Merge tag 'sound-6.0-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "As the diffstat shows, we've had lots of developments in a wide range
  at this time; the majority of changes are about ASoC, including
  subsystem-wide cleanups, continued SOF / Intel updates and a bunch of
  new drivers (as usual), while there have been some significant (but
  almost invisible) improvements in ALSA core side, too.

  Below are some highlights:

  Core:

   - Faster lookups of control elements with Xarray; normal user won't
     notice, but on the devices with tons of control elements, it can be
     visibly faster

   - Support for input validation for controls; this will harden for
     badly written drivers in general with a slight overhead

   - Deferred async signal handling for working around the potential
     deadlocks

   - Cleanup / refactoring raw MIDI locking code

  ASoC:

   - Restructing of the set_fmt() callbacks for making things clearer in
     situations like CODEC to CODEC links

   - Clean up and modernizing the DAI naming scheme setups

   - Merge of more of the Intel AVS driver stack, including some board
     integrations

   - New version 4 mechanism for communication with SOF DSPs

   - Suppoort for dynamically selecting the PLL to use at runtime on
     i.MX platforms

   - Improvements for CODEC to CODEC support in the generic cards

   - Support for AMD Jadeite and various machines, AMD RPL, Intel
     MetorLake DSPs, Mediatek MT8186 DSPs and MT6366, nVidia Tegra
     MDDRC, OPE and PEQ, NXP TFA9890, Qualcomm SDM845, WCD9335 and
     WAS883x, and Texas Instruments TAS2780

  HD- and USB-audio:

   - Continued improvement for CS35L41 (sub)codec support

   - More quirks for various devices (HP, Lenovo, Dell, Clevo)"

* tag 'sound-6.0-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (778 commits)
  ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
  ALSA: line6: Replace sprintf() with sysfs_emit()
  ALSA: hda: Replace sprintf() with sysfs_emit()
  ALSA: pcm: Replace sprintf() with sysfs_emit()
  ALSA: core: Replace scnprintf() with sysfs_emit()
  ALSA: control-led: Replace sprintf() with sysfs_emit()
  ALSA: aoa: Replace sprintf() with sysfs_emit()
  ALSA: ac97: Replace sprintf() with sysfs_emit()
  ALSA: hda/realtek: Add quirk for Clevo NV45PZ
  ALSA: hda/realtek: Add quirk for Lenovo Yoga9 14IAP7
  ALSA: control: Use deferred fasync helper
  ALSA: pcm: Use deferred fasync helper
  ALSA: timer: Use deferred fasync helper
  ALSA: core: Add async signal helpers
  ASoC: q6asm: use kcalloc() instead of kzalloc()
  ACPI: scan: Add CLSA0101 Laptop Support
  ALSA: hda: cs35l41: Support CLSA0101
  ALSA: hda: cs35l41: Use the CS35L41 HDA internal define
  ASoC: dt-bindings: use spi-peripheral-props.yaml
  ASoC: codecs: va-macro: use fsgen as clock
  ...

1 files changed, 9 insertions, 11 deletions

diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index b2e59581c17a..0847302121f6 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -1033,8 +1033,8 @@ static int aic31xx_clock_master_routes(struct snd_soc_component *component,
 	struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component);
 	int ret;
 
-	fmt &= SND_SOC_DAIFMT_MASTER_MASK;
-	if (fmt == SND_SOC_DAIFMT_CBS_CFS &&
+	fmt &= SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK;
+	if (fmt == SND_SOC_DAIFMT_CBC_CFC &&
 	    aic31xx->master_dapm_route_applied) {
 		/*
 		 * Remove the DAPM route(s) for codec clock master modes,
@@ -1051,7 +1051,7 @@ static int aic31xx_clock_master_routes(struct snd_soc_component *component,
 			return ret;
 
 		aic31xx->master_dapm_route_applied = false;
-	} else if (fmt != SND_SOC_DAIFMT_CBS_CFS &&
+	} else if (fmt != SND_SOC_DAIFMT_CBC_CFC &&
 		   !aic31xx->master_dapm_route_applied) {
 		/*
 		 * Add the needed DAPM route(s) for codec clock master modes,
@@ -1083,21 +1083,20 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
 
 	dev_dbg(component->dev, "## %s: fmt = 0x%x\n", __func__, fmt);
 
-	/* set master/slave audio interface */
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBM_CFM:
+	switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
+	case SND_SOC_DAIFMT_CBP_CFP:
 		iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER;
 		break;
-	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBC_CFP:
 		iface_reg1 |= AIC31XX_WCLK_MASTER;
 		break;
-	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBP_CFC:
 		iface_reg1 |= AIC31XX_BCLK_MASTER;
 		break;
-	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBC_CFC:
 		break;
 	default:
-		dev_err(component->dev, "Invalid DAI master/slave interface\n");
+		dev_err(component->dev, "Invalid DAI clock provider\n");
 		return -EINVAL;
 	}
 
@@ -1418,7 +1417,6 @@ static const struct snd_soc_component_driver soc_codec_driver_aic31xx = {
 	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
 	.endianness		= 1,
-	.non_legacy_dai_naming	= 1,
 };
 
 static const struct snd_soc_dai_ops aic31xx_dai_ops = {