Merge tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been major modernization with the standard bus: in ALSA
  sequencer core and HD-audio.  Also, HD-audio receives the regmap
  support replacing the in-house cache register cache code.  These
  changes shouldn't impact the existing behavior, but rather
  refactoring.

  In addition, HD-audio got the code split to a core library part and
  the "legacy" driver parts.  This is a preliminary work for adapting
  the upcoming ASoC HD-audio driver, and the whole transition is still
  work in progress, likely finished in 4.1.

  Along with them, there are many updates in ASoC area as usual, too:
  lots of cleanups, Intel code shuffling, etc.

  Here are some highlights:

  ALSA core:
   - PCM: the audio timestamp / wallclock enhancement
   - PCM: fixes in DPCM management
   - Fixes / cleanups of user-space control element management
   - Sequencer: modernization using the standard bus

  HD-audio:
   - Modernization using the standard bus
   - Regmap support
   - Use standard runtime PM for codec power saving
   - Widget-path based power-saving for IDT, VIA and Realtek codecs
   - Reorganized sysfs entries for each codec object
   - More Dell headset support

  ASoC:
   - Move of jack registration to the card level
   - Lots of ASoC cleanups, mainly moving things from the CODEC level to
     the card level
   - Support for DAPM routes specified by both the machine driver and DT
   - Continuing improvements to rcar
   - pcm512x enhacements
   - Intel platforms updates
   - rt5670 updates / fixes
   - New platforms / devices: some non-DSP Qualcomm platforms, Google's
     Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC

  Misc:
   - ice1724: Improved ESI W192M support
   - emu10k1: Emu 1010 fixes/enhancement"

* tag 'sound-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
  ALSA: hda - set GET bit when adding a vendor verb to the codec regmap
  ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
  ALSA: hda - Fix another race in runtime PM refcounting
  ALSA: hda - Expose codec type sysfs
  ALSA: ctl: fix to handle several elements added by one operation for userspace element
  ASoC: Intel: fix array_size.cocci warnings
  ASoC: n810: Automatically disconnect non-connected pins
  ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
  ASoC: davinci-evm: Use card DAPM context to access widgets
  ASoC: mop500_ab8500: Use card DAPM context to access widgets
  ASoC: wm1133-ev1: Use card DAPM context to access widgets
  ASoC: atmel: Improve machine driver compile test coverage
  ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
  ALSA: control: Fix a typo of SNDRV_CTL_ELEM_ACCESS_TLV_* with SNDRV_CTL_TLV_OP_*
  ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
  ASoC: rnsd: fix build regression without CONFIG_OF
  ALSA: emu10k1: add toggles for E-mu 1010 optical ports
  ALSA: ctl: fill identical information to return value when adding userspace elements
  ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls
  ALSA: ctl: confirm to return all identical information in 'activate' event
  ...

43 files changed, 2845 insertions, 656 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index ea9f0e31f9d4..061c46587628 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -70,6 +70,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98090 if I2C
 	select SND_SOC_MAX98095 if I2C
 	select SND_SOC_MAX98357A if GPIOLIB
+	select SND_SOC_MAX98925 if I2C
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9768 if I2C
 	select SND_SOC_MAX9877 if I2C
@@ -141,7 +142,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8770 if SPI_MASTER
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8782
-	select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8804_I2C if I2C
+	select SND_SOC_WM8804_SPI if SPI_MASTER
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
 	select SND_SOC_WM8904 if I2C
@@ -460,6 +462,9 @@ config SND_SOC_MAX98095
 config SND_SOC_MAX98357A
        tristate
 
+config SND_SOC_MAX98925
+       tristate
+
 config SND_SOC_MAX9850
 	tristate
 
@@ -744,8 +749,19 @@ config SND_SOC_WM8782
 	tristate
 
 config SND_SOC_WM8804
-	tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver"
-	depends on SND_SOC_I2C_AND_SPI
+	tristate
+
+config SND_SOC_WM8804_I2C
+	tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C"
+	depends on I2C
+	select SND_SOC_WM8804
+	select REGMAP_I2C
+
+config SND_SOC_WM8804_SPI
+	tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI"
+	depends on SPI_MASTER
+	select SND_SOC_WM8804
+	select REGMAP_SPI
 
 config SND_SOC_WM8900
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 69b8666d187a..abe2d7edf65c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -65,6 +65,7 @@ snd-soc-max98088-objs := max98088.o
 snd-soc-max98090-objs := max98090.o
 snd-soc-max98095-objs := max98095.o
 snd-soc-max98357a-objs := max98357a.o
+snd-soc-max98925-objs := max98925.o
 snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
@@ -145,6 +146,8 @@ snd-soc-wm8770-objs := wm8770.o
 snd-soc-wm8776-objs := wm8776.o
 snd-soc-wm8782-objs := wm8782.o
 snd-soc-wm8804-objs := wm8804.o
+snd-soc-wm8804-i2c-objs := wm8804-i2c.o
+snd-soc-wm8804-spi-objs := wm8804-spi.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
 snd-soc-wm8904-objs := wm8904.o
@@ -247,6 +250,7 @@ obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98090)	+= snd-soc-max98090.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
 obj-$(CONFIG_SND_SOC_MAX98357A)	+= snd-soc-max98357a.o
+obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
@@ -323,6 +327,8 @@ obj-$(CONFIG_SND_SOC_WM8770)	+= snd-soc-wm8770.o
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
 obj-$(CONFIG_SND_SOC_WM8782)	+= snd-soc-wm8782.o
 obj-$(CONFIG_SND_SOC_WM8804)	+= snd-soc-wm8804.o
+obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o
+obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8904)	+= snd-soc-wm8904.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 7895689588da..88ca9cb0ce79 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2003,7 +2003,6 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 
 	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)
@@ -2036,7 +2035,6 @@ static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec,
 
 	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)
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index 70ab35744aba..7ad8e156e2df 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap,
 		adau1977->dvdd_reg = NULL;
 	}
 
-	adau1977->reset_gpio = devm_gpiod_get(dev, "reset");
-	if (IS_ERR(adau1977->reset_gpio)) {
-		ret = PTR_ERR(adau1977->reset_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return PTR_ERR(adau1977->reset_gpio);
-		adau1977->reset_gpio = NULL;
-	}
+	adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset",
+						       GPIOD_OUT_LOW);
+	if (IS_ERR(adau1977->reset_gpio))
+		return PTR_ERR(adau1977->reset_gpio);
 
 	dev_set_drvdata(dev, adau1977);
 
-	if (adau1977->reset_gpio) {
-		ret = gpiod_direction_output(adau1977->reset_gpio, 0);
-		if (ret)
-			return ret;
+	if (adau1977->reset_gpio)
 		ndelay(100);
-	}
 
 	ret = adau1977_power_enable(adau1977);
 	if (ret)
diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c
index 16ce9f9fefa1..298dedc05140 100644
--- a/sound/soc/codecs/ak4554.c
+++ b/sound/soc/codecs/ak4554.c
@@ -84,7 +84,7 @@ static int ak4554_soc_remove(struct platform_device *pdev)
 	return 0;
 }
 
-static struct of_device_id ak4554_of_match[] = {
+static const struct of_device_id ak4554_of_match[] = {
 	{ .compatible = "asahi-kasei,ak4554" },
 	{},
 };
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index dde8b49c19ad..13585e88f597 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -97,6 +97,9 @@
 #define PMMP		(1 << 2) /* MPWR pin Power Management */
 #define MGAIN0		(1 << 0) /* MIC amp gain*/
 
+/* SG_SL2 */
+#define LOPS		(1 << 6) /* Stero Line-out Power Save Mode */
+
 /* TIMER */
 #define ZTM(param)	((param & 0x3) << 4) /* ALC Zero Crossing TimeOut */
 #define WTM(param)	(((param & 0x4) << 4) | ((param & 0x3) << 2))
@@ -168,6 +171,29 @@ static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
 	SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
 };
 
+/* event handlers */
+static int ak4642_lout_event(struct snd_soc_dapm_widget *w,
+			     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMD:
+	case SND_SOC_DAPM_PRE_PMU:
+		/* Power save mode ON */
+		snd_soc_update_bits(codec, SG_SL2, LOPS, LOPS);
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+	case SND_SOC_DAPM_POST_PMD:
+		/* Power save mode OFF */
+		mdelay(300);
+		snd_soc_update_bits(codec, SG_SL2, LOPS, 0);
+		break;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 
 	/* Outputs */
@@ -182,12 +208,15 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 
 	SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
 
-	SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
+	SND_SOC_DAPM_MIXER_E("LINEOUT Mixer", PW_MGMT1, 3, 0,
 			   &ak4642_lout_mixer_controls[0],
-			   ARRAY_SIZE(ak4642_lout_mixer_controls)),
+			   ARRAY_SIZE(ak4642_lout_mixer_controls),
+			   ak4642_lout_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+			   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
 
 	/* DAC */
-	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
+	SND_SOC_DAPM_DAC("DAC", NULL, PW_MGMT1, 2, 0),
 };
 
 static const struct snd_soc_dapm_route ak4642_intercon[] = {
@@ -205,6 +234,8 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = {
 	{"DACH", NULL, "DAC"},
 
 	{"LINEOUT Mixer", "DACL", "DAC"},
+
+	{ "DAC", NULL, "Playback" },
 };
 
 /*
@@ -468,13 +499,13 @@ static struct snd_soc_dai_driver ak4642_dai = {
 	.name = "ak4642-hifi",
 	.playback = {
 		.stream_name = "Playback",
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE },
 	.capture = {
 		.stream_name = "Capture",
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE },
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 95d31d6291ac..57da0ceda03f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1902,7 +1902,7 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll)
 static int arizona_enable_fll(struct arizona_fll *fll)
 {
 	struct arizona *arizona = fll->arizona;
-	int ret;
+	unsigned long time_left;
 	bool use_sync = false;
 	int already_enabled = arizona_is_enabled_fll(fll);
 	struct arizona_fll_cfg cfg;
@@ -1978,9 +1978,9 @@ static int arizona_enable_fll(struct arizona_fll *fll)
 		regmap_update_bits_async(arizona->regmap, fll->base + 1,
 					 ARIZONA_FLL1_FREERUN, 0);
 
-	ret = wait_for_completion_timeout(&fll->ok,
+	time_left = wait_for_completion_timeout(&fll->ok,
 					  msecs_to_jiffies(250));
-	if (ret == 0)
+	if (time_left == 0)
 		arizona_fll_warn(fll, "Timed out waiting for lock\n");
 
 	return 0;
diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c
index f2b8aad21274..60598b230341 100644
--- a/sound/soc/codecs/cs35l32.c
+++ b/sound/soc/codecs/cs35l32.c
@@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client,
 	}
 
 	/* Reset the Device */
-	cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev,
-		"reset-gpios");
-	if (IS_ERR(cs35l32->reset_gpio)) {
-		ret = PTR_ERR(cs35l32->reset_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
-
-		cs35l32->reset_gpio = NULL;
-	} else {
-		ret = gpiod_direction_output(cs35l32->reset_gpio, 0);
-		if (ret)
-			return ret;
+	cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
+		"reset", GPIOD_OUT_LOW);
+	if (IS_ERR(cs35l32->reset_gpio))
+		return PTR_ERR(cs35l32->reset_gpio);
+
+	if (cs35l32->reset_gpio)
 		gpiod_set_value_cansleep(cs35l32->reset_gpio, 1);
-	}
 
 	/* initialize codec */
 	ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, &reg);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index ce6086835ebd..cac48ddf3ba6 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client,
 		return ret;
 	}
 
-	cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev,
-		"reset-gpios");
-	if (IS_ERR(cs4265->reset_gpio)) {
-		ret = PTR_ERR(cs4265->reset_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
-
-		cs4265->reset_gpio = NULL;
-	} else {
-		ret = gpiod_direction_output(cs4265->reset_gpio, 0);
-		if (ret)
-			return ret;
+	cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
+		"reset", GPIOD_OUT_LOW);
+	if (IS_ERR(cs4265->reset_gpio))
+		return PTR_ERR(cs4265->reset_gpio);
+
+	if (cs4265->reset_gpio) {
 		mdelay(1);
 		gpiod_set_value_cansleep(cs4265->reset_gpio, 1);
-
 	}
 
 	i2c_set_clientdata(i2c_client, cs4265);
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 7d3a6accaf9a..e770ee6f36da 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -561,10 +561,10 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec)
 	if (gpio_is_valid(cs4271->gpio_nreset)) {
 		/* Reset codec */
 		gpio_direction_output(cs4271->gpio_nreset, 0);
-		udelay(1);
+		mdelay(1);
 		gpio_set_value(cs4271->gpio_nreset, 1);
 		/* Give the codec time to wake up */
-		udelay(1);
+		mdelay(1);
 	}
 
 	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 0b10979513c4..0f334bc1b63c 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -420,7 +420,7 @@ static int cx20442_platform_probe(struct platform_device *pdev)
 			&cx20442_codec_dev, &cx20442_dai, 1);
 }
 
-static int __exit cx20442_platform_remove(struct platform_device *pdev)
+static int cx20442_platform_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_codec(&pdev->dev);
 	return 0;
@@ -431,7 +431,7 @@ static struct platform_driver cx20442_platform_driver = {
 		.name = "cx20442-codec",
 		},
 	.probe = cx20442_platform_probe,
-	.remove = __exit_p(cx20442_platform_remove),
+	.remove = cx20442_platform_remove,
 };
 
 module_platform_driver(cx20442_platform_driver);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index b112b1c2c394..3e33ef2acf3c 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2605,8 +2605,24 @@ err_enable:
 	return ret;
 }
 
+static void max98090_i2c_shutdown(struct i2c_client *i2c)
+{
+	struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev);
+
+	/*
+	 * Enable volume smoothing, disable zero cross.  This will cause
+	 * a quick 40ms ramp to mute on shutdown.
+	 */
+	regmap_write(max98090->regmap,
+		M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK);
+	regmap_write(max98090->regmap,
+		M98090_REG_DEVICE_SHUTDOWN, 0x00);
+	msleep(40);
+}
+
 static int max98090_i2c_remove(struct i2c_client *client)
 {
+	max98090_i2c_shutdown(client);
 	snd_soc_unregister_codec(&client->dev);
 	return 0;
 }
@@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = {
 		.acpi_match_table = ACPI_PTR(max98090_acpi_match),
 	},
 	.probe  = max98090_i2c_probe,
+	.shutdown = max98090_i2c_shutdown,
 	.remove = max98090_i2c_remove,
 	.id_table = max98090_i2c_id,
 };
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index e9e6efbc21dd..bf3e933ee895 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -26,8 +26,6 @@
 #include <sound/soc-dai.h>
 #include <sound/soc-dapm.h>
 
-#define DRV_NAME "max98357a"
-
 static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 		int cmd, struct snd_soc_dai *dai)
 {
@@ -87,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = {
 };
 
 static struct snd_soc_dai_driver max98357a_dai_driver = {
-	.name = DRV_NAME,
+	.name = "HiFi",
 	.playback = {
-		.stream_name	= DRV_NAME "-playback",
+		.stream_name	= "HiFi Playback",
 		.formats	= SNDRV_PCM_FMTBIT_S16 |
 					SNDRV_PCM_FMTBIT_S24 |
 					SNDRV_PCM_FMTBIT_S32,
@@ -127,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev)
 
 #ifdef CONFIG_OF
 static const struct of_device_id max98357a_device_id[] = {
-	{ .compatible = "maxim," DRV_NAME, },
+	{ .compatible = "maxim,max98357a" },
 	{}
 };
 MODULE_DEVICE_TABLE(of, max98357a_device_id);
@@ -135,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id);
 
 static struct platform_driver max98357a_platform_driver = {
 	.driver = {
-		.name = DRV_NAME,
+		.name = "max98357a",
 		.of_match_table = of_match_ptr(max98357a_device_id),
 	},
 	.probe	= max98357a_platform_probe,
@@ -145,4 +143,3 @@ module_platform_driver(max98357a_platform_driver);
 
 MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver");
 MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c
new file mode 100644
index 000000000000..9b5a17de4690
--- /dev/null
+++ b/sound/soc/codecs/max98925.c
@@ -0,0 +1,655 @@
+/*
+ * max98925.c -- ALSA SoC Stereo MAX98925 driver
+ * Copyright 2013-15 Maxim Integrated Products
+ * 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/delay.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/cdev.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "max98925.h"
+
+static const char *const dai_text[] = {
+	"Left", "Right", "LeftRight", "LeftRightDiv2",
+};
+
+static const char * const max98925_boost_voltage_text[] = {
+	"8.5V", "8.25V", "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V",
+	"6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V",	"6.5V", "6.5V"
+};
+
+static SOC_ENUM_SINGLE_DECL(max98925_boost_voltage,
+	MAX98925_CONFIGURATION, M98925_BST_VOUT_SHIFT,
+	max98925_boost_voltage_text);
+
+static const char *const hpf_text[] = {
+	"Disable", "DC Block", "100Hz",	"200Hz", "400Hz", "800Hz",
+};
+
+static const struct reg_default max98925_reg[] = {
+	{ 0x0B, 0x00 }, /* IRQ Enable0 */
+	{ 0x0C, 0x00 }, /* IRQ Enable1 */
+	{ 0x0D, 0x00 }, /* IRQ Enable2 */
+	{ 0x0E, 0x00 }, /* IRQ Clear0 */
+	{ 0x0F, 0x00 }, /* IRQ Clear1 */
+	{ 0x10, 0x00 }, /* IRQ Clear2 */
+	{ 0x11, 0xC0 }, /* Map0 */
+	{ 0x12, 0x00 }, /* Map1 */
+	{ 0x13, 0x00 }, /* Map2 */
+	{ 0x14, 0xF0 }, /* Map3 */
+	{ 0x15, 0x00 }, /* Map4 */
+	{ 0x16, 0xAB }, /* Map5 */
+	{ 0x17, 0x89 }, /* Map6 */
+	{ 0x18, 0x00 }, /* Map7 */
+	{ 0x19, 0x00 }, /* Map8 */
+	{ 0x1A, 0x06 }, /* DAI Clock Mode 1 */
+	{ 0x1B, 0xC0 }, /* DAI Clock Mode 2 */
+	{ 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */
+	{ 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */
+	{ 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */
+	{ 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */
+	{ 0x20, 0x50 }, /* Format */
+	{ 0x21, 0x00 }, /* TDM Slot Select */
+	{ 0x22, 0x00 }, /* DOUT Configuration VMON */
+	{ 0x23, 0x00 }, /* DOUT Configuration IMON */
+	{ 0x24, 0x00 }, /* DOUT Configuration VBAT */
+	{ 0x25, 0x00 }, /* DOUT Configuration VBST */
+	{ 0x26, 0x00 }, /* DOUT Configuration FLAG */
+	{ 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */
+	{ 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */
+	{ 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */
+	{ 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */
+	{ 0x2B, 0x02 }, /* DOUT Drive Strength */
+	{ 0x2C, 0x90 }, /* Filters */
+	{ 0x2D, 0x00 }, /* Gain */
+	{ 0x2E, 0x02 }, /* Gain Ramping */
+	{ 0x2F, 0x00 }, /* Speaker Amplifier */
+	{ 0x30, 0x0A }, /* Threshold */
+	{ 0x31, 0x00 }, /* ALC Attack */
+	{ 0x32, 0x80 }, /* ALC Atten and Release */
+	{ 0x33, 0x00 }, /* ALC Infinite Hold Release */
+	{ 0x34, 0x92 }, /* ALC Configuration */
+	{ 0x35, 0x01 }, /* Boost Converter */
+	{ 0x36, 0x00 }, /* Block Enable */
+	{ 0x37, 0x00 }, /* Configuration */
+	{ 0x38, 0x00 }, /* Global Enable */
+	{ 0x3A, 0x00 }, /* Boost Limiter */
+};
+
+static const struct soc_enum max98925_dai_enum =
+	SOC_ENUM_SINGLE(MAX98925_GAIN, 5, ARRAY_SIZE(dai_text), dai_text);
+
+static const struct soc_enum max98925_hpf_enum =
+	SOC_ENUM_SINGLE(MAX98925_FILTERS, 0, ARRAY_SIZE(hpf_text), hpf_text);
+
+static const struct snd_kcontrol_new max98925_hpf_sel_mux =
+	SOC_DAPM_ENUM("Rc Filter MUX Mux", max98925_hpf_enum);
+
+static const struct snd_kcontrol_new max98925_dai_sel_mux =
+	SOC_DAPM_ENUM("DAI IN MUX Mux", max98925_dai_enum);
+
+static int max98925_dac_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		regmap_update_bits(max98925->regmap,
+			MAX98925_BLOCK_ENABLE,
+			M98925_BST_EN_MASK |
+			M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK,
+			M98925_BST_EN_MASK |
+			M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		regmap_update_bits(max98925->regmap,
+			MAX98925_BLOCK_ENABLE, M98925_BST_EN_MASK |
+			M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, 0);
+		break;
+	default:
+		return 0;
+	}
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget max98925_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MUX("DAI IN MUX", SND_SOC_NOPM, 0, 0,
+				&max98925_dai_sel_mux),
+	SND_SOC_DAPM_MUX("Rc Filter MUX", SND_SOC_NOPM, 0, 0,
+				&max98925_hpf_sel_mux),
+	SND_SOC_DAPM_DAC_E("Amp Enable", NULL, MAX98925_BLOCK_ENABLE,
+			M98925_SPK_EN_SHIFT, 0, max98925_dac_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("Global Enable", MAX98925_GLOBAL_ENABLE,
+			M98925_EN_SHIFT, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("BE_OUT"),
+};
+
+static const struct snd_soc_dapm_route max98925_audio_map[] = {
+	{"DAI IN MUX", "Left", "DAI_OUT"},
+	{"DAI IN MUX", "Right", "DAI_OUT"},
+	{"DAI IN MUX", "LeftRight", "DAI_OUT"},
+	{"DAI IN MUX", "LeftRightDiv2", "DAI_OUT"},
+	{"Rc Filter MUX", "Disable", "DAI IN MUX"},
+	{"Rc Filter MUX", "DC Block", "DAI IN MUX"},
+	{"Rc Filter MUX", "100Hz", "DAI IN MUX"},
+	{"Rc Filter MUX", "200Hz", "DAI IN MUX"},
+	{"Rc Filter MUX", "400Hz", "DAI IN MUX"},
+	{"Rc Filter MUX", "800Hz", "DAI IN MUX"},
+	{"Amp Enable", NULL, "Rc Filter MUX"},
+	{"BE_OUT", NULL, "Amp Enable"},
+	{"BE_OUT", NULL, "Global Enable"},
+};
+
+static bool max98925_volatile_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX98925_VBAT_DATA:
+	case MAX98925_VBST_DATA:
+	case MAX98925_LIVE_STATUS0:
+	case MAX98925_LIVE_STATUS1:
+	case MAX98925_LIVE_STATUS2:
+	case MAX98925_STATE0:
+	case MAX98925_STATE1:
+	case MAX98925_STATE2:
+	case MAX98925_FLAG0:
+	case MAX98925_FLAG1:
+	case MAX98925_FLAG2:
+	case MAX98925_REV_VERSION:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool max98925_readable_register(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case MAX98925_IRQ_CLEAR0:
+	case MAX98925_IRQ_CLEAR1:
+	case MAX98925_IRQ_CLEAR2:
+	case MAX98925_ALC_HOLD_RLS:
+		return false;
+	default:
+		return true;
+	}
+}
+
+static DECLARE_TLV_DB_SCALE(max98925_spk_tlv, -600, 100, 0);
+
+static const struct snd_kcontrol_new max98925_snd_controls[] = {
+	SOC_SINGLE_TLV("Speaker Volume", MAX98925_GAIN,
+		M98925_SPK_GAIN_SHIFT, (1<<M98925_SPK_GAIN_WIDTH