Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...

4 files changed, 395 insertions, 6 deletions

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index fe3995ce9b38..e7b83179aca2 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,6 +1,6 @@
 config SND_SOC_SAMSUNG
 	tristate "ASoC support for Samsung"
-	depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4
+	depends on PLAT_SAMSUNG
 	select S3C64XX_DMA if ARCH_S3C64XX
 	select S3C2410_DMA if ARCH_S3C24XX
 	help
@@ -191,6 +191,7 @@ config SND_SOC_SPEYSIDE
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8996
 	select SND_SOC_WM9081
+	select SND_SOC_WM0010
 	select SND_SOC_WM1250_EV1
 
 config SND_SOC_TOBERMORY
@@ -199,6 +200,14 @@ config SND_SOC_TOBERMORY
 	select SND_SAMSUNG_I2S
 	select SND_SOC_WM8962
 
+config SND_SOC_BELLS
+	tristate "Audio support for Wolfson Bells"
+	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+	select SND_SAMSUNG_I2S
+	select SND_SOC_WM5102
+	select SND_SOC_WM5110
+	select SND_SOC_WM9081
+
 config SND_SOC_LOWLAND
 	tristate "Audio support for Wolfson Lowland"
 	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 9d03beb40c86..709f6059ad67 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -42,6 +42,7 @@ snd-soc-speyside-objs := speyside.o
 snd-soc-tobermory-objs := tobermory.o
 snd-soc-lowland-objs := lowland.o
 snd-soc-littlemill-objs := littlemill.o
+snd-soc-bells-objs := bells.o
 
 obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -65,3 +66,4 @@ obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
 obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o
 obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o
 obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
+obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
new file mode 100644
index 000000000000..5dc10dfc0d42
--- /dev/null
+++ b/sound/soc/samsung/bells.c
@@ -0,0 +1,346 @@
+/*
+ * Bells audio support
+ *
+ * Copyright 2012 Wolfson Microelectronics
+ *
+ * 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 <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+
+#include "../codecs/wm5102.h"
+#include "../codecs/wm9081.h"
+
+/*
+ * 44.1kHz based clocks for the SYSCLK domain, use a very high clock
+ * to allow all the DSP functionality to be enabled if desired.
+ */
+#define SYSCLK_RATE (44100 * 1024)
+
+/* 48kHz based clocks for the ASYNC domain */
+#define ASYNCCLK_RATE (48000 * 512)
+
+/* BCLK2 is fixed at this currently */
+#define BCLK2_RATE (64 * 8000)
+
+/*
+ * Expect a 24.576MHz crystal if one is fitted (the driver will function
+ * if this is not fitted).
+ */
+#define MCLK_RATE 24576000
+
+#define WM9081_AUDIO_RATE 44100
+#define WM9081_MCLK_RATE  (WM9081_AUDIO_RATE * 256)
+
+static int bells_set_bias_level(struct snd_soc_card *card,
+				struct snd_soc_dapm_context *dapm,
+				enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+			ret = snd_soc_codec_set_pll(codec, WM5102_FLL1,
+						    ARIZONA_FLL_SRC_MCLK1,
+						    MCLK_RATE,
+						    SYSCLK_RATE);
+			if (ret < 0)
+				pr_err("Failed to start FLL: %d\n", ret);
+
+			ret = snd_soc_codec_set_pll(codec, WM5102_FLL2,
+						    ARIZONA_FLL_SRC_AIF2BCLK,
+						    BCLK2_RATE,
+						    ASYNCCLK_RATE);
+			if (ret < 0)
+				pr_err("Failed to start FLL: %d\n", ret);
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int bells_set_bias_level_post(struct snd_soc_card *card,
+				     struct snd_soc_dapm_context *dapm,
+				     enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_STANDBY:
+		ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, 0, 0, 0);
+		if (ret < 0) {
+			pr_err("Failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, 0, 0, 0);
+		if (ret < 0) {
+			pr_err("Failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	dapm->bias_level = level;
+
+	return 0;
+}
+
+static int bells_late_probe(struct snd_soc_card *card)
+{
+	struct snd_soc_codec *codec = card->rtd[0].codec;
+	struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai;
+	struct snd_soc_dai *aif3_dai = card->rtd[2].cpu_dai;
+	struct snd_soc_dai *wm9081_dai = card->rtd[2].codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0);
+	if (ret != 0) {
+		dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0);
+	if (ret != 0) {
+		dev_err(aif2_dai->dev, "Failed to set AIF2 clock: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0);
+	if (ret != 0) {
+		dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK,
+				       ARIZONA_CLK_SRC_FLL1, SYSCLK_RATE,
+				       SND_SOC_CLOCK_IN);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_OPCLK, 0,
+				       WM9081_MCLK_RATE, SND_SOC_CLOCK_OUT);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set OPCLK: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK,
+				       ARIZONA_CLK_SRC_FLL2, ASYNCCLK_RATE,
+				       SND_SOC_CLOCK_IN);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_codec_set_sysclk(wm9081_dai->codec, WM9081_SYSCLK_MCLK,
+				       0, WM9081_MCLK_RATE, 0);
+	if (ret != 0) {
+		dev_err(wm9081_dai->dev, "Failed to set MCLK: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_pcm_stream baseband_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static const struct snd_soc_pcm_stream sub_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = WM9081_AUDIO_RATE,
+	.rate_max = WM9081_AUDIO_RATE,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static struct snd_soc_dai_link bells_dai_wm5102[] = {
+	{
+		.name = "CPU",
+		.stream_name = "CPU",
+		.cpu_dai_name = "samsung-i2s.0",
+		.codec_dai_name = "wm5102-aif1",
+		.platform_name = "samsung-audio",
+		.codec_name = "wm5102-codec",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+	},
+	{
+		.name = "Baseband",
+		.stream_name = "Baseband",
+		.cpu_dai_name = "wm5102-aif2",
+		.codec_dai_name = "wm1250-ev1",
+		.codec_name = "wm1250-ev1.1-0027",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.ignore_suspend = 1,
+		.params = &baseband_params,
+	},
+	{
+		.name = "Sub",
+		.stream_name = "Sub",
+		.cpu_dai_name = "wm5102-aif3",
+		.codec_dai_name = "wm9081-hifi",
+		.codec_name = "wm9081.1-006c",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.params = &sub_params,
+	},
+};
+
+static struct snd_soc_dai_link bells_dai_wm5110[] = {
+	{
+		.name = "CPU",
+		.stream_name = "CPU",
+		.cpu_dai_name = "samsung-i2s.0",
+		.codec_dai_name = "wm5110-aif1",
+		.platform_name = "samsung-audio",
+		.codec_name = "wm5110-codec",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+	},
+	{
+		.name = "Baseband",
+		.stream_name = "Baseband",
+		.cpu_dai_name = "wm5110-aif2",
+		.codec_dai_name = "wm1250-ev1",
+		.codec_name = "wm1250-ev1.1-0027",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.ignore_suspend = 1,
+		.params = &baseband_params,
+	},
+	{
+		.name = "Sub",
+		.stream_name = "Sub",
+		.cpu_dai_name = "wm5102-aif3",
+		.codec_dai_name = "wm9081-hifi",
+		.codec_name = "wm9081.1-006c",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBS_CFS,
+		.ignore_suspend = 1,
+		.params = &sub_params,
+	},
+};
+
+static struct snd_soc_codec_conf bells_codec_conf[] = {
+	{
+		.dev_name = "wm9081.1-006c",
+		.name_prefix = "Sub",
+	},
+};
+
+static struct snd_soc_dapm_route bells_routes[] = {
+	{ "Sub CLK_SYS", NULL, "OPCLK" },
+};
+
+static struct snd_soc_card bells_cards[] = {
+	{
+		.name = "Bells WM5102",
+		.owner = THIS_MODULE,
+		.dai_link = bells_dai_wm5102,
+		.num_links = ARRAY_SIZE(bells_dai_wm5102),
+		.codec_conf = bells_codec_conf,
+		.num_configs = ARRAY_SIZE(bells_codec_conf),
+
+		.late_probe = bells_late_probe,
+
+		.dapm_routes = bells_routes,
+		.num_dapm_routes = ARRAY_SIZE(bells_routes),
+
+		.set_bias_level = bells_set_bias_level,
+		.set_bias_level_post = bells_set_bias_level_post,
+	},
+	{
+		.name = "Bells WM5110",
+		.owner = THIS_MODULE,
+		.dai_link = bells_dai_wm5110,
+		.num_links = ARRAY_SIZE(bells_dai_wm5110),
+		.codec_conf = bells_codec_conf,
+		.num_configs = ARRAY_SIZE(bells_codec_conf),
+
+		.late_probe = bells_late_probe,
+
+		.dapm_routes = bells_routes,
+		.num_dapm_routes = ARRAY_SIZE(bells_routes),
+
+		.set_bias_level = bells_set_bias_level,
+		.set_bias_level_post = bells_set_bias_level_post,
+	},
+};
+
+
+static __devinit int bells_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	bells_cards[pdev->id].dev = &pdev->dev;
+
+	ret = snd_soc_register_card(&bells_cards[pdev->id]);
+	if (ret) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card(%s) failed: %d\n",
+			bells_cards[pdev->id].name, ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit bells_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&bells_cards[pdev->id]);
+
+	return 0;
+}
+
+static struct platform_driver bells_driver = {
+	.driver = {
+		.name = "bells",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bells_probe,
+	.remove = __devexit_p(bells_remove),
+};
+
+module_platform_driver(bells_driver);
+
+MODULE_DESCRIPTION("Bells audio support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bells");
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index a4a9fc7e8c76..c7e1c28528a4 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -25,7 +25,7 @@ static int speyside_set_bias_level(struct snd_soc_card *card,
 				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level)
 {
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai;
 	int ret;
 
 	if (dapm->dev != codec_dai->dev)
@@ -57,7 +57,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card,
 					struct snd_soc_dapm_context *dapm,
 					enum snd_soc_bias_level level)
 {
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai;
 	int ret;
 
 	if (dapm->dev != codec_dai->dev)
@@ -126,6 +126,18 @@ static void speyside_set_polarity(struct snd_soc_codec *codec,
 	snd_soc_dapm_sync(&codec->dapm);
 }
 
+static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *dai = rtd->codec_dai;
@@ -172,17 +184,37 @@ static int speyside_late_probe(struct snd_soc_card *card)
 	return 0;
 }
 
+static const struct snd_soc_pcm_stream dsp_codec_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 48000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
 static struct snd_soc_dai_link speyside_dai[] = {
 	{
-		.name = "CPU",
-		.stream_name = "CPU",
+		.name = "CPU-DSP",
+		.stream_name = "CPU-DSP",
 		.cpu_dai_name = "samsung-i2s.0",
-		.codec_dai_name = "wm8996-aif1",
+		.codec_dai_name = "wm0010-sdi1",
 		.platform_name = "samsung-audio",
+		.codec_name = "spi0.0",
+		.init = speyside_wm0010_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+	},
+	{
+		.name = "DSP-CODEC",
+		.stream_name = "DSP-CODEC",
+		.cpu_dai_name = "wm0010-sdi2",
+		.codec_dai_name = "wm8996-aif1",
 		.codec_name = "wm8996.1-001a",
 		.init = speyside_wm8996_init,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dsp_codec_params,
+		.ignore_suspend = 1,
 	},
 	{
 		.name = "Baseband",