00001000
+#define DMA_RQ_C2_AC_MONO_TO_STEREO             0x00002000
+#define DMA_RQ_C2_AC_ENDIAN_CONVERT             0x00004000
+#define DMA_RQ_C2_AC_SIGNED_CONVERT             0x00008000
+#define DMA_RQ_C2_LOOP_END_MASK                 0x0FFF0000
+#define DMA_RQ_C2_LOOP_MASK                     0x30000000
+#define DMA_RQ_C2_NO_LOOP                       0x00000000
+#define DMA_RQ_C2_ONE_PAGE_LOOP                 0x10000000
+#define DMA_RQ_C2_TWO_PAGE_LOOP                 0x20000000
+#define DMA_RQ_C2_MULTI_PAGE_LOOP               0x30000000
+#define DMA_RQ_C2_SIGNAL_LOOP_BACK              0x40000000
+#define DMA_RQ_C2_SIGNAL_POST_BEGIN_PAGE        0x80000000
+#define DMA_RQ_C2_VIRTUAL_CHANNEL_SHIFT         0
+#define DMA_RQ_C2_LOOP_END_SHIFT                16
+
+/*
+ *  The following defines are for the flags in the source and destination words
+ *  of the on-chip generic DMA requestor.
+ */
+#define DMA_RQ_SD_ADDRESS_MASK                  0x0000FFFF
+#define DMA_RQ_SD_MEMORY_ID_MASK                0x000F0000
+#define DMA_RQ_SD_SP_PARAM_ADDR                 0x00000000
+#define DMA_RQ_SD_SP_SAMPLE_ADDR                0x00010000
+#define DMA_RQ_SD_SP_PROGRAM_ADDR               0x00020000
+#define DMA_RQ_SD_SP_DEBUG_ADDR                 0x00030000
+#define DMA_RQ_SD_OMNIMEM_ADDR                  0x000E0000
+#define DMA_RQ_SD_END_FLAG                      0x40000000
+#define DMA_RQ_SD_ERROR_FLAG                    0x80000000
+#define DMA_RQ_SD_ADDRESS_SHIFT                 0
+
+/*
+ *  The following defines are for the flags in the page map address word of the
+ *  on-chip generic DMA requestor.
+ */
+#define DMA_RQ_PMA_LOOP_THIRD_PAGE_ENTRY_MASK   0x00000FF8
+#define DMA_RQ_PMA_PAGE_TABLE_MASK              0xFFFFF000
+#define DMA_RQ_PMA_LOOP_THIRD_PAGE_ENTRY_SHIFT  3
+#define DMA_RQ_PMA_PAGE_TABLE_SHIFT             12
+
+#define BA1_VARIDEC_BUF_1       0x000
+
+#define BA1_PDTC                0x0c0    /* BA1_PLAY_DMA_TRANSACTION_COUNT_REG */
+#define BA1_PFIE                0x0c4    /* BA1_PLAY_FORMAT_&_INTERRUPT_ENABLE_REG */
+#define BA1_PBA                 0x0c8    /* BA1_PLAY_BUFFER_ADDRESS */
+#define BA1_PVOL                0x0f8    /* BA1_PLAY_VOLUME_REG */
+#define BA1_PSRC                0x288    /* BA1_PLAY_SAMPLE_RATE_CORRECTION_REG */
+#define BA1_PCTL                0x2a4    /* BA1_PLAY_CONTROL_REG */
+#define BA1_PPI                 0x2b4    /* BA1_PLAY_PHASE_INCREMENT_REG */
+
+#define BA1_CCTL                0x064    /* BA1_CAPTURE_CONTROL_REG */
+#define BA1_CIE                 0x104    /* BA1_CAPTURE_INTERRUPT_ENABLE_REG */
+#define BA1_CBA                 0x10c    /* BA1_CAPTURE_BUFFER_ADDRESS */
+#define BA1_CSRC                0x2c8    /* BA1_CAPTURE_SAMPLE_RATE_CORRECTION_REG */
+#define BA1_CCI                 0x2d8    /* BA1_CAPTURE_COEFFICIENT_INCREMENT_REG */
+#define BA1_CD                  0x2e0    /* BA1_CAPTURE_DELAY_REG */
+#define BA1_CPI                 0x2f4    /* BA1_CAPTURE_PHASE_INCREMENT_REG */
+#define BA1_CVOL                0x2f8    /* BA1_CAPTURE_VOLUME_REG */
+
+#define BA1_CFG1                0x134    /* BA1_CAPTURE_FRAME_GROUP_1_REG */
+#define BA1_CFG2                0x138    /* BA1_CAPTURE_FRAME_GROUP_2_REG */
+#define BA1_CCST                0x13c    /* BA1_CAPTURE_CONSTANT_REG */
+#define BA1_CSPB                0x340    /* BA1_CAPTURE_SPB_ADDRESS */
+
+/*
+ *
+ */
+
+#define CS46XX_MODE_OUTPUT	(1<<0)	 /* MIDI UART - output */ 
+#define CS46XX_MODE_INPUT	(1<<1)	 /* MIDI UART - input */
+
+/*
+ *
+ */
+
+#define SAVE_REG_MAX             0x10
+#define POWER_DOWN_ALL         0x7f0f
+
+/* maxinum number of AC97 codecs connected, AC97 2.0 defined 4 */
+#define MAX_NR_AC97				            4
+#define CS46XX_PRIMARY_CODEC_INDEX          0
+#define CS46XX_SECONDARY_CODEC_INDEX		1
+#define CS46XX_SECONDARY_CODEC_OFFSET		0x80
+#define CS46XX_DSP_CAPTURE_CHANNEL          1
+
+/* capture */
+#define CS46XX_DSP_CAPTURE_CHANNEL          1
+
+/* mixer */
+#define CS46XX_MIXER_SPDIF_INPUT_ELEMENT    1
+#define CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT   2
+
+
+struct snd_cs46xx_pcm {
+	struct snd_dma_buffer hw_buf;
+  
+	unsigned int ctl;
+	unsigned int shift;	/* Shift count to trasform frames in bytes */
+	struct snd_pcm_indirect pcm_rec;
+	struct snd_pcm_substream *substream;
+
+	struct dsp_pcm_channel_descriptor * pcm_channel;
+
+	int pcm_channel_id;    /* Fron Rear, Center Lfe  ... */
+};
+
+struct snd_cs46xx_region {
+	char name[24];
+	unsigned long base;
+	void __iomem *remap_addr;
+	unsigned long size;
+	struct resource *resource;
+};
+
+struct snd_cs46xx {
+	int irq;
+	unsigned long ba0_addr;
+	unsigned long ba1_addr;
+	union {
+		struct {
+			struct snd_cs46xx_region ba0;
+			struct snd_cs46xx_region data0;
+			struct snd_cs46xx_region data1;
+			struct snd_cs46xx_region pmem;
+			struct snd_cs46xx_region reg;
+		} name;
+		struct snd_cs46xx_region idx[5];
+	} region;
+
+	unsigned int mode;
+	
+	struct {
+		struct snd_dma_buffer hw_buf;
+
+		unsigned int ctl;
+		unsigned int shift;	/* Shift count to trasform frames in bytes */
+		struct snd_pcm_indirect pcm_rec;
+		struct snd_pcm_substream *substream;
+	} capt;
+
+
+	int nr_ac97_codecs;
+	struct snd_ac97_bus *ac97_bus;
+	struct snd_ac97 *ac97[MAX_NR_AC97];
+
+	struct pci_dev *pci;
+	struct snd_card *card;
+	struct snd_pcm *pcm;
+
+	struct snd_rawmidi *rmidi;
+	struct snd_rawmidi_substream *midi_input;
+	struct snd_rawmidi_substream *midi_output;
+
+	spinlock_t reg_lock;
+	unsigned int midcr;
+	unsigned int uartm;
+
+	int amplifier;
+	void (*amplifier_ctrl)(struct snd_cs46xx *, int);
+	void (*active_ctrl)(struct snd_cs46xx *, int);
+  	void (*mixer_init)(struct snd_cs46xx *);
+
+	int acpi_port;
+	struct snd_kcontrol *eapd_switch; /* for amplifier hack */
+	int accept_valid;	/* accept mmap valid (for OSS) */
+	int in_suspend;
+
+	struct gameport *gameport;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+	struct mutex spos_mutex;
+
+	struct dsp_spos_instance * dsp_spos_instance;
+
+	struct snd_pcm *pcm_rear;
+	struct snd_pcm *pcm_center_lfe;
+	struct snd_pcm *pcm_iec958;
+#else /* for compatibility */
+	struct snd_cs46xx_pcm *playback_pcm;
+	unsigned int play_ctl;
+#endif
+
+#ifdef CONFIG_PM
+	u32 *saved_regs;
+#endif
+};
+
+int snd_cs46xx_create(struct snd_card *card,
+		      struct pci_dev *pci,
+		      int external_amp, int thinkpad,
+		      struct snd_cs46xx **rcodec);
+extern const struct dev_pm_ops snd_cs46xx_pm;
+
+int snd_cs46xx_pcm(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm);
+int snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm);
+int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm);
+int snd_cs46xx_pcm_center_lfe(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm);
+int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device);
+int snd_cs46xx_midi(struct snd_cs46xx *chip, int device, struct snd_rawmidi **rmidi);
+int snd_cs46xx_start_dsp(struct snd_cs46xx *chip);
+int snd_cs46xx_gameport(struct snd_cs46xx *chip);
+
+#endif /* __SOUND_CS46XX_H */
diff --git a/sound/pci/cs46xx/cs46xx_dsp_scb_types.h b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h
new file mode 100644
index 000000000000..080857ad0ca2
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h
@@ -0,0 +1,1213 @@
+/*
+ *  The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ *
+ * NOTE: comments are copy/paste from cwcemb80.lst 
+ * provided by Tom Woller at Cirrus (my only
+ * documentation about the SP OS running inside
+ * the DSP) 
+ */
+
+#ifndef __CS46XX_DSP_SCB_TYPES_H__
+#define __CS46XX_DSP_SCB_TYPES_H__
+
+#include <asm/byteorder.h>
+
+#ifndef ___DSP_DUAL_16BIT_ALLOC
+#if   defined(__LITTLE_ENDIAN)
+#define ___DSP_DUAL_16BIT_ALLOC(a,b) u16 a; u16 b;
+#elif defined(__BIG_ENDIAN)
+#define ___DSP_DUAL_16BIT_ALLOC(a,b) u16 b; u16 a;
+#else
+#error Not __LITTLE_ENDIAN and not __BIG_ENDIAN, then what ???
+#endif
+#endif
+
+/* This structs are used internally by the SP */
+
+struct dsp_basic_dma_req {
+	/* DMA Requestor Word 0 (DCW)  fields:
+
+	   31 [30-28]27  [26:24] 23 22 21 20 [19:18] [17:16] 15 14 13  12  11 10 9 8 7 6  [5:0]
+	   _______________________________________________________________________________________	
+	   |S| SBT  |D|  DBT    |wb|wb|  |  |  LS  |  SS   |Opt|Do|SSG|DSG|  |  | | | | | Dword   |
+	   |H|_____ |H|_________|S_|D |__|__|______|_______|___|ne|__ |__ |__|__|_|_|_|_|_Count -1|
+	*/
+	u32 dcw;                 /* DMA Control Word */
+	u32 dmw;                 /* DMA Mode Word */
+	u32 saw;                 /* Source Address Word */
+	u32 daw;                 /* Destination Address Word  */
+};
+
+struct dsp_scatter_gather_ext {
+	u32 npaw;                /* Next-Page Address Word */
+
+	/* DMA Requestor Word 5 (NPCW)  fields:
+     
+	   31-30 29 28          [27:16]              [15:12]             [11:3]                [2:0] 				
+	   _________________________________________________________________________________________	
+	   |SV  |LE|SE|   Sample-end byte offset   |         | Page-map entry offset for next  |    | 
+	   |page|__|__| ___________________________|_________|__page, if !sample-end___________|____|
+	*/
+	u32 npcw;                /* Next-Page Control Word */
+	u32 lbaw;                /* Loop-Begin Address Word */
+	u32 nplbaw;              /* Next-Page after Loop-Begin Address Word */
+	u32 sgaw;                /* Scatter/Gather Address Word */
+};
+
+struct dsp_volume_control {
+	___DSP_DUAL_16BIT_ALLOC(
+	   rightTarg,  /* Target volume for left & right channels */
+	   leftTarg
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	   rightVol,   /* Current left & right channel volumes */
+	   leftVol
+	)
+};
+
+/* Generic stream control block (SCB) structure definition */
+struct dsp_generic_scb {
+	/* For streaming I/O, the DSP should never alter any words in the DMA
+	   requestor or the scatter/gather extension.  Only ad hoc DMA request
+	   streams are free to alter the requestor (currently only occur in the
+	   DOS-based MIDI controller and in debugger-inserted code).
+    
+	   If an SCB does not have any associated DMA requestor, these 9 ints
+	   may be freed for use by other tasks, but the pointer to the SCB must
+	   still be such that the insOrd:nextSCB appear at offset 9 from the
+	   SCB pointer.
+     
+	   Basic (non scatter/gather) DMA requestor (4 ints)
+	*/
+  
+	/* Initialized by the host, only modified by DMA 
+	   R/O for the DSP task */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+
+	/* Scatter/gather DMA requestor extension   (5 ints) 
+	   Initialized by the host, only modified by DMA
+	   DSP task never needs to even read these.
+	*/
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+
+	/* Sublist pointer & next stream control block (SCB) link.
+	   Initialized & modified by the host R/O for the DSP task
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,     /* REQUIRED */
+	    sub_list_ptr  /* REQUIRED */
+	)
+  
+	/* Pointer to this tasks parameter block & stream function pointer 
+	   Initialized by the host  R/O for the DSP task */
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,  /* REQUIRED */
+	    this_spb      /* REQUIRED */
+	)
+
+	/* rsConfig register for stream buffer (rsDMA reg. 
+	   is loaded from basicReq.daw for incoming streams, or 
+	   basicReq.saw, for outgoing streams) 
+
+	   31 30 29  [28:24]     [23:16] 15 14 13 12 11 10 9 8 7 6  5      4      [3:0]
+	   ______________________________________________________________________________
+	   |DMA  |D|maxDMAsize| streamNum|dir|p|  |  |  |  | | |ds |shr 1|rev Cy | mod   |
+	   |prio |_|__________|__________|___|_|__|__|__|__|_|_|___|_____|_______|_______|
+	   31 30 29  [28:24]     [23:16] 15 14 13 12 11 10 9 8 7 6  5      4      [3:0]
+
+
+	   Initialized by the host R/O for the DSP task
+	*/
+	u32  strm_rs_config; /* REQUIRED */
+               // 
+	/* On mixer input streams: indicates mixer input stream configuration
+	   On Tees, this is copied from the stream being snooped
+
+	   Stream sample pointer & MAC-unit mode for this stream 
+     
+	   Initialized by the host Updated by the DSP task
+	*/
+	u32  strm_buf_ptr; /* REQUIRED  */
+
+	/* On mixer input streams: points to next mixer input and is updated by the
+                                   mixer subroutine in the "parent" DSP task
+				   (least-significant 16 bits are preserved, unused)
+    
+           On Tees, the pointer is copied from the stream being snooped on
+	   initialization, and, subsequently, it is copied into the
+	   stream being snooped.
+
+	   On wavetable/3D voices: the strmBufPtr will use all 32 bits to allow for
+                                   fractional phase accumulation
+
+	   Fractional increment per output sample in the input sample buffer
+
+	   (Not used on mixer input streams & redefined on Tees)
+	   On wavetable/3D voices: this 32-bit word specifies the integer.fractional 
+	   increment per output sample.
+	*/
+	u32  strmPhiIncr;
+
+
+	/* Standard stereo volume control
+	   Initialized by the host (host updates target volumes) 
+
+	   Current volumes update by the DSP task
+	   On mixer input streams: required & updated by the mixer subroutine in the
+                                   "parent" DSP task
+
+	   On Tees, both current & target volumes are copied up on initialization,
+	   and, subsequently, the target volume is copied up while the current
+	   volume is copied down.
+     
+	   These two 32-bit words are redefined for wavetable & 3-D voices.    
+	*/
+	struct dsp_volume_control vol_ctrl_t;   /* Optional */
+};
+
+
+struct dsp_spos_control_block {
+	/* WARNING: Certain items in this structure are modified by the host
+	            Any dword that can be modified by the host, must not be
+		    modified by the SP as the host can only do atomic dword
+		    writes, and to do otherwise, even a read modify write, 
+		    may lead to corrupted data on the SP.
+  
+		    This rule does not apply to one off boot time initialisation prior to starting the SP
+	*/
+
+
+	___DSP_DUAL_16BIT_ALLOC( 
+	/* First element on the Hyper forground task tree */
+	    hfg_tree_root_ptr,  /* HOST */			    
+	/* First 3 dwords are written by the host and read-only on the DSP */
+	    hfg_stack_base      /* HOST */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Point to this data structure to enable easy access */
+	    spos_cb_ptr,	 /* SP */
+	    prev_task_tree_ptr   /* SP && HOST */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Currently Unused */
+	    xxinterval_timer_period,
+	/* Enable extension of SPOS data structure */
+	    HFGSPB_ptr
+	)
+
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    xxnum_HFG_ticks_thisInterval,
+	/* Modified by the DSP */
+	    xxnum_tntervals
+	)
+
+
+	/* Set by DSP upon encountering a trap (breakpoint) or a spurious
+	   interrupt.  The host must clear this dword after reading it
+	   upon receiving spInt1. */
+	___DSP_DUAL_16BIT_ALLOC(
+	    spurious_int_flag,	 /* (Host & SP) Nature of the spurious interrupt */
+	    trap_flag            /* (Host & SP) Nature of detected Trap */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused2,					
+	    invalid_IP_flag        /* (Host & SP ) Indicate detection of invalid instruction pointer */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	/* pointer to forground task tree header for use in next task search */
+	    fg_task_tree_hdr_ptr,	  /* HOST */		
+	/* Data structure for controlling synchronous link update */
+	    hfg_sync_update_ptr           /* HOST */
+	)
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	     begin_foreground_FCNT,  /* SP */
+	/* Place holder for holding sleep timing */
+	     last_FCNT_before_sleep  /* SP */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused7,           /* SP */
+	    next_task_treePtr  /* SP */
+	)
+
+	u32 unused5;        
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    active_flags,   /* SP */
+	/* State flags, used to assist control of execution of Hyper Forground */
+	    HFG_flags       /* SP */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused9,
+	    unused8
+	)
+                              
+	/* Space for saving enough context so that we can set up enough 
+	   to save some more context.
+	*/
+	u32 rFE_save_for_invalid_IP;
+	u32 r32_save_for_spurious_int;
+	u32 r32_save_for_trap;
+	u32 r32_save_for_HFG;
+};
+
+/* SPB for MIX_TO_OSTREAM algorithm family */
+struct dsp_mix2_ostream_spb
+{
+	/* 16b.16b integer.frac approximation to the
+	   number of 3 sample triplets to output each
+	   frame. (approximation must be floor, to
+	   insure that the fractional error is always
+	   positive)
+	*/
+	u32 outTripletsPerFrame;
+
+	/* 16b.16b integer.frac accumulated number of
+	   output triplets since the start of group 
+	*/
+	u32 accumOutTriplets;  
+};
+
+/* SCB for Timing master algorithm */
+struct dsp_timing_master_scb {
+	/* First 12 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,     /* REQUIRED */
+	    sub_list_ptr  /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,  /* REQUIRED */
+	    this_spb      /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Initial values are 0000:xxxx */
+ 	    reserved,
+	    extra_sample_accum
+	)
+
+  
+	/* Initial values are xxxx:0000
+	   hi: Current CODEC output FIFO pointer
+	       (0 to 0x0f)
+           lo: Flag indicating that the CODEC
+	       FIFO is sync'd (host clears to
+	       resynchronize the FIFO pointer
+	       upon start/restart) 
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    codec_FIFO_syncd, 
+	    codec_FIFO_ptr
+	)
+  
+	/* Init. 8000:0005 for 44.1k
+                 8000:0001 for 48k
+	   hi: Fractional sample accumulator 0.16b
+	   lo: Number of frames remaining to be
+	       processed in the current group of
+	       frames
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    frac_samp_accum_qm1,
+	    TM_frms_left_in_group
+	) 
+
+	/* Init. 0001:0005 for 44.1k
+                 0000:0001 for 48k
+	   hi: Fractional sample correction factor 0.16b
+	       to be added every frameGroupLength frames
+	       to correct for truncation error in
+	       nsamp_per_frm_q15
+	   lo: Number of frames in the group
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    frac_samp_correction_qm1,
+	    TM_frm_group_length  
+	)
+
+	/* Init. 44.1k*65536/8k = 0x00058333 for 44.1k
+                 48k*65536/8k = 0x00060000 for 48k
+	   16b.16b integer.frac approximation to the
+	   number of samples to output each frame.
+	   (approximation must be floor, to insure */
+	u32 nsamp_per_frm_q15;
+};
+
+/* SCB for CODEC output algorithm */
+struct dsp_codec_output_scb {
+	/* First 13 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,       /* REQUIRED */
+	    sub_list_ptr    /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,    /* REQUIRED */
+	    this_spb        /* REQUIRED */
+	)
+
+	u32 strm_rs_config; /* REQUIRED */
+
+	u32 strm_buf_ptr;   /* REQUIRED */
+
+	/* NOTE: The CODEC output task reads samples from the first task on its
+                 sublist at the stream buffer pointer (init. to lag DMA destination
+		 address word).  After the required number of samples is transferred,
+		 the CODEC output task advances sub_list_ptr->strm_buf_ptr past the samples
+		 consumed.
+	*/
+
+	/* Init. 0000:0010 for SDout
+                 0060:0010 for SDout2
+		 0080:0010 for SDout3
+	   hi: Base IO address of FIFO to which
+	       the left-channel samples are to
+	       be written.
+	   lo: Displacement for the base IO
+	       address for left-channel to obtain
+	       the base IO address for the FIFO
+	       to which the right-channel samples
+	       are to be written.
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    left_chan_base_IO_addr,
+	    right_chan_IO_disp
+	)
+
+
+	/* Init: 0x0080:0004 for non-AC-97
+	   Init: 0x0080:0000 for AC-97
+	   hi: Exponential volume change rate
+	       for input stream
+	   lo: Positive shift count to shift the
+	       16-bit input sample to obtain the
+	       32-bit output word
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    CO_scale_shift_count, 
+	    CO_exp_vol_change_rate
+	)
+
+	/* Pointer to SCB at end of input chain */
+	___DSP_DUAL_16BIT_ALLOC(
+	    reserved,
+	    last_sub_ptr
+	)
+};
+
+/* SCB for CODEC input algorithm */
+struct dsp_codec_input_scb {
+	/* First 13 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,       /* REQUIRED */
+	    sub_list_ptr    /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,    /* REQUIRED */
+	    this_spb        /* REQUIRED */
+	)
+
+	u32 strm_rs_config; /* REQUIRED */
+	u32 strm_buf_ptr;   /* REQUIRED */
+
+	/* NOTE: The CODEC input task reads samples from the hardware FIFO 
+                 sublist at the DMA source address word (sub_list_ptr->basic_req.saw).
+                 After the required number of samples is transferred, the CODEC
+                 output task advances sub_list_ptr->basic_req.saw past the samples
+                 consumed.  SPuD must initialize the sub_list_ptr->basic_req.saw
+                 to point half-way around from the initial sub_list_ptr->strm_nuf_ptr
+                 to allow for lag/lead.
+	*/
+
+	/* Init. 0000:0010 for SDout
+                 0060:0010 for SDout2
+		 0080:0010 for SDout3
+	   hi: Base IO address of FIFO to which
+	       the left-channel samples are to
+	       be written.
+	   lo: Displacement for the base IO
+	       address for left-channel to obtain
+	       the base IO address for the FIFO
+	       to which the right-channel samples
+	       are to be written.
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    rightChanINdisp, 
+	    left_chan_base_IN_addr
+	)
+	/* Init. ?:fffc
+	   lo: Negative shift count to shift the
+	       32-bit input dword to obtain the
+	       16-bit sample msb-aligned (count
+	       is negative to shift left)
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    scaleShiftCount, 
+	    reserver1
+	)
+
+	u32  reserved2;
+};
+
+
+struct dsp_pcm_serial_input_scb {
+	/* First 13 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,       /* REQUIRED */
+	    sub_list_ptr    /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,    /* REQUIRED */
+	    this_spb        /* REQUIRED */
+	)
+
+	u32 strm_buf_ptr;   /* REQUIRED */
+	u32 strm_rs_config; /* REQUIRED */
+  
+	/* Init. Ptr to CODEC input SCB
+	   hi: Pointer to the SCB containing the
+	       input buffer to which CODEC input
+	       samples are written
+	   lo: Flag indicating the link to the CODEC
+	       input task is to be initialized
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    init_codec_input_link,
+	    codec_input_buf_scb
+	)
+
+	/* Initialized by the host (host updates target volumes) */
+	struct dsp_volume_control psi_vol_ctrl;   
+  
+};
+
+struct dsp_src_task_scb {
+	___DSP_DUAL_16BIT_ALLOC(
+	    frames_left_in_gof,
+	    gofs_left_in_sec
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    const2_thirds,
+	    num_extra_tnput_samples
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    cor_per_gof,
+	    correction_per_sec 
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    output_buf_producer_ptr,  
+	    junk_DMA_MID
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    gof_length,  
+	    gofs_per_sec
+	)
+
+	u32 input_buf_strm_config;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    reserved_for_SRC_use,
+	    input_buf_consumer_ptr
+	)
+
+	u32 accum_phi;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    exp_src_vol_change_rate,
+	    input_buf_producer_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    src_next_scb,
+	    src_sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    src_entry_point,
+	    src_this_sbp
+	)
+
+	u32  src_strm_rs_config;
+	u32  src_strm_buf_ptr;
+  
+	u32   phiIncr6int_26frac;
+  
+	struct dsp_volume_control src_vol_ctrl;
+};
+
+struct dsp_decimate_by_pow2_scb {
+	/* decimationFactor = 2, 4, or 8 (larger factors waste too much memory
+	                                  when compared to cascading decimators)
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_coef_base_ptr,
+	    dec2_coef_increment
+	)
+
+	/* coefIncrement = 128 / decimationFactor (for our ROM filter)
+	   coefBasePtr = 0x8000 (for our ROM filter)
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_in_samples_per_out_triplet,
+	    dec2_extra_in_samples
+	)
+	/* extraInSamples: # of accumulated, unused input samples (init. to 0)
+	   inSamplesPerOutTriplet = 3 * decimationFactor
+	*/
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_const2_thirds,
+	    dec2_half_num_taps_mp5
+	)
+	/* halfNumTapsM5: (1/2 number of taps in decimation filter) minus 5
+	   const2thirds: constant 2/3 in 16Q0 format (sign.15)
+	*/
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_output_buf_producer_ptr,
+	    dec2_junkdma_mid
+	)
+
+	u32  dec2_reserved2;
+
+	u32  dec2_input_nuf_strm_config;
+	/* inputBufStrmConfig: rsConfig for the input buffer to the decimator
+	   (buffer size = decimationFactor * 32 dwords)
+	*/
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_phi_incr,
+	    dec2_input_buf_consumer_ptr
+	)
+	/* inputBufConsumerPtr: Input buffer read pointer (into SRC filter)
+	   phiIncr = decimationFactor * 4
+	*/
+
+	u32 dec2_reserved3;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_exp_vol_change_rate,
+	    dec2_input_buf_producer_ptr
+	)
+	/* inputBufProducerPtr: Input buffer write pointer
+	   expVolChangeRate: Exponential volume change rate for possible
+	                     future mixer on input streams
+	*/
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_next_scb,
+	    dec2_sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    dec2_entry_point,
+	    dec2_this_spb
+	)
+
+	u32  dec2_strm_rs_config;
+	u32  dec2_strm_buf_ptr;
+
+	u32  dec2_reserved4;
+
+	struct dsp_volume_control dec2_vol_ctrl; /* Not used! */
+};
+
+struct dsp_vari_decimate_scb {
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_frames_left_in_gof,
+	    vdec_gofs_left_in_sec
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_const2_thirds,
+	    vdec_extra_in_samples
+	)
+	/* extraInSamples: # of accumulated, unused input samples (init. to 0)
+	   const2thirds: constant 2/3 in 16Q0 format (sign.15) */
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_cor_per_gof,
+	    vdec_correction_per_sec
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_output_buf_producer_ptr,
+	    vdec_input_buf_consumer_ptr
+	)
+	/* inputBufConsumerPtr: Input buffer read pointer (into SRC filter) */
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_gof_length,
+	    vdec_gofs_per_sec
+	)
+
+	u32  vdec_input_buf_strm_config;
+	/* inputBufStrmConfig: rsConfig for the input buffer to the decimator
+	   (buffer size = 64 dwords) */
+	u32  vdec_coef_increment;
+	/* coefIncrement = - 128.0 / decimationFactor (as a 32Q15 number) */
+
+	u32  vdec_accumphi;
+	/* accumPhi: accumulated fractional phase increment (6.26) */
+
+	___DSP_DUAL_16BIT_ALLOC(
+ 	    vdec_exp_vol_change_rate,
+	    vdec_input_buf_producer_ptr
+	)
+	/* inputBufProducerPtr: Input buffer write pointer
+	   expVolChangeRate: Exponential volume change rate for possible
+	   future mixer on input streams */
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_next_scb,
+	    vdec_sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    vdec_entry_point,
+	    vdec_this_spb
+	)
+
+	u32 vdec_strm_rs_config;
+	u32 vdec_strm_buf_ptr;
+
+	u32 vdec_phi_incr_6int_26frac;
+
+	struct dsp_volume_control vdec_vol_ctrl;
+};
+
+
+/* SCB for MIX_TO_OSTREAM algorithm family */
+struct dsp_mix2_ostream_scb {
+	/* First 13 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,       /* REQUIRED */
+	    sub_list_ptr    /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,    /* REQUIRED */
+	    this_spb        /* REQUIRED */
+	)
+
+	u32 strm_rs_config; /* REQUIRED */
+	u32 strm_buf_ptr;   /* REQUIRED */
+
+
+	/* hi: Number of mixed-down input triplets
+	       computed since start of group
+	   lo: Number of frames remaining to be
+	       processed in the current group of
+	       frames
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    frames_left_in_group,
+	    accum_input_triplets
+	)
+
+	/* hi: Exponential volume change rate
+	       for mixer on input streams
+	   lo: Number of frames in the group
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    frame_group_length,
+	    exp_vol_change_rate
+	)
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	    const_FFFF,
+	    const_zero
+	)
+};
+
+
+/* SCB for S16_MIX algorithm */
+struct dsp_mix_only_scb {
+	/* First 13 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,       /* REQUIRED */
+	    sub_list_ptr    /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,    /* REQUIRED */
+	    this_spb        /* REQUIRED */
+	)
+
+	u32 strm_rs_config; /* REQUIRED */
+	u32 strm_buf_ptr;   /* REQUIRED */
+
+	u32 reserved;
+	struct dsp_volume_control vol_ctrl;
+};
+
+/* SCB for the async. CODEC input algorithm */
+struct dsp_async_codec_input_scb {
+	u32 io_free2;     
+  
+	u32 io_current_total;
+	u32 io_previous_total;
+  
+	u16 io_count;
+	u16 io_count_limit;
+  
+	u16 o_fifo_base_addr;            
+	u16 ost_mo_format;
+	/* 1 = stereo; 0 = mono 
+	   xxx for ASER 1 (not allowed); 118 for ASER2 */
+
+	u32  ostrm_rs_config;
+	u32  ostrm_buf_ptr;
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	    io_sclks_per_lr_clk,
+	    io_io_enable
+	)
+
+	u32  io_free4;
+
+	___DSP_DUAL_16BIT_ALLOC(  
+	    io_next_scb,
+	    io_sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    io_entry_point,
+	    io_this_spb
+	)
+
+	u32 istrm_rs_config;
+	u32 istrm_buf_ptr;
+
+	/* Init. 0000:8042: for ASER1
+                 0000:8044: for ASER2  */
+	___DSP_DUAL_16BIT_ALLOC(
+	    io_stat_reg_addr,
+	    iofifo_pointer
+	)
+
+	/* Init 1 stero:100 ASER1
+	   Init 0 mono:110 ASER2 
+	*/
+	___DSP_DUAL_16BIT_ALLOC(
+	    ififo_base_addr,            
+	    ist_mo_format
+	)
+
+	u32 i_free;
+};
+
+
+/* SCB for the SP/DIF CODEC input and output */
+struct dsp_spdifiscb {
+	___DSP_DUAL_16BIT_ALLOC(
+	    status_ptr,     
+	    status_start_ptr
+	)
+
+	u32 current_total;
+	u32 previous_total;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    count,
+	    count_limit
+	)
+
+	u32 status_data;
+
+	___DSP_DUAL_16BIT_ALLOC(  
+	    status,
+	    free4
+	)
+
+	u32 free3;
+
+	___DSP_DUAL_16BIT_ALLOC(  
+	    free2,
+	    bit_count
+	)
+
+	u32  temp_status;
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_SCB,
+	    sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,
+	    this_spb
+	)
+
+	u32  strm_rs_config;
+	u32  strm_buf_ptr;
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	    stat_reg_addr, 
+	    fifo_pointer
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    fifo_base_addr, 
+	    st_mo_format
+	)
+
+	u32  free1;
+};
+
+
+/* SCB for the SP/DIF CODEC input and output  */
+struct dsp_spdifoscb {		 
+
+	u32 free2;     
+
+	u32 free3[4];             
+
+	/* Need to be here for compatibility with AsynchFGTxCode */
+	u32 strm_rs_config;
+                               
+	u32 strm_buf_ptr;
+
+	___DSP_DUAL_16BIT_ALLOC(  
+	    status,
+	    free5
+	)
+
+	u32 free4;
+
+	___DSP_DUAL_16BIT_ALLOC(  
+	    next_scb,
+	    sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,
+	    this_spb
+	)
+
+	u32 free6[2];
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	    stat_reg_addr, 
+	    fifo_pointer
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    fifo_base_addr,
+	    st_mo_format
+	)
+
+	u32  free1;                                         
+};
+
+
+struct dsp_asynch_fg_rx_scb {
+	___DSP_DUAL_16BIT_ALLOC(
+	    bot_buf_mask,
+	    buf_Mask
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    max,
+	    min
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    old_producer_pointer,
+	    hfg_scb_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    delta,
+	    adjust_count
+	)
+
+	u32 unused2[5];  
+
+	___DSP_DUAL_16BIT_ALLOC(  
+	    sibling_ptr,  
+	    child_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    code_ptr,
+	    this_ptr
+	)
+
+	u32 strm_rs_config; 
+
+	u32 strm_buf_ptr;
+  
+	u32 unused_phi_incr;
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	    right_targ,   
+	    left_targ
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    right_vol,
+	    left_vol
+	)
+};
+
+
+struct dsp_asynch_fg_tx_scb {
+	___DSP_DUAL_16BIT_ALLOC(
+	    not_buf_mask,
+	    buf_mask
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    max,
+	    min
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused1,
+	    hfg_scb_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    delta,
+	    adjust_count
+	)
+
+	u32 accum_phi;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused2,
+	    const_one_third
+	)
+
+	u32 unused3[3];
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    sibling_ptr,
+	    child_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    codePtr,
+	    this_ptr
+	)
+
+	u32 strm_rs_config;
+
+	u32 strm_buf_ptr;
+
+	u32 phi_incr;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused_right_targ,
+	    unused_left_targ
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused_right_vol,
+	    unused_left_vol
+	)
+};
+
+
+struct dsp_output_snoop_scb {
+	/* First 13 dwords from generic_scb_t */
+	struct dsp_basic_dma_req  basic_req;  /* Optional */
+	struct dsp_scatter_gather_ext sg_ext;  /* Optional */
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,       /* REQUIRED */
+	    sub_list_ptr    /* REQUIRED */
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,    /* REQUIRED */
+	    this_spb        /* REQUIRED */
+	)
+
+	u32 strm_rs_config; /* REQUIRED */
+	u32 strm_buf_ptr;   /* REQUIRED */
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    init_snoop_input_link,
+	    snoop_child_input_scb
+	)
+
+	u32 snoop_input_buf_ptr;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    reserved,
+	    input_scb
+	)
+};
+
+struct dsp_spio_write_scb {
+	___DSP_DUAL_16BIT_ALLOC(
+	    address1,
+	    address2
+	)
+
+	u32 data1;
+
+	u32 data2;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    address3,
+	    address4
+	)
+
+	u32 data3;
+
+	u32 data4;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    unused1,
+	    data_ptr
+	)
+
+	u32 unused2[2];
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    sibling_ptr,
+	    child_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,
+	    this_ptr
+	)
+
+	u32 unused3[5];
+};
+
+struct dsp_magic_snoop_task {
+	u32 i0;
+	u32 i1;
+
+	u32 strm_buf_ptr1;
+  
+	u16 i2;
+	u16 snoop_scb;
+
+	u32 i3;
+	u32 i4;
+	u32 i5;
+	u32 i6;
+
+	u32 i7;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    next_scb,
+	    sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    entry_point,
+	    this_ptr
+	)
+
+	u32 strm_buf_config;
+	u32 strm_buf_ptr2;
+
+	u32 i8;
+
+	struct dsp_volume_control vdec_vol_ctrl;
+};
+
+
+struct dsp_filter_scb {
+	___DSP_DUAL_16BIT_ALLOC(
+	      a0_right,          /* 0x00 */
+	      a0_left
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	      a1_right,          /* 0x01 */
+	      a1_left
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	      a2_right,          /* 0x02 */
+	      a2_left
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	      output_buf_ptr,    /* 0x03 */
+	      init
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	      filter_unused3,    /* 0x04 */
+	      filter_unused2
+	)
+
+	u32 prev_sample_output1; /* 0x05 */
+	u32 prev_sample_output2; /* 0x06 */
+	u32 prev_sample_input1;  /* 0x07 */
+	u32 prev_sample_input2;  /* 0x08 */
+
+	___DSP_DUAL_16BIT_ALLOC(
+	      next_scb_ptr,      /* 0x09 */
+	      sub_list_ptr
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	      entry_point,       /* 0x0A */
+	      spb_ptr
+	)
+
+	u32  strm_rs_config;     /* 0x0B */
+	u32  strm_buf_ptr;       /* 0x0C */
+
+	___DSP_DUAL_16BIT_ALLOC(
+              b0_right,          /* 0x0D */
+	      b0_left
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+              b1_right,          /* 0x0E */
+	      b1_left
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+              b2_right,          /* 0x0F */
+	      b2_left
+	)
+};
+#endif /* __DSP_SCB_TYPES_H__ */
diff --git a/sound/pci/cs46xx/cs46xx_dsp_spos.h b/sound/pci/cs46xx/cs46xx_dsp_spos.h
new file mode 100644
index 000000000000..8008c59288a6
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx_dsp_spos.h
@@ -0,0 +1,234 @@
+/*
+ *  The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#ifndef __CS46XX_DSP_SPOS_H__
+#define __CS46XX_DSP_SPOS_H__
+
+#include "cs46xx_dsp_scb_types.h"
+#include "cs46xx_dsp_task_types.h"
+
+#define SYMBOL_CONSTANT  0x0
+#define SYMBOL_SAMPLE    0x1
+#define SYMBOL_PARAMETER 0x2
+#define SYMBOL_CODE      0x3
+
+#define SEGTYPE_SP_PROGRAM              0x00000001
+#define SEGTYPE_SP_PARAMETER            0x00000002
+#define SEGTYPE_SP_SAMPLE               0x00000003
+#define SEGTYPE_SP_COEFFICIENT          0x00000004
+
+#define DSP_SPOS_UU      0x0deadul     /* unused */
+#define DSP_SPOS_DC      0x0badul      /* don't care */
+#define DSP_SPOS_DC_DC   0x0bad0badul  /* don't care */
+#define DSP_SPOS_UUUU    0xdeadc0edul  /* unused */
+#define DSP_SPOS_UUHI    0xdeadul
+#define DSP_SPOS_UULO    0xc0edul
+#define DSP_SPOS_DCDC    0x0badf1d0ul  /* don't care */
+#define DSP_SPOS_DCDCHI  0x0badul
+#define DSP_SPOS_DCDCLO  0xf1d0ul
+
+#define DSP_MAX_TASK_NAME   60
+#define DSP_MAX_SYMBOL_NAME 100
+#define DSP_MAX_SCB_NAME    60
+#define DSP_MAX_SCB_DESC    200
+#define DSP_MAX_TASK_DESC   50
+
+#define DSP_MAX_PCM_CHANNELS 32
+#define DSP_MAX_SRC_NR       14
+
+#define DSP_PCM_MAIN_CHANNEL        1
+#define DSP_PCM_REAR_CHANNEL        2
+#define DSP_PCM_CENTER_LFE_CHANNEL  3
+#define DSP_PCM_S71_CHANNEL         4 /* surround 7.1 */
+#define DSP_IEC958_CHANNEL          5
+
+#define DSP_SPDIF_STATUS_OUTPUT_ENABLED       1
+#define DSP_SPDIF_STATUS_PLAYBACK_OPEN        2
+#define DSP_SPDIF_STATUS_HW_ENABLED           4
+#define DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED   8
+
+struct dsp_symbol_entry {
+	u32 address;
+	char symbol_name[DSP_MAX_SYMBOL_NAME];
+	int symbol_type;
+
+	/* initialized by driver */
+	struct dsp_module_desc * module;
+	int deleted;
+};
+
+struct dsp_symbol_desc {
+	int nsymbols;
+
+	struct dsp_symbol_entry *symbols;
+
+	/* initialized by driver */
+	int highest_frag_index;
+};
+
+struct dsp_segment_desc {
+	int segment_type;
+	u32 offset;
+	u32 size;
+	u32 * data;
+};
+
+struct dsp_module_desc {
+	char * module_name;
+	struct dsp_symbol_desc symbol_table;
+	int nsegments;
+	struct dsp_segment_desc * segments;
+
+	/* initialized by driver */
+	u32 overlay_begin_address;
+	u32 load_address;
+	int nfixups;
+};
+
+struct dsp_scb_descriptor {
+	char scb_name[DSP_MAX_SCB_NAME];
+	u32 address;
+	int index;
+	u32 *data;
+
+	struct dsp_scb_descriptor * sub_list_ptr;
+	struct dsp_scb_descriptor * next_scb_ptr;
+	struct dsp_scb_descriptor * parent_scb_ptr;
+
+	struct dsp_symbol_entry * task_entry;
+	struct dsp_symbol_entry * scb_symbol;
+
+	struct snd_info_entry *proc_info;
+	int ref_count;
+
+	u16 volume[2];
+	unsigned int deleted :1;
+	unsigned int updated :1;
+	unsigned int volume_set :1;
+};
+
+struct dsp_task_descriptor {
+	char task_name[DSP_MAX_TASK_NAME];
+	int size;
+	u32 address;
+	int index;
+	u32 *data;
+};
+
+struct dsp_pcm_channel_descriptor {
+	int active;
+	int src_slot;
+	int pcm_slot;
+	u32 sample_rate;
+	u32 unlinked;
+	struct dsp_scb_descriptor * pcm_reader_scb;
+	struct dsp_scb_descriptor * src_scb;
+	struct dsp_scb_descriptor * mixer_scb;
+
+	void * private_data;
+};
+
+struct dsp_spos_instance {
+	struct dsp_symbol_desc symbol_table; /* currently available loaded symbols in SP */
+
+	int nmodules;
+	struct dsp_module_desc * modules; /* modules loaded into SP */
+
+	struct dsp_segment_desc code;
+
+	/* Main PCM playback mixer */
+	struct dsp_scb_descriptor * master_mix_scb;
+	u16 dac_volume_right;
+	u16 dac_volume_left;
+
+	/* Rear/surround PCM playback mixer */
+	struct dsp_scb_descriptor * rear_mix_scb;
+
+	/* Center/LFE mixer */
+	struct dsp_scb_descriptor * center_lfe_mix_scb;
+
+	int npcm_channels;
+	int nsrc_scb;
+	struct dsp_pcm_channel_descriptor pcm_channels[DSP_MAX_PCM_CHANNELS];
+	int src_scb_slots[DSP_MAX_SRC_NR];
+
+	/* cache this symbols */
+	struct dsp_symbol_entry * null_algorithm; /* used by PCMreaderSCB's */
+	struct dsp_symbol_entry * s16_up;         /* used by SRCtaskSCB's */
+
+	/* proc fs */  
+	struct snd_card *snd_card;
+	struct snd_info_entry * proc_dsp_dir;
+	struct snd_info_entry * proc_sym_info_entry;
+	struct snd_info_entry * proc_modules_info_entry;
+	struct snd_info_entry * proc_parameter_dump_info_entry;
+	struct snd_info_entry * proc_sample_dump_info_entry;
+
+	/* SCB's descriptors */
+	int nscb;
+	int scb_highest_frag_index;
+	struct dsp_scb_descriptor scbs[DSP_MAX_SCB_DESC];
+	struct snd_info_entry * proc_scb_info_entry;
+	struct dsp_scb_descriptor * the_null_scb;
+
+	/* Task's descriptors */
+	int ntask;
+	struct dsp_task_descriptor tasks[DSP_MAX_TASK_DESC];
+	struct snd_info_entry * proc_task_info_entry;
+
+	/* SPDIF status */
+	int spdif_status_out;
+	int spdif_status_in;
+	u16 spdif_input_volume_right;
+	u16 spdif_input_volume_left;
+	/* spdif channel status,
+	   left right and user validity bits */
+	unsigned int spdif_csuv_default;
+	unsigned int spdif_csuv_stream;
+
+	/* SPDIF input sample rate converter */
+	struct dsp_scb_descriptor * spdif_in_src;
+	/* SPDIF input asynch. receiver */
+	struct dsp_scb_descriptor * asynch_rx_scb;
+
+	/* Capture record mixer SCB */
+	struct dsp_scb_descriptor * record_mixer_scb;
+    
+	/* CODEC input SCB */
+	struct dsp_scb_descriptor * codec_in_scb;
+
+	/* reference snooper */
+	struct dsp_scb_descriptor * ref_snoop_scb;
+
+	/* SPDIF output  PCM reference  */
+	struct dsp_scb_descriptor * spdif_pcm_input_scb;
+
+	/* asynch TX task */
+	struct dsp_scb_descriptor * asynch_tx_scb;
+
+	/* record sources */
+	struct dsp_scb_descriptor * pcm_input;
+	struct dsp_scb_descriptor * adc_input;
+
+	int spdif_in_sample_rate;
+};
+
+#endif /* __DSP_SPOS_H__ */
diff --git a/sound/pci/cs46xx/cs46xx_dsp_task_types.h b/sound/pci/cs46xx/cs46xx_dsp_task_types.h
new file mode 100644
index 000000000000..5cf920bfda27
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx_dsp_task_types.h
@@ -0,0 +1,252 @@
+/*
+ *  The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ *
+ * NOTE: comments are copy/paste from cwcemb80.lst 
+ * provided by Tom Woller at Cirrus (my only
+ * documentation about the SP OS running inside
+ * the DSP) 
+ */
+
+#ifndef __CS46XX_DSP_TASK_TYPES_H__
+#define __CS46XX_DSP_TASK_TYPES_H__
+
+#include "cs46xx_dsp_scb_types.h"
+
+/*********************************************************************************************
+Example hierarchy of stream control blocks in the SP
+
+hfgTree
+Ptr____Call (c)
+       \
+ -------+------         -------------      -------------      -------------      -----
+| SBlaster IF  |______\| Foreground  |___\| Middlegr'nd |___\| Background  |___\| Nul |
+|              |Goto  /| tree header |g  /| tree header |g  /| tree header |g  /| SCB |r
+ -------------- (g)     -------------      -------------      -------------      -----
+       |c                     |c                 |c                 |c
+       |                      |                  |                  |
+      \/                  -------------      -------------      -------------   
+                       | Foreground  |_\  | Middlegr'nd |_\  | Background  |_\
+                       |     tree    |g/  |    tree     |g/  |     tree    |g/
+                        -------------      -------------      -------------   
+                              |c                 |c                 |c
+                              |                  |                  | 
+                             \/                 \/                 \/ 
+
+*********************************************************************************************/
+
+#define		HFG_FIRST_EXECUTE_MODE			0x0001
+#define		HFG_FIRST_EXECUTE_MODE_BIT		0
+#define		HFG_CONTEXT_SWITCH_MODE			0x0002
+#define		HFG_CONTEXT_SWITCH_MODE_BIT		1
+
+#define MAX_FG_STACK_SIZE 	32			/* THESE NEED TO BE COMPUTED PROPERLY */
+#define MAX_MG_STACK_SIZE 	16
+#define MAX_BG_STACK_SIZE 	9
+#define MAX_HFG_STACK_SIZE	4
+
+#define SLEEP_ACTIVE_INCREMENT		0		/* Enable task tree thread to go to sleep
+											   This should only ever be used on the Background thread */
+#define STANDARD_ACTIVE_INCREMENT	1		/* Task tree thread normal operation */
+#define SUSPEND_ACTIVE_INCREMENT	2		/* Cause execution to suspend in the task tree thread
+                                               This should only ever be used on the Background thread */
+
+#define HOSTFLAGS_DISABLE_BG_SLEEP  0       /* Host-controlled flag that determines whether we go to sleep
+                                               at the end of BG */
+
+/* Minimal context save area for Hyper Forground */
+struct dsp_hf_save_area {
+	u32	r10_save;
+	u32	r54_save;
+	u32	r98_save;
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    status_save,
+	    ind_save
+	)
+
+	___DSP_DUAL_16BIT_ALLOC(
+	    rci1_save,
+	    rci0_save
+	)
+
+	u32	r32_save;
+	u32	r76_save;
+	u32	rsd2_save;
+
+       	___DSP_DUAL_16BIT_ALLOC(
+	      rsi2_save,	  /* See TaskTreeParameterBlock for 
+				     remainder of registers  */
+	      rsa2Save
+	)
+	/* saved as part of HFG context  */
+};
+
+
+/* Task link data structure */
+struct dsp_tree_link {
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Pointer to sibling task control block */
+	    next_scb,
+	/* Pointer to child task control block */
+	    sub_ptr
+	)
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Pointer to code entry point */
+	    entry_point, 
+	/* Pointer to local data */
+	    this_spb
+	)
+};
+
+
+struct dsp_task_tree_data {
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Initial tock count; controls task tree execution rate */
+	    tock_count_limit,
+	/* Tock down counter */
+	    tock_count
+	)
+
+	/* Add to ActiveCount when TockCountLimit reached: 
+	   Subtract on task tree termination */
+	___DSP_DUAL_16BIT_ALLOC(
+	    active_tncrement,		
+	/* Number of pending activations for task tree */
+	    active_count
+	)
+
+        ___DSP_DUAL_16BIT_ALLOC(
+	/* BitNumber to enable modification of correct bit in ActiveTaskFlags */
+	    active_bit,	    
+	/* Pointer to OS location for indicating current activity on task level */
+	    active_task_flags_ptr
+	)
+
+	/* Data structure for controlling movement of memory blocks:- 
+	   currently unused */
+	___DSP_DUAL_16BIT_ALLOC(
+	    mem_upd_ptr,
+	/* Data structure for controlling synchronous link update */
+	    link_upd_ptr
+	)
+  
+	___DSP_DUAL_16BIT_ALLOC(
+	/* Save area for remainder of full context. */
+	    save_area,
+	/* Address of start of local stack for data storage */
+	    data_stack_base_ptr
+	)
+
+};
+
+
+struct dsp_interval_timer_data
+{
+	/* These data items have the same relative locations to those */
+	___DSP_DUAL_16BIT_ALLOC(
+	     interval_timer_period,
+	     itd_unused
+	)
+
+	/* used for this data in the SPOS control block for SPOS 1.0 */
+	___DSP_DUAL_16BIT_ALLOC(
+	     num_FG_ticks_this_interval,        
+	     num_intervals
+	)
+};
+
+
+/* This structure contains extra storage for the task tree
+   Currently, this additional data is related only to a full context save */
+struct dsp_task_tree_context_block {
+	/* Up to 10 values are saved onto the stack.  8 for the task tree, 1 for
+	   The access to the context switch (call or interrupt), and 1 spare that
+	   users should never use.  This last may be required by the system */
+	___DSP_DUAL_16BIT_ALLOC(
+	     stack1,
+	     stack0
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	     stack3,
+	     stack2
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	     stack5,
+	     stack4
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	     stack7,
+	     stack6
+	)
+	___DSP_DUAL_16BIT_ALLOC(
+	     stack9,
+	     stack8
+	)
+
+	u32	  saverfe;					
+
+	/* Value may be overwriten by stack save algorithm.
+	   Retain the size of the stack data saved here if used */
+	___DSP_DUAL_16BIT_ALLOC(
+             reserved1,	
+  	     stack_size
+	)
+	u32		saverba;	  /* (HFG) */
+	u32		saverdc;
+	u32		savers_config_23; /* (HFG) */
+	u32		savers_DMA23;	  /* (HFG) */
+	u32		saversa0;
+	u32		saversi0;
+	u32		saversa1;
+	u32		saversi1;
+	u32		saversa3;
+	u32		saversd0;
+	u32		saversd1;
+	u32		saversd3;
+	u32		savers_config01;
+	u32		savers_DMA01;
+	u32		saveacc0hl;
+	u32		saveacc1hl;
+	u32		saveacc0xacc1x;
+	u32		saveacc2hl;
+	u32		saveacc3hl;
+	u32		saveacc2xacc3x;
+	u32		saveaux0hl;
+	u32		saveaux1hl;
+	u32		saveaux0xaux1x;
+	u32		saveaux2hl;
+	u32		saveaux3hl;
+	u32		saveaux2xaux3x;
+	u32		savershouthl;
+	u32		savershoutxmacmode;
+};
+                
+
+struct dsp_task_tree_control_block {
+	struct dsp_hf_save_area			context;
+	struct dsp_tree_link			links;
+	struct dsp_task_tree_data		data;
+	struct dsp_task_tree_context_block	context_blk;
+	struct dsp_interval_timer_data		int_timer;
+};
+
+
+#endif /* __DSP_TASK_TYPES_H__ */
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 28b9747becc9..f75f5ffdfdfb 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -61,7 +61,7 @@
 #include <sound/info.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 
 #include <asm/io.h>
 
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index e377287192aa..56fec0bc0efb 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -32,7 +32,7 @@
 #include <sound/control.h>
 #include <sound/info.h>
 #include <sound/asoundef.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 
 #include "cs46xx_lib.h"
 #include "dsp_spos.h"
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 00b148a10239..c2c695b07f8c 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -31,7 +31,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/info.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 
 #include "cs46xx_lib.h"
 #include "dsp_spos.h"
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index f61346a555bb..d36e6ca147e1 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -26,7 +26,7 @@
 #include <linux/time.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include <sound/trident.h>
+#include "trident.h"
 #include <sound/initval.h>
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, <audio@tridentmicro.com>");
diff --git a/sound/pci/trident/trident.h b/sound/pci/trident/trident.h
new file mode 100644
index 000000000000..5f110eb56e47
--- /dev/null
+++ b/sound/pci/trident/trident.h
@@ -0,0 +1,444 @@
+#ifndef __SOUND_TRIDENT_H
+#define __SOUND_TRIDENT_H
+
+/*
+ *  audio@tridentmicro.com
+ *  Fri Feb 19 15:55:28 MST 1999
+ *  Definitions for Trident 4DWave DX/NX chips
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <sound/pcm.h>
+#include <sound/mpu401.h>
+#include <sound/ac97_codec.h>
+#include <sound/util_mem.h>
+
+#define TRIDENT_DEVICE_ID_DX		((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_DX)
+#define TRIDENT_DEVICE_ID_NX		((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_NX)
+#define TRIDENT_DEVICE_ID_SI7018	((PCI_VENDOR_ID_SI<<16)|PCI_DEVICE_ID_SI_7018)
+
+#define SNDRV_TRIDENT_VOICE_TYPE_PCM		0
+#define SNDRV_TRIDENT_VOICE_TYPE_SYNTH		1
+#define SNDRV_TRIDENT_VOICE_TYPE_MIDI		2
+
+#define SNDRV_TRIDENT_VFLG_RUNNING		(1<<0)
+
+/* TLB code constants */
+#define SNDRV_TRIDENT_PAGE_SIZE			4096
+#define SNDRV_TRIDENT_PAGE_SHIFT			12
+#define SNDRV_TRIDENT_PAGE_MASK			((1<<SNDRV_TRIDENT_PAGE_SHIFT)-1)
+#define SNDRV_TRIDENT_MAX_PAGES			4096
+
+/*
+ * Direct registers
+ */
+
+#define TRID_REG(trident, x) ((trident)->port + (x))
+
+#define ID_4DWAVE_DX        0x2000
+#define ID_4DWAVE_NX        0x2001
+
+/* Bank definitions */
+
+#define T4D_BANK_A	0
+#define T4D_BANK_B	1
+#define T4D_NUM_BANKS	2
+
+/* Register definitions */
+
+/* Global registers */
+
+enum global_control_bits {
+	CHANNEL_IDX	= 0x0000003f,
+	OVERRUN_IE	= 0x00000400,	/* interrupt enable: capture overrun */
+	UNDERRUN_IE	= 0x00000800,	/* interrupt enable: playback underrun */
+	ENDLP_IE	= 0x00001000,	/* interrupt enable: end of buffer */
+	MIDLP_IE	= 0x00002000,	/* interrupt enable: middle buffer */
+	ETOG_IE		= 0x00004000,	/* interrupt enable: envelope toggling */
+	EDROP_IE	= 0x00008000,	/* interrupt enable: envelope drop */
+	BANK_B_EN	= 0x00010000,	/* SiS: enable bank B (64 channels) */
+	PCMIN_B_MIX	= 0x00020000,	/* SiS: PCM IN B mixing enable */
+	I2S_OUT_ASSIGN	= 0x00040000,	/* SiS: I2S Out contains surround PCM */
+	SPDIF_OUT_ASSIGN= 0x00080000,	/* SiS: 0=S/PDIF L/R | 1=PCM Out FIFO */
+	MAIN_OUT_ASSIGN = 0x00100000,	/* SiS: 0=PCM Out FIFO | 1=MMC Out buffer */
+};
+
+enum miscint_bits {
+	PB_UNDERRUN_IRQ = 0x00000001, REC_OVERRUN_IRQ = 0x00000002,
+	SB_IRQ		= 0x00000004, MPU401_IRQ      = 0x00000008,
+	OPL3_IRQ        = 0x00000010, ADDRESS_IRQ     = 0x00000020,
+	ENVELOPE_IRQ    = 0x00000040, PB_UNDERRUN     = 0x00000100,
+	REC_OVERRUN	= 0x00000200, MIXER_UNDERFLOW = 0x00000400,
+	MIXER_OVERFLOW  = 0x00000800, NX_SB_IRQ_DISABLE = 0x00001000,
+        ST_TARGET_REACHED = 0x00008000,
+	PB_24K_MODE     = 0x00010000, ST_IRQ_EN       = 0x00800000,
+	ACGPIO_IRQ	= 0x01000000
+};
+
+/* T2 legacy dma control registers. */
+#define LEGACY_DMAR0                0x00  // ADR0
+#define LEGACY_DMAR4                0x04  // CNT0
+#define LEGACY_DMAR6		    0x06  // CNT0 - High bits
+#define LEGACY_DMAR11               0x0b  // MOD 
+#define LEGACY_DMAR15               0x0f  // MMR 
+
+#define T4D_START_A		     0x80
+#define T4D_STOP_A		     0x84
+#define T4D_DLY_A		     0x88
+#define T4D_SIGN_CSO_A		     0x8c
+#define T4D_CSPF_A		     0x90
+#define T4D_CSPF_B		     0xbc
+#define T4D_CEBC_A		     0x94
+#define T4D_AINT_A		     0x98
+#define T4D_AINTEN_A		     0x9c
+#define T4D_LFO_GC_CIR               0xa0
+#define T4D_MUSICVOL_WAVEVOL         0xa8
+#define T4D_SBDELTA_DELTA_R          0xac
+#define T4D_MISCINT                  0xb0
+#define T4D_START_B                  0xb4
+#define T4D_STOP_B                   0xb8
+#define T4D_SBBL_SBCL                0xc0
+#define T4D_SBCTRL_SBE2R_SBDD        0xc4
+#define T4D_STIMER		     0xc8
+#define T4D_AINT_B                   0xd8
+#define T4D_AINTEN_B                 0xdc
+#define T4D_RCI                      0x70
+
+/* MPU-401 UART */
+#define T4D_MPU401_BASE             0x20
+#define T4D_MPUR0                   0x20
+#define T4D_MPUR1                   0x21
+#define T4D_MPUR2                   0x22
+#define T4D_MPUR3                   0x23
+
+/* S/PDIF Registers */
+#define NX_SPCTRL_SPCSO             0x24
+#define NX_SPLBA                    0x28
+#define NX_SPESO                    0x2c
+#define NX_SPCSTATUS                0x64
+
+/* Joystick */
+#define GAMEPORT_GCR                0x30
+#define GAMEPORT_MODE_ADC           0x80
+#define GAMEPORT_LEGACY             0x31
+#define GAMEPORT_AXES               0x34
+
+/* NX Specific Registers */
+#define NX_TLBC                     0x6c
+
+/* Channel Registers */
+
+#define CH_START		    0xe0
+
+#define CH_DX_CSO_ALPHA_FMS         0xe0
+#define CH_DX_ESO_DELTA             0xe8
+#define CH_DX_FMC_RVOL_CVOL         0xec
+
+#define CH_NX_DELTA_CSO             0xe0
+#define CH_NX_DELTA_ESO             0xe8
+#define CH_NX_ALPHA_FMS_FMC_RVOL_CVOL 0xec
+
+#define CH_LBA                      0xe4
+#define CH_GVSEL_PAN_VOL_CTRL_EC    0xf0
+#define CH_EBUF1                    0xf4
+#define CH_EBUF2                    0xf8
+
+/* AC-97 Registers */
+
+#define DX_ACR0_AC97_W              0x40
+#define DX_ACR1_AC97_R              0x44
+#define DX_ACR2_AC97_COM_STAT       0x48
+
+#define NX_ACR0_AC97_COM_STAT       0x40
+#define NX_ACR1_AC97_W              0x44
+#define NX_ACR2_AC97_R_PRIMARY      0x48
+#define NX_ACR3_AC97_R_SECONDARY    0x4c
+
+#define SI_AC97_WRITE		    0x40
+#define SI_AC97_READ		    0x44
+#define SI_SERIAL_INTF_CTRL	    0x48
+#define SI_AC97_GPIO		    0x4c
+#define SI_ASR0			    0x50
+#define SI_SPDIF_CS		    0x70
+#define SI_GPIO			    0x7c
+
+enum trident_nx_ac97_bits {
+	/* ACR1-3 */
+	NX_AC97_BUSY_WRITE 	= 0x0800,
+	NX_AC97_BUSY_READ	= 0x0800,
+	NX_AC97_BUSY_DATA 	= 0x0400,
+	NX_AC97_WRITE_SECONDARY = 0x0100,
+	/* ACR0 */
+	NX_AC97_SECONDARY_READY = 0x0040,
+	NX_AC97_SECONDARY_RECORD = 0x0020,
+	NX_AC97_SURROUND_OUTPUT = 0x0010,
+	NX_AC97_PRIMARY_READY	= 0x0008,
+	NX_AC97_PRIMARY_RECORD	= 0x0004,
+	NX_AC97_PCM_OUTPUT	= 0x0002,
+	NX_AC97_WARM_RESET	= 0x0001
+};
+
+enum trident_dx_ac97_bits {
+	DX_AC97_BUSY_WRITE	= 0x8000,
+	DX_AC97_BUSY_READ	= 0x8000,
+	DX_AC97_READY		= 0x0010,
+	DX_AC97_RECORD		= 0x0008,
+	DX_AC97_PLAYBACK	= 0x0002
+};
+
+enum sis7018_ac97_bits {
+	SI_AC97_BUSY_WRITE =	0x00008000,
+	SI_AC97_AUDIO_BUSY =	0x00004000,
+	SI_AC97_MODEM_BUSY =	0x00002000,
+	SI_AC97_BUSY_READ =	0x00008000,
+	SI_AC97_SECONDARY =	0x00000080,
+};
+
+enum serial_intf_ctrl_bits {
+	WARM_RESET	= 0x00000001,
+	COLD_RESET	= 0x00000002,
+	I2S_CLOCK	= 0x00000004,
+	PCM_SEC_AC97	= 0x00000008,
+	AC97_DBL_RATE	= 0x00000010,
+	SPDIF_EN	= 0x00000020,
+	I2S_OUTPUT_EN	= 0x00000040,
+	I2S_INPUT_EN	= 0x00000080,
+	PCMIN		= 0x00000100,
+	LINE1IN		= 0x00000200,
+	MICIN		= 0x00000400,
+	LINE2IN		= 0x00000800,
+	HEAD_SET_IN	= 0x00001000,
+	GPIOIN		= 0x00002000,
+	/* 7018 spec says id = 01 but the demo board routed to 10
+	   SECONDARY_ID= 0x00004000, */
+	SECONDARY_ID	= 0x00004000,
+	PCMOUT		= 0x00010000,
+	SURROUT		= 0x00020000,
+	CENTEROUT	= 0x00040000,
+	LFEOUT		= 0x00080000,
+	LINE1OUT	= 0x00100000,
+	LINE2OUT	= 0x00200000,
+	GPIOOUT		= 0x00400000,
+	SI_AC97_PRIMARY_READY = 0x01000000,
+	SI_AC97_SECONDARY_READY = 0x02000000,
+	SI_AC97_POWERDOWN = 0x04000000,
+};
+                                                                                                                                   
+/* PCM defaults */
+
+#define T4D_DEFAULT_PCM_VOL	10	/* 0 - 255 */
+#define T4D_DEFAULT_PCM_PAN	0	/* 0 - 127 */
+#define T4D_DEFAULT_PCM_RVOL	127	/* 0 - 127 */
+#define T4D_DEFAULT_PCM_CVOL	127	/* 0 - 127 */
+
+struct snd_trident;
+struct snd_trident_voice;
+struct snd_trident_pcm_mixer;
+
+struct snd_trident_port {
+	struct snd_midi_channel_set * chset;
+	struct snd_trident * trident;
+	int mode;		/* operation mode */
+	int client;		/* sequencer client number */
+	int port;		/* sequencer port number */
+	unsigned int midi_has_voices: 1;
+};
+
+struct snd_trident_memblk_arg {
+	short first_page, last_page;
+};
+
+struct snd_trident_tlb {
+	unsigned int * entries;		/* 16k-aligned TLB table */
+	dma_addr_t entries_dmaaddr;	/* 16k-aligned PCI address to TLB table */
+	unsigned long * shadow_entries;	/* shadow entries with virtual addresses */
+	struct snd_dma_buffer buffer;
+	struct snd_util_memhdr * memhdr;	/* page allocation list */
+	struct snd_dma_buffer silent_page;
+};
+
+struct snd_trident_voice {
+	unsigned int number;
+	unsigned int use: 1,
+	    pcm: 1,
+	    synth:1,
+	    midi: 1;
+	unsigned int flags;
+	unsigned char client;
+	unsigned char port;
+	unsigned char index;
+
+	struct snd_trident_sample_ops *sample_ops;
+
+	/* channel parameters */
+	unsigned int CSO;		/* 24 bits (16 on DX) */
+	unsigned int ESO;		/* 24 bits (16 on DX) */
+	unsigned int LBA;		/* 30 bits */
+	unsigned short EC;		/* 12 bits */
+	unsigned short Alpha;		/* 12 bits */
+	unsigned short Delta;		/* 16 bits */
+	unsigned short Attribute;	/* 16 bits - SiS 7018 */
+	unsigned short Vol;		/* 12 bits (6.6) */
+	unsigned char Pan;		/* 7 bits (1.4.2) */
+	unsigned char GVSel;		/* 1 bit */
+	unsigned char RVol;		/* 7 bits (5.2) */
+	unsigned char CVol;		/* 7 bits (5.2) */
+	unsigned char FMC;		/* 2 bits */
+	unsigned char CTRL;		/* 4 bits */
+	unsigned char FMS;		/* 4 bits */
+	unsigned char LFO;		/* 8 bits */
+
+	unsigned int negCSO;	/* nonzero - use negative CSO */
+
+	struct snd_util_memblk *memblk;	/* memory block if TLB enabled */
+
+	/* PCM data */
+
+	struct snd_trident *trident;
+	struct snd_pcm_substream *substream;
+	struct snd_trident_voice *extra;	/* extra PCM voice (acts as interrupt generator) */
+	unsigned int running: 1,
+            capture: 1,
+            spdif: 1,
+            foldback: 1,
+            isync: 1,
+            isync2: 1,
+            isync3: 1;
+	int foldback_chan;		/* foldback subdevice number */
+	unsigned int stimer;		/* global sample timer (to detect spurious interrupts) */
+	unsigned int spurious_threshold; /* spurious threshold */
+	unsigned int isync_mark;
+	unsigned int isync_max;
+	unsigned int isync_ESO;
+
+	/* --- */
+
+	void *private_data;
+	void (*private_free)(struct snd_trident_voice *voice);
+};
+
+struct snd_4dwave {
+	int seq_client;
+
+	struct snd_trident_port seq_ports[4];
+	struct snd_trident_voice voices[64];	
+
+	int ChanSynthCount;		/* number of allocated synth channels */
+	int max_size;			/* maximum synth memory size in bytes */
+	int current_size;		/* current allocated synth mem in bytes */
+};
+
+struct snd_trident_pcm_mixer {
+	struct snd_trident_voice *voice;	/* active voice */
+	unsigned short vol;		/* front volume */
+	unsigned char pan;		/* pan control */
+	unsigned char rvol;		/* rear volume */
+	unsigned char cvol;		/* center volume */
+	unsigned char pad;
+};
+
+struct snd_trident {
+	int irq;
+
+	unsigned int device;	/* device ID */
+
+        unsigned char  bDMAStart;
+
+	unsigned long port;
+	unsigned long midi_port;
+
+	unsigned int spurious_irq_count;
+	unsigned int spurious_irq_max_delta;
+
+        struct snd_trident_tlb tlb;	/* TLB entries for NX cards */
+
+	unsigned char spdif_ctrl;
+	unsigned char spdif_pcm_ctrl;
+	unsigned int spdif_bits;
+	unsigned int spdif_pcm_bits;
+	struct snd_kcontrol *spdif_pcm_ctl;	/* S/PDIF settings */
+	unsigned int ac97_ctrl;
+        
+        unsigned int ChanMap[2];	/* allocation map for hardware channels */
+        
+        int ChanPCM;			/* max number of PCM channels */
+	int ChanPCMcnt;			/* actual number of PCM channels */
+
+	unsigned int ac97_detect: 1;	/* 1 = AC97 in detection phase */
+	unsigned int in_suspend: 1;	/* 1 during suspend/resume */
+
+	struct snd_4dwave synth;	/* synth specific variables */
+
+	spinlock_t event_lock;
+	spinlock_t voice_alloc;
+
+	struct snd_dma_device dma_dev;
+
+	struct pci_dev *pci;
+	struct snd_card *card;
+	struct snd_pcm *pcm;		/* ADC/DAC PCM */
+	struct snd_pcm *foldback;	/* Foldback PCM */
+	struct snd_pcm *spdif;	/* SPDIF PCM */
+	struct snd_rawmidi *rmidi;
+
+	struct snd_ac97_bus *ac97_bus;
+	struct snd_ac97 *ac97;
+	struct snd_ac97 *ac97_sec;
+
+	unsigned int musicvol_wavevol;
+	struct snd_trident_pcm_mixer pcm_mixer[32];
+	struct snd_kcontrol *ctl_vol;	/* front volume */
+	struct snd_kcontrol *ctl_pan;	/* pan */
+	struct snd_kcontrol *ctl_rvol;	/* rear volume */
+	struct snd_kcontrol *ctl_cvol;	/* center volume */
+
+	spinlock_t reg_lock;
+
+	struct gameport *gameport;
+};
+
+int snd_trident_create(struct snd_card *card,
+		       struct pci_dev *pci,
+		       int pcm_streams,
+		       int pcm_spdif_device,
+		       int max_wavetable_size,
+		       struct snd_trident ** rtrident);
+int snd_trident_create_gameport(struct snd_trident *trident);
+
+int snd_trident_pcm(struct snd_trident * trident, int device, struct snd_pcm **rpcm);
+int snd_trident_foldback_pcm(struct snd_trident * trident, int device, struct snd_pcm **rpcm);
+int snd_trident_spdif_pcm(struct snd_trident * trident, int device, struct snd_pcm **rpcm);
+int snd_trident_attach_synthesizer(struct snd_trident * trident);
+struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, int type,
+					     int client, int port);
+void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice);
+void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice);
+void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice);
+void snd_trident_write_voice_regs(struct snd_trident * trident, struct snd_trident_voice *voice);
+extern const struct dev_pm_ops snd_trident_pm;
+
+/* TLB memory allocation */
+struct snd_util_memblk *snd_trident_alloc_pages(struct snd_trident *trident,
+						struct snd_pcm_substream *substream);
+int snd_trident_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk);
+struct snd_util_memblk *snd_trident_synth_alloc(struct snd_trident *trident, unsigned int size);
+int snd_trident_synth_free(struct snd_trident *trident, struct snd_util_memblk *blk);
+int snd_trident_synth_copy_from_user(struct snd_trident *trident, struct snd_util_memblk *blk,
+				     int offset, const char __user *data, int size);
+
+#endif /* __SOUND_TRIDENT_H */
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index b4430c093bad..94011dcae731 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -41,7 +41,7 @@
 #include <sound/info.h>
 #include <sound/control.h>
 #include <sound/tlv.h>
-#include <sound/trident.h>
+#include "trident.h"
 #include <sound/asoundef.h>
 
 #include <asm/io.h>
diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c
index f9779e23fe57..3102a579660b 100644
--- a/sound/pci/trident/trident_memory.c
+++ b/sound/pci/trident/trident_memory.c
@@ -29,7 +29,7 @@
 #include <linux/mutex.h>
 
 #include <sound/core.h>
-#include <sound/trident.h>
+#include "trident.h"
 
 /* page arguments of these two macros are Trident page (4096 bytes), not like
  * aligned pages in others
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 7e20ddb9123a..4810356b97ba 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -24,7 +24,7 @@
 #include <linux/time.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include <sound/ymfpci.h>
+#include "ymfpci.h"
 #include <sound/mpu401.h>
 #include <sound/opl3.h>
 #include <sound/initval.h>
diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h
new file mode 100644
index 000000000000..bddc4052286b
--- /dev/null
+++ b/sound/pci/ymfpci/ymfpci.h
@@ -0,0 +1,389 @@
+#ifndef __SOUND_YMFPCI_H
+#define __SOUND_YMFPCI_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *  Definitions for Yahama YMF724/740/744/754 chips
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/ac97_codec.h>
+#include <sound/timer.h>
+#include <linux/gameport.h>
+
+/*
+ *  Direct registers
+ */
+
+#define YMFREG(chip, reg)		(chip->port + YDSXGR_##reg)
+
+#define	YDSXGR_INTFLAG			0x0004
+#define	YDSXGR_ACTIVITY			0x0006
+#define	YDSXGR_GLOBALCTRL		0x0008
+#define	YDSXGR_ZVCTRL			0x000A
+#define	YDSXGR_TIMERCTRL		0x0010
+#define	YDSXGR_TIMERCOUNT		0x0012
+#define	YDSXGR_SPDIFOUTCTRL		0x0018
+#define	YDSXGR_SPDIFOUTSTATUS		0x001C
+#define	YDSXGR_EEPROMCTRL		0x0020
+#define	YDSXGR_SPDIFINCTRL		0x0034
+#define	YDSXGR_SPDIFINSTATUS		0x0038
+#define	YDSXGR_DSPPROGRAMDL		0x0048
+#define	YDSXGR_DLCNTRL			0x004C
+#define	YDSXGR_GPIOININTFLAG		0x0050
+#define	YDSXGR_GPIOININTENABLE		0x0052
+#define	YDSXGR_GPIOINSTATUS		0x0054
+#define	YDSXGR_GPIOOUTCTRL		0x0056
+#define	YDSXGR_GPIOFUNCENABLE		0x0058
+#define	YDSXGR_GPIOTYPECONFIG		0x005A
+#define	YDSXGR_AC97CMDDATA		0x0060
+#define	YDSXGR_AC97CMDADR		0x0062
+#define	YDSXGR_PRISTATUSDATA		0x0064
+#define	YDSXGR_PRISTATUSADR		0x0066
+#define	YDSXGR_SECSTATUSDATA		0x0068
+#define	YDSXGR_SECSTATUSADR		0x006A
+#define	YDSXGR_SECCONFIG		0x0070
+#define	YDSXGR_LEGACYOUTVOL		0x0080
+#define	YDSXGR_LEGACYOUTVOLL		0x0080
+#define	YDSXGR_LEGACYOUTVOLR		0x0082
+#define	YDSXGR_NATIVEDACOUTVOL		0x0084
+#define	YDSXGR_NATIVEDACOUTVOLL		0x0084
+#define	YDSXGR_NATIVEDACOUTVOLR		0x0086
+#define	YDSXGR_ZVOUTVOL			0x0088
+#define	YDSXGR_ZVOUTVOLL		0x0088
+#define	YDSXGR_ZVOUTVOLR		0x008A
+#define	YDSXGR_SECADCOUTVOL		0x008C
+#define	YDSXGR_SECADCOUTVOLL		0x008C
+#define	YDSXGR_SECADCOUTVOLR		0x008E
+#define	YDSXGR_PRIADCOUTVOL		0x0090
+#define	YDSXGR_PRIADCOUTVOLL		0x0090
+#define	YDSXGR_PRIADCOUTVOLR		0x0092
+#define	YDSXGR_LEGACYLOOPVOL		0x0094
+#define	YDSXGR_LEGACYLOOPVOLL		0x0094
+#define	YDSXGR_LEGACYLOOPVOLR		0x0096
+#define	YDSXGR_NATIVEDACLOOPVOL		0x0098
+#define	YDSXGR_NATIVEDACLOOPVOLL	0x0098
+#define	YDSXGR_NATIVEDACLOOPVOLR	0x009A
+#define	YDSXGR_ZVLOOPVOL		0x009C
+#define	YDSXGR_ZVLOOPVOLL		0x009E
+#define	YDSXGR_ZVLOOPVOLR		0x009E
+#define	YDSXGR_SECADCLOOPVOL		0x00A0
+#define	YDSXGR_SECADCLOOPVOLL		0x00A0
+#define	YDSXGR_SECADCLOOPVOLR		0x00A2
+#define	YDSXGR_PRIADCLOOPVOL		0x00A4
+#define	YDSXGR_PRIADCLOOPVOLL		0x00A4
+#define	YDSXGR_PRIADCLOOPVOLR		0x00A6
+#define	YDSXGR_NATIVEADCINVOL		0x00A8
+#define	YDSXGR_NATIVEADCINVOLL		0x00A8
+#define	YDSXGR_NATIVEADCINVOLR		0x00AA
+#define	YDSXGR_NATIVEDACINVOL		0x00AC
+#define	YDSXGR_NATIVEDACINVOLL		0x00AC
+#define	YDSXGR_NATIVEDACINVOLR		0x00AE
+#define	YDSXGR_BUF441OUTVOL		0x00B0
+#define	YDSXGR_BUF441OUTVOLL		0x00B0
+#define	YDSXGR_BUF441OUTVOLR		0x00B2
+#define	YDSXGR_BUF441LOOPVOL		0x00B4
+#define	YDSXGR_BUF441LOOPVOLL		0x00B4
+#define	YDSXGR_BUF441LOOPVOLR		0x00B6
+#define	YDSXGR_SPDIFOUTVOL		0x00B8
+#define	YDSXGR_SPDIFOUTVOLL		0x00B8
+#define	YDSXGR_SPDIFOUTVOLR		0x00BA
+#define	YDSXGR_SPDIFLOOPVOL		0x00BC
+#define	YDSXGR_SPDIFLOOPVOLL		0x00BC
+#define	YDSXGR_SPDIFLOOPVOLR		0x00BE
+#define	YDSXGR_ADCSLOTSR		0x00C0
+#define	YDSXGR_RECSLOTSR		0x00C4
+#define	YDSXGR_ADCFORMAT		0x00C8
+#define	YDSXGR_RECFORMAT		0x00CC
+#define	YDSXGR_P44SLOTSR		0x00D0
+#define	YDSXGR_STATUS			0x0100
+#define	YDSXGR_CTRLSELECT		0x0104
+#define	YDSXGR_MODE			0x0108
+#define	YDSXGR_SAMPLECOUNT		0x010C
+#define	YDSXGR_NUMOFSAMPLES		0x0110
+#define	YDSXGR_CONFIG			0x0114
+#define	YDSXGR_PLAYCTRLSIZE		0x0140
+#define	YDSXGR_RECCTRLSIZE		0x0144
+#define	YDSXGR_EFFCTRLSIZE		0x0148
+#define	YDSXGR_WORKSIZE			0x014C
+#define	YDSXGR_MAPOFREC			0x0150
+#define	YDSXGR_MAPOFEFFECT		0x0154
+#define	YDSXGR_PLAYCTRLBASE		0x0158
+#define	YDSXGR_RECCTRLBASE		0x015C
+#define	YDSXGR_EFFCTRLBASE		0x0160
+#define	YDSXGR_WORKBASE			0x0164
+#define	YDSXGR_DSPINSTRAM		0x1000
+#define	YDSXGR_CTRLINSTRAM		0x4000
+
+#define YDSXG_AC97READCMD		0x8000
+#define YDSXG_AC97WRITECMD		0x0000
+
+#define PCIR_DSXG_LEGACY		0x40
+#define PCIR_DSXG_ELEGACY		0x42
+#define PCIR_DSXG_CTRL			0x48
+#define PCIR_DSXG_PWRCTRL1		0x4a
+#define PCIR_DSXG_PWRCTRL2		0x4e
+#define PCIR_DSXG_FMBASE		0x60
+#define PCIR_DSXG_SBBASE		0x62
+#define PCIR_DSXG_MPU401BASE		0x64
+#define PCIR_DSXG_JOYBASE		0x66
+
+#define YDSXG_DSPLENGTH			0x0080
+#define YDSXG_CTRLLENGTH		0x3000
+
+#define YDSXG_DEFAULT_WORK_SIZE		0x0400
+
+#define YDSXG_PLAYBACK_VOICES		64
+#define YDSXG_CAPTURE_VOICES		2
+#define YDSXG_EFFECT_VOICES		5
+
+#define YMFPCI_LEGACY_SBEN	(1 << 0)	/* soundblaster enable */
+#define YMFPCI_LEGACY_FMEN	(1 << 1)	/* OPL3 enable */
+#define YMFPCI_LEGACY_JPEN	(1 << 2)	/* joystick enable */
+#define YMFPCI_LEGACY_MEN	(1 << 3)	/* MPU401 enable */
+#define YMFPCI_LEGACY_MIEN	(1 << 4)	/* MPU RX irq enable */
+#define YMFPCI_LEGACY_IOBITS	(1 << 5)	/* i/o bits range, 0 = 16bit, 1 =10bit */
+#define YMFPCI_LEGACY_SDMA	(3 << 6)	/* SB DMA select */
+#define YMFPCI_LEGACY_SBIRQ	(7 << 8)	/* SB IRQ select */
+#define YMFPCI_LEGACY_MPUIRQ	(7 << 11)	/* MPU IRQ select */
+#define YMFPCI_LEGACY_SIEN	(1 << 14)	/* serialized IRQ */
+#define YMFPCI_LEGACY_LAD	(1 << 15)	/* legacy audio disable */
+
+#define YMFPCI_LEGACY2_FMIO	(3 << 0)	/* OPL3 i/o address (724/740) */
+#define YMFPCI_LEGACY2_SBIO	(3 << 2)	/* SB i/o address (724/740) */
+#define YMFPCI_LEGACY2_MPUIO	(3 << 4)	/* MPU401 i/o address (724/740) */
+#define YMFPCI_LEGACY2_JSIO	(3 << 6)	/* joystick i/o address (724/740) */
+#define YMFPCI_LEGACY2_MAIM	(1 << 8)	/* MPU401 ack intr mask */
+#define YMFPCI_LEGACY2_SMOD	(3 << 11)	/* SB DMA mode */
+#define YMFPCI_LEGACY2_SBVER	(3 << 13)	/* SB version select */
+#define YMFPCI_LEGACY2_IMOD	(1 << 15)	/* legacy IRQ mode */
+/* SIEN:IMOD 0:0 = legacy irq, 0:1 = INTA, 1:0 = serialized IRQ */
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK
+#endif
+
+/*
+ *
+ */
+
+struct snd_ymfpci_playback_bank {
+	u32 format;
+	u32 loop_default;
+	u32 base;			/* 32-bit address */
+	u32 loop_start;			/* 32-bit offset */
+	u32 loop_end;			/* 32-bit offset */
+	u32 loop_frac;			/* 8-bit fraction - loop_start */
+	u32 delta_end;			/* pitch delta end */
+	u32 lpfK_end;
+	u32 eg_gain_end;
+	u32 left_gain_end;
+	u32 right_gain_end;
+	u32 eff1_gain_end;
+	u32 eff2_gain_end;
+	u32 eff3_gain_end;
+	u32 lpfQ;
+	u32 status;
+	u32 num_of_frames;
+	u32 loop_count;
+	u32 start;
+	u32 start_frac;
+	u32 delta;
+	u32 lpfK;
+	u32 eg_gain;
+	u32 left_gain;
+	u32 right_gain;
+	u32 eff1_gain;
+	u32 eff2_gain;
+	u32 eff3_gain;
+	u32 lpfD1;
+	u32 lpfD2;
+ };
+
+struct snd_ymfpci_capture_bank {
+	u32 base;			/* 32-bit address */
+	u32 loop_end;			/* 32-bit offset */
+	u32 start;			/* 32-bit offset */
+	u32 num_of_loops;		/* counter */
+};
+
+struct snd_ymfpci_effect_bank {
+	u32 base;			/* 32-bit address */
+	u32 loop_end;			/* 32-bit offset */
+	u32 start;			/* 32-bit offset */
+	u32 temp;
+};
+
+struct snd_ymfpci_pcm;
+struct snd_ymfpci;
+
+enum snd_ymfpci_voice_type {
+	YMFPCI_PCM,
+	YMFPCI_SYNTH,
+	YMFPCI_MIDI
+};
+
+struct snd_ymfpci_voice {
+	struct snd_ymfpci *chip;
+	int number;
+	unsigned int use: 1,
+	    pcm: 1,
+	    synth: 1,
+	    midi: 1;
+	struct snd_ymfpci_playback_bank *bank;
+	dma_addr_t bank_addr;
+	void (*interrupt)(struct snd_ymfpci *chip, struct snd_ymfpci_voice *voice);
+	struct snd_ymfpci_pcm *ypcm;
+};
+
+enum snd_ymfpci_pcm_type {
+	PLAYBACK_VOICE,
+	CAPTURE_REC,
+	CAPTURE_AC97,
+	EFFECT_DRY_LEFT,
+	EFFECT_DRY_RIGHT,
+	EFFECT_EFF1,
+	EFFECT_EFF2,
+	EFFECT_EFF3
+};
+
+struct snd_ymfpci_pcm {
+	struct snd_ymfpci *chip;
+	enum snd_ymfpci_pcm_type type;
+	struct snd_pcm_substream *substream;
+	struct snd_ymfpci_voice *voices[2];	/* playback only */
+	unsigned int running: 1,
+		     use_441_slot: 1,
+	             output_front: 1,
+	             output_rear: 1,
+	             swap_rear: 1;
+	unsigned int update_pcm_vol;
+	u32 period_size;		/* cached from runtime->period_size */
+	u32 buffer_size;		/* cached from runtime->buffer_size */
+	u32 period_pos;
+	u32 last_pos;
+	u32 capture_bank_number;
+	u32 shift;
+};
+
+struct snd_ymfpci {
+	int irq;
+
+	unsigned int device_id;	/* PCI device ID */
+	unsigned char rev;	/* PCI revision */
+	unsigned long reg_area_phys;
+	void __iomem *reg_area_virt;
+	struct resource *res_reg_area;
+	struct resource *fm_res;
+	struct resource *mpu_res;
+
+	unsigned short old_legacy_ctrl;
+#ifdef SUPPORT_JOYSTICK
+	struct gameport *gameport;
+#endif
+
+	struct snd_dma_buffer work_ptr;
+
+	unsigned int bank_size_playback;
+	unsigned int bank_size_capture;
+	unsigned int bank_size_effect;
+	unsigned int work_size;
+
+	void *bank_base_playback;
+	void *bank_base_capture;
+	void *bank_base_effect;
+	void *work_base;
+	dma_addr_t bank_base_playback_addr;
+	dma_addr_t bank_base_capture_addr;
+	dma_addr_t bank_base_effect_addr;
+	dma_addr_t work_base_addr;
+	struct snd_dma_buffer ac3_tmp_base;
+
+	u32 *ctrl_playback;
+	struct snd_ymfpci_playback_bank *bank_playback[YDSXG_PLAYBACK_VOICES][2];
+	struct snd_ymfpci_capture_bank *bank_capture[YDSXG_CAPTURE_VOICES][2];
+	struct snd_ymfpci_effect_bank *bank_effect[YDSXG_EFFECT_VOICES][2];
+
+	int start_count;
+
+	u32 active_bank;
+	struct snd_ymfpci_voice voices[64];
+	int src441_used;
+
+	struct snd_ac97_bus *ac97_bus;
+	struct snd_ac97 *ac97;
+	struct snd_rawmidi *rawmidi;
+	struct snd_timer *timer;
+	unsigned int timer_ticks;
+
+	struct pci_dev *pci;
+	struct snd_card *card;
+	struct snd_pcm *pcm;
+	struct snd_pcm *pcm2;
+	struct snd_pcm *pcm_spdif;
+	struct snd_pcm *pcm_4ch;
+	struct snd_pcm_substream *capture_substream[YDSXG_CAPTURE_VOICES];
+	struct snd_pcm_substream *effect_substream[YDSXG_EFFECT_VOICES];
+	struct snd_kcontrol *ctl_vol_recsrc;
+	struct snd_kcontrol *ctl_vol_adcrec;
+	struct snd_kcontrol *ctl_vol_spdifrec;
+	unsigned short spdif_bits, spdif_pcm_bits;
+	struct snd_kcontrol *spdif_pcm_ctl;
+	int mode_dup4ch;
+	int rear_opened;
+	int spdif_opened;
+	struct snd_ymfpci_pcm_mixer {
+		u16 left;
+		u16 right;
+		struct snd_kcontrol *ctl;
+	} pcm_mixer[32];
+
+	spinlock_t reg_lock;
+	spinlock_t voice_lock;
+	wait_queue_head_t interrupt_sleep;
+	atomic_t interrupt_sleep_count;
+	struct snd_info_entry *proc_entry;
+	const struct firmware *dsp_microcode;
+	const struct firmware *controller_microcode;
+
+#ifdef CONFIG_PM
+	u32 *saved_regs;
+	u32 saved_ydsxgr_mode;
+	u16 saved_dsxg_legacy;
+	u16 saved_dsxg_elegacy;
+#endif
+};
+
+int snd_ymfpci_create(struct snd_card *card,
+		      struct pci_dev *pci,
+		      unsigned short old_legacy_ctrl,
+		      struct snd_ymfpci ** rcodec);
+void snd_ymfpci_free_gameport(struct snd_ymfpci *chip);
+
+extern const struct dev_pm_ops snd_ymfpci_pm;
+
+int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm);
+int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm);
+int snd_ymfpci_pcm_spdif(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm);
+int snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm);
+int snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch);
+int snd_ymfpci_timer(struct snd_ymfpci *chip, int device);
+
+#endif /* __SOUND_YMFPCI_H */
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index c706901d6ff6..62b23635b754 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -33,7 +33,7 @@
 #include <sound/control.h>
 #include <sound/info.h>
 #include <sound/tlv.h>
-#include <sound/ymfpci.h>
+#include "ymfpci.h"
 #include <sound/asoundef.h>
 #include <sound/mpu401.h>
 
-- 
cgit v1.2.3


From 0920c9b4c4d896025a560e4510d473dfd41c3dcd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 3 Jul 2012 16:58:48 +0200
Subject: ALSA: hda - Remove beep_mode=2

The beep_mode=2 option was introduced to make the beep mixer
controlling the beep input allocation/deallocation dynamically, so
that a user can switch between HD-audio codec digital beep and the
system beep only via mixer API.  This was necessary because the
keyboard driver took only the first input beep instance at that time.

However, the recent keyboard driver already processes the multiple
input instances, thus there is no point to keep this mode.

Let's remove it.

Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig     |  7 +++----
 sound/pci/hda/hda_beep.c  | 37 +------------------------------------
 sound/pci/hda/hda_beep.h  |  4 ----
 sound/pci/hda/hda_intel.c |  6 +++---
 4 files changed, 7 insertions(+), 47 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index d03079764189..194d625c1f83 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -53,15 +53,14 @@ config SND_HDA_INPUT_BEEP
 	  driver. This interface is used to generate digital beeps.
 
 config SND_HDA_INPUT_BEEP_MODE
-	int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)"
+	int "Digital beep registration mode (0=off, 1=on)"
 	depends on SND_HDA_INPUT_BEEP=y
 	default "1"
-	range 0 2
+	range 0 1
 	help
 	  Set 0 to disable the digital beep interface for HD-audio by default.
 	  Set 1 to always enable the digital beep interface for HD-audio by
-	  default. Set 2 to control the beep device registration to input
-	  layer using a "Beep Switch" in mixer applications.
+	  default.
 
 config SND_HDA_INPUT_JACK
 	bool "Support jack plugging notification via input layer"
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 60738e52b8f9..662de6e58b6f 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -162,28 +162,6 @@ static int snd_hda_do_attach(struct hda_beep *beep)
 	return 0;
 }
 
-static void snd_hda_do_register(struct work_struct *work)
-{
-	struct hda_beep *beep =
-		container_of(work, struct hda_beep, register_work);
-
-	mutex_lock(&beep->mutex);
-	if (beep->enabled && !beep->dev)
-		snd_hda_do_attach(beep);
-	mutex_unlock(&beep->mutex);
-}
-
-static void snd_hda_do_unregister(struct work_struct *work)
-{
-	struct hda_beep *beep =
-		container_of(work, struct hda_beep, unregister_work.work);
-
-	mutex_lock(&beep->mutex);
-	if (!beep->enabled && beep->dev)
-		snd_hda_do_detach(beep);
-	mutex_unlock(&beep->mutex);
-}
-
 int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 {
 	struct hda_beep *beep = codec->beep;
@@ -197,15 +175,6 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 			snd_hda_codec_write(beep->codec, beep->nid, 0,
 						  AC_VERB_SET_BEEP_CONTROL, 0);
 		}
-		if (beep->mode == HDA_BEEP_MODE_SWREG) {
-			if (enable) {
-				cancel_delayed_work(&beep->unregister_work);
-				schedule_work(&beep->register_work);
-			} else {
-				schedule_delayed_work(&beep->unregister_work,
-									   HZ);
-			}
-		}
 		return 1;
 	}
 	return 0;
@@ -235,12 +204,10 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	beep->mode = codec->beep_mode;
 	codec->beep = beep;
 
-	INIT_WORK(&beep->register_work, &snd_hda_do_register);
-	INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister);
 	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
 	mutex_init(&beep->mutex);
 
-	if (beep->mode == HDA_BEEP_MODE_ON) {
+	if (beep->mode) {
 		int err = snd_hda_do_attach(beep);
 		if (err < 0) {
 			kfree(beep);
@@ -257,8 +224,6 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 {
 	struct hda_beep *beep = codec->beep;
 	if (beep) {
-		cancel_work_sync(&beep->register_work);
-		cancel_delayed_work(&beep->unregister_work);
 		if (beep->dev)
 			snd_hda_do_detach(beep);
 		codec->beep = NULL;
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 55f0647458c7..30e79d16f9f8 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -26,7 +26,6 @@
 
 #define HDA_BEEP_MODE_OFF	0
 #define HDA_BEEP_MODE_ON	1
-#define HDA_BEEP_MODE_SWREG	2
 
 /* beep information */
 struct hda_beep {
@@ -37,10 +36,7 @@ struct hda_beep {
 	int tone;
 	hda_nid_t nid;
 	unsigned int enabled:1;
-	unsigned int request_enable:1;
 	unsigned int linear_tone:1;	/* linear tone for IDT/STAC codec */
-	struct work_struct register_work; /* registration work */
-	struct delayed_work unregister_work; /* unregistration work */
 	struct work_struct beep_work; /* scheduled task for beep event */
 	struct mutex mutex;
 };
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7757536b9d5f..334c0ba7d04b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -72,7 +72,7 @@ static int enable_msi = -1;
 static char *patch[SNDRV_CARDS];
 #endif
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] =
+static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] =
 					CONFIG_SND_HDA_INPUT_BEEP_MODE};
 #endif
 
@@ -103,9 +103,9 @@ module_param_array(patch, charp, NULL, 0444);
 MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface.");
 #endif
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-module_param_array(beep_mode, int, NULL, 0444);
+module_param_array(beep_mode, bool, NULL, 0444);
 MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode "
-			    "(0=off, 1=on, 2=mute switch on/off) (default=1).");
+			    "(0=off, 1=on) (default=1).");
 #endif
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-- 
cgit v1.2.3


From 0401e8548eace5bdb8adfa3e82f56165982cb3ad Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 3 Jul 2012 17:27:57 +0200
Subject: ALSA: hda - Move beep helper functions to hda_beep.c

Move snd_hda_mixer_amp_switch_put_beep() to hda_beep.c as a clean up
to remove one more ifdef.

Also add the corresponding get callback to return consistently the
digital beep state independently from the mixer amp value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c  | 28 ++++++++++++++++++++++++++++
 sound/pci/hda/hda_codec.c | 19 -------------------
 sound/pci/hda/hda_local.h |  4 +++-
 3 files changed, 31 insertions(+), 20 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 662de6e58b6f..336b4b3a80b9 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -231,3 +231,31 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
+
+/* get/put callbacks for beep mute mixer switches */
+int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_beep *beep = codec->beep;
+	if (beep) {
+		ucontrol->value.integer.value[0] =
+			ucontrol->value.integer.value[1] =
+			beep->enabled;
+		return 0;
+	}
+	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
+}
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get_beep);
+
+int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_beep *beep = codec->beep;
+	if (beep)
+		snd_hda_enable_beep_device(codec,
+					   *ucontrol->value.integer.value);
+	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+}
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 51cb2a2e4fce..ddac4288615d 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2676,25 +2676,6 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put);
 
-#ifdef CONFIG_SND_HDA_INPUT_BEEP
-/**
- * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch
- *
- * This function calls snd_hda_enable_beep_device(), which behaves differently
- * depending on beep_mode option.
- */
-int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	long *valp = ucontrol->value.integer.value;
-
-	snd_hda_enable_beep_device(codec, *valp);
-	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-}
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
-#endif /* CONFIG_SND_HDA_INPUT_BEEP */
-
 /*
  * bound volume controls
  *
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 9a096a8e0fc5..1b4c12941baa 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -89,7 +89,7 @@
 	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
 	  .subdevice = HDA_SUBDEV_AMP_FLAG, \
 	  .info = snd_hda_mixer_amp_switch_info, \
-	  .get = snd_hda_mixer_amp_switch_get, \
+	  .get = snd_hda_mixer_amp_switch_get_beep, \
 	  .put = snd_hda_mixer_amp_switch_put_beep, \
 	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
 #else
@@ -121,6 +121,8 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
 int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol);
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol);
 int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol);
 #endif
-- 
cgit v1.2.3


From 257dfb410070b48e377c7894222b73ca41d662e0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 3 Jul 2012 17:35:05 +0200
Subject: ALSA: hda - Get rid of superfluous beep->mode field

It's no longer necessary since beep_mode=2 option was dropped.
It can be checked simply via codec->beep != NULL.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 14 ++++++--------
 sound/pci/hda/hda_beep.h |  1 -
 2 files changed, 6 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 336b4b3a80b9..e6cf2a22c407 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -184,6 +184,7 @@ EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device);
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 {
 	struct hda_beep *beep;
+	int err;
 
 	if (!snd_hda_get_bool_hint(codec, "beep"))
 		return 0; /* disabled explicitly by hints */
@@ -201,19 +202,16 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 
 	beep->nid = nid;
 	beep->codec = codec;
-	beep->mode = codec->beep_mode;
 	codec->beep = beep;
 
 	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
 	mutex_init(&beep->mutex);
 
-	if (beep->mode) {
-		int err = snd_hda_do_attach(beep);
-		if (err < 0) {
-			kfree(beep);
-			codec->beep = NULL;
-			return err;
-		}
+	err = snd_hda_do_attach(beep);
+	if (err < 0) {
+		kfree(beep);
+		codec->beep = NULL;
+		return err;
 	}
 
 	return 0;
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 30e79d16f9f8..4dc6933bc655 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -31,7 +31,6 @@
 struct hda_beep {
 	struct input_dev *dev;
 	struct hda_codec *codec;
-	unsigned int mode;
 	char phys[32];
 	int tone;
 	hda_nid_t nid;
-- 
cgit v1.2.3


From 3fd877d32cac31292628fb8f443543fc1989b49b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 3 Jul 2012 17:36:35 +0200
Subject: ALSA: hda - Avoid possible race of beep on/off

Call cancel_work_sync() when turning off the beep switch so that the
mute call is executed in a proper order.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index e6cf2a22c407..0bc2315b181d 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -165,12 +165,13 @@ static int snd_hda_do_attach(struct hda_beep *beep)
 int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 {
 	struct hda_beep *beep = codec->beep;
-	enable = !!enable;
-	if (beep == NULL)
+	if (!beep)
 		return 0;
+	enable = !!enable;
 	if (beep->enabled != enable) {
 		beep->enabled = enable;
 		if (!enable) {
+			cancel_work_sync(&beep->beep_work);
 			/* turn off beep */
 			snd_hda_codec_write(beep->codec, beep->nid, 0,
 						  AC_VERB_SET_BEEP_CONTROL, 0);
-- 
cgit v1.2.3


From 9c9acc91561221c30a530c9b84056609d0307c7c Mon Sep 17 00:00:00 2001
From: Sachin Kamat <sachin.kamat@linaro.org>
Date: Tue, 3 Jul 2012 14:04:04 +0530
Subject: ASoC: smdk_wm8994: Convert to use snd_soc_register_card()

Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/samsung/smdk_wm8994.c | 36 +++++++++++++++++++++++-------------
 1 file changed, 23 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 8eb309f23d18..48dd4dd9ee08 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -149,31 +149,41 @@ static struct snd_soc_card smdk = {
 	.num_links = ARRAY_SIZE(smdk_dai),
 };
 
-static struct platform_device *smdk_snd_device;
 
-static int __init smdk_audio_init(void)
+static int __devinit smdk_audio_probe(struct platform_device *pdev)
 {
 	int ret;
+	struct snd_soc_card *card = &smdk;
 
-	smdk_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!smdk_snd_device)
-		return -ENOMEM;
+	card->dev = &pdev->dev;
+	ret = snd_soc_register_card(card);
 
-	platform_set_drvdata(smdk_snd_device, &smdk);
-
-	ret = platform_device_add(smdk_snd_device);
 	if (ret)
-		platform_device_put(smdk_snd_device);
+		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
 
 	return ret;
 }
-module_init(smdk_audio_init);
 
-static void __exit smdk_audio_exit(void)
+static int __devexit smdk_audio_remove(struct platform_device *pdev)
 {
-	platform_device_unregister(smdk_snd_device);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
 }
-module_exit(smdk_audio_exit);
+
+static struct platform_driver smdk_audio_driver = {
+	.driver		= {
+		.name	= "smdk-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= smdk_audio_probe,
+	.remove		= __devexit_p(smdk_audio_remove),
+};
+
+module_platform_driver(smdk_audio_driver);
 
 MODULE_DESCRIPTION("ALSA SoC SMDK WM8994");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:smdk-audio");
-- 
cgit v1.2.3


From da602ab8a10e47c59be1a7ce524aaa76b77c23b6 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Beno=C3=AEt=20Th=C3=A9baudeau?=
 <benoit.thebaudeau@advansee.com>
Date: Tue, 3 Jul 2012 20:18:17 +0200
Subject: ASoC: dapm: Remove incomplete stereo code
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Stereo is not yet supported by dapm widgets, so remove stereo code from
snd_soc_dapm_get_volsw(), and warn if stereo controls are detected.

Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 22 ++++++++++++----------
 1 file changed, 12 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 967066873aad..912330b147e0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2464,23 +2464,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;
 	unsigned int shift = mc->shift;
-	unsigned int rshift = mc->rshift;
 	int max = mc->max;
-	unsigned int invert = mc->invert;
 	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
+
+	if (snd_soc_volsw_is_stereo(mc))
+		dev_warn(widget->dapm->dev,
+			 "Control '%s' is stereo, which is not supported\n",
+			 kcontrol->id.name);
 
 	ucontrol->value.integer.value[0] =
 		(snd_soc_read(widget->codec, reg) >> shift) & mask;
-	if (shift != rshift)
-		ucontrol->value.integer.value[1] =
-			(snd_soc_read(widget->codec, reg) >> rshift) & mask;
-	if (invert) {
+	if (invert)
 		ucontrol->value.integer.value[0] =
 			max - ucontrol->value.integer.value[0];
-		if (shift != rshift)
-			ucontrol->value.integer.value[1] =
-				max - ucontrol->value.integer.value[1];
-	}
 
 	return 0;
 }
@@ -2514,6 +2511,11 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_update update;
 	int wi;
 
+	if (snd_soc_volsw_is_stereo(mc))
+		dev_warn(widget->dapm->dev,
+			 "Control '%s' is stereo, which is not supported\n",
+			 kcontrol->id.name);
+
 	val = (ucontrol->value.integer.value[0] & mask);
 	connect = !!val;
 
-- 
cgit v1.2.3


From df79f55df3992fdd5dd206de6aa9af6a8ec1f86f Mon Sep 17 00:00:00 2001
From: Laxman Dewangan <ldewangan@nvidia.com>
Date: Fri, 29 Jun 2012 17:04:33 +0530
Subject: ASoC: tegra: use dmaengine based dma driver

Use the dmaengine based Tegra APB DMA driver for
data transfer between SPI fifo and memory in
place of legacy Tegra APB DMA.

Because generic soc-dmaengine-pcm uses the DMAs API
based on dmaengine, using the exported APIs provided
by this generic driver.

The new driver is selected if legacy driver is not
selected and new dma driver is enabled through config
file.

Signed-off-by: Laxman Dewangan <ldewangan@nvidia.com>
Acked-by: Stephen Warren <swarren@wwwdotorg.org>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/tegra/Kconfig     |   3 +-
 sound/soc/tegra/tegra_pcm.c | 115 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/tegra/tegra_pcm.h |   2 +
 3 files changed, 119 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index c1c8e955f4d3..7b6a1ebd197a 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -1,7 +1,8 @@
 config SND_SOC_TEGRA
 	tristate "SoC Audio for the Tegra System-on-Chip"
-	depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA
+	depends on ARCH_TEGRA && (TEGRA_SYSTEM_DMA || TEGRA20_APB_DMA)
 	select REGMAP_MMIO
+	select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA
 	help
 	  Say Y or M here if you want support for SoC audio on Tegra.
 
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 127348dc09b1..5658bcec1931 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -36,6 +36,7 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "tegra_pcm.h"
 
@@ -56,6 +57,7 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = {
 	.fifo_size		= 4,
 };
 
+#if defined(CONFIG_TEGRA_SYSTEM_DMA)
 static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd)
 {
 	struct snd_pcm_substream *substream = prtd->substream;
@@ -285,6 +287,119 @@ static struct snd_pcm_ops tegra_pcm_ops = {
 	.pointer	= tegra_pcm_pointer,
 	.mmap		= tegra_pcm_mmap,
 };
+#else
+static int tegra_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->platform->dev;
+	int ret;
+
+	/* Set HW params now that initialization is complete */
+	snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware);
+
+	ret = snd_dmaengine_pcm_open(substream, NULL, NULL);
+	if (ret) {
+		dev_err(dev, "dmaengine pcm open failed with err %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tegra_pcm_close(struct snd_pcm_substream *substream)
+{
+	snd_dmaengine_pcm_close(substream);
+	return 0;
+}
+
+static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->platform->dev;
+	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	struct tegra_pcm_dma_params *dmap;
+	struct dma_slave_config slave_config;
+	int ret;
+
+	dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	ret = snd_hwparams_to_dma_slave_config(substream, params,
+						&slave_config);
+	if (ret) {
+		dev_err(dev, "hw params config failed with err %d\n", ret);
+		return ret;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+		slave_config.dst_addr = dmap->addr;
+		slave_config.src_maxburst = 0;
+	} else {
+		slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+		slave_config.src_addr = dmap->addr;
+		slave_config.dst_maxburst = 0;
+	}
+	slave_config.slave_id = dmap->req_sel;
+
+	ret = dmaengine_slave_config(chan, &slave_config);
+	if (ret < 0) {
+		dev_err(dev, "dma slave config failed with err %d\n", ret);
+		return ret;
+	}
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	return 0;
+}
+
+static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+	return 0;
+}
+
+static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		return snd_dmaengine_pcm_trigger(substream,
+					SNDRV_PCM_TRIGGER_START);
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		return snd_dmaengine_pcm_trigger(substream,
+					SNDRV_PCM_TRIGGER_STOP);
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
+				struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+					runtime->dma_area,
+					runtime->dma_addr,
+					runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops tegra_pcm_ops = {
+	.open		= tegra_pcm_open,
+	.close		= tegra_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= tegra_pcm_hw_params,
+	.hw_free	= tegra_pcm_hw_free,
+	.trigger	= tegra_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= tegra_pcm_mmap,
+};
+#endif
 
 static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h
index 985d418a35e7..a3a450352dcf 100644
--- a/sound/soc/tegra/tegra_pcm.h
+++ b/sound/soc/tegra/tegra_pcm.h
@@ -40,6 +40,7 @@ struct tegra_pcm_dma_params {
 	unsigned long req_sel;
 };
 
+#if defined(CONFIG_TEGRA_SYSTEM_DMA)
 struct tegra_runtime_data {
 	struct snd_pcm_substream *substream;
 	spinlock_t lock;
@@ -51,6 +52,7 @@ struct tegra_runtime_data {
 	struct tegra_dma_req dma_req[2];
 	struct tegra_dma_channel *dma_chan;
 };
+#endif
 
 int tegra_pcm_platform_register(struct device *dev);
 void tegra_pcm_platform_unregister(struct device *dev);
-- 
cgit v1.2.3


From 32fee7afe763344ef53bbd4e737aa6168a9308aa Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Beno=C3=AEt=20Th=C3=A9baudeau?=
 <benoit.thebaudeau@advansee.com>
Date: Mon, 2 Jul 2012 13:45:21 +0200
Subject: ASoC: dapm: Fix dapm_set_path_status() connect
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

dapm_set_path_status() sets connect incorrectly in the case max > 1 with invert.
In that case, the raw disconnect value should be max, which corresponds to the
userspace value 0.

This use case currently does not appear upstream, but it could break
SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future.

This patch completes commit 3a9abe8.

Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <alsa-devel@alsa-project.org>
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 7 +++----
 1 file changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 89eae93445cf..5be4f9a2edb8 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -321,11 +321,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 
 		val = soc_widget_read(w, reg);
 		val = (val >> shift) & mask;
+		if (invert)
+			val = max - val;
 
-		if ((invert && !val) || (!invert && val))
-			p->connect = 1;
-		else
-			p->connect = 0;
+		p->connect = !!val;
 	}
 	break;
 	case snd_soc_dapm_mux: {
-- 
cgit v1.2.3


From 081413f206876e9d3755e1673828c7742fd00184 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 2 Jul 2012 18:19:58 +0100
Subject: ASoC: wm8962: Log AIF configuration requested by hw_params()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8962.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 27da4d722edc..beb709bd56cd 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2580,6 +2580,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
 			    WM8962_SAMPLE_RATE_INT_MODE |
 			    WM8962_SAMPLE_RATE_MASK, adctl3);
 
+	dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n",
+		wm8962->bclk, wm8962->lrclk);
+
 	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
 		wm8962_configure_bclk(codec);
 
-- 
cgit v1.2.3


From 38cbf9598feba97de9f9b43efa9153fd7c1a2ec9 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 22 Jun 2012 13:04:02 +0100
Subject: ASoC: core: Try to use regmap if the driver doesn't set up any I/O

Since most new drivers are expected to use regmap and since frequently the
only thing we need to do in the CODEC probe function is configure the I/O
try to initialise the register I/O using regmap if the driver hasn't done
so after probe().

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

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fe16135250f8..64b464ca3bc5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1095,6 +1095,10 @@ static int soc_probe_codec(struct snd_soc_card *card,
 		}
 	}
 
+	/* If the driver didn't set I/O up try regmap */
+	if (!codec->control_data)
+		snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+
 	if (driver->controls)
 		snd_soc_add_codec_controls(codec, driver->controls,
 				     driver->num_controls);
-- 
cgit v1.2.3


From 2974d6b1aa5261d8db1b614437cc6bafd3ddf0f2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 26 Jun 2012 11:06:53 +0100
Subject: ASoC: wm8994: Don't suspend accessory detection

Leave it up to the machine driver to disable accessory detection if
desired, the common pattern is to have accessory detection be a wake
source.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 37 -------------------------------------
 1 file changed, 37 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index a6e82d0a8e37..7bb875230dc0 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2970,23 +2970,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
 static int wm8994_codec_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	struct wm8994 *control = wm8994->wm8994;
 	int i, ret;
 
-	switch (control->type) {
-	case WM8994:
-		snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
-		break;
-	case WM1811:
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-				    WM1811_JACKDET_MODE_MASK, 0);
-		/* Fall through */
-	case WM8958:
-		snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
-				    WM8958_MICD_ENA, 0);
-		break;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
 		memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
 		       sizeof(struct wm8994_fll_config));
@@ -3036,28 +3021,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec)
 				 i + 1, ret);
 	}
 
-	switch (control->type) {
-	case WM8994:
-		if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
-			snd_soc_update_bits(codec, WM8994_MICBIAS,
-					    WM8994_MICD_ENA, WM8994_MICD_ENA);
-		break;
-	case WM1811:
-		if (wm8994->jackdet && wm8994->jack_cb) {
-			/* Restart from idle */
-			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-					    WM1811_JACKDET_MODE_MASK,
-					    WM1811_JACKDET_MODE_JACK);
-			break;
-		}
-		break;
-	case WM8958:
-		if (wm8994->jack_cb)
-			snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
-					    WM8958_MICD_ENA, WM8958_MICD_ENA);
-		break;
-	}
-
 	return 0;
 }
 #else
-- 
cgit v1.2.3


From 784a897e2310410ed169b5b331f2b7f06b7d58b7 Mon Sep 17 00:00:00 2001
From: Jiri Prchal <jiri.prchal@aksignal.cz>
Date: Wed, 4 Jul 2012 08:12:50 +0200
Subject: ASoC: tlv320aic3x: add missing registers and bits

Adds register and bit shift definitions in header file.
Changes are for TLV320AIC310x based on data sheet.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.h | 19 ++++++++++++++++++-
 1 file changed, 18 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 6f097fb60683..5da5eb3f4cc0 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -13,7 +13,7 @@
 #define _AIC3X_H
 
 /* AIC3X register space */
-#define AIC3X_CACHEREGNUM		103
+#define AIC3X_CACHEREGNUM		110
 
 /* Page select register */
 #define AIC3X_PAGE_SELECT		0
@@ -74,6 +74,8 @@
 #define HPLCOM_CFG			37
 /* Right High Power Output control registers */
 #define HPRCOM_CFG			38
+/* High Power Output Stage Control Register */
+#define HPOUT_SC			40
 /* DAC Output Switching control registers */
 #define DAC_LINE_MUX			41
 /* High Power Output Driver Pop Reduction registers */
@@ -148,6 +150,17 @@
 #define AIC3X_GPIOB_REG			101
 /* Clock generation control register */
 #define AIC3X_CLKGEN_CTRL_REG		102
+/* New AGC registers */
+#define LAGCN_ATTACK			103
+#define LAGCN_DECAY			104
+#define RAGCN_ATTACK			105
+#define RAGCN_DECAY			106
+/* New Programmable ADC Digital Path and I2C Bus Condition Register */
+#define NEW_ADC_DIGITALPATH		107
+/* Passive Analog Signal Bypass Selection During Powerdown Register */
+#define PASSIVE_BYPASS			108
+/* DAC Quiescent Current Adjustment Register */
+#define DAC_ICC_ADJ			109
 
 /* Page select register bits */
 #define PAGE0_SELECT		0
@@ -163,6 +176,10 @@
 #define DUAL_RATE_MODE		((1 << 5) | (1 << 6))
 #define LDAC2LCH		(0x1 << 3)
 #define RDAC2RCH		(0x1 << 1)
+#define LDAC2RCH		(0x2 << 3)
+#define RDAC2LCH		(0x2 << 1)
+#define LDAC2MONOMIX		(0x3 << 3)
+#define RDAC2MONOMIX		(0x3 << 1)
 
 /* PLL registers bitfields */
 #define PLLP_SHIFT		0
-- 
cgit v1.2.3


From c9e8e8d2541cbf0331400e2fa2fdca404e3569d4 Mon Sep 17 00:00:00 2001
From: Jiri Prchal <jiri.prchal@aksignal.cz>
Date: Wed, 4 Jul 2012 08:12:51 +0200
Subject: ASoC: tlv320aic3x: extending registers cache

Adds missing register default values to cache.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 58ef59dfbae9..174de6650563 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -118,7 +118,9 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
 	0x00, 0x00, 0x00, 0x00,	/* 88 */
 	0x00, 0x00, 0x00, 0x00,	/* 92 */
 	0x00, 0x00, 0x00, 0x00,	/* 96 */
-	0x00, 0x00, 0x02,	/* 100 */
+	0x00, 0x00, 0x02, 0x00,	/* 100 */
+	0x00, 0x00, 0x00, 0x00,	/* 104 */
+	0x00, 0x00,            	/* 108 */
 };
 
 #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
-- 
cgit v1.2.3


From 3be58dbb92871442191188ae51b449e1a9f0fe64 Mon Sep 17 00:00:00 2001
From: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Date: Wed, 4 Jul 2012 16:11:12 +0530
Subject: ASoC: STA529: Add support for STA529 Audio Codec

The STA529 is a digital stereo class-D audio amplifier. It includes an
audio DSP, an ST proprietary high-efficiency class-D driver and CMOS
power output stage. It is intended for high-efficiency
digital-to-power-audio conversion for portable applications.

Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   4 +
 sound/soc/codecs/Makefile |   2 +
 sound/soc/codecs/sta529.c | 441 ++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 447 insertions(+)
 create mode 100644 sound/soc/codecs/sta529.c

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1de24ccfe1c3..bbcb03863503 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_STA32X if I2C
+	select SND_SOC_STA529 if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -284,6 +285,9 @@ config SND_SOC_SSM2602
 config SND_SOC_STA32X
 	tristate
 
+config SND_SOC_STA529
+	tristate
+
 config SND_SOC_STAC9766
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index acf80888790c..8da3d22a7d1c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -49,6 +49,7 @@ snd-soc-spdif-tx-objs := spdif_transciever.o
 snd-soc-spdif-rx-objs := spdif_receiver.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-sta32x-objs := sta32x.o
+snd-soc-sta529-objs := sta529.o
 snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -163,6 +164,7 @@ obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif-rx.o snd-soc-spdif-tx.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
+obj-$(CONFIG_SND_SOC_STA529)   += snd-soc-sta529.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
new file mode 100644
index 000000000000..a9f34c736bfa
--- /dev/null
+++ b/sound/soc/codecs/sta529.c
@@ -0,0 +1,441 @@
+/*
+ * ASoC codec driver for spear platform
+ *
+ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+/* STA529 Register offsets */
+#define	 STA529_FFXCFG0		0x00
+#define	 STA529_FFXCFG1		0x01
+#define	 STA529_MVOL		0x02
+#define	 STA529_LVOL		0x03
+#define	 STA529_RVOL		0x04
+#define	 STA529_TTF0		0x05
+#define	 STA529_TTF1		0x06
+#define	 STA529_TTP0		0x07
+#define	 STA529_TTP1		0x08
+#define	 STA529_S2PCFG0		0x0A
+#define	 STA529_S2PCFG1		0x0B
+#define	 STA529_P2SCFG0		0x0C
+#define	 STA529_P2SCFG1		0x0D
+#define	 STA529_PLLCFG0		0x14
+#define	 STA529_PLLCFG1		0x15
+#define	 STA529_PLLCFG2		0x16
+#define	 STA529_PLLCFG3		0x17
+#define	 STA529_PLLPFE		0x18
+#define	 STA529_PLLST		0x19
+#define	 STA529_ADCCFG		0x1E /*mic_select*/
+#define	 STA529_CKOCFG		0x1F
+#define	 STA529_MISC		0x20
+#define	 STA529_PADST0		0x21
+#define	 STA529_PADST1		0x22
+#define	 STA529_FFXST		0x23
+#define	 STA529_PWMIN1		0x2D
+#define	 STA529_PWMIN2		0x2E
+#define	 STA529_POWST		0x32
+
+#define STA529_MAX_REGISTER	0x32
+
+#define STA529_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)
+
+#define STA529_FORMAT		(SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_S24_LE | \
+				SNDRV_PCM_FMTBIT_S32_LE)
+#define	S2PC_VALUE		0x98
+#define CLOCK_OUT		0x60
+#define LEFT_J_DATA_FORMAT	0x10
+#define I2S_DATA_FORMAT		0x12
+#define RIGHT_J_DATA_FORMAT	0x14
+#define CODEC_MUTE_VAL		0x80
+
+#define POWER_CNTLMSAK		0x40
+#define POWER_STDBY		0x40
+#define FFX_MASK		0x80
+#define FFX_OFF			0x80
+#define POWER_UP		0x00
+#define FFX_CLK_ENB		0x01
+#define FFX_CLK_DIS		0x00
+#define FFX_CLK_MSK		0x01
+#define PLAY_FREQ_RANGE_MSK	0x70
+#define CAP_FREQ_RANGE_MSK	0x0C
+#define PDATA_LEN_MSK		0xC0
+#define BCLK_TO_FS_MSK		0x30
+#define AUDIO_MUTE_MSK		0x80
+
+static const struct reg_default sta529_reg_defaults[] = {
+	{ 0,  0x35 },     /* R0   - FFX Configuration reg 0 */
+	{ 1,  0xc8 },     /* R1   - FFX Configuration reg 1 */
+	{ 2,  0x50 },     /* R2   - Master Volume */
+	{ 3,  0x00 },     /* R3   - Left Volume */
+	{ 4,  0x00 },     /* R4  -  Right Volume */
+	{ 10, 0xb2 },     /* R10  - S2P Config Reg 0 */
+	{ 11, 0x41 },     /* R11  - S2P Config Reg 1 */
+	{ 12, 0x92 },     /* R12  - P2S Config Reg 0 */
+	{ 13, 0x41 },     /* R13  - P2S Config Reg 1 */
+	{ 30, 0xd2 },     /* R30  - ADC Config Reg */
+	{ 31, 0x40 },     /* R31  - clock Out Reg */
+	{ 32, 0x21 },     /* R32  - Misc Register */
+};
+
+struct sta529 {
+	struct regmap *regmap;
+};
+
+static bool sta529_readable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+
+	case STA529_FFXCFG0:
+	case STA529_FFXCFG1:
+	case STA529_MVOL:
+	case STA529_LVOL:
+	case STA529_RVOL:
+	case STA529_S2PCFG0:
+	case STA529_S2PCFG1:
+	case STA529_P2SCFG0:
+	case STA529_P2SCFG1:
+	case STA529_ADCCFG:
+	case STA529_CKOCFG:
+	case STA529_MISC:
+		return true;
+	default:
+		return false;
+	}
+}
+
+
+static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary",
+	"Phase-shift"};
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0);
+static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
+
+static const struct snd_kcontrol_new sta529_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0,
+			127, 0, out_gain_tlv),
+	SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1,
+			master_vol_tlv),
+	SOC_ENUM("PWM Select", pwm_src),
+};
+
+static int sta529_set_bias_level(struct snd_soc_codec *codec, enum
+		snd_soc_bias_level level)
+{
+	struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK,
+				POWER_UP);
+		snd_soc_update_bits(codec, STA529_MISC,	FFX_CLK_MSK,
+				FFX_CLK_ENB);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			regcache_sync(sta529->regmap);
+		snd_soc_update_bits(codec, STA529_FFXCFG0,
+					POWER_CNTLMSAK, POWER_STDBY);
+		/* Making FFX output to zero */
+		snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK,
+				FFX_OFF);
+		snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK,
+				FFX_CLK_DIS);
+		break;
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+
+	/*
+	 * store the label for powers down audio subsystem for suspend.This is
+	 * used by soc core layer
+	 */
+	codec->dapm.bias_level = level;
+
+	return 0;
+
+}
+
+static int sta529_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_codec *codec = rtd->codec;
+	int pdata, play_freq_val, record_freq_val;
+	int bclk_to_fs_ratio;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		pdata = 1;
+		bclk_to_fs_ratio = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		pdata = 2;
+		bclk_to_fs_ratio = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		pdata = 3;
+		bclk_to_fs_ratio = 2;
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported format\n");
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 11025:
+		play_freq_val = 0;
+		record_freq_val = 2;
+		break;
+	case 16000:
+	case 22050:
+		play_freq_val = 1;
+		record_freq_val = 0;
+		break;
+
+	case 32000:
+	case 44100:
+	case 48000:
+		play_freq_val = 2;
+		record_freq_val = 0;
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported rate\n");
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK,
+				pdata << 6);
+		snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK,
+				bclk_to_fs_ratio << 4);
+		snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK,
+				play_freq_val << 4);
+	} else {
+		snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK,
+				pdata << 6);
+		snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK,
+				bclk_to_fs_ratio << 4);
+		snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK,
+				record_freq_val << 2);
+	}
+
+	return 0;
+}
+
+static int sta529_mute(struct snd_soc_dai *dai, int mute)
+{
+	u8 val = 0;
+
+	if (mute)
+		val |= CODEC_MUTE_VAL;
+
+	snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val);
+
+	return 0;
+}
+
+static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 mode = 0;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode = LEFT_J_DATA_FORMAT;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		mode = I2S_DATA_FORMAT;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode = RIGHT_J_DATA_FORMAT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops sta529_dai_ops = {
+	.hw_params	=	sta529_hw_params,
+	.set_fmt	=	sta529_set_dai_fmt,
+	.digital_mute	=	sta529_mute,
+};
+
+static struct snd_soc_dai_driver sta529_dai = {
+	.name = "sta529-audio",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = STA529_RATES,
+		.formats = STA529_FORMAT,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = STA529_RATES,
+		.formats = STA529_FORMAT,
+	},
+	.ops	= &sta529_dai_ops,
+};
+
+static int sta529_probe(struct snd_soc_codec *codec)
+{
+	struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	codec->control_data = sta529->regmap;
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+	sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+/* power down chip */
+static int sta529_remove(struct snd_soc_codec *codec)
+{
+	sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int sta529_suspend(struct snd_soc_codec *codec)
+{
+	sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int sta529_resume(struct snd_soc_codec *codec)
+{
+	sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+struct snd_soc_codec_driver sta529_codec_driver = {
+	.probe = sta529_probe,
+	.remove = sta529_remove,
+	.set_bias_level = sta529_set_bias_level,
+	.suspend = sta529_suspend,
+	.resume = sta529_resume,
+	.controls = sta529_snd_controls,
+	.num_controls = ARRAY_SIZE(sta529_snd_controls),
+};
+
+static const struct regmap_config sta529_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = STA529_MAX_REGISTER,
+	.readable_reg = sta529_readable,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = sta529_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults),
+};
+
+static __devinit int sta529_i2c_probe(struct i2c_client *i2c,
+		const struct i2c_device_id *id)
+{
+	struct sta529 *sta529;
+	int ret;
+
+	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+		return -EINVAL;
+
+	sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL);
+	if (sta529 == NULL) {
+		dev_err(&i2c->dev, "Can not allocate memory\n");
+		return -ENOMEM;
+	}
+
+	sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
+	if (IS_ERR(sta529->regmap)) {
+		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+		return PTR_ERR(sta529->regmap);
+	}
+
+	i2c_set_clientdata(i2c, sta529);
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&sta529_codec_driver, &sta529_dai, 1);
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+
+	return ret;
+}
+
+static int __devexit sta529_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+
+	return 0;
+}
+
+static const struct i2c_device_id sta529_i2c_id[] = {
+	{ "sta529", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, sta529_i2c_id);
+
+static struct i2c_driver sta529_i2c_driver = {
+	.driver = {
+		.name = "sta529",
+		.owner = THIS_MODULE,
+	},
+	.probe		= sta529_i2c_probe,
+	.remove		= __devexit_p(sta529_i2c_remove),
+	.id_table	= sta529_i2c_id,
+};
+
+module_i2c_driver(sta529_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC STA529 codec driver");
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From e584f9b4c2a919eb665fea8536ecd2bd7260e876 Mon Sep 17 00:00:00 2001
From: Vipin Kumar <vipin.kumar@st.com>
Date: Wed, 4 Jul 2012 16:11:13 +0530
Subject: ASoC: SPEAr spdif_out: Add spdif out support

This patch implements the spdif out driver for ST peripheral. This
peripheral implements IEC60958 standard

Signed-off-by: Vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/spear/spdif_out.c      | 389 +++++++++++++++++++++++++++++++++++++++
 sound/soc/spear/spdif_out_regs.h |  79 ++++++++
 2 files changed, 468 insertions(+)
 create mode 100644 sound/soc/spear/spdif_out.c
 create mode 100644 sound/soc/spear/spdif_out_regs.h

(limited to 'sound')

diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
new file mode 100644
index 000000000000..5eac4cda2fd7
--- /dev/null
+++ b/sound/soc/spear/spdif_out.c
@@ -0,0 +1,389 @@
+/*
+ * ALSA SoC SPDIF Out Audio Layer for spear processors
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Vipin Kumar <vipin.kumar@st.com>
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/ioport.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+#include <sound/spear_spdif.h>
+#include "spdif_out_regs.h"
+
+struct spdif_out_params {
+	u32 rate;
+	u32 core_freq;
+	u32 mute;
+};
+
+struct spdif_out_dev {
+	struct clk *clk;
+	struct spear_dma_data dma_params;
+	struct spdif_out_params saved_params;
+	u32 running;
+	void __iomem *io_base;
+};
+
+static void spdif_out_configure(struct spdif_out_dev *host)
+{
+	writel(SPDIF_OUT_RESET, host->io_base + SPDIF_OUT_SOFT_RST);
+	mdelay(1);
+	writel(readl(host->io_base + SPDIF_OUT_SOFT_RST) & ~SPDIF_OUT_RESET,
+			host->io_base + SPDIF_OUT_SOFT_RST);
+
+	writel(SPDIF_OUT_FDMA_TRIG_16 | SPDIF_OUT_MEMFMT_16_16 |
+			SPDIF_OUT_VALID_HW | SPDIF_OUT_USER_HW |
+			SPDIF_OUT_CHNLSTA_HW | SPDIF_OUT_PARITY_HW,
+			host->io_base + SPDIF_OUT_CFG);
+
+	writel(0x7F, host->io_base + SPDIF_OUT_INT_STA_CLR);
+	writel(0x7F, host->io_base + SPDIF_OUT_INT_EN_CLR);
+}
+
+static int spdif_out_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -EINVAL;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params);
+
+	ret = clk_enable(host->clk);
+	if (ret)
+		return ret;
+
+	host->running = true;
+	spdif_out_configure(host);
+
+	return 0;
+}
+
+static void spdif_out_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return;
+
+	clk_disable(host->clk);
+	host->running = false;
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq,
+		u32 rate)
+{
+	u32 divider, ctrl;
+
+	clk_set_rate(host->clk, core_freq);
+	divider = DIV_ROUND_CLOSEST(clk_get_rate(host->clk), (rate * 128));
+
+	ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+	ctrl &= ~SPDIF_DIVIDER_MASK;
+	ctrl |= (divider << SPDIF_DIVIDER_SHIFT) & SPDIF_DIVIDER_MASK;
+	writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+}
+
+static int spdif_out_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 rate, core_freq;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -EINVAL;
+
+	rate = params_rate(params);
+
+	switch (rate) {
+	case 8000:
+	case 16000:
+	case 32000:
+	case 64000:
+		/*
+		 * The clock is multiplied by 10 to bring it to feasible range
+		 * of frequencies for sscg
+		 */
+		core_freq = 64000 * 128 * 10;	/* 81.92 MHz */
+		break;
+	case 5512:
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+	case 176400:
+		core_freq = 176400 * 128;	/* 22.5792 MHz */
+		break;
+	case 48000:
+	case 96000:
+	case 192000:
+	default:
+		core_freq = 192000 * 128;	/* 24.576 MHz */
+		break;
+	}
+
+	spdif_out_clock(host, core_freq, rate);
+	host->saved_params.core_freq = core_freq;
+	host->saved_params.rate = rate;
+
+	return 0;
+}
+
+static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 ctrl;
+	int ret = 0;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -EINVAL;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+			ctrl &= ~SPDIF_OPMODE_MASK;
+			if (!host->saved_params.mute)
+				ctrl |= SPDIF_OPMODE_AUD_DATA |
+					SPDIF_STATE_NORMAL;
+			else
+				ctrl |= SPDIF_OPMODE_MUTE_PCM;
+			writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+		ctrl &= ~SPDIF_OPMODE_MASK;
+		ctrl |= SPDIF_OPMODE_OFF;
+		writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
+static int spdif_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 val;
+
+	host->saved_params.mute = mute;
+	val = readl(host->io_base + SPDIF_OUT_CTRL);
+	val &= ~SPDIF_OPMODE_MASK;
+
+	if (mute)
+		val |= SPDIF_OPMODE_MUTE_PCM;
+	else {
+		if (host->running)
+			val |= SPDIF_OPMODE_AUD_DATA | SPDIF_STATE_NORMAL;
+		else
+			val |= SPDIF_OPMODE_OFF;
+	}
+
+	writel(val, host->io_base + SPDIF_OUT_CTRL);
+	return 0;
+}
+
+static int spdif_mute_get(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = codec->card;
+	struct snd_soc_pcm_runtime *rtd = card->rtd;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ucontrol->value.integer.value[0] = host->saved_params.mute;
+	return 0;
+}
+
+static int spdif_mute_put(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = codec->card;
+	struct snd_soc_pcm_runtime *rtd = card->rtd;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (host->saved_params.mute == ucontrol->value.integer.value[0])
+		return 0;
+
+	spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]);
+
+	return 1;
+}
+static const struct snd_kcontrol_new spdif_out_controls[] = {
+	SOC_SINGLE_BOOL_EXT("IEC958 Playback Switch", 0,
+			spdif_mute_get, spdif_mute_put),
+};
+
+int spdif_soc_dai_probe(struct snd_soc_dai *dai)
+{
+	return snd_soc_add_dai_controls(dai, spdif_out_controls,
+				ARRAY_SIZE(spdif_out_controls));
+}
+
+static const struct snd_soc_dai_ops spdif_out_dai_ops = {
+	.digital_mute	= spdif_digital_mute,
+	.startup	= spdif_out_startup,
+	.shutdown	= spdif_out_shutdown,
+	.trigger	= spdif_out_trigger,
+	.hw_params	= spdif_out_hw_params,
+};
+
+static struct snd_soc_dai_driver spdif_out_dai = {
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+				 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+				 SNDRV_PCM_RATE_192000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.probe = spdif_soc_dai_probe,
+	.ops = &spdif_out_dai_ops,
+};
+
+static int spdif_out_probe(struct platform_device *pdev)
+{
+	struct spdif_out_dev *host;
+	struct spear_spdif_platform_data *pdata;
+	struct resource *res;
+	int ret;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res)
+		return -EINVAL;
+
+	if (!devm_request_mem_region(&pdev->dev, res->start,
+				resource_size(res), pdev->name)) {
+		dev_warn(&pdev->dev, "Failed to get memory resourse\n");
+		return -ENOENT;
+	}
+
+	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
+	if (!host) {
+		dev_warn(&pdev->dev, "kzalloc fail\n");
+		return -ENOMEM;
+	}
+
+	host->io_base = devm_ioremap(&pdev->dev, res->start,
+				resource_size(res));
+	if (!host->io_base) {
+		dev_warn(&pdev->dev, "ioremap failed\n");
+		return -ENOMEM;
+	}
+
+	host->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(host->clk))
+		return PTR_ERR(host->clk);
+
+	pdata = dev_get_platdata(&pdev->dev);
+
+	host->dma_params.data = pdata->dma_params;
+	host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA;
+	host->dma_params.max_burst = 16;
+	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	host->dma_params.filter = pdata->filter;
+
+	dev_set_drvdata(&pdev->dev, host);
+
+	ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai);
+	if (ret != 0) {
+		clk_put(host->clk);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int spdif_out_remove(struct platform_device *pdev)
+{
+	struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	clk_put(host->clk);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int spdif_out_suspend(struct device *dev)
+{
+	struct platform_device *pdev = to_platform_device(dev);
+	struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+	if (host->running)
+		clk_disable(host->clk);
+
+	return 0;
+}
+
+static int spdif_out_resume(struct device *dev)
+{
+	struct platform_device *pdev = to_platform_device(dev);
+	struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+	if (host->running) {
+		clk_enable(host->clk);
+		spdif_out_configure(host);
+		spdif_out_clock(host, host->saved_params.core_freq,
+				host->saved_params.rate);
+	}
+	return 0;
+}
+
+static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \
+		spdif_out_resume);
+
+#define SPDIF_OUT_DEV_PM_OPS (&spdif_out_dev_pm_ops)
+
+#else
+#define SPDIF_OUT_DEV_PM_OPS NULL
+
+#endif
+
+static struct platform_driver spdif_out_driver = {
+	.probe		= spdif_out_probe,
+	.remove		= spdif_out_remove,
+	.driver		= {
+		.name	= "spdif-out",
+		.owner	= THIS_MODULE,
+		.pm	= SPDIF_OUT_DEV_PM_OPS,
+	},
+};
+
+module_platform_driver(spdif_out_driver);
+
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr SPDIF OUT SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spdif_out");
diff --git a/sound/soc/spear/spdif_out_regs.h b/sound/soc/spear/spdif_out_regs.h
new file mode 100644
index 000000000000..a5e53324b452
--- /dev/null
+++ b/sound/soc/spear/spdif_out_regs.h
@@ -0,0 +1,79 @@
+/*
+ * SPEAr SPDIF OUT controller header file
+ *
+ * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef SPDIF_OUT_REGS_H
+#define SPDIF_OUT_REGS_H
+
+#define SPDIF_OUT_SOFT_RST	0x00
+	#define SPDIF_OUT_RESET		(1 << 0)
+#define SPDIF_OUT_FIFO_DATA	0x04
+#define SPDIF_OUT_INT_STA	0x08
+#define SPDIF_OUT_INT_STA_CLR	0x0C
+	#define SPDIF_INT_UNDERFLOW	(1 << 0)
+	#define SPDIF_INT_EODATA	(1 << 1)
+	#define SPDIF_INT_EOBLOCK	(1 << 2)
+	#define SPDIF_INT_EOLATENCY	(1 << 3)
+	#define SPDIF_INT_EOPD_DATA	(1 << 4)
+	#define SPDIF_INT_MEMFULLREAD	(1 << 5)
+	#define SPDIF_INT_EOPD_PAUSE	(1 << 6)
+
+#define SPDIF_OUT_INT_EN	0x10
+#define SPDIF_OUT_INT_EN_SET	0x14
+#define SPDIF_OUT_INT_EN_CLR	0x18
+#define SPDIF_OUT_CTRL		0x1C
+	#define SPDIF_OPMODE_MASK	(7 << 0)
+	#define SPDIF_OPMODE_OFF	(0 << 0)
+	#define SPDIF_OPMODE_MUTE_PCM	(1 << 0)
+	#define SPDIF_OPMODE_MUTE_PAUSE	(2 << 0)
+	#define SPDIF_OPMODE_AUD_DATA	(3 << 0)
+	#define SPDIF_OPMODE_ENCODE	(4 << 0)
+	#define SPDIF_STATE_NORMAL	(1 << 3)
+	#define SPDIF_DIVIDER_MASK	(0xff << 5)
+	#define SPDIF_DIVIDER_SHIFT	(5)
+	#define SPDIF_SAMPLEREAD_MASK	(0x1ffff << 15)
+	#define SPDIF_SAMPLEREAD_SHIFT	(15)
+#define SPDIF_OUT_STA		0x20
+#define SPDIF_OUT_PA_PB		0x24
+#define SPDIF_OUT_PC_PD		0x28
+#define SPDIF_OUT_CL1		0x2C
+#define SPDIF_OUT_CR1		0x30
+#define SPDIF_OUT_CL2_CR2_UV	0x34
+#define SPDIF_OUT_PAUSE_LAT	0x38
+#define SPDIF_OUT_FRMLEN_BRST	0x3C
+#define SPDIF_OUT_CFG		0x40
+	#define SPDIF_OUT_MEMFMT_16_0	(0 << 5)
+	#define SPDIF_OUT_MEMFMT_16_16	(1 << 5)
+	#define SPDIF_OUT_VALID_DMA	(0 << 3)
+	#define SPDIF_OUT_VALID_HW	(1 << 3)
+	#define SPDIF_OUT_USER_DMA	(0 << 2)
+	#define SPDIF_OUT_USER_HW	(1 << 2)
+	#define SPDIF_OUT_CHNLSTA_DMA	(0 << 1)
+	#define SPDIF_OUT_CHNLSTA_HW	(1 << 1)
+	#define SPDIF_OUT_PARITY_HW	(0 << 0)
+	#define SPDIF_OUT_PARITY_DMA	(1 << 0)
+	#define SPDIF_OUT_FDMA_TRIG_2	(2 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_6	(6 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_8	(8 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_10	(10 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_12	(12 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_16	(16 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_18	(18 << 8)
+
+#endif /* SPDIF_OUT_REGS_H */
-- 
cgit v1.2.3


From 1520ffd218f4aa53bc7652c0f6454da3cb428337 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 4 Jul 2012 19:04:11 +0100
Subject: ASoC: dwc: Staticise non-exported i2s_start()

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/dwc/designware_i2s.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 1bd042b15aef..1aa51300c564 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -107,7 +107,8 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
 	}
 }
 
-void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream)
+static void i2s_start(struct dw_i2s_dev *dev,
+		      struct snd_pcm_substream *substream)
 {
 
 	i2s_write_reg(dev->i2s_base, IER, 1);
-- 
cgit v1.2.3


From 949e6bc75fea779b433679601641ea641456283b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 4 Jul 2012 18:58:04 +0100
Subject: ASoC: arizona: Rename current rates tables to bclk_rates

They're the rates for the BCLK, not for the sample rate, so rename so that
we don't confuse ourselves.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/arizona.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 3b5730b90686..67760b4ea24c 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -363,7 +363,7 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static const int arizona_48k_rates[] = {
+static const int arizona_48k_bclk_rates[] = {
 	-1,
 	48000,
 	64000,
@@ -385,7 +385,7 @@ static const int arizona_48k_rates[] = {
 	24576000,
 };
 
-static const int arizona_44k1_rates[] = {
+static const int arizona_44k1_bclk_rates[] = {
 	-1,
 	44100,
 	58800,
@@ -445,17 +445,17 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	int bclk, lrclk, wl, frame, sr_val;
 
 	if (params_rate(params) % 8000)
-		rates = &arizona_44k1_rates[0];
+		rates = &arizona_44k1_bclk_rates[0];
 	else
-		rates = &arizona_48k_rates[0];
+		rates = &arizona_48k_bclk_rates[0];
 
-	for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) {
+	for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
 		if (rates[i] == snd_soc_params_to_bclk(params)) {
 			bclk = i;
 			break;
 		}
 	}
-	if (i == ARRAY_SIZE(arizona_44k1_rates)) {
+	if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) {
 		arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
 				params_rate(params));
 		return -EINVAL;
-- 
cgit v1.2.3


From 5b2eec3f98e08a8442ada41c4a63658b95a355f2 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 4 Jul 2012 17:32:05 +0100
Subject: ASoC: arizona: Implement AIF clock configuration

Allow the user to select which of the system clocks each AIF is referenced
to and constran the DAI to the set of frequencies which can be generated
from that clock.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/arizona.c | 106 +++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/arizona.h |   9 ++++
 sound/soc/codecs/wm5102.c  |   3 ++
 3 files changed, 118 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 67760b4ea24c..8e5246ca5550 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -385,6 +385,29 @@ static const int arizona_48k_bclk_rates[] = {
 	24576000,
 };
 
+static const unsigned int arizona_48k_rates[] = {
+	12000,
+	24000,
+	48000,
+	96000,
+	192000,
+	384000,
+	768000,
+	4000,
+	8000,
+	16000,
+	32000,
+	64000,
+	128000,
+	256000,
+	512000,
+};
+
+static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = {
+	.count	= ARRAY_SIZE(arizona_48k_rates),
+	.list	= arizona_48k_rates,
+};
+
 static const int arizona_44k1_bclk_rates[] = {
 	-1,
 	44100,
@@ -407,6 +430,21 @@ static const int arizona_44k1_bclk_rates[] = {
 	22579200,
 };
 
+static const unsigned int arizona_44k1_rates[] = {
+	11025,
+	22050,
+	44100,
+	88200,
+	176400,
+	352800,
+	705600,
+};
+
+static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = {
+	.count	= ARRAY_SIZE(arizona_44k1_rates),
+	.list	= arizona_44k1_rates,
+};
+
 static int arizona_sr_vals[] = {
 	0,
 	12000,
@@ -434,6 +472,36 @@ static int arizona_sr_vals[] = {
 	512000,
 };
 
+static int arizona_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+	const struct snd_pcm_hw_constraint_list *constraint;
+	unsigned int base_rate;
+
+	switch (dai_priv->clk) {
+	case ARIZONA_CLK_SYSCLK:
+		base_rate = priv->sysclk;
+		break;
+	case ARIZONA_CLK_ASYNCCLK:
+		base_rate = priv->asyncclk;
+		break;
+	default:
+		return 0;
+	}
+
+	if (base_rate % 8000)
+		constraint = &arizona_44k1_constraint;
+	else
+		constraint = &arizona_48k_constraint;
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  constraint);
+}
+
 static int arizona_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params,
 			     struct snd_soc_dai *dai)
@@ -501,11 +569,49 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+
+	switch (clk_id) {
+	case ARIZONA_CLK_SYSCLK:
+	case ARIZONA_CLK_ASYNCCLK:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (clk_id != dai_priv->clk && dai->active) {
+		dev_err(codec->dev, "Can't change clock on active DAI %d\n",
+			dai->id);
+		return -EBUSY;
+	}
+
+	dai_priv->clk = clk_id;
+
+	return 0;
+}
+
 const struct snd_soc_dai_ops arizona_dai_ops = {
+	.startup = arizona_startup,
 	.set_fmt = arizona_set_fmt,
 	.hw_params = arizona_hw_params,
+	.set_sysclk = arizona_dai_set_sysclk,
 };
 
+int arizona_init_dai(struct arizona_priv *priv, int id)
+{
+	struct arizona_dai_priv *dai_priv = &priv->dai[id];
+
+	dai_priv->clk = ARIZONA_CLK_SYSCLK;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_dai);
+
 static irqreturn_t arizona_fll_lock(int irq, void *data)
 {
 	struct arizona_fll *fll = data;
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 8c2ca1d9dbae..896711e19baa 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -44,12 +44,19 @@
 #define ARIZONA_MIXER_VOL_SHIFT                 1
 #define ARIZONA_MIXER_VOL_WIDTH                 7
 
+#define ARIZONA_MAX_DAI 3
+
 struct arizona;
 
+struct arizona_dai_priv {
+	int clk;
+};
+
 struct arizona_priv {
 	struct arizona *arizona;
 	int sysclk;
 	int asyncclk;
+	struct arizona_dai_priv dai[ARIZONA_MAX_DAI];
 };
 
 #define ARIZONA_NUM_MIXER_INPUTS 55
@@ -146,4 +153,6 @@ extern int arizona_init_fll(struct arizona *arizona, int id, int base,
 extern int arizona_set_fll(struct arizona_fll *fll, int source,
 			   unsigned int Fref, unsigned int Fout);
 
+extern int arizona_init_dai(struct arizona_priv *priv, int dai);
+
 #endif
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e76c41e1f847..be74a78e1aea 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -832,6 +832,9 @@ static int __devinit wm5102_probe(struct platform_device *pdev)
 			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
 			 &wm5102->fll[1]);
 
+	for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++)
+		arizona_init_dai(&wm5102->core, i);
+
 	/* Latch volume update bits */
 	for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++)
 		regmap_update_bits(arizona->regmap, wm5102_digital_vu[i],
-- 
cgit v1.2.3


From 5001765f992423fdfb82f42f548d3a51b9590186 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 4 Jul 2012 19:07:09 +0100
Subject: ASoC: arizona: Be more forgiving in BCLK selection

Allow any BCLK which can be divided down to generate LRCLK, not just the
lowest possible BCLK to clock out the samples.

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

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8e5246ca5550..8e066ebf1227 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -518,7 +518,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		rates = &arizona_48k_bclk_rates[0];
 
 	for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
-		if (rates[i] == snd_soc_params_to_bclk(params)) {
+		if (rates[i] >= snd_soc_params_to_bclk(params) &&
+		    rates[i] % params_rate(params) == 0) {
 			bclk = i;
 			break;
 		}
-- 
cgit v1.2.3


From a7a0a64daba5105215b79fe27b2d1ebbdcf5eebb Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 5 Jul 2012 12:00:12 +0200
Subject: ALSA: hda - Always call standard unsolicited event for Realtek codecs

With the model parsers out of the way, we have no custom unsol
events to worry about, we can therefore simplify the code.

In addition, this fixes a bug on the ASUS TC710, which has only
a headphone jack and a mic jack, but no internal mic or speakers.
Therefore the unsol_event pointer was not set, and as a result,
the jack kcontrols were not correctly updated.

BugLink: https://bugs.launchpad.net/bugs/1021192
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 21 +++------------------
 1 file changed, 3 insertions(+), 18 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f912d74438a6..a5b0b50b6a92 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -174,7 +174,6 @@ struct alc_spec {
 
 	/* hooks */
 	void (*init_hook)(struct hda_codec *codec);
-	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	void (*power_hook)(struct hda_codec *codec);
 #endif
@@ -688,7 +687,7 @@ static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid)
 }
 
 /* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
 	int action;
 
@@ -1024,11 +1023,9 @@ static void alc_init_automute(struct hda_codec *codec)
 	spec->automute_lo = spec->automute_lo_possible;
 	spec->automute_speaker = spec->automute_speaker_possible;
 
-	if (spec->automute_speaker_possible || spec->automute_lo_possible) {
+	if (spec->automute_speaker_possible || spec->automute_lo_possible)
 		/* create a control for automute mode */
 		alc_add_automute_mode_enum(codec);
-		spec->unsol_event = alc_sku_unsol_event;
-	}
 }
 
 /* return the position of NID in the list, or -1 if not found */
@@ -1191,7 +1188,6 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 
 	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n",
 		    ext, fixed, dock);
-	spec->unsol_event = alc_sku_unsol_event;
 }
 
 /* check the availabilities of auto-mute and auto-mic switches */
@@ -2062,14 +2058,6 @@ static int alc_init(struct hda_codec *codec)
 	return 0;
 }
 
-static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	struct alc_spec *spec = codec->spec;
-
-	if (spec->unsol_event)
-		spec->unsol_event(codec, res);
-}
-
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 {
@@ -4271,14 +4259,12 @@ static void set_capture_mixer(struct hda_codec *codec)
  */
 static void alc_auto_init_std(struct hda_codec *codec)
 {
-	struct alc_spec *spec = codec->spec;
 	alc_auto_init_multi_out(codec);
 	alc_auto_init_extra_out(codec);
 	alc_auto_init_analog_input(codec);
 	alc_auto_init_input_src(codec);
 	alc_auto_init_digital(codec);
-	if (spec->unsol_event)
-		alc_inithook(codec);
+	alc_inithook(codec);
 }
 
 /*
@@ -4879,7 +4865,6 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
 		spec->automute_speaker = 1;
 		spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
 		snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
-		spec->unsol_event = alc_sku_unsol_event;
 		snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs);
 	}
 }
-- 
cgit v1.2.3


From 1464189f8c2a5341722437ef916786afaf241c44 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 5 Jul 2012 12:15:01 +0100
Subject: ALSA: pcm: Make constraints lists const

They aren't modified by the core so the drivers can declare them const.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/pcm_lib.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 8f312fa6c282..7ae671923393 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1250,10 +1250,10 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
 int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
 			       unsigned int cond,
 			       snd_pcm_hw_param_t var,
-			       struct snd_pcm_hw_constraint_list *l)
+			       const struct snd_pcm_hw_constraint_list *l)
 {
 	return snd_pcm_hw_rule_add(runtime, cond, var,
-				   snd_pcm_hw_rule_list, l,
+				   snd_pcm_hw_rule_list, (void *)l,
 				   var, -1);
 }
 
-- 
cgit v1.2.3


From ef3207c503519bf33a114af3a780dfd00cfd5ce4 Mon Sep 17 00:00:00 2001
From: Hui Wang <jason77.wang@gmail.com>
Date: Wed, 4 Jul 2012 15:38:27 +0800
Subject: ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000

If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR
register (means Tx clock pin is input), we don't need to set
IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and
i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL
settings are ignored".

Signed-off-by: Hui Wang <jason77.wang@gmail.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/imx-sgtl5000.c | 3 +--
 1 file changed, 1 insertion(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 3a729caeb8c8..fb21b17f17f5 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -95,8 +95,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
 		return ret;
 	}
 	imx_audmux_v2_configure_port(ext_port,
-			IMX_AUDMUX_V2_PTCR_SYN |
-			IMX_AUDMUX_V2_PTCR_TCSEL(int_port),
+			IMX_AUDMUX_V2_PTCR_SYN,
 			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
 	if (ret) {
 		dev_err(&pdev->dev, "audmux external port setup failed\n");
-- 
cgit v1.2.3


From 42810d16220484a104317007e3d8fe5269df017b Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Tue, 3 Jul 2012 15:44:58 -0300
Subject: ASoC: imx-mc13783: Add audmux settings for mx27pdk

mx27pdk board also has a mc13783 codec.

Add support for it and do a run-time machine type check to perform the correct
audiomux settings.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/imx-mc13783.c | 49 ++++++++++++++++++++++++++++++---------------
 1 file changed, 33 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index f59c34943662..549b31fdc9dd 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -111,22 +111,39 @@ static int __devinit imx_mc13783_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
-		IMX_AUDMUX_V2_PTCR_SYN,
-		IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
-		IMX_AUDMUX_V2_PDCR_MODE(1) |
-		IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
-	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
-		IMX_AUDMUX_V2_PTCR_SYN |
-		IMX_AUDMUX_V2_PTCR_TFSDIR |
-		IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
-		IMX_AUDMUX_V2_PTCR_TCLKDIR |
-		IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
-		IMX_AUDMUX_V2_PTCR_RFSDIR |
-		IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
-		IMX_AUDMUX_V2_PTCR_RCLKDIR |
-		IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
-		IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+	if (machine_is_mx31_3ds()) {
+		imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
+			IMX_AUDMUX_V2_PDCR_MODE(1) |
+			IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
+		imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR |
+			IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+			IMX_AUDMUX_V2_PTCR_RFSDIR |
+			IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+			IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
+			IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+	} else if (machine_is_mx27_3ds()) {
+		imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+			IMX_AUDMUX_V1_PCR_SYN |
+			IMX_AUDMUX_V1_PCR_TFSDIR |
+			IMX_AUDMUX_V1_PCR_TCLKDIR |
+			IMX_AUDMUX_V1_PCR_RFSDIR |
+			IMX_AUDMUX_V1_PCR_RCLKDIR |
+			IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+			IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+			IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4)
+		);
+		imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4,
+			IMX_AUDMUX_V1_PCR_SYN |
+			IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+		);
+	}
 
 	return ret;
 }
-- 
cgit v1.2.3


From c013b27a174e8a83d3c8df799aa37c897842efcb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 4 Jul 2012 20:05:57 +0100
Subject: ASoC: arizona: Enable ASYNCCLK domain for audio interfaces

If an audio interface is configured to use ASYNCCLK then update the
asynchronous sample rate rather than one of our primary sample rates.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/arizona.c | 30 ++++++++++++++++++++++++------
 1 file changed, 24 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8e066ebf1227..d0bcca959111 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -507,6 +507,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
 	int base = dai->driver->base;
 	const int *rates;
 	int i;
@@ -530,10 +532,6 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	/*
-	 * We will need to be more flexible than this in future,
-	 * currently we use a single sample rate for the chip.
-	 */
 	for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++)
 		if (arizona_sr_vals[i] == params_rate(params))
 			break;
@@ -552,8 +550,28 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	wl = snd_pcm_format_width(params_format(params));
 	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
 
-	snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
-			    ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
+	/*
+	 * We will need to be more flexible than this in future,
+	 * currently we use a single sample rate for SYSCLK.
+	 */
+	switch (dai_priv->clk) {
+	case ARIZONA_CLK_SYSCLK:
+		snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
+				    ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
+		snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+				    ARIZONA_AIF1_RATE_MASK, 0);
+		break;
+	case ARIZONA_CLK_ASYNCCLK:
+		snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
+				    ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
+		snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+				    ARIZONA_AIF1_RATE_MASK, 8);
+		break;
+	default:
+		arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
+		return -EINVAL;
+	}
+
 	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
 			    ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
 	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
-- 
cgit v1.2.3


From 9498822d753d241fc93fbeebc17e668cf3023cf7 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 4 Jul 2012 20:07:03 +0100
Subject: ASoC: wm5102: Allow routing through the ASRCs

This enables common telephony use cases.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5102.c | 30 +++++++++++++++++++++++++++++-
 1 file changed, 29 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index be74a78e1aea..3827fa2af70a 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -275,6 +275,11 @@ ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
 
+ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
 static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = {
 SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
 		    0, NULL, 0),
@@ -363,6 +368,15 @@ SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
 SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
 		 0, NULL, 0),
 
+SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
 SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
 		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
 SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
@@ -491,6 +505,11 @@ ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
 ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
 ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
 
+ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"),
+
 SND_SOC_DAPM_OUTPUT("HPOUT1L"),
 SND_SOC_DAPM_OUTPUT("HPOUT1R"),
 SND_SOC_DAPM_OUTPUT("HPOUT2L"),
@@ -539,7 +558,11 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
 	{ name, "LHPF1", "LHPF1" }, \
 	{ name, "LHPF2", "LHPF2" }, \
 	{ name, "LHPF3", "LHPF3" }, \
-	{ name, "LHPF4", "LHPF4" }
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ASRC1L", "ASRC1L" }, \
+	{ name, "ASRC1R", "ASRC1R" }, \
+	{ name, "ASRC2L", "ASRC2L" }, \
+	{ name, "ASRC2R", "ASRC2R" }
 
 static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "DBVDD2" },
@@ -660,6 +683,11 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
 	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
 
+	ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+
 	{ "HPOUT1L", NULL, "OUT1L" },
 	{ "HPOUT1R", NULL, "OUT1R" },
 
-- 
cgit v1.2.3


From bcbf4a69ee6ca68d62440bc936a3c977c2141a66 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 5 Jul 2012 14:30:59 +0100
Subject: ASoC: wm1250-ev1: Flag all supported rates in the DAI

Not previously noticed due to normal usage being with CODEC<->CODEC links.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm1250-ev1.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index e0b51e9f8b12..951d7b49476a 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -121,20 +121,23 @@ static const struct snd_soc_dai_ops wm1250_ev1_ops = {
 	.hw_params = wm1250_ev1_hw_params,
 };
 
+#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+			  SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000)
+
 static struct snd_soc_dai_driver wm1250_ev1_dai = {
 	.name = "wm1250-ev1",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000,
+		.rates = WM1250_EV1_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000,
+		.rates = WM1250_EV1_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.ops = &wm1250_ev1_ops,
-- 
cgit v1.2.3


From 01005a729a17ab419f61a366e22f3419e7a2c3fe Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@ti.com>
Date: Fri, 6 Jul 2012 16:57:05 +0100
Subject: ASoC: dapm: Fix locking during codec shutdown

Codec shutdown performs a DAPM power sequence that might cause conflicts
and/or race conditions if another stream power event is running simultaneously.
Use card's dapm mutex to protect any potential race condition between them.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
 sound/soc/soc-dapm.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 5be4f9a2edb8..114f2af5f304 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3537,10 +3537,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
 
 static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 {
+	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	LIST_HEAD(down_list);
 	int powerdown = 0;
 
+	mutex_lock(&card->dapm_mutex);
+
 	list_for_each_entry(w, &dapm->card->widgets, list) {
 		if (w->dapm != dapm)
 			continue;
@@ -3563,6 +3566,8 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 			snd_soc_dapm_set_bias_level(dapm,
 						    SND_SOC_BIAS_STANDBY);
 	}
+
+	mutex_unlock(&card->dapm_mutex);
 }
 
 /*
-- 
cgit v1.2.3


From 5cb9b7482270972421a1f2d4145efc60d7ee1176 Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@ti.com>
Date: Fri, 6 Jul 2012 16:54:52 +0100
Subject: ASoC: pcm: Clean up logging in soc_new_pcm()

Use dev_ style logging throughout soc_new_pcm()

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

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 48fd15b312c1..7063b8f926c6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2003,7 +2003,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
 /* create a new pcm */
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
-	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
@@ -2042,7 +2041,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			capture, &pcm);
 	}
 	if (ret < 0) {
-		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+		dev_err(rtd->card->dev, "can't create pcm for %s\n",
+			rtd->dai_link->name);
 		return ret;
 	}
 	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name);
@@ -2099,14 +2099,14 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	if (platform->driver->pcm_new) {
 		ret = platform->driver->pcm_new(rtd);
 		if (ret < 0) {
-			pr_err("asoc: platform pcm constructor failed\n");
+			dev_err(platform->dev, "pcm constructor failed\n");
 			return ret;
 		}
 	}
 
 	pcm->private_free = platform->driver->pcm_free;
 out:
-	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+	dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name,
 		cpu_dai->name);
 	return ret;
 }
-- 
cgit v1.2.3


From 4123128ee4854a955dd4a94b31991f8cc38c9b5e Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@ti.com>
Date: Fri, 6 Jul 2012 16:56:16 +0100
Subject: ASoC: dapm: Make sure all dapm contexts are updated

Make sure we set the bias level for all DAPM contexts when
changing level.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 114f2af5f304..7c9cd276c2fc 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -288,9 +288,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 		if (dapm->codec->driver->set_bias_level)
 			ret = dapm->codec->driver->set_bias_level(dapm->codec,
 								  level);
-		else
-			dapm->bias_level = level;
-	}
+	} else
+		dapm->bias_level = level;
+
 	if (ret != 0)
 		goto out;
 
-- 
cgit v1.2.3


From 3e4536546beb5295e6e0459e745327ebffeade9d Mon Sep 17 00:00:00 2001
From: Simon Wilson <simonwilson@google.com>
Date: Fri, 6 Jul 2012 17:04:17 +0100
Subject: ASoC: twl6040: fix spelling mistake

Fix spelling mistake in "High-Performance" option of twl6040 power mode.

Signed-off-by: Simon Wilson <simonwilson@google.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/twl6040.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index a36e9fcdf184..0ff1e70b7770 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -553,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls =
 
 /* Headset power mode */
 static const char *twl6040_power_mode_texts[] = {
-	"Low-Power", "High-Perfomance",
+	"Low-Power", "High-Performance",
 };
 
 static const struct soc_enum twl6040_power_mode_enum =
-- 
cgit v1.2.3


From 3ac3f5ca91afc03587b1d2d642f126efc5be37ca Mon Sep 17 00:00:00 2001
From: Liam Girdwood <lrg@ti.com>
Date: Fri, 6 Jul 2012 17:07:00 +0100
Subject: ASoC: dpcm: Allow FE to be opened without valid BE routes.

Some userspace will open a PCM device and then configure mixers
for routing before triggering. This patch allows userspace to do
this sequence.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-pcm.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 7063b8f926c6..ef22d0bd9e9e 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1955,10 +1955,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 	fe->dpcm[stream].runtime = fe_substream->runtime;
 
 	if (dpcm_path_get(fe, stream, &list) <= 0) {
-		dev_warn(fe->dev, "asoc: %s no valid %s route\n",
+		dev_dbg(fe->dev, "asoc: %s no valid %s route\n",
 			fe->dai_link->name, stream ? "capture" : "playback");
-			mutex_unlock(&fe->card->mutex);
-			return -EINVAL;
 	}
 
 	/* calculate valid and active FE <-> BE dpcms */
-- 
cgit v1.2.3


From fabd03842b77b1eb6c9b08c79be86fa38afbe310 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 5 Jul 2012 17:20:06 +0100
Subject: ASoC: dapm: Mark widgets as dirty when a route is added

If we add a new route at runtime then we'll need to recheck the connections
to the affected widgets.

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

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 912330b147e0..19fda1339510 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2251,6 +2251,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 		path->connect = 0;
 		return 0;
 	}
+
+	dapm_mark_dirty(wsource, "Route added");
+	dapm_mark_dirty(wsink, "Route added");
+
 	return 0;
 
 err:
-- 
cgit v1.2.3


From efcc3c61b9b1e4f764e14c48c553e6d477f40512 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 5 Jul 2012 17:24:19 +0100
Subject: ASoC: dapm: Allow routes to be deleted at runtime

Since we're now relying on DAPM for things like enabling clocks when we
reparent the clocks for widgets we need to either use conditional routes
(which are expensive) or remove routes at runtime. Add a route removal
API to support this use case.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
 sound/soc/soc-dapm.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 77 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 19fda1339510..4ba47aab9801 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2264,6 +2264,59 @@ err:
 	return ret;
 }
 
+static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm,
+				  const struct snd_soc_dapm_route *route)
+{
+	struct snd_soc_dapm_path *path, *p;
+	const char *sink;
+	const char *source;
+	char prefixed_sink[80];
+	char prefixed_source[80];
+
+	if (route->control) {
+		dev_err(dapm->dev,
+			"Removal of routes with controls not supported\n");
+		return -EINVAL;
+	}
+
+	if (dapm->codec && dapm->codec->name_prefix) {
+		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
+			 dapm->codec->name_prefix, route->sink);
+		sink = prefixed_sink;
+		snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
+			 dapm->codec->name_prefix, route->source);
+		source = prefixed_source;
+	} else {
+		sink = route->sink;
+		source = route->source;
+	}
+
+	path = NULL;
+	list_for_each_entry(p, &dapm->card->paths, list) {
+		if (strcmp(p->source->name, source) != 0)
+			continue;
+		if (strcmp(p->sink->name, sink) != 0)
+			continue;
+		path = p;
+		break;
+	}
+
+	if (path) {
+		dapm_mark_dirty(path->source, "Route removed");
+		dapm_mark_dirty(path->sink, "Route removed");
+
+		list_del(&path->list);
+		list_del(&path->list_sink);
+		list_del(&path->list_source);
+		kfree(path);
+	} else {
+		dev_warn(dapm->dev, "Route %s->%s does not exist\n",
+			 source, sink);
+	}
+
+	return 0;
+}
+
 /**
  * snd_soc_dapm_add_routes - Add routes between DAPM widgets
  * @dapm: DAPM context
@@ -2298,6 +2351,30 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
 
+/**
+ * snd_soc_dapm_del_routes - Remove routes between DAPM widgets
+ * @dapm: DAPM context
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Removes routes from the DAPM context.
+ */
+int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
+			    const struct snd_soc_dapm_route *route, int num)
+{
+	int i, ret = 0;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
+	for (i = 0; i < num; i++) {
+		snd_soc_dapm_del_route(dapm, route);
+		route++;
+	}
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes);
+
 static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm,
 				   const struct snd_soc_dapm_route *route)
 {
-- 
cgit v1.2.3


From 410837a7a29efa2402f496215244569c988bf0db Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 5 Jul 2012 17:26:59 +0100
Subject: ASoC: arizona: Change DAPM routes for AIF clocks when we change them

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
 sound/soc/codecs/arizona.c | 32 +++++++++++++++++++++++++++++---
 1 file changed, 29 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index d0bcca959111..901b53e1d7bc 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -588,12 +588,25 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static const char *arizona_dai_clk_str(int clk_id)
+{
+	switch (clk_id) {
+	case ARIZONA_CLK_SYSCLK:
+		return "SYSCLK";
+	case ARIZONA_CLK_ASYNCCLK:
+		return "ASYNCCLK";
+	default:
+		return "Unknown clock";
+	}
+}
+
 static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
 				  int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+	struct snd_soc_dapm_route routes[2];
 
 	switch (clk_id) {
 	case ARIZONA_CLK_SYSCLK:
@@ -603,15 +616,28 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	if (clk_id != dai_priv->clk && dai->active) {
+	if (clk_id == dai_priv->clk)
+		return 0;
+
+	if (dai->active) {
 		dev_err(codec->dev, "Can't change clock on active DAI %d\n",
 			dai->id);
 		return -EBUSY;
 	}
 
-	dai_priv->clk = clk_id;
+	memset(&routes, 0, sizeof(routes));
+	routes[0].sink = dai->driver->capture.stream_name;
+	routes[1].sink = dai->driver->playback.stream_name;
 
-	return 0;
+	routes[0].source = arizona_dai_clk_str(dai_priv->clk);
+	routes[1].source = arizona_dai_clk_str(dai_priv->clk);
+	snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+
+	routes[0].source = arizona_dai_clk_str(clk_id);
+	routes[1].source = arizona_dai_clk_str(clk_id);
+	snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+
+	return snd_soc_dapm_sync(&codec->dapm);
 }
 
 const struct snd_soc_dai_ops arizona_dai_ops = {
-- 
cgit v1.2.3


From d66a547cddb9124cea6308c33e1f54c7c8db288f Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Fri, 6 Jul 2012 12:19:10 +0200
Subject: ASoC: omap-mcpdm: Add missing MODULE_ALIAS

The MODULE_ALIAS() was missing from the driver.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap-mcpdm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 59d47ab5b15d..2c66e2498a45 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -527,6 +527,7 @@ static struct platform_driver asoc_mcpdm_driver = {
 
 module_platform_driver(asoc_mcpdm_driver);
 
+MODULE_ALIAS("platform:omap-mcpdm");
 MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
 MODULE_DESCRIPTION("OMAP PDM SoC Interface");
 MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From a837987e7b36a9056cd17c0967efe1ce73a102ff Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 9 Jul 2012 12:16:41 +0100
Subject: ASoC: arizona: Export dai_ops

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

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 901b53e1d7bc..0be04b526588 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -646,6 +646,7 @@ const struct snd_soc_dai_ops arizona_dai_ops = {
 	.hw_params = arizona_hw_params,
 	.set_sysclk = arizona_dai_set_sysclk,
 };
+EXPORT_SYMBOL_GPL(arizona_dai_ops);
 
 int arizona_init_dai(struct arizona_priv *priv, int id)
 {
-- 
cgit v1.2.3


From c9c56fd0b766f6f3cd19c83945954ff5b06afc5f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 9 Jul 2012 19:09:01 +0100
Subject: ASoC: arizona: Add IN4 to the mixer tables

Some devices have four input structures rather than three.

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

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 0be04b526588..f3680c374347 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -79,6 +79,8 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"IN2R",
 	"IN3L",
 	"IN3R",
+	"IN4L",
+	"IN4R",
 	"AIF1RX1",
 	"AIF1RX2",
 	"AIF1RX3",
@@ -138,6 +140,8 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
 	0x13,
 	0x14,
 	0x15,
+	0x16,
+	0x17,
 	0x20,  /* AIF1RX1 */
 	0x21,
 	0x22,
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 896711e19baa..b894b64e8f5c 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -59,7 +59,7 @@ struct arizona_priv {
 	struct arizona_dai_priv dai[ARIZONA_MAX_DAI];
 };
 
-#define ARIZONA_NUM_MIXER_INPUTS 55
+#define ARIZONA_NUM_MIXER_INPUTS 57
 
 extern const unsigned int arizona_mixer_tlv[];
 extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
-- 
cgit v1.2.3


From bc9dce5853ced3b7a5bc79f1101a4c4b0a752f3e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 9 Jul 2012 19:08:23 +0100
Subject: ASoC: wm5102: Fix cut'n'paste for digital volume registers

The analogue PGA shifts were used; this makes no practical difference as
the values are the same.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm5102.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 3827fa2af70a..7a6a11a323ff 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -68,13 +68,13 @@ SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
 	     ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
 
 SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
-		 ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_PGA_VOL_SHIFT,
+		 ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
-		 ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_PGA_VOL_SHIFT,
+		 ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
-		 ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_PGA_VOL_SHIFT,
+		 ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 
 ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
-- 
cgit v1.2.3


From 774441915de8402103ffe5bf68656f160fefc86f Mon Sep 17 00:00:00 2001
From: Jiri Prchal <jiri.prchal@aksignal.cz>
Date: Mon, 9 Jul 2012 09:48:44 +0200
Subject: ASoC: tlv320aic3x: add deemphasis switch

This patch adds missing deemphasis switch.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 174de6650563..7933b8c720af 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -356,6 +356,9 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	*/
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
 
+	/* De-emphasis */
+	SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
+
 	/* Input */
 	SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
 			 0, 119, 0, adc_tlv),
-- 
cgit v1.2.3


From bb1daa803c733462248421dd9beed84fecf1745e Mon Sep 17 00:00:00 2001
From: Jiri Prchal <jiri.prchal@aksignal.cz>
Date: Tue, 10 Jul 2012 14:35:11 +0200
Subject: ASoC: tlv320aic3x: add AGC settings

This patch adds AGC target level and times settings for TLV320AIC3x.
Enums uses small arrays of two channels left and right since it uses different registers.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++
 1 file changed, 25 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 7933b8c720af..0d2f8c44999d 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -231,6 +231,25 @@ static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
 };
 
+static const char *aic3x_agc_level[] =
+	{ "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" };
+static const struct soc_enum aic3x_agc_level_enum[] = {
+	SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level),
+	SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level),
+};
+
+static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" };
+static const struct soc_enum aic3x_agc_attack_enum[] = {
+	SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack),
+	SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack),
+};
+
+static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" };
+static const struct soc_enum aic3x_agc_decay_enum[] = {
+	SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay),
+	SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
+};
+
 /*
  * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
  */
@@ -355,6 +374,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	 * adjust PGA to max value when ADC is on and will never go back.
 	*/
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
+	SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]),
+	SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]),
+	SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]),
+	SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]),
+	SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]),
+	SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]),
 
 	/* De-emphasis */
 	SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
-- 
cgit v1.2.3


From a1f34af0ec35e3131d65e0ae4cec6b048cba3e88 Mon Sep 17 00:00:00 2001
From: Jiri Prchal <jiri.prchal@aksignal.cz>
Date: Tue, 10 Jul 2012 14:36:58 +0200
Subject: ASoC: tlv320aic3x: add input clock selection

This patch adds input selection of main codec clock - from what pin.
Both registers set same value since codec uses clock divider or pll at one time.

Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 6 ++++++
 sound/soc/codecs/tlv320aic3x.h | 8 ++++++++
 2 files changed, 14 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 0d2f8c44999d..b94f81ffed34 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1002,6 +1002,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 
+	/* set clock on MCLK or GPIO2 or BCLK */
+	snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK,
+				clk_id << PLLCLK_IN_SHIFT);
+	snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK,
+				clk_id << CLKDIV_IN_SHIFT);
+
 	aic3x->sysclk = freq;
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 5da5eb3f4cc0..149338b254f6 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -195,6 +195,14 @@
 #define PLL_CLKIN_SHIFT		4
 #define MCLK_SOURCE		0x0
 #define PLL_CLKDIV_SHIFT	0
+#define PLLCLK_IN_MASK		0x30
+#define PLLCLK_IN_SHIFT		4
+#define CLKDIV_IN_MASK		0xc0
+#define CLKDIV_IN_SHIFT		6
+/* clock in source */
+#define CLKIN_MCLK		0
+#define CLKIN_GPIO2		1
+#define CLKIN_BCLK		2
 
 /* Software reset register bits */
 #define SOFT_RESET		0x80
-- 
cgit v1.2.3


From 2b4d39fc2a80e271ac8d44fccd02277a4b63c557 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 10 Jul 2012 17:03:46 +0100
Subject: ASoC: arizona: Support variable FLL VCO multipliers

Some Arizona chips have a higher frequency for the FLL VCO, support this
in the common code.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/arizona.c | 6 +++---
 sound/soc/codecs/arizona.h | 1 +
 sound/soc/codecs/wm5102.c  | 3 +++
 3 files changed, 7 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index f3680c374347..5c9cacaf2d52 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -734,9 +734,9 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 	/* Apply the division for our remaining calculations */
 	Fref /= div;
 
-	/* Fvco should be 90-100MHz; don't check the upper bound */
+	/* Fvco should be over the targt; don't check the upper bound */
 	div = 1;
-	while (Fout * div < 90000000) {
+	while (Fout * div < 90000000 * fll->vco_mult) {
 		div++;
 		if (div > 7) {
 			arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
@@ -744,7 +744,7 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 			return -EINVAL;
 		}
 	}
-	target = Fout * div;
+	target = Fout * div / fll->vco_mult;
 	cfg->outdiv = div;
 
 	arizona_fll_dbg(fll, "Fvco=%dHz\n", target);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index b894b64e8f5c..59caca8865e8 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -141,6 +141,7 @@ struct arizona_fll {
 	struct arizona *arizona;
 	int id;
 	unsigned int base;
+	unsigned int vco_mult;
 	struct completion lock;
 	struct completion ok;
 
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 7a6a11a323ff..6537f16d383e 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -853,6 +853,9 @@ static int __devinit wm5102_probe(struct platform_device *pdev)
 
 	wm5102->core.arizona = arizona;
 
+	for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++)
+		wm5102->fll[i].vco_mult = 1;
+
 	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
 			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
 			 &wm5102->fll[0]);
-- 
cgit v1.2.3


From f96985e3b3cfcd2d21faca79863fb34533d575aa Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Wed, 11 Jul 2012 09:41:23 +0300
Subject: ASoC: STA529: fix an error message

GCC complains that "ret" is uninitialized here.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-By: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/sta529.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index a9f34c736bfa..0c225cd569d2 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -397,8 +397,9 @@ static __devinit int sta529_i2c_probe(struct i2c_client *i2c,
 
 	sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
 	if (IS_ERR(sta529->regmap)) {
+		ret = PTR_ERR(sta529->regmap);
 		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
-		return PTR_ERR(sta529->regmap);
+		return ret;
 	}
 
 	i2c_set_clientdata(i2c, sta529);
-- 
cgit v1.2.3


From 5c6af635fd77251b753cb1c07a6a6f306ba4e287 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 9 Jul 2012 19:09:41 +0100
Subject: ASoC: wm5110: Add audio CODEC driver

The WM5110 is a highly integrated low power audio subsystem for
smartphones, tablets and other portable audio devices. It combines
an advanced DSP feature set with a flexible, high performance audio
hub CODEC.

This patch adds the audio CODEC driver for the device.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |   6 +
 sound/soc/codecs/Makefile |   2 +
 sound/soc/codecs/wm5110.c | 950 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm5110.h |  21 +
 4 files changed, 979 insertions(+)
 create mode 100644 sound/soc/codecs/wm5110.c
 create mode 100644 sound/soc/codecs/wm5110.h

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bbcb03863503..9f8e8594aeb9 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM2200 if I2C
 	select SND_SOC_WM5100 if I2C
 	select SND_SOC_WM5102 if MFD_WM5102
+	select SND_SOC_WM5110 if MFD_WM5110
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
@@ -134,7 +135,9 @@ config SND_SOC_88PM860X
 config SND_SOC_ARIZONA
 	tristate
 	default y if SND_SOC_WM5102=y
+	default y if SND_SOC_WM5110=y
 	default m if SND_SOC_WM5102=m
+	default m if SND_SOC_WM5110=m
 
 config SND_SOC_WM_HUBS
 	tristate
@@ -338,6 +341,9 @@ config SND_SOC_WM5100
 config SND_SOC_WM5102
 	tristate
 
+config SND_SOC_WM5110
+	tristate
+
 config SND_SOC_WM8350
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 8da3d22a7d1c..34148bb59c68 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -66,6 +66,7 @@ snd-soc-wm2000-objs := wm2000.o
 snd-soc-wm2200-objs := wm2200.o
 snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
 snd-soc-wm5102-objs := wm5102.o
+snd-soc-wm5110-objs := wm5110.o
 snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
@@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o
 obj-$(CONFIG_SND_SOC_WM2200)	+= snd-soc-wm2200.o
 obj-$(CONFIG_SND_SOC_WM5100)	+= snd-soc-wm5100.o
 obj-$(CONFIG_SND_SOC_WM5102)	+= snd-soc-wm5102.o
+obj-$(CONFIG_SND_SOC_WM5110)	+= snd-soc-wm5110.o
 obj-$(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
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
new file mode 100644
index 000000000000..8033f7065189
--- /dev/null
+++ b/sound/soc/codecs/wm5110.c
@@ -0,0 +1,950 @@
+/*
+ * wm5110.c  --  WM5110 ALSA SoC Audio driver
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm5110.h"
+
+struct wm5110_priv {
+	struct arizona_priv core;
+	struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new wm5110_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
+	   ARIZONA_IN3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL,
+	   ARIZONA_IN4_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL,
+		       ARIZONA_IN1R_CONTROL,
+		       ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+		       ARIZONA_IN2R_CONTROL,
+		       ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL,
+		       ARIZONA_IN3R_CONTROL,
+		       ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
+		   ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+	       ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+	   ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
+	   ARIZONA_OUT6_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	       ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
+		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
+		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+		     ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
+	   ARIZONA_SPK2R_MUTE_SHIFT, 1, 1),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT2L, ARIZONA_OUT6LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT2R, ARIZONA_OUT6RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+		    0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+		    ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+SND_SOC_DAPM_INPUT("IN4L"),
+SND_SOC_DAPM_INPUT("IN4R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+		 ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+		 ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT6L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT6L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT6R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT6R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+ARIZONA_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+SND_SOC_DAPM_OUTPUT("SPKDAT2L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT2R"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "IN1L", "IN1L PGA" }, \
+	{ name, "IN1R", "IN1R PGA" }, \
+	{ name, "IN2L", "IN2L PGA" }, \
+	{ name, "IN2R", "IN2R PGA" }, \
+	{ name, "IN3L", "IN3L PGA" }, \
+	{ name, "IN3R", "IN3R PGA" }, \
+	{ name, "IN4L", "IN4L PGA" }, \
+	{ name, "IN4R", "IN4R PGA" }, \
+	{ name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF1RX7", "AIF1RX7" }, \
+	{ name, "AIF1RX8", "AIF1RX8" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "DRC2L", "DRC2L" }, \
+	{ name, "DRC2R", "DRC2R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ASRC1L", "ASRC1L" }, \
+	{ name, "ASRC1R", "ASRC1R" }, \
+	{ name, "ASRC2L", "ASRC2L" }, \
+	{ name, "ASRC2R", "ASRC2R" }
+
+static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
+	{ "AIF2 Capture", NULL, "DBVDD2" },
+	{ "AIF2 Playback", NULL, "DBVDD2" },
+
+	{ "AIF3 Capture", NULL, "DBVDD3" },
+	{ "AIF3 Playback", NULL, "DBVDD3" },
+
+	{ "OUT1L", NULL, "CPVDD" },
+	{ "OUT1R", NULL, "CPVDD" },
+	{ "OUT2L", NULL, "CPVDD" },
+	{ "OUT2R", NULL, "CPVDD" },
+	{ "OUT3L", NULL, "CPVDD" },
+
+	{ "OUT4L", NULL, "SPKVDDL" },
+	{ "OUT4R", NULL, "SPKVDDR" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT2L", NULL, "SYSCLK" },
+	{ "OUT2R", NULL, "SYSCLK" },
+	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+	{ "OUT4R", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+	{ "OUT6L", NULL, "SYSCLK" },
+	{ "OUT6R", NULL, "SYSCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+	{ "MICBIAS3", NULL, "MICVDD" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "Mic Mute Mixer", NULL, "Noise Mixer" },
+	{ "Mic Mute Mixer", NULL, "Mic Mixer" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+	{ "AIF1 Capture", NULL, "AIF1TX7" },
+	{ "AIF1 Capture", NULL, "AIF1TX8" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+	{ "AIF1RX7", NULL, "AIF1 Playback" },
+	{ "AIF1RX8", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+
+	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+	ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+	ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+	ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+	ARIZONA_MIXER_ROUTES("OUT6L", "SPKDAT2L"),
+	ARIZONA_MIXER_ROUTES("OUT6R", "SPKDAT2R"),
+
+	ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+	ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+	ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+	ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+	ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+	ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+	ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+	ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+	ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+	ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
+	ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+
+	{ "HPOUT1L", NULL, "OUT1L" },
+	{ "HPOUT1R", NULL, "OUT1R" },
+
+	{ "HPOUT2L", NULL, "OUT2L" },
+	{ "HPOUT2R", NULL, "OUT2R" },
+
+	{ "EPOUTN", NULL, "OUT3L" },
+	{ "EPOUTP", NULL, "OUT3L" },
+
+	{ "SPKOUTLN", NULL, "OUT4L" },
+	{ "SPKOUTLP", NULL, "OUT4L" },
+
+	{ "SPKOUTRN", NULL, "OUT4R" },
+	{ "SPKOUTRP", NULL, "OUT4R" },
+
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+
+	{ "SPKDAT2L", NULL, "OUT6L" },
+	{ "SPKDAT2R", NULL, "OUT6R" },
+};
+
+static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fll_id) {
+	case WM5110_FLL1:
+		return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout);
+	case WM5110_FLL2:
+		return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+#define WM5110_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM5110_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_driver wm5110_dai[] = {
+	{
+		.name = "wm5110-aif1",
+		.id = 1,
+		.base = ARIZONA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 8,
+			 .rates = WM5110_RATES,
+			 .formats = WM5110_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5110-aif2",
+		.id = 2,
+		.base = ARIZONA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5110_RATES,
+			 .formats = WM5110_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5110-aif3",
+		.id = 3,
+		.base = ARIZONA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF3 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5110_RATES,
+			 .formats = WM5110_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static int wm5110_codec_probe(struct snd_soc_codec *codec)
+{
+	struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	codec->control_data = priv->core.arizona->regmap;
+	return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+}
+
+#define WM5110_DIG_VU 0x0200
+
+static unsigned int wm5110_digital_vu[] = {
+	ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	ARIZONA_ADC_DIGITAL_VOLUME_1R,
+	ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	ARIZONA_ADC_DIGITAL_VOLUME_2R,
+	ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	ARIZONA_ADC_DIGITAL_VOLUME_3R,
+
+	ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	ARIZONA_DAC_DIGITAL_VOLUME_1R,
+	ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	ARIZONA_DAC_DIGITAL_VOLUME_2R,
+	ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	ARIZONA_DAC_DIGITAL_VOLUME_3R,
+	ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4R,
+	ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm5110 = {
+	.probe = wm5110_codec_probe,
+
+	.idle_bias_off = true,
+
+	.set_sysclk = arizona_set_sysclk,
+	.set_pll = wm5110_set_fll,
+
+	.controls = wm5110_snd_controls,
+	.num_controls = ARRAY_SIZE(wm5110_snd_controls),
+	.dapm_widgets = wm5110_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm5110_dapm_widgets),
+	.dapm_routes = wm5110_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes),
+};
+
+static int __devinit wm5110_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct wm5110_priv *wm5110;
+	int i;
+
+	wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv),
+			      GFP_KERNEL);
+	if (wm5110 == NULL)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, wm5110);
+
+	wm5110->core.arizona = arizona;
+
+	for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++)
+		wm5110->fll[i].vco_mult = 3;
+
+	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+			 &wm5110->fll[0]);
+	arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+			 &wm5110->fll[1]);
+
+	for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++)
+		arizona_init_dai(&wm5110->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(wm5110_digital_vu); i++)
+		regmap_update_bits(arizona->regmap, wm5110_digital_vu[i],
+				   WM5110_DIG_VU, WM5110_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110,
+				      wm5110_dai, ARRAY_SIZE(wm5110_dai));
+}
+
+static int __devexit wm5110_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver wm5110_codec_driver = {
+	.driver = {
+		.name = "wm5110-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm5110_probe,
+	.remove = __devexit_p(wm5110_remove),
+};
+
+module_platform_driver(wm5110_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM5110 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm5110-codec");
diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h
new file mode 100644
index 000000000000..75e9351ccab0
--- /dev/null
+++ b/sound/soc/codecs/wm5110.h
@@ -0,0 +1,21 @@
+/*
+ * wm5110.h  --  WM5110 ALSA SoC Audio driver
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM5110_H
+#define _WM5110_H
+
+#include "arizona.h"
+
+#define WM5110_FLL1 1
+#define WM5110_FLL2 2
+
+#endif
-- 
cgit v1.2.3


From b761c0ca2e964a240d74e50da9e27dc0b3be0649 Mon Sep 17 00:00:00 2001
From: Matthias Kaehlcke <matthias@kaehlcke.net>
Date: Wed, 11 Jul 2012 17:36:34 +0200
Subject: ASoC: Free memory in the error paths of soc_of_parse_audio_routing()

Release the memory of the routing table before leaving the function upon errors
in the device tree

Signed-off-by: Matthias Kaehlcke <matthias@kaehlcke.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 64b464ca3bc5..f219b2f7ee68 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -4136,6 +4136,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			dev_err(card->dev,
 				"Property '%s' index %d could not be read: %d\n",
 				propname, 2 * i, ret);
+			kfree(routes);
 			return -EINVAL;
 		}
 		ret = of_property_read_string_index(np, propname,
@@ -4144,6 +4145,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			dev_err(card->dev,
 				"Property '%s' index %d could not be read: %d\n",
 				propname, (2 * i) + 1, ret);
+			kfree(routes);
 			return -EINVAL;
 		}
 	}
-- 
cgit v1.2.3


From e4dd76788c7e5b27165890d712c8c4f6f0abd645 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 11 Jul 2012 19:03:48 +0100
Subject: ASoC: wm8962: Redo early init of the part on resume

Ensure robust startup of the part by going through the reset procedure
prior to resyncing the full register cache, avoiding potential intermittent
faults in some designs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
---
 sound/soc/codecs/wm8962.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index beb709bd56cd..eaf65863ec21 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3725,6 +3725,9 @@ static int wm8962_runtime_resume(struct device *dev)
 	}
 
 	regcache_cache_only(wm8962->regmap, false);
+
+	wm8962_reset(wm8962);
+
 	regcache_sync(wm8962->regmap);
 
 	regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
-- 
cgit v1.2.3


From 98b3cf1290d2d6fbc926dc410d3713c8244e7604 Mon Sep 17 00:00:00 2001
From: Marek Belisko <marek.belisko@open-nandra.com>
Date: Thu, 12 Jul 2012 23:00:16 +0200
Subject: ASoC: dapm: Fix compilation warning
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fix following:
sound/soc/soc-dapm.c: In function ‘dapm_clock_event’:
sound/soc/soc-dapm.c:1021:1: warning: control reaches end of non-void function [-Wreturn-type]

Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 4ba47aab9801..f7a13f720529 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1019,6 +1019,7 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w,
 		return 0;
 	}
 #endif
+	return 0;
 }
 EXPORT_SYMBOL_GPL(dapm_clock_event);
 
-- 
cgit v1.2.3


From 0eed8a18696af4e6cf0315f935a730521b54725e Mon Sep 17 00:00:00 2001
From: Sylwester Nawrocki <sylvester.nawrocki@gmail.com>
Date: Fri, 13 Jul 2012 19:22:44 +0200
Subject: ASoC: Convert S3C24XX I2S driver to gpiolib API

The s3c2410_gpio* calls are obsolete and have been scheduled for
removal since several kernel releases. Remove them and use common
gpiolib API.
This patch is a prerequisite for removal of the obsolete S3C24XX
SoC GPIO definitions.

Tested on Micro2440-SDK.

Cc: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Sylwester Nawrocki <sylvester.nawrocki@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/samsung/s3c24xx-i2s.c | 10 +++-------
 1 file changed, 3 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index c4aa4d412fbf..0aae3a3883dc 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -23,7 +23,6 @@
 #include <sound/soc.h>
 #include <sound/pcm_params.h>
 
-#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 #include <plat/regs-iis.h>
 
@@ -391,12 +390,9 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai)
 	}
 	clk_enable(s3c24xx_i2s.iis_clk);
 
-	/* Configure the I2S pins in correct mode */
-	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
-	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+	/* Configure the I2S pins (GPE0...GPE4) in correct mode */
+	s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
+			      S3C_GPIO_PULL_NONE);
 
 	writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON);
 
-- 
cgit v1.2.3


From 601787c232306e0bb84fff9fc7c2be5a5c7b87a0 Mon Sep 17 00:00:00 2001
From: Sylwester Nawrocki <sylvester.nawrocki@gmail.com>
Date: Fri, 13 Jul 2012 19:22:45 +0200
Subject: ASoC: Convert S3C2412 I2S driver to gpiolib API

The s3c2410_gpio* calls are obsolete and have been scheduled for
removal since several kernel releases. Remove them and use common
gpiolib API.
This patch is a prerequisite for removal of the obsolete S3C24XX
SoC GPIO definitions.

Compile tested only.

Cc: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Sylwester Nawrocki <sylvester.nawrocki@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/samsung/s3c2412-i2s.c | 10 +++-------
 1 file changed, 3 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 79fbeea99d46..ac7701b3c5dc 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -25,7 +25,6 @@
 #include <sound/soc.h>
 #include <sound/pcm_params.h>
 
-#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
 #include "dma.h"
@@ -83,12 +82,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
 
 	s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
 
-	/* Configure the I2S pins in correct mode */
-	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
-	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+	/* Configure the I2S pins (GPE0...GPE4) in correct mode */
+	s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
+			      S3C_GPIO_PULL_NONE);
 
 	return 0;
 }
-- 
cgit v1.2.3


From 093eef416642c84265cced12335ff125f0db7313 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Sun, 20 Nov 2011 17:21:34 +0100
Subject: ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE

Instead of the hard-to-mantain TLV_DB_RANGE_HEAD macro, use
DECLARE_TLV_DB_RANGE, which computes its size automatically.

(Also make this data const on the way.)

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/es1938.c | 25 ++++++++++---------------
 1 file changed, 10 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 82c8d8c5c52a..a41106d745ca 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1321,35 +1321,30 @@ static int snd_es1938_put_double(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static unsigned int db_scale_master[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_master,
 	0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1),
 	54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0),
-};
+);
 
-static unsigned int db_scale_audio1[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_audio1,
 	0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0),
-};
+);
 
-static unsigned int db_scale_audio2[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_audio2,
 	0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0),
-};
+);
 
-static unsigned int db_scale_mic[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_mic,
 	0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(0, 150, 0),
-};
+);
 
-static unsigned int db_scale_line[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_line,
 	0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0),
-};
+);
 
 static const DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0);
 
-- 
cgit v1.2.3


From ca3273fb594ae4e953042c956eb40141ba46ad2c Mon Sep 17 00:00:00 2001
From: Ondrej Zary <linux@rainbow-software.org>
Date: Thu, 12 Jul 2012 23:30:31 +0200
Subject: ALSA: wss_lib: fix suspend/resume

By setting SNDRV_PCM_INFO_RESUME, wss_lib claims that it can restore the card
state fully on resume. But in fact, it can't as DMA is not restored so any
playback/capture running during suspend will fail to continue after resume.

Remove SNDRV_PCM_INFO_RESUME flag from pcm info field to fix the problem.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/wss/wss_lib.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound')

diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 49c8a0c2442c..00ca9fc4de0a 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1456,7 +1456,6 @@ static struct snd_pcm_hardware snd_wss_playback =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID |
-				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
 				 SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
-- 
cgit v1.2.3


From b4e2a16f992288f1c5cddda1948c53ba3dde9cb3 Mon Sep 17 00:00:00 2001
From: Ondrej Zary <linux@rainbow-software.org>
Date: Mon, 16 Jul 2012 08:30:18 +0200
Subject: ALSA: wss_lib: Fix resume on Yamaha OPL3-SAx

Yamaha OPL3-SAx chips don't resume properly when playback is running -
garbage is played after resume. Restoring the CS4231_PLAYBK_FORMAT register
last fixes the problem.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/wss/wss_lib.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 00ca9fc4de0a..360b08b03e1d 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1656,6 +1656,10 @@ static void snd_wss_resume(struct snd_wss *chip)
 			break;
 		}
 	}
+	/* Yamaha needs this to resume properly */
+	if (chip->hardware == WSS_HW_OPL3SA2)
+		snd_wss_out(chip, CS4231_PLAYBK_FORMAT,
+			    chip->image[CS4231_PLAYBK_FORMAT]);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 #if 1
 	snd_wss_mce_down(chip);
-- 
cgit v1.2.3


From e926f2c850c472f813f9bab486c68a3fe0b03ae4 Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Wed, 13 Jun 2012 10:23:51 +0800
Subject: ALSA: hda - Add DeviceID for Haswell HDA

this patch add proper id for Haswell HDA Controller.

[Added AZX_DCAPS_POSFIX_COMBO flag by tiwai]

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 796472d1ff5a..238653d55868 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, CPT},"
 			 "{Intel, PPT},"
 			 "{Intel, LPT},"
+			 "{Intel, HPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
 			 "{ATI, SB450},"
@@ -3261,6 +3262,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
 	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Haswell */
+	{ PCI_DEVICE(0x8086, 0x0c0c),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
-- 
cgit v1.2.3


From 1c76684d2752b3a24bb7da183cc18e5d126dbcc9 Mon Sep 17 00:00:00 2001
From: Wang Xingchao <xingchao.wang@intel.com>
Date: Wed, 13 Jun 2012 10:23:52 +0800
Subject: ALSA: hda - add Haswell HDMI codec id

0x80862807 is HDMI id for Haswell HDA.

Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index d6a8260a6f74..a57cf3665df9 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1927,6 +1927,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x80862804, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862805, .name = "CougarPoint HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
+{ .id = 0x80862807, .name = "Haswell HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862880, .name = "CedarTrail HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x808629fb, .name = "Crestline HDMI",	.patch = patch_generic_hdmi },
 {} /* terminator */
@@ -1978,6 +1979,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803");
 MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:80862805");
 MODULE_ALIAS("snd-hda-codec-id:80862806");
+MODULE_ALIAS("snd-hda-codec-id:80862807");
 MODULE_ALIAS("snd-hda-codec-id:80862880");
 MODULE_ALIAS("snd-hda-codec-id:808629fb");
 
-- 
cgit v1.2.3


From bdbe34dece4942f4d8df9865dba7785bb813366a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 Jul 2012 16:17:10 +0200
Subject: ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCH

According to Xingchao,
  This works for HDMI audio, otherwise there's blocking issue.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 238653d55868..b4f3c7295a53 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -3264,7 +3264,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* Haswell */
 	{ PCI_DEVICE(0x8086, 0x0c0c),
-	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
 	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
-- 
cgit v1.2.3


From 7ae48b56f8d9c836259bc02f3e2ea4962d6b5d1b Mon Sep 17 00:00:00 2001
From: Aaron Plattner <aplattner@nvidia.com>
Date: Mon, 16 Jul 2012 17:10:04 -0700
Subject: ALSA: hda - Add new GPU codec ID to snd-hda

Vendor ID 0x10de0051 is used by a yet-to-be-named GPU chip.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Andy Ritger <aritger@nvidia.com>
Reviewed-by: Daniel Dadap <ddadap@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index a57cf3665df9..0b4a1ea147c6 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1914,6 +1914,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
 { .id = 0x10de0042, .name = "GPU 42 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0043, .name = "GPU 43 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x11069f80, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
@@ -1965,6 +1966,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0041");
 MODULE_ALIAS("snd-hda-codec-id:10de0042");
 MODULE_ALIAS("snd-hda-codec-id:10de0043");
 MODULE_ALIAS("snd-hda-codec-id:10de0044");
+MODULE_ALIAS("snd-hda-codec-id:10de0051");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
 MODULE_ALIAS("snd-hda-codec-id:10de8001");
 MODULE_ALIAS("snd-hda-codec-id:11069f80");
-- 
cgit v1.2.3


From 5dd250728aa4cad49cfa18eb8ed11ba470ce382a Mon Sep 17 00:00:00 2001
From: Ondrej Zary <linux@rainbow-software.org>
Date: Tue, 17 Jul 2012 09:16:44 +0200
Subject: ALSA: snd-opti9xx: Implement suspend/resume

Implement suspend/resume support for Opti 92x and 93x chips.
Tested with Opti 929A+AD1848 and Opti 931.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/opti9xx/opti92x-ad1848.c | 67 +++++++++++++++++++++++++++++++++++---
 1 file changed, 63 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index ecc68dfe7b54..d7ce0125dcf2 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -136,8 +136,8 @@ struct snd_opti9xx {
 #ifdef OPTi93X
 	unsigned long mc_indir_index;
 	struct resource *res_mc_indir;
-	struct snd_wss *codec;
 #endif	/* OPTi93X */
+	struct snd_wss *codec;
 	unsigned long pwd_reg;
 
 	spinlock_t lock;
@@ -870,9 +870,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 			       &codec);
 	if (error < 0)
 		return error;
-#ifdef OPTi93X
 	chip->codec = codec;
-#endif
 	error = snd_wss_pcm(codec, 0, &pcm);
 	if (error < 0)
 		return error;
@@ -1053,11 +1051,55 @@ static int __devexit snd_opti9xx_isa_remove(struct device *devptr,
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int snd_opti9xx_suspend(struct snd_card *card)
+{
+	struct snd_opti9xx *chip = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	chip->codec->suspend(chip->codec);
+	return 0;
+}
+
+static int snd_opti9xx_resume(struct snd_card *card)
+{
+	struct snd_opti9xx *chip = card->private_data;
+	int error, xdma2;
+#if defined(CS4231) || defined(OPTi93X)
+	xdma2 = dma2;
+#else
+	xdma2 = -1;
+#endif
+
+	error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2,
+				      mpu_port, mpu_irq);
+	if (error)
+		return error;
+	chip->codec->resume(chip->codec);
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+
+static int snd_opti9xx_isa_suspend(struct device *dev, unsigned int n,
+				   pm_message_t state)
+{
+	return snd_opti9xx_suspend(dev_get_drvdata(dev));
+}
+
+static int snd_opti9xx_isa_resume(struct device *dev, unsigned int n)
+{
+	return snd_opti9xx_resume(dev_get_drvdata(dev));
+}
+#endif
+
 static struct isa_driver snd_opti9xx_driver = {
 	.match		= snd_opti9xx_isa_match,
 	.probe		= snd_opti9xx_isa_probe,
 	.remove		= __devexit_p(snd_opti9xx_isa_remove),
-	/* FIXME: suspend/resume */
+#ifdef CONFIG_PM
+	.suspend	= snd_opti9xx_isa_suspend,
+	.resume		= snd_opti9xx_isa_resume,
+#endif
 	.driver		= {
 		.name	= DEV_NAME
 	},
@@ -1123,12 +1165,29 @@ static void __devexit snd_opti9xx_pnp_remove(struct pnp_card_link * pcard)
 	snd_opti9xx_pnp_is_probed = 0;
 }
 
+#ifdef CONFIG_PM
+static int snd_opti9xx_pnp_suspend(struct pnp_card_link *pcard,
+				   pm_message_t state)
+{
+	return snd_opti9xx_suspend(pnp_get_card_drvdata(pcard));
+}
+
+static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard)
+{
+	return snd_opti9xx_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
 static struct pnp_card_driver opti9xx_pnpc_driver = {
 	.flags		= PNP_DRIVER_RES_DISABLE,
 	.name		= "opti9xx",
 	.id_table	= snd_opti9xx_pnpids,
 	.probe		= snd_opti9xx_pnp_probe,
 	.remove		= __devexit_p(snd_opti9xx_pnp_remove),
+#ifdef CONFIG_PM
+	.suspend	= snd_opti9xx_pnp_suspend,
+	.resume		= snd_opti9xx_pnp_resume,
+#endif
 };
 #endif
 
-- 
cgit v1.2.3


From 59b1f084abd8690ffe68c67758ad08bbcb7d1af0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Jul 2012 12:14:17 +0200
Subject: ALSA: opti9xx: Fix section mismatch by PM support

In the previous commit, snd_opti9xx_configure() is called from the
resume handler but it's still marked as __devinit.  Fix it.

Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/isa/opti9xx/opti92x-ad1848.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index d7ce0125dcf2..f8fbe22515c9 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -348,7 +348,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
 		(snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
 
 
-static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip,
+static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 					   long port,
 					   int irq, int dma1, int dma2,
 					   long mpu_port, int mpu_irq)
-- 
cgit v1.2.3


From f46c329644b1f7144d336fce037dd9f84ee1995f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Jul 2012 16:48:27 +0200
Subject: ALSA: hda - Fix index number conflicts of phantom jacks

Since some jack controls may be renamed as phantom jacks, the existing
check for index conflicts doesn't work because it simply compares the
name with the last used name, assuming that the controls with the same
name continue.  Thus, it would result in the duplicated controls when
two or more phantom jacks with the very same type exist, and the
driver gives up with an error.

This patch fixes the problem by checking the index number conflicts
more intensively (but dumbly).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_jack.c | 45 ++++++++++++++++++++++++++++++---------------
 1 file changed, 30 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 60c976f06280..aaccc0236bda 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -314,9 +314,28 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
 
+/* get the unique index number for the given kctl name */
+static int get_unique_index(struct hda_codec *codec, const char *name, int idx)
+{
+	struct hda_jack_tbl *jack;
+	int i, len = strlen(name);
+ again:
+	jack = codec->jacktbl.list;
+	for (i = 0; i < codec->jacktbl.used; i++, jack++) {
+		/* jack->kctl.id contains "XXX Jack" name string with index */
+		if (jack->kctl &&
+		    !strncmp(name, jack->kctl->id.name, len) &&
+		    !strcmp(" Jack", jack->kctl->id.name + len) &&
+		    jack->kctl->id.index == idx) {
+			idx++;
+			goto again;
+		}
+	}
+	return idx;
+}
+
 static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
-			 const struct auto_pin_cfg *cfg,
-			 char *lastname, int *lastidx)
+			 const struct auto_pin_cfg *cfg)
 {
 	unsigned int def_conf, conn;
 	char name[44];
@@ -336,10 +355,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
 	if (phantom_jack)
 		/* Example final name: "Internal Mic Phantom Jack" */
 		strncat(name, " Phantom", sizeof(name) - strlen(name) - 1);
-	if (!strcmp(name, lastname) && idx == *lastidx)
-		idx++;
-	strncpy(lastname, name, sizeof(name));
-	*lastidx = idx;
+	idx = get_unique_index(codec, name, idx);
 	err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack);
 	if (err < 0)
 		return err;
@@ -356,42 +372,41 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
 			   const struct auto_pin_cfg *cfg)
 {
 	const hda_nid_t *p;
-	int i, err, lastidx = 0;
-	char lastname[44] = "";
+	int i, err;
 
 	for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
 		if (*p == *cfg->line_out_pins) /* might be duplicated */
 			break;
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
 		if (*p == *cfg->line_out_pins) /* might be duplicated */
 			break;
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0; i < cfg->num_inputs; i++) {
-		err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
-	err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx);
+	err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
 	if (err < 0)
 		return err;
-	err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx);
+	err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
 	if (err < 0)
 		return err;
 	return 0;
-- 
cgit v1.2.3


From 4e01ec636e64707d202a1ca21a47bbc6d53085b7 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 18 Jul 2012 07:38:46 +0200
Subject: ALSA: hda - Add support for Realtek ALC282

This codec has a separate dmic path (separate dmic only ADC),
and thus it looks mostly like ALC275.

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1025377
Tested-by: Ray Chen <ray.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a5b0b50b6a92..aef31392aee1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6988,6 +6988,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
 	{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
 	{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
+	{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
-- 
cgit v1.2.3


From 639aa4bd58582f3015ae5621b7e9e754dcb58e6b Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 18 Jul 2012 18:02:53 +0200
Subject: ALSA: hda - make sure alc268 does not OOPS on codec parse

A recent commit made patch_alc268 call snd_hda_pick_fixup with
NULL quirk pointer. Make sure we do not reference that NULL pointer.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index f7520b9f909c..647218d69f68 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -727,7 +727,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 			models++;
 		}
 	}
-	if (id < 0) {
+	if (id < 0 && quirk) {
 		q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
 		if (q) {
 			id = q->value;
@@ -736,7 +736,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 #endif
 		}
 	}
-	if (id < 0) {
+	if (id < 0 && quirk) {
 		for (q = quirk; q->subvendor; q++) {
 			unsigned int vendorid =
 				q->subdevice | (q->subvendor << 16);
-- 
cgit v1.2.3


From b4046d013b5b9a7cab835def403f7f421cdf0cb6 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Jul 2012 19:11:30 +0100
Subject: ASoC: wm8994: Update micdet for irqdomain conversion

The conversion of the core driver to irqdomains means that we don't need
and irq_base to have working interrupts so use wm8994_request_irq() to
deal with looking up the interrupt number for the micdet IRQ.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8994.c | 7 ++++---
 1 file changed, 4 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 7bb875230dc0..65763388649c 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3695,9 +3695,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 	if (wm8994->pdata && wm8994->pdata->micdet_irq)
 		wm8994->micdet_irq = wm8994->pdata->micdet_irq;
-	else if (wm8994->pdata && wm8994->pdata->irq_base)
-		wm8994->micdet_irq = wm8994->pdata->irq_base +
-				     WM8994_IRQ_MIC1_DET;
 
 	pm_runtime_enable(codec->dev);
 	pm_runtime_idle(codec->dev);
@@ -3836,6 +3833,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 				dev_warn(codec->dev,
 					 "Failed to request Mic detect IRQ: %d\n",
 					 ret);
+		} else {
+			wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET,
+					   wm8958_mic_irq, "Mic detect",
+					   wm8994);
 		}
 	}
 
-- 
cgit v1.2.3


From 31a2239a5a77c48b12c54210aa250ce76c8f9535 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 18 Jul 2012 19:16:06 +0100
Subject: ASoC: littlemill: Add userspace control of the WM1250 I/O

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

(limited to 'sound')

diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index c82c646b8a08..ee52c8a00779 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -211,6 +211,11 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("WM1250 Input"),
+	SOC_DAPM_PIN_SWITCH("WM1250 Output"),
+};
+
 static struct snd_soc_dapm_widget widgets[] = {
 	SND_SOC_DAPM_HP("Headphone", NULL),
 
@@ -282,6 +287,8 @@ static struct snd_soc_card littlemill = {
 	.set_bias_level = littlemill_set_bias_level,
 	.set_bias_level_post = littlemill_set_bias_level_post,
 
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
 	.dapm_widgets = widgets,
 	.num_dapm_widgets = ARRAY_SIZE(widgets),
 	.dapm_routes = audio_paths,
-- 
cgit v1.2.3


From 409b78cc17a4a3d07a541037575da648ced99437 Mon Sep 17 00:00:00 2001
From: Torben Hohn <torbenh@linutronix.de>
Date: Wed, 18 Jul 2012 15:01:17 +0200
Subject: ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define

The MX31 Audmux also has 7 Ports.
This patch adds the missing define, and makes the debugfs code iterate
over that port too.

Signed-off-by: Torben Hohn <torbenh@linutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/fsl/imx-audmux.c | 2 +-
 sound/soc/fsl/imx-audmux.h | 1 +
 2 files changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 080327414c6b..e7c800ebbd75 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void)
 		return;
 	}
 
-	for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) {
+	for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
 		snprintf(buf, sizeof(buf), "ssi%d", i);
 		if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
 					 (void *)i, &audmux_debugfs_fops))
diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
index 04ebbab8d7b9..b8ff44b9dafa 100644
--- a/sound/soc/fsl/imx-audmux.h
+++ b/sound/soc/fsl/imx-audmux.h
@@ -14,6 +14,7 @@
 #define MX31_AUDMUX_PORT4_SSI_PINS_4	3
 #define MX31_AUDMUX_PORT5_SSI_PINS_5	4
 #define MX31_AUDMUX_PORT6_SSI_PINS_6	5
+#define MX31_AUDMUX_PORT7_SSI_PINS_7	6
 
 #define MX51_AUDMUX_PORT1_SSI0		0
 #define MX51_AUDMUX_PORT2_SSI1		1
-- 
cgit v1.2.3


From 9e76e6d031482194a5b24d8e9ab88063fbd6b4b5 Mon Sep 17 00:00:00 2001
From: Dylan Reid <dgreid@chromium.org>
Date: Thu, 19 Jul 2012 17:52:58 -0700
Subject: ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is
enabled in open, but is disabled in hdmi_init_pin which is called during
system resume.  This causes a system suspend/resume during playback to
mute HDMI/DP. Enabling the pin in prepare instead of open allows calling
snd_pcm_prepare after a system resume to restore audio.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 0b4a1ea147c6..641408dc28c0 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -876,7 +876,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	struct hdmi_spec_per_pin *per_pin;
 	struct hdmi_eld *eld;
 	struct hdmi_spec_per_cvt *per_cvt = NULL;
-	int pinctl;
 
 	/* Validate hinfo */
 	pin_idx = hinfo_to_pin_index(spec, hinfo);
@@ -912,11 +911,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	snd_hda_codec_write(codec, per_pin->pin_nid, 0,
 			    AC_VERB_SET_CONNECT_SEL,
 			    mux_idx);
-	pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
-				    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-	snd_hda_codec_write(codec, per_pin->pin_nid, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pinctl | PIN_OUT);
 	snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid);
 
 	/* Initially set the converter's capabilities */
@@ -1153,11 +1147,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx = hinfo_to_pin_index(spec, hinfo);
 	hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid;
+	int pinctl;
 
 	hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels);
 
 	hdmi_setup_audio_infoframe(codec, pin_idx, substream);
 
+	pinctl = snd_hda_codec_read(codec, pin_nid, 0,
+				    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+	snd_hda_codec_write(codec, pin_nid, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT);
+
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
 
-- 
cgit v1.2.3


From 108cc108a3bb42fe4705df1317ff98e1e29428a6 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 20 Jul 2012 10:37:25 +0200
Subject: ALSA: hda - add dock support for Thinkpad X230 Tablet

Also add a model/fixup string "lenovo-dock", so that other Thinkpad
users will be able to test this fixup easily, to see if it enables
dock I/O for them as well.

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1026953
Tested-by: John McCarron <john.mccarron@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++
 1 file changed, 27 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 537c365716a6..f141395dfee6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5881,6 +5881,15 @@ static int alc269_resume(struct hda_codec *codec)
 }
 #endif /* CONFIG_PM */
 
+static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec,
+						 const struct alc_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == ALC_FIXUP_ACT_PRE_PROBE)
+		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+}
+
 static void alc269_fixup_hweq(struct hda_codec *codec,
 			       const struct alc_fixup *fix, int action)
 {
@@ -6007,6 +6016,8 @@ enum {
 	ALC269VB_FIXUP_DMIC,
 	ALC269_FIXUP_MIC2_MUTE_LED,
 	ALC269_FIXUP_INV_DMIC,
+	ALC269_FIXUP_LENOVO_DOCK,
+	ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -6135,6 +6146,20 @@ static const struct alc_fixup alc269_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc_fixup_inv_dmic_0x12,
 	},
+	[ALC269_FIXUP_LENOVO_DOCK] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x19, 0x23a11040 }, /* dock mic */
+			{ 0x1b, 0x2121103f }, /* dock headphone */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT
+	},
+	[ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -6161,6 +6186,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -6222,6 +6248,7 @@ static const struct alc_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"},
 	{.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"},
 	{.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
 	{}
 };
 
-- 
cgit v1.2.3


From 0ff97ebf0804d2e519d578fcb4db03f104d2ca8c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 20 Jul 2012 17:29:34 +0100
Subject: ASoC: dapm: Fix _PRE and _POST events for DAPM performance
 improvements

Ever since the DAPM performance improvements we've been marking all widgets
as not dirty after each DAPM run. Since _PRE and _POST events aren't part
of the DAPM graph this has rendered them non-functional, they will never be
marked dirty again and thus will never be run again.

Fix this by skipping them when marking widgets as not dirty.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@vger.kernel.org
---
 sound/soc/soc-dapm.c | 10 +++++++++-
 1 file changed, 9 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f7a13f720529..025060b26fb7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1598,7 +1598,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 	}
 
 	list_for_each_entry(w, &card->widgets, list) {
-		list_del_init(&w->dirty);
+		switch (w->id) {
+		case snd_soc_dapm_pre:
+		case snd_soc_dapm_post:
+			/* These widgets always need to be powered */
+			break;
+		default:
+			list_del_init(&w->dirty);
+			break;
+		}
 
 		if (w->power) {
 			d = w->dapm;
-- 
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