Merge tag 'sound-5.13-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A bit more commits than expected at this time, but likely it's the
  last shot before the final.

  Many of changes are device-specific fix-ups for various ASoC drivers,
  while a few usual HD-audio quirks and a FireWire fix, as well as a
  couple of ALSA / ASoC core fixes.

  All look nice and small, and nothing to scare much"

* tag 'sound-5.13-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: seq: Fix race of snd_seq_timer_open()
  ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8
  ALSA: hda/realtek: headphone and mic don't work on an Acer laptop
  ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
  ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun()
  ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook 840 Aero G8
  ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP EliteBook x360 1040 G8
  ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite Dragonfly G2
  ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
  ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
  ASoC: meson: gx-card: fix sound-dai dt schema
  ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
  ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
  ASoC: SOF: reset enabled_cores state at suspend
  ASoC: fsl-asoc-card: Set .owner attribute when registering card.
  ASoC: topology: Fix spelling mistake "vesion" -> "version"
  ASoC: rt5659: Fix the lost powers for the HDA header
  ASoC: core: Fix Null-point-dereference in fmt_single_name()

9 files changed, 120 insertions, 16 deletions

diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index 87f5709fe2cc..4a50b169fe03 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0,
 		NULL, 0),
-	SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
-		NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
+		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL,
 		RT5659_PWR_MIC_DET_BIT, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+	SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
+		NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1,
 		RT5659_PWR_VREF3_BIT, 0, NULL, 0),
 
@@ -2464,8 +2469,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
 		RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0),
 
 	/* Input Side */
-	SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
-		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT,
 		0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT,
@@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component,
 
 static int rt5659_probe(struct snd_soc_component *component)
 {
+	struct snd_soc_dapm_context *dapm =
+		snd_soc_component_get_dapm(component);
 	struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
 
 	rt5659->component = component;
 
+	switch (rt5659->pdata.jd_src) {
+	case RT5659_JD_HDA_HEADER:
+		break;
+
+	default:
+		snd_soc_dapm_new_controls(dapm,
+			rt5659_particular_dapm_widgets,
+			ARRAY_SIZE(rt5659_particular_dapm_widgets));
+		break;
+	}
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c
index fed80c8f994f..e78ba3b064c4 100644
--- a/sound/soc/codecs/rt5682-sdw.c
+++ b/sound/soc/codecs/rt5682-sdw.c
@@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave)
 
 	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2,
 		RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL);
-	regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042);
+	regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142);
+	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600);
 	regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3,
 		RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN);
 	regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1,
diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h
index 81866aeb3fbf..55b2a1f52ca3 100644
--- a/sound/soc/codecs/tas2562.h
+++ b/sound/soc/codecs/tas2562.h
@@ -57,13 +57,13 @@
 #define TAS2562_TDM_CFG0_RAMPRATE_MASK		BIT(5)
 #define TAS2562_TDM_CFG0_RAMPRATE_44_1		BIT(5)
 #define TAS2562_TDM_CFG0_SAMPRATE_MASK		GENMASK(3, 1)
-#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ	0x0
-#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ	0x1
-#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ	0x2
-#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ	0x3
-#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ	0x4
-#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ	0x5
-#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ	0x6
+#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ	(0x0 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ	(0x1 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ	(0x2 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ	(0x3 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ	(0x4 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ	(0x5 << 1)
+#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ	(0x6 << 1)
 
 #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY	BIT(6)
 
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index c62bfd1c3ac7..4f55b316cf0f 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	/* Initialize sound card */
 	priv->pdev = pdev;
 	priv->card.dev = &pdev->dev;
+	priv->card.owner = THIS_MODULE;
 	ret = snd_soc_of_parse_card_name(&priv->card, "model");
 	if (ret) {
 		snprintf(priv->name, sizeof(priv->name), "%s-audio",
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index 28c7497344e3..a6e95db6b3fb 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
 	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
+	struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
+	unsigned int id = dai->driver->id;
 
 	clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+	/*
+	 * Ensure LRCLK is disabled even in device node validation.
+	 * Will not impact if disabled in lpass_cpu_daiops_trigger()
+	 * suspend.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE);
+	else
+		regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE);
+
+	/*
+	 * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before
+	 * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in
+	 * lpass_cpu_daiops_prepare.
+	 */
+	if (drvdata->mi2s_was_prepared[dai->driver->id]) {
+		drvdata->mi2s_was_prepared[dai->driver->id] = false;
+		clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]);
+	}
+
 	clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
 }
 
@@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		/*
+		 * Ensure lpass BCLK/LRCLK is enabled during
+		 * device resume as lpass_cpu_daiops_prepare() is not called
+		 * after the device resumes. We don't check mi2s_was_prepared before
+		 * enable/disable BCLK in trigger events because:
+		 *  1. These trigger events are paired, so the BCLK
+		 *     enable_count is balanced.
+		 *  2. the BCLK can be shared (ex: headset and headset mic),
+		 *     we need to increase the enable_count so that we don't
+		 *     turn off the shared BCLK while other devices are using
+		 *     it.
+		 */
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			ret = regmap_fields_write(i2sctl->spken, id,
 						 LPAIF_I2SCTL_SPKEN_ENABLE);
@@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		/*
+		 * To ensure lpass BCLK/LRCLK is disabled during
+		 * device suspend.
+		 */
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			ret = regmap_fields_write(i2sctl->spken, id,
 						 LPAIF_I2SCTL_SPKEN_DISABLE);
@@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
+	struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
+	unsigned int id = dai->driver->id;
+	int ret;
+
+	/*
+	 * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture
+	 * data flow starts. This allows other codec to have some delay before
+	 * the data flow.
+	 * (ex: to drop start up pop noise before capture starts).
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE);
+	else
+		ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE);
+
+	if (ret) {
+		dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret);
+		return ret;
+	}
+
+	/*
+	 * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can
+	 * be called multiple times. It's paired with the clk_disable in
+	 * lpass_cpu_daiops_shutdown.
+	 */
+	if (!drvdata->mi2s_was_prepared[dai->driver->id]) {
+		ret = clk_enable(drvdata->mi2s_bit_clk[id]);
+		if (ret) {
+			dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
+			return ret;
+		}
+		drvdata->mi2s_was_prepared[dai->driver->id] = true;
+	}
+	return 0;
+}
+
 const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = {
 	.set_sysclk	= lpass_cpu_daiops_set_sysclk,
 	.startup	= lpass_cpu_daiops_startup,
 	.shutdown	= lpass_cpu_daiops_shutdown,
 	.hw_params	= lpass_cpu_daiops_hw_params,
 	.trigger	= lpass_cpu_daiops_trigger,
+	.prepare	= lpass_cpu_daiops_prepare,
 };
 EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops);
 
diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h
index 83b2e08ade06..7f72214404ba 100644
--- a/sound/soc/qcom/lpass.h
+++ b/sound/soc/qcom/lpass.h
@@ -67,6 +67,10 @@ struct lpass_data {
 	/* MI2S SD lines to use for playback/capture */
 	unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS];
 	unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS];
+
+	/* The state of MI2S prepare dai_ops was called */
+	bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS];
+
 	int hdmi_port_enable;
 
 	/* low-power audio interface (LPAIF) registers */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1c0904acb935..a76974ccfce1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2225,6 +2225,8 @@ static char *fmt_single_name(struct device *dev, int *id)
 		return NULL;
 
 	name = devm_kstrdup(dev, devname, GFP_KERNEL);
+	if (!name)
+		return NULL;
 
 	/* are we a "%s.%d" name (platform and SPI components) */
 	found = strstr(name, dev->driver->name);
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 73076d425efb..4893a56208e0 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1901,7 +1901,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest,
  * @src: older version of pcm as a source
  * @pcm: latest version of pcm created from the source
  *
- * Support from vesion 4. User should free the returned pcm manually.
+ * Support from version 4. User should free the returned pcm manually.
  */
 static int pcm_new_ver(struct soc_tplg *tplg,
 		       struct snd_soc_tplg_pcm *src,
@@ -2089,7 +2089,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link,
  * @src: old version of phyical link config as a source
  * @link: latest version of physical link config created from the source
  *
- * Support from vesion 4. User need free the returned link config manually.
+ * Support from version 4. User need free the returned link config manually.
  */
 static int link_new_ver(struct soc_tplg *tplg,
 			struct snd_soc_tplg_link_config *src,
@@ -2400,7 +2400,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
  * @src: old version of manifest as a source
  * @manifest: latest version of manifest created from the source
  *
- * Support from vesion 4. Users need free the returned manifest manually.
+ * Support from version 4. Users need free the returned manifest manually.
  */
 static int manifest_new_ver(struct soc_tplg *tplg,
 			    struct snd_soc_tplg_manifest *src,
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index fd265803f7bc..c83fb6255961 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -256,6 +256,7 @@ suspend:
 
 	/* reset FW state */
 	sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+	sdev->enabled_cores_mask = 0;
 
 	return ret;
 }