Merge tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This looks like a relatively calm development cycle; there have been
  only few changes in ALSA and ASoC core sides while we get lots of
  device-specific fixes and updates as usual. Most of commits are about
  ASoC, including Intel SOF/AVS and many device tree updates.

  Below are some highlights:

  Core:
   - Improvement in memalloc helper for fallback allocations
   - More cleanups of ASoC DAPM code

  ASoC:
   - Factoring out of mapping hw_params onto SoundWire configuration
   - The ever ongoing overhauls of the Intel DSP code continue,
     including support for loading libraries and probes with IPC4 on
     SOF.
   - Support for more sample formats on JZ4740
   - Lots of device tree conversions and fixups
   - Support for Allwinner D1, a range of AMD and Intel systems,
     Mediatek systems with multiple DMICs, Nuvoton NAU8318, NXP
     fsl_rpmsg and i.MX93, Qualcomm AudioReach Enable, MFC and SAL,
     RealTek RT1318 and Rockchip RK3588

  ALSA:
   - Addition of PCM kselftest; still minimalistic but can be extended
     in future
   - Fixes for corner-case XRUNs with USB-audio implicit feedback mode
   - Usual device-specific quirk updates for USB- and HD-audio
   - FireWire DICE updates

  This also contains a few cross-tree updates:
   - Some OMAP board file updates for removal of relevant OMAP platforms
   - A new I2C API update for I2C probe API adaption
   - A DRM update for the further hdmi-codec updates"

* tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (417 commits)
  ALSA: mts64: fix possible null-ptr-defer in snd_mts64_interrupt
  ALSA: patch_realtek: Fix Dell Inspiron Plus 16
  ALSA: hda/cirrus: Add extra 10 ms delay to allow PLL settle and lock.
  ASoC: dt-bindings: Correct Alexandre Belloni email
  ASoC: dt-bindings: maxim,max98504: Convert to DT schema
  ASoC: dt-bindings: maxim,max98357a: Convert to DT schema
  ASoC: dt-bindings: Reference common DAI properties
  ASoC: dt-bindings: Extend name-prefix.yaml into common DAI properties
  ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
  ASoC: uniphier: aio-core: Make some read-only arrays static const
  ASoC: wcd938x: Make read-only array minCode_param static const
  ASoC: qcom: lpass-sc7280: Add maybe_unused tag for system PM ops
  ASoC : SOF: amd: Add support for IPC and DSP dumps
  ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER
  ALSA: usb-audio: Workaround for XRUN at prepare
  ALSA: pcm: Handle XRUN at trigger START
  ALSA: pcm: Set missing stop_operating flag at undoing trigger start
  drm: tda99x: Don't advertise non-existent capture support
  ASoC: hdmi-codec: Allow playback and capture to be disabled
  kselftest/alsa: Add more coverage of sample rates and channel counts
  ...

1 files changed, 13 insertions, 18 deletions

diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 01f9b9f0c12b..473e9fe5d0bf 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -26,6 +26,7 @@
 #define AVS_RT5682_SSP_CODEC_MASK	(GENMASK(2, 0))
 #define AVS_RT5682_MCLK_EN		BIT(3)
 #define AVS_RT5682_MCLK_24MHZ		BIT(4)
+#define AVS_RT5682_CODEC_DAI_NAME	"rt5682-aif1"
 
 /* Default: MCLK on, MCLK 19.2M, SSP0 */
 static unsigned long avs_rt5682_quirk = AVS_RT5682_MCLK_EN | AVS_RT5682_SSP_CODEC(0);
@@ -119,6 +120,11 @@ static int avs_rt5682_codec_init(struct snd_soc_pcm_runtime *runtime)
 	return 0;
 };
 
+static void avs_rt5682_codec_exit(struct snd_soc_pcm_runtime *rtd)
+{
+	snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL);
+}
+
 static int
 avs_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params)
 {
@@ -184,7 +190,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 
 	dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port);
 	dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10EC5682:00");
-	dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt5682-aif1");
+	dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, AVS_RT5682_CODEC_DAI_NAME);
 	if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name)
 		return -ENOMEM;
 
@@ -194,6 +200,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->init = avs_rt5682_codec_init;
+	dl->exit = avs_rt5682_codec_exit;
 	dl->ops = &avs_rt5682_ops;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
@@ -237,30 +244,19 @@ static int avs_create_dapm_routes(struct device *dev, int ssp_port,
 	return 0;
 }
 
-static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack)
-{
-	struct snd_soc_component *component;
-
-	for_each_card_components(card, component)
-		snd_soc_component_set_jack(component, jack, NULL);
-	return 0;
-}
-
-static int avs_card_remove(struct snd_soc_card *card)
-{
-	return avs_card_set_jack(card, NULL);
-}
-
 static int avs_card_suspend_pre(struct snd_soc_card *card)
 {
-	return avs_card_set_jack(card, NULL);
+	struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, AVS_RT5682_CODEC_DAI_NAME);
+
+	return snd_soc_component_set_jack(codec_dai->component, NULL, NULL);
 }
 
 static int avs_card_resume_post(struct snd_soc_card *card)
 {
+	struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, AVS_RT5682_CODEC_DAI_NAME);
 	struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card);
 
-	return avs_card_set_jack(card, jack);
+	return snd_soc_component_set_jack(codec_dai->component, jack, NULL);
 }
 
 static int avs_rt5682_probe(struct platform_device *pdev)
@@ -304,7 +300,6 @@ static int avs_rt5682_probe(struct platform_device *pdev)
 	card->name = "avs_rt5682";
 	card->dev = dev;
 	card->owner = THIS_MODULE;
-	card->remove = avs_card_remove;
 	card->suspend_pre = avs_card_suspend_pre;
 	card->resume_post = avs_card_resume_post;
 	card->dai_link = dai_link;