Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became wide and scattered updates all over the sound tree as
  diffstat shows: lots of (still ongoing) refactoring works in ASoC,
  fixes and cleanups caught by static analysis, inclusive term
  conversions as well as lots of new drivers. Below are highlights:

  ASoC core:
   - API cleanups and conversions to the unified mute_stream() call
   - Simplify I/O helper functions
   - Use helper macros to retrieve RTD from substreams

  ASoC drivers:
   - Lots of fixes and cleanups in Intel ASoC drivers
   - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
     Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
     nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
     boards, TI J721e EVM

  ALSA core:
   - Minor code refacotring for SG-buffer handling

  HD-audio:
   - Generalization of mute-LED handling with LED classdev
   - Intel silent stream support for HDMI
   - Device-specific fixes: CA0132, Loongson-3

  Others:
   - Usual USB- and HD-audio quirks for various devices
   - Fixes for echoaudio DMA position handling
   - Various documents and trivial fixes for sparse warnings
   - Conversion to adopt inclusive terms"

* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
  ALSA: pci: delete repeated words in comments
  ALSA: isa: delete repeated words in comments
  ALSA: hda/tegra: Add 100us dma stop delay
  ALSA: hda: Add dma stop delay variable
  ASoC: hda/tegra: Set buffer alignment to 128 bytes
  ALSA: seq: oss: Serialize ioctls
  ALSA: hda/hdmi: Add quirk to force connectivity
  ALSA: usb-audio: add startech usb audio dock name
  ALSA: usb-audio: Add support for Lenovo ThinkStation P620
  Revert "ALSA: hda: call runtime_allow() for all hda controllers"
  ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
  ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
  ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
  ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
  ALSA: docs: fix typo
  ALSA: doc: use correct config variable name
  ASoC: core: Two step component registration
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
  ...

1 files changed, 23 insertions, 22 deletions

diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index f8375d67e901..16bc8609d0d2 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -139,7 +139,7 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
 		return ret;
 
 	/* now hit the volume update bits (always bit 8) */
-	val = snd_soc_component_read32(component, reg);
+	val = snd_soc_component_read(component, reg);
 	return snd_soc_component_write(component, reg, val | 0x0100);
 }
 
@@ -364,7 +364,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
 
 	switch (reg_shift) {
 	case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8):
-		reg = snd_soc_component_read32(component, WM8991_OUTPUT_MIXER1);
+		reg = snd_soc_component_read(component, WM8991_OUTPUT_MIXER1);
 		if (reg & WM8991_LDLO) {
 			printk(KERN_WARNING
 			       "Cannot set as Output Mixer 1 LDLO Set\n");
@@ -373,7 +373,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
 		break;
 
 	case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8):
-		reg = snd_soc_component_read32(component, WM8991_OUTPUT_MIXER2);
+		reg = snd_soc_component_read(component, WM8991_OUTPUT_MIXER2);
 		if (reg & WM8991_RDRO) {
 			printk(KERN_WARNING
 			       "Cannot set as Output Mixer 2 RDRO Set\n");
@@ -382,7 +382,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
 		break;
 
 	case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8):
-		reg = snd_soc_component_read32(component, WM8991_SPEAKER_MIXER);
+		reg = snd_soc_component_read(component, WM8991_SPEAKER_MIXER);
 		if (reg & WM8991_LDSPK) {
 			printk(KERN_WARNING
 			       "Cannot set as Speaker Mixer LDSPK Set\n");
@@ -391,7 +391,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
 		break;
 
 	case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8):
-		reg = snd_soc_component_read32(component, WM8991_SPEAKER_MIXER);
+		reg = snd_soc_component_read(component, WM8991_SPEAKER_MIXER);
 		if (reg & WM8991_RDSPK) {
 			printk(KERN_WARNING
 			       "Cannot set as Speaker Mixer RDSPK Set\n");
@@ -922,12 +922,12 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
 		pll_factors(&pll_div, freq_out * 4, freq_in);
 
 		/* Turn on PLL */
-		reg = snd_soc_component_read32(component, WM8991_POWER_MANAGEMENT_2);
+		reg = snd_soc_component_read(component, WM8991_POWER_MANAGEMENT_2);
 		reg |= WM8991_PLL_ENA;
 		snd_soc_component_write(component, WM8991_POWER_MANAGEMENT_2, reg);
 
 		/* sysclk comes from PLL */
-		reg = snd_soc_component_read32(component, WM8991_CLOCKING_2);
+		reg = snd_soc_component_read(component, WM8991_CLOCKING_2);
 		snd_soc_component_write(component, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
 
 		/* set up N , fractional mode and pre-divisor if necessary */
@@ -937,7 +937,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
 		snd_soc_component_write(component, WM8991_PLL3, (u8)(pll_div.k & 0xFF));
 	} else {
 		/* Turn on PLL */
-		reg = snd_soc_component_read32(component, WM8991_POWER_MANAGEMENT_2);
+		reg = snd_soc_component_read(component, WM8991_POWER_MANAGEMENT_2);
 		reg &= ~WM8991_PLL_ENA;
 		snd_soc_component_write(component, WM8991_POWER_MANAGEMENT_2, reg);
 	}
@@ -953,8 +953,8 @@ static int wm8991_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_component *component = codec_dai->component;
 	u16 audio1, audio3;
 
-	audio1 = snd_soc_component_read32(component, WM8991_AUDIO_INTERFACE_1);
-	audio3 = snd_soc_component_read32(component, WM8991_AUDIO_INTERFACE_3);
+	audio1 = snd_soc_component_read(component, WM8991_AUDIO_INTERFACE_1);
+	audio3 = snd_soc_component_read(component, WM8991_AUDIO_INTERFACE_3);
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -1008,22 +1008,22 @@ static int wm8991_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 
 	switch (div_id) {
 	case WM8991_MCLK_DIV:
-		reg = snd_soc_component_read32(component, WM8991_CLOCKING_2) &
+		reg = snd_soc_component_read(component, WM8991_CLOCKING_2) &
 		      ~WM8991_MCLK_DIV_MASK;
 		snd_soc_component_write(component, WM8991_CLOCKING_2, reg | div);
 		break;
 	case WM8991_DACCLK_DIV:
-		reg = snd_soc_component_read32(component, WM8991_CLOCKING_2) &
+		reg = snd_soc_component_read(component, WM8991_CLOCKING_2) &
 		      ~WM8991_DAC_CLKDIV_MASK;
 		snd_soc_component_write(component, WM8991_CLOCKING_2, reg | div);
 		break;
 	case WM8991_ADCCLK_DIV:
-		reg = snd_soc_component_read32(component, WM8991_CLOCKING_2) &
+		reg = snd_soc_component_read(component, WM8991_CLOCKING_2) &
 		      ~WM8991_ADC_CLKDIV_MASK;
 		snd_soc_component_write(component, WM8991_CLOCKING_2, reg | div);
 		break;
 	case WM8991_BCLK_DIV:
-		reg = snd_soc_component_read32(component, WM8991_CLOCKING_1) &
+		reg = snd_soc_component_read(component, WM8991_CLOCKING_1) &
 		      ~WM8991_BCLK_DIV_MASK;
 		snd_soc_component_write(component, WM8991_CLOCKING_1, reg | div);
 		break;
@@ -1042,7 +1042,7 @@ static int wm8991_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_component *component = dai->component;
-	u16 audio1 = snd_soc_component_read32(component, WM8991_AUDIO_INTERFACE_1);
+	u16 audio1 = snd_soc_component_read(component, WM8991_AUDIO_INTERFACE_1);
 
 	audio1 &= ~WM8991_AIF_WL_MASK;
 	/* bit size */
@@ -1064,12 +1064,12 @@ static int wm8991_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8991_mute(struct snd_soc_dai *dai, int mute)
+static int wm8991_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 val;
 
-	val  = snd_soc_component_read32(component, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE;
+	val  = snd_soc_component_read(component, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE;
 	if (mute)
 		snd_soc_component_write(component, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE);
 	else
@@ -1089,7 +1089,7 @@ static int wm8991_set_bias_level(struct snd_soc_component *component,
 
 	case SND_SOC_BIAS_PREPARE:
 		/* VMID=2*50k */
-		val = snd_soc_component_read32(component, WM8991_POWER_MANAGEMENT_1) &
+		val = snd_soc_component_read(component, WM8991_POWER_MANAGEMENT_1) &
 		      ~WM8991_VMID_MODE_MASK;
 		snd_soc_component_write(component, WM8991_POWER_MANAGEMENT_1, val | 0x2);
 		break;
@@ -1146,7 +1146,7 @@ static int wm8991_set_bias_level(struct snd_soc_component *component,
 		}
 
 		/* VMID=2*250k */
-		val = snd_soc_component_read32(component, WM8991_POWER_MANAGEMENT_1) &
+		val = snd_soc_component_read(component, WM8991_POWER_MANAGEMENT_1) &
 		      ~WM8991_VMID_MODE_MASK;
 		snd_soc_component_write(component, WM8991_POWER_MANAGEMENT_1, val | 0x4);
 		break;
@@ -1162,7 +1162,7 @@ static int wm8991_set_bias_level(struct snd_soc_component *component,
 			      WM8991_BUFIOEN);
 
 		/* mute DAC */
-		val = snd_soc_component_read32(component, WM8991_DAC_CTRL);
+		val = snd_soc_component_read(component, WM8991_DAC_CTRL);
 		snd_soc_component_write(component, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE);
 
 		/* Enable any disabled outputs */
@@ -1196,10 +1196,11 @@ static int wm8991_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8991_ops = {
 	.hw_params = wm8991_hw_params,
-	.digital_mute = wm8991_mute,
+	.mute_stream = wm8991_mute,
 	.set_fmt = wm8991_set_dai_fmt,
 	.set_clkdiv = wm8991_set_dai_clkdiv,
-	.set_pll = wm8991_set_dai_pll
+	.set_pll = wm8991_set_dai_pll,
+	.no_capture_mute = 1,
 };
 
 /*