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, 22 insertions, 18 deletions

diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index f70b9f7e68bb..653da3eaf355 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -64,7 +64,7 @@ static const struct reg_default rt5631_reg[] = {
 	{ RT5631_PSEUDO_SPATL_CTRL, 0x0553 },
 };
 
-/**
+/*
  * rt5631_write_index - write index register of 2nd layer
  */
 static void rt5631_write_index(struct snd_soc_component *component,
@@ -74,7 +74,7 @@ static void rt5631_write_index(struct snd_soc_component *component,
 	snd_soc_component_write(component, RT5631_INDEX_DATA, value);
 }
 
-/**
+/*
  * rt5631_read_index - read index register of 2nd layer
  */
 static unsigned int rt5631_read_index(struct snd_soc_component *component,
@@ -83,7 +83,7 @@ static unsigned int rt5631_read_index(struct snd_soc_component *component,
 	unsigned int value;
 
 	snd_soc_component_write(component, RT5631_INDEX_ADD, reg);
-	value = snd_soc_component_read32(component, RT5631_INDEX_DATA);
+	value = snd_soc_component_read(component, RT5631_INDEX_DATA);
 
 	return value;
 }
@@ -285,7 +285,7 @@ static int check_sysclk1_source(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_GLOBAL_CLK_CTRL);
+	reg = snd_soc_component_read(component, RT5631_GLOBAL_CLK_CTRL);
 	return reg & RT5631_SYSCLK_SOUR_SEL_PLL;
 }
 
@@ -303,7 +303,7 @@ static int check_dacl_to_outmixl(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_OUTMIXER_L_CTRL);
+	reg = snd_soc_component_read(component, RT5631_OUTMIXER_L_CTRL);
 	return !(reg & RT5631_M_DAC_L_TO_OUTMIXER_L);
 }
 
@@ -313,7 +313,7 @@ static int check_dacr_to_outmixr(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_OUTMIXER_R_CTRL);
+	reg = snd_soc_component_read(component, RT5631_OUTMIXER_R_CTRL);
 	return !(reg & RT5631_M_DAC_R_TO_OUTMIXER_R);
 }
 
@@ -323,7 +323,7 @@ static int check_dacl_to_spkmixl(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_SPK_MIXER_CTRL);
+	reg = snd_soc_component_read(component, RT5631_SPK_MIXER_CTRL);
 	return !(reg & RT5631_M_DAC_L_TO_SPKMIXER_L);
 }
 
@@ -333,7 +333,7 @@ static int check_dacr_to_spkmixr(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_SPK_MIXER_CTRL);
+	reg = snd_soc_component_read(component, RT5631_SPK_MIXER_CTRL);
 	return !(reg & RT5631_M_DAC_R_TO_SPKMIXER_R);
 }
 
@@ -343,7 +343,7 @@ static int check_adcl_select(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_ADC_REC_MIXER);
+	reg = snd_soc_component_read(component, RT5631_ADC_REC_MIXER);
 	return !(reg & RT5631_M_MIC1_TO_RECMIXER_L);
 }
 
@@ -353,12 +353,13 @@ static int check_adcr_select(struct snd_soc_dapm_widget *source,
 	struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm);
 	unsigned int reg;
 
-	reg = snd_soc_component_read32(component, RT5631_ADC_REC_MIXER);
+	reg = snd_soc_component_read(component, RT5631_ADC_REC_MIXER);
 	return !(reg & RT5631_M_MIC2_TO_RECMIXER_R);
 }
 
 /**
  * onebit_depop_power_stage - auto depop in power stage.
+ * @component: ASoC component
  * @enable: power on/off
  *
  * When power on/off headphone, the depop sequence is done by hardware.
@@ -372,9 +373,9 @@ static void onebit_depop_power_stage(struct snd_soc_component *component, int en
 				RT5631_EN_ONE_BIT_DEPOP, 0);
 
 	/* keep soft volume and zero crossing setting */
-	soft_vol = snd_soc_component_read32(component, RT5631_SOFT_VOL_CTRL);
+	soft_vol = snd_soc_component_read(component, RT5631_SOFT_VOL_CTRL);
 	snd_soc_component_write(component, RT5631_SOFT_VOL_CTRL, 0);
-	hp_zc = snd_soc_component_read32(component, RT5631_INT_ST_IRQ_CTRL_2);
+	hp_zc = snd_soc_component_read(component, RT5631_INT_ST_IRQ_CTRL_2);
 	snd_soc_component_write(component, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff);
 	if (enable) {
 		/* config one-bit depop parameter */
@@ -397,6 +398,7 @@ static void onebit_depop_power_stage(struct snd_soc_component *component, int en
 
 /**
  * onebit_depop_mute_stage - auto depop in mute stage.
+ * @component: ASoC component
  * @enable: mute/unmute
  *
  * When mute/unmute headphone, the depop sequence is done by hardware.
@@ -410,9 +412,9 @@ static void onebit_depop_mute_stage(struct snd_soc_component *component, int ena
 				RT5631_EN_ONE_BIT_DEPOP, 0);
 
 	/* keep soft volume and zero crossing setting */
-	soft_vol = snd_soc_component_read32(component, RT5631_SOFT_VOL_CTRL);
+	soft_vol = snd_soc_component_read(component, RT5631_SOFT_VOL_CTRL);
 	snd_soc_component_write(component, RT5631_SOFT_VOL_CTRL, 0);
-	hp_zc = snd_soc_component_read32(component, RT5631_INT_ST_IRQ_CTRL_2);
+	hp_zc = snd_soc_component_read(component, RT5631_INT_ST_IRQ_CTRL_2);
 	snd_soc_component_write(component, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff);
 	if (enable) {
 		schedule_timeout_uninterruptible(msecs_to_jiffies(10));
@@ -435,6 +437,7 @@ static void onebit_depop_mute_stage(struct snd_soc_component *component, int ena
 
 /**
  * onebit_depop_power_stage - step by step depop sequence in power stage.
+ * @component: ASoC component
  * @enable: power on/off
  *
  * When power on/off headphone, the depop sequence is done in step by step.
@@ -448,9 +451,9 @@ static void depop_seq_power_stage(struct snd_soc_component *component, int enabl
 		RT5631_EN_ONE_BIT_DEPOP, RT5631_EN_ONE_BIT_DEPOP);
 
 	/* keep soft volume and zero crossing setting */
-	soft_vol = snd_soc_component_read32(component, RT5631_SOFT_VOL_CTRL);
+	soft_vol = snd_soc_component_read(component, RT5631_SOFT_VOL_CTRL);
 	snd_soc_component_write(component, RT5631_SOFT_VOL_CTRL, 0);
-	hp_zc = snd_soc_component_read32(component, RT5631_INT_ST_IRQ_CTRL_2);
+	hp_zc = snd_soc_component_read(component, RT5631_INT_ST_IRQ_CTRL_2);
 	snd_soc_component_write(component, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff);
 	if (enable) {
 		/* config depop sequence parameter */
@@ -507,6 +510,7 @@ static void depop_seq_power_stage(struct snd_soc_component *component, int enabl
 
 /**
  * depop_seq_mute_stage - step by step depop sequence in mute stage.
+ * @component: ASoC component
  * @enable: mute/unmute
  *
  * When mute/unmute headphone, the depop sequence is done in step by step.
@@ -520,9 +524,9 @@ static void depop_seq_mute_stage(struct snd_soc_component *component, int enable
 		RT5631_EN_ONE_BIT_DEPOP, RT5631_EN_ONE_BIT_DEPOP);
 
 	/* keep soft volume and zero crossing setting */
-	soft_vol = snd_soc_component_read32(component, RT5631_SOFT_VOL_CTRL);
+	soft_vol = snd_soc_component_read(component, RT5631_SOFT_VOL_CTRL);
 	snd_soc_component_write(component, RT5631_SOFT_VOL_CTRL, 0);
-	hp_zc = snd_soc_component_read32(component, RT5631_INT_ST_IRQ_CTRL_2);
+	hp_zc = snd_soc_component_read(component, RT5631_INT_ST_IRQ_CTRL_2);
 	snd_soc_component_write(component, RT5631_INT_ST_IRQ_CTRL_2, hp_zc & 0xf7ff);
 	if (enable) {
 		schedule_timeout_uninterruptible(msecs_to_jiffies(10));