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, 19 deletions

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index e172913d04a4..3d05c37f676e 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -330,7 +330,7 @@ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol,
 
 	if (ucontrol->value.integer.value[0]) {
 		/* Check if noise suppression is enabled */
-		if (snd_soc_component_read32(component, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) {
+		if (snd_soc_component_read(component, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) {
 			dev_dbg(component->dev,
 				"Disable noise suppression to enable ALC\n");
 			return -EINVAL;
@@ -354,27 +354,27 @@ static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol,
 
 	if (ucontrol->value.integer.value[0]) {
 		/* Check if ALC is enabled */
-		if (snd_soc_component_read32(component, DA7210_ADC) & DA7210_ADC_ALC_EN)
+		if (snd_soc_component_read(component, DA7210_ADC) & DA7210_ADC_ALC_EN)
 			goto err;
 
 		/* Check ZC for HP and AUX1 PGA */
-		if ((snd_soc_component_read32(component, DA7210_ZERO_CROSS) &
+		if ((snd_soc_component_read(component, DA7210_ZERO_CROSS) &
 			(DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC |
 			DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC |
 			DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC))
 			goto err;
 
 		/* Check INPGA_L_VOL and INPGA_R_VOL */
-		val = snd_soc_component_read32(component, DA7210_IN_GAIN);
+		val = snd_soc_component_read(component, DA7210_IN_GAIN);
 		if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) ||
 			(((val & DA7210_INPGA_R_VOL) >> 4) <
 			DA7210_INPGA_MIN_VOL_NS))
 			goto err;
 
 		/* Check AUX1_L_VOL and AUX1_R_VOL */
-		if (((snd_soc_component_read32(component, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) <
+		if (((snd_soc_component_read(component, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) <
 		    DA7210_AUX1_MIN_VOL_NS) ||
-		    ((snd_soc_component_read32(component, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) <
+		    ((snd_soc_component_read(component, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) <
 		    DA7210_AUX1_MIN_VOL_NS))
 			goto err;
 	}
@@ -767,7 +767,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	/* Enable DAI */
 	snd_soc_component_write(component, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN);
 
-	dai_cfg1 = 0xFC & snd_soc_component_read32(component, DA7210_DAI_CFG1);
+	dai_cfg1 = 0xFC & snd_soc_component_read(component, DA7210_DAI_CFG1);
 
 	switch (params_width(params)) {
 	case 16:
@@ -874,11 +874,11 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 	u32 dai_cfg1;
 	u32 dai_cfg3;
 
-	dai_cfg1 = 0x7f & snd_soc_component_read32(component, DA7210_DAI_CFG1);
-	dai_cfg3 = 0xfc & snd_soc_component_read32(component, DA7210_DAI_CFG3);
+	dai_cfg1 = 0x7f & snd_soc_component_read(component, DA7210_DAI_CFG1);
+	dai_cfg3 = 0xfc & snd_soc_component_read(component, DA7210_DAI_CFG3);
 
-	if ((snd_soc_component_read32(component, DA7210_PLL) & DA7210_PLL_EN) &&
-		(!(snd_soc_component_read32(component, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
+	if ((snd_soc_component_read(component, DA7210_PLL) & DA7210_PLL_EN) &&
+		(!(snd_soc_component_read(component, DA7210_PLL_DIV3) & DA7210_PLL_BYP)))
 		return -EINVAL;
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -924,10 +924,10 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 	return 0;
 }
 
-static int da7210_mute(struct snd_soc_dai *dai, int mute)
+static int da7210_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
-	u8 mute_reg = snd_soc_component_read32(component, DA7210_DAC_HPF) & 0xFB;
+	u8 mute_reg = snd_soc_component_read(component, DA7210_DAC_HPF) & 0xFB;
 
 	if (mute)
 		snd_soc_component_write(component, DA7210_DAC_HPF, mute_reg | 0x4);
@@ -971,14 +971,16 @@ static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 /**
  * da7210_set_dai_pll	:Configure the codec PLL
- * @param codec_dai	: pointer to codec DAI
- * @param pll_id	: da7210 has only one pll, so pll_id is always zero
- * @param fref		: MCLK frequency, should be < 20MHz
- * @param fout		: FsDM value, Refer page 44 & 45 of datasheet
- * @return int		: Zero for success, negative error code for error
+ * @codec_dai: pointer to codec DAI
+ * @pll_id: da7210 has only one pll, so pll_id is always zero
+ * @source: clock source
+ * @fref: MCLK frequency, should be < 20MHz
+ * @fout: FsDM value, Refer page 44 & 45 of datasheet
  *
  * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz,
  *       19.2MHz, 19.6MHz and 19.8MHz
+ *
+ * Return: Zero for success, negative error code for error
  */
 static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 			      int source, unsigned int fref, unsigned int fout)
@@ -1034,7 +1036,8 @@ static const struct snd_soc_dai_ops da7210_dai_ops = {
 	.set_fmt	= da7210_set_dai_fmt,
 	.set_sysclk	= da7210_set_dai_sysclk,
 	.set_pll	= da7210_set_dai_pll,
-	.digital_mute	= da7210_mute,
+	.mute_stream	= da7210_mute,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver da7210_dai = {