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, 12 insertions, 10 deletions

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 00b2c43d28a1..cac7e557edc8 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -274,10 +274,10 @@ static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
 	unsigned int reg2 = mc->rreg;
 	int val[2], val2[2], i;
 
-	val[0] = snd_soc_component_read32(component, reg) & 0x3f;
-	val[1] = (snd_soc_component_read32(component, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
-	val2[0] = snd_soc_component_read32(component, reg2) & 0x3f;
-	val2[1] = (snd_soc_component_read32(component, PM860X_SIDETONE_SHIFT)) & 0xf;
+	val[0] = snd_soc_component_read(component, reg) & 0x3f;
+	val[1] = (snd_soc_component_read(component, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
+	val2[0] = snd_soc_component_read(component, reg2) & 0x3f;
+	val2[1] = (snd_soc_component_read(component, PM860X_SIDETONE_SHIFT)) & 0xf;
 
 	for (i = 0; i < ARRAY_SIZE(st_table); i++) {
 		if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
@@ -333,8 +333,8 @@ static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
 	int max = mc->max, val, val2;
 	unsigned int mask = (1 << fls(max)) - 1;
 
-	val = snd_soc_component_read32(component, reg) >> shift;
-	val2 = snd_soc_component_read32(component, reg2) >> shift;
+	val = snd_soc_component_read(component, reg) >> shift;
+	val2 = snd_soc_component_read(component, reg2) >> shift;
 	ucontrol->value.integer.value[0] = (max - val) & mask;
 	ucontrol->value.integer.value[1] = (max - val2) & mask;
 
@@ -426,7 +426,7 @@ static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
 			snd_soc_component_update_bits(component, PM860X_EAR_CTRL_2,
 					    RSYNC_CHANGE, RSYNC_CHANGE);
 			/* update dac */
-			data = snd_soc_component_read32(component, PM860X_DAC_EN_2);
+			data = snd_soc_component_read(component, PM860X_DAC_EN_2);
 			data &= ~dac;
 			if (!(data & (DAC_LEFT | DAC_RIGHT)))
 				data &= ~MODULATOR;
@@ -902,7 +902,7 @@ static const struct snd_soc_dapm_route pm860x_dapm_routes[] = {
  * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
  * These bits can also be used to mute.
  */
-static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int pm860x_mute_stream(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
@@ -1136,17 +1136,19 @@ static int pm860x_set_bias_level(struct snd_soc_component *component,
 }
 
 static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
-	.digital_mute	= pm860x_digital_mute,
+	.mute_stream	= pm860x_mute_stream,
 	.hw_params	= pm860x_pcm_hw_params,
 	.set_fmt	= pm860x_pcm_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
+	.no_capture_mute = 1,
 };
 
 static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
-	.digital_mute	= pm860x_digital_mute,
+	.mute_stream	= pm860x_mute_stream,
 	.hw_params	= pm860x_i2s_hw_params,
 	.set_fmt	= pm860x_i2s_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
+	.no_capture_mute = 1,
 };
 
 #define PM860X_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |	\