ASoC: Fixes for v6.6

Quite a large collection of fixes, with numbers boosted by multiple
 vendors sending multi-patch serieses.  Nothing super major, and also one
 device quirk.
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Merge tag 'asoc-fix-v6.6-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.6

Quite a large collection of fixes, with numbers boosted by multiple
vendors sending multi-patch serieses.  Nothing super major, and also one
device quirk.

drivers/firmware/cirrus/cs_dsp.c

@@ -1863,15 +1863,15 @@ static int cs_dsp_adsp2_setup_algs(struct cs_dsp *dsp)
 		return PTR_ERR(adsp2_alg);
 
 	for (i = 0; i < n_algs; i++) {
-		cs_dsp_info(dsp,
+		cs_dsp_dbg(dsp,
-			    "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n",
+			   "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n",
-			    i, be32_to_cpu(adsp2_alg[i].alg.id),
+			   i, be32_to_cpu(adsp2_alg[i].alg.id),
-			    (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16,
+			   (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16,
-			    (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8,
+			   (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8,
-			    be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff,
+			   be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff,
-			    be32_to_cpu(adsp2_alg[i].xm),
+			   be32_to_cpu(adsp2_alg[i].xm),
-			    be32_to_cpu(adsp2_alg[i].ym),
+			   be32_to_cpu(adsp2_alg[i].ym),
-			    be32_to_cpu(adsp2_alg[i].zm));
+			   be32_to_cpu(adsp2_alg[i].zm));
 
 		alg_region = cs_dsp_create_region(dsp, WMFW_ADSP2_XM,
 						  adsp2_alg[i].alg.id,
@@ -1996,14 +1996,14 @@ static int cs_dsp_halo_setup_algs(struct cs_dsp *dsp)
 		return PTR_ERR(halo_alg);
 
 	for (i = 0; i < n_algs; i++) {
-		cs_dsp_info(dsp,
+		cs_dsp_dbg(dsp,
-			    "%d: ID %x v%d.%d.%d XM@%x YM@%x\n",
+			   "%d: ID %x v%d.%d.%d XM@%x YM@%x\n",
-			    i, be32_to_cpu(halo_alg[i].alg.id),
+			   i, be32_to_cpu(halo_alg[i].alg.id),
-			    (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16,
+			   (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16,
-			    (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8,
+			   (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8,
-			    be32_to_cpu(halo_alg[i].alg.ver) & 0xff,
+			   be32_to_cpu(halo_alg[i].alg.ver) & 0xff,
-			    be32_to_cpu(halo_alg[i].xm_base),
+			   be32_to_cpu(halo_alg[i].xm_base),
-			    be32_to_cpu(halo_alg[i].ym_base));
+			   be32_to_cpu(halo_alg[i].ym_base));
 
 		ret = cs_dsp_halo_create_regions(dsp, halo_alg[i].alg.id,
 						 halo_alg[i].alg.ver,

sound/soc/amd/yc/acp6x-mach.c

@@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "21J6"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82QF"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
@@ -220,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "82TL"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82UG"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
@@ -262,6 +276,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

sound/soc/codecs/aw88395/aw88395_lib.c

@@ -452,11 +452,13 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev,
 	if ((aw_bin->all_bin_parse_num != 1) ||
 		(aw_bin->header_info[0].bin_data_type != DATA_TYPE_REGISTER)) {
 		dev_err(aw_dev->dev, "bin num or type error");
+		ret = -EINVAL;
 		goto parse_bin_failed;
 	}
 
 	if (aw_bin->header_info[0].valid_data_len % 4) {
 		dev_err(aw_dev->dev, "bin data len get error!");
+		ret = -EINVAL;
 		goto parse_bin_failed;
 	}
 

sound/soc/codecs/cs35l56-i2c.c

@@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client)
 		return -ENOMEM;
 
 	cs35l56->base.dev = dev;
-	cs35l56->base.can_hibernate = true;
 
 	i2c_set_clientdata(client, cs35l56);
 	cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config);

sound/soc/codecs/cs35l56.c

@@ -1207,6 +1207,7 @@ void cs35l56_remove(struct cs35l56_private *cs35l56)
 	flush_workqueue(cs35l56->dsp_wq);
 	destroy_workqueue(cs35l56->dsp_wq);
 
+	pm_runtime_dont_use_autosuspend(cs35l56->base.dev);
 	pm_runtime_suspend(cs35l56->base.dev);
 	pm_runtime_disable(cs35l56->base.dev);
 

sound/soc/codecs/cs42l42-sdw.c

@@ -344,6 +344,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral,
 	switch (status) {
 	case SDW_SLAVE_ATTACHED:
 		dev_dbg(cs42l42->dev, "ATTACHED\n");
+
+		/*
+		 * The SoundWire core can report stale ATTACH notifications
+		 * if we hard-reset CS42L42 in probe() but it had already been
+		 * enumerated. Reject the ATTACH if we haven't yet seen an
+		 * UNATTACH report for the device being in reset.
+		 */
+		if (cs42l42->sdw_waiting_first_unattach)
+			break;
+
 		/*
 		 * Initialise codec, this only needs to be done once.
 		 * When resuming from suspend, resume callback will handle re-init of codec,
@@ -354,6 +364,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral,
 		break;
 	case SDW_SLAVE_UNATTACHED:
 		dev_dbg(cs42l42->dev, "UNATTACHED\n");
+
+		if (cs42l42->sdw_waiting_first_unattach) {
+			/*
+			 * SoundWire core has seen that CS42L42 is not on
+			 * the bus so release RESET and wait for ATTACH.
+			 */
+			cs42l42->sdw_waiting_first_unattach = false;
+			gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+		}
+
 		break;
 	default:
 		break;

sound/soc/codecs/cs42l42.c

@@ -2320,7 +2320,26 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42,
 
 	if (cs42l42->reset_gpio) {
 		dev_dbg(cs42l42->dev, "Found reset GPIO\n");
-		gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+
+		/*
+		 * ACPI can override the default GPIO state we requested
+		 * so ensure that we start with RESET low.
+		 */
+		gpiod_set_value_cansleep(cs42l42->reset_gpio, 0);
+
+		/* Ensure minimum reset pulse width */
+		usleep_range(10, 500);
+
+		/*
+		 * On SoundWire keep the chip in reset until we get an UNATTACH
+		 * notification from the SoundWire core. This acts as a
+		 * synchronization point to reject stale ATTACH notifications
+		 * if the chip was already enumerated before we reset it.
+		 */
+		if (cs42l42->sdw_peripheral)
+			cs42l42->sdw_waiting_first_unattach = true;
+		else
+			gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
 	}
 	usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
 

sound/soc/codecs/cs42l42.h

@@ -53,6 +53,7 @@ struct  cs42l42_private {
 	u8 stream_use;
 	bool hp_adc_up_pending;
 	bool suspended;
+	bool sdw_waiting_first_unattach;
 	bool init_done;
 };
 

sound/soc/codecs/cs42l43.c

@@ -2077,7 +2077,8 @@ static const struct cs42l43_irq cs42l43_irqs[] = {
 
 static int cs42l43_request_irq(struct cs42l43_codec *priv,
 			       struct irq_domain *dom, const char * const name,
-			       unsigned int irq, irq_handler_t handler)
+			       unsigned int irq, irq_handler_t handler,
+			       unsigned long flags)
 {
 	int ret;
 
@@ -2087,8 +2088,8 @@ static int cs42l43_request_irq(struct cs42l43_codec *priv,
 
 	dev_dbg(priv->dev, "Request IRQ %d for %s\n", ret, name);
 
-	ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, IRQF_ONESHOT,
+	ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler,
-					name, priv);
+					IRQF_ONESHOT | flags, name, priv);
 	if (ret)
 		return dev_err_probe(priv->dev, ret, "Failed to request IRQ %s\n", name);
 
@@ -2124,11 +2125,11 @@ static int cs42l43_shutter_irq(struct cs42l43_codec *priv,
 		return 0;
 	}
 
-	ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler);
+	ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler, IRQF_SHARED);
 	if (ret)
 		return ret;
 
-	return cs42l43_request_irq(priv, dom, open_name, open_irq, handler);
+	return cs42l43_request_irq(priv, dom, open_name, open_irq, handler, IRQF_SHARED);
 }
 
 static int cs42l43_codec_probe(struct platform_device *pdev)
@@ -2178,7 +2179,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev)
 
 	for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) {
 		ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name,
-					  cs42l43_irqs[i].irq, cs42l43_irqs[i].handler);
+					  cs42l43_irqs[i].irq,
+					  cs42l43_irqs[i].handler, 0);
 		if (ret)
 			goto err_pm;
 	}

sound/soc/codecs/rt5640.c

@@ -2403,13 +2403,11 @@ static irqreturn_t rt5640_irq(int irq, void *data)
 	struct rt5640_priv *rt5640 = data;
 	int delay = 0;
 
-	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
+	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
-		cancel_delayed_work_sync(&rt5640->jack_work);
 		delay = 100;
-	}
 
 	if (rt5640->jack)
-		queue_delayed_work(system_long_wq, &rt5640->jack_work, delay);
+		mod_delayed_work(system_long_wq, &rt5640->jack_work, delay);
 
 	return IRQ_HANDLED;
 }
@@ -2565,10 +2563,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component,
 	if (jack_data && jack_data->use_platform_clock)
 		rt5640->use_platform_clock = jack_data->use_platform_clock;
 
-	ret = devm_request_threaded_irq(component->dev, rt5640->irq,
+	ret = request_irq(rt5640->irq, rt5640_irq,
-					NULL, rt5640_irq,
+			  IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
-					IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
+			  "rt5640", rt5640);
-					"rt5640", rt5640);
 	if (ret) {
 		dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret);
 		rt5640_disable_jack_detect(component);
@@ -2621,14 +2618,14 @@ static void rt5640_enable_hda_jack_detect(
 
 	rt5640->jack = jack;
 
-	ret = devm_request_threaded_irq(component->dev, rt5640->irq,
+	ret = request_irq(rt5640->irq, rt5640_irq,
-					NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT,
+			  IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640);
-					"rt5640", rt5640);
 	if (ret) {
 		dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret);
-		rt5640->irq = -ENXIO;
+		rt5640->jack = NULL;
 		return;
 	}
+	rt5640->irq_requested = true;
 
 	/* sync initial jack state */
 	queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
@@ -2801,12 +2798,12 @@ static int rt5640_suspend(struct snd_soc_component *component)
 {
 	struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component);
 
-	if (rt5640->irq) {
+	if (rt5640->jack) {
 		/* disable jack interrupts during system suspend */
 		disable_irq(rt5640->irq);
+		rt5640_cancel_work(rt5640);
 	}
 
-	rt5640_cancel_work(rt5640);
 	snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
 	rt5640_reset(component);
 	regcache_cache_only(rt5640->regmap, true);
@@ -2829,9 +2826,6 @@ static int rt5640_resume(struct snd_soc_component *component)
 	regcache_cache_only(rt5640->regmap, false);
 	regcache_sync(rt5640->regmap);
 
-	if (rt5640->irq)
-		enable_irq(rt5640->irq);
-
 	if (rt5640->jack) {
 		if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) {
 			snd_soc_component_update_bits(component,
@@ -2859,6 +2853,7 @@ static int rt5640_resume(struct snd_soc_component *component)
 			}
 		}
 
+		enable_irq(rt5640->irq);
 		queue_delayed_work(system_long_wq, &rt5640->jack_work, 0);
 	}
 

sound/soc/codecs/wm8960.c

@@ -1468,8 +1468,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 	}
 
 	wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap);
-	if (IS_ERR(wm8960->regmap))
+	if (IS_ERR(wm8960->regmap)) {
-		return PTR_ERR(wm8960->regmap);
+		ret = PTR_ERR(wm8960->regmap);
+		goto bulk_disable;
+	}
 
 	if (pdata)
 		memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data));
@@ -1479,13 +1481,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 	ret = i2c_master_recv(i2c, &val, sizeof(val));
 	if (ret >= 0) {
 		dev_err(&i2c->dev, "Not wm8960, wm8960 reg can not read by i2c\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto bulk_disable;
 	}
 
 	ret = wm8960_reset(wm8960->regmap);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to issue reset\n");
-		return ret;
+		goto bulk_disable;
 	}
 
 	if (wm8960->pdata.shared_lrclk) {
@@ -1494,7 +1497,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 		if (ret != 0) {
 			dev_err(&i2c->dev, "Failed to enable LRCM: %d\n",
 				ret);
-			return ret;
+			goto bulk_disable;
 		}
 	}
 
@@ -1528,7 +1531,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c)
 
 	ret = devm_snd_soc_register_component(&i2c->dev,
 			&soc_component_dev_wm8960, &wm8960_dai, 1);
+	if (ret)
+		goto bulk_disable;
 
+	return 0;
+
+bulk_disable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8960->supplies), wm8960->supplies);
 	return ret;
 }
 

sound/soc/codecs/wm_adsp.c

@@ -687,7 +687,10 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type,
 	struct wm_coeff_ctl *ctl;
 	int ret;
 
+	mutex_lock(&dsp->cs_dsp.pwr_lock);
 	ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len);
+	mutex_unlock(&dsp->cs_dsp.pwr_lock);
+
 	if (ret < 0)
 		return ret;
 
@@ -703,8 +706,14 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl);
 int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type,
 		     unsigned int alg, void *buf, size_t len)
 {
-	return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg),
+	int ret;
-				      0, buf, len);
+
+	mutex_lock(&dsp->cs_dsp.pwr_lock);
+	ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg),
+				     0, buf, len);
+	mutex_unlock(&dsp->cs_dsp.pwr_lock);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(wm_adsp_read_ctl);
 

sound/soc/fsl/imx-audmix.c

@@ -315,7 +315,7 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	if (IS_ERR(priv->cpu_mclk)) {
 		ret = PTR_ERR(priv->cpu_mclk);
 		dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret);
-		return -EINVAL;
+		return ret;
 	}
 
 	priv->audmix_pdev = audmix_pdev;

sound/soc/fsl/imx-pcm-rpmsg.c

@@ -19,6 +19,7 @@
 static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = {
 	.info = SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_BATCH |
 		SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_MMAP_VALID |
 		SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |

sound/soc/fsl/imx-rpmsg.c

@@ -89,6 +89,14 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
 			    SND_SOC_DAIFMT_NB_NF |
 			    SND_SOC_DAIFMT_CBC_CFC;
 
+	/*
+	 * i.MX rpmsg sound cards work on codec slave mode. MCLK will be
+	 * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK
+	 * present at power up/down sequence. So need to set ignore_pmdown_time
+	 * to power down codec immediately before MCLK is turned off.
+	 */
+	data->dai.ignore_pmdown_time = 1;
+
 	/* Optional codec node */
 	ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args);
 	if (ret) {

sound/soc/intel/avs/boards/hdaudio.c

@@ -55,6 +55,9 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int
 			return -ENOMEM;
 
 		dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL);
+		if (!dl[i].codecs->name)
+			return -ENOMEM;
+
 		dl[i].codecs->dai_name = pcm->name;
 		dl[i].num_codecs = 1;
 		dl[i].num_cpus = 1;

sound/soc/meson/axg-spdifin.c

@@ -112,34 +112,6 @@ static int axg_spdifin_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int axg_spdifin_startup(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
-{
-	struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai);
-	int ret;
-
-	ret = clk_prepare_enable(priv->refclk);
-	if (ret) {
-		dev_err(dai->dev,
-			"failed to enable spdifin reference clock\n");
-		return ret;
-	}
-
-	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN,
-			   SPDIFIN_CTRL0_EN);
-
-	return 0;
-}
-
-static void axg_spdifin_shutdown(struct snd_pcm_substream *substream,
-				 struct snd_soc_dai *dai)
-{
-	struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai);
-
-	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0);
-	clk_disable_unprepare(priv->refclk);
-}
-
 static void axg_spdifin_write_mode_param(struct regmap *map, int mode,
 					 unsigned int val,
 					 unsigned int num_per_reg,
@@ -251,17 +223,32 @@ static int axg_spdifin_dai_probe(struct snd_soc_dai *dai)
 	ret = axg_spdifin_sample_mode_config(dai, priv);
 	if (ret) {
 		dev_err(dai->dev, "mode configuration failed\n");
-		clk_disable_unprepare(priv->pclk);
+		goto pclk_err;
-		return ret;
 	}
 
+	ret = clk_prepare_enable(priv->refclk);
+	if (ret) {
+		dev_err(dai->dev,
+			"failed to enable spdifin reference clock\n");
+		goto pclk_err;
+	}
+
+	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN,
+			   SPDIFIN_CTRL0_EN);
+
 	return 0;
+
+pclk_err:
+	clk_disable_unprepare(priv->pclk);
+	return ret;
 }
 
 static int axg_spdifin_dai_remove(struct snd_soc_dai *dai)
 {
 	struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai);
 
+	regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0);
+	clk_disable_unprepare(priv->refclk);
 	clk_disable_unprepare(priv->pclk);
 	return 0;
 }
@@ -270,8 +257,6 @@ static const struct snd_soc_dai_ops axg_spdifin_ops = {
 	.probe		= axg_spdifin_dai_probe,
 	.remove		= axg_spdifin_dai_remove,
 	.prepare	= axg_spdifin_prepare,
-	.startup	= axg_spdifin_startup,
-	.shutdown	= axg_spdifin_shutdown,
 };
 
 static int axg_spdifin_iec958_info(struct snd_kcontrol *kcontrol,

sound/soc/sh/rcar/core.c

@@ -1303,6 +1303,7 @@ audio_graph:
 		if (i >= RSND_MAX_COMPONENT) {
 			dev_info(dev, "reach to max component\n");
 			of_node_put(node);
+			of_node_put(ports);
 			break;
 		}
 	}

sound/soc/soc-pcm.c

@@ -985,6 +985,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_soc_dai *cpu_dai;
 	struct snd_soc_dai *codec_dai;
+	struct snd_pcm_hw_params tmp_params;
 	int i, ret = 0;
 
 	snd_soc_dpcm_mutex_assert_held(rtd);
@@ -998,7 +999,6 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 		goto out;
 
 	for_each_rtd_codec_dais(rtd, i, codec_dai) {
-		struct snd_pcm_hw_params codec_params;
 		unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream);
 
 		/*
@@ -1019,23 +1019,22 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 			continue;
 
 		/* copy params for each codec */
-		codec_params = *params;
+		tmp_params = *params;
 
 		/* fixup params based on TDM slot masks */
 		if (tdm_mask)
-			soc_pcm_codec_params_fixup(&codec_params, tdm_mask);
+			soc_pcm_codec_params_fixup(&tmp_params, tdm_mask);
 
 		ret = snd_soc_dai_hw_params(codec_dai, substream,
-					    &codec_params);
+					    &tmp_params);
 		if(ret < 0)
 			goto out;
 
-		soc_pcm_set_dai_params(codec_dai, &codec_params);
+		soc_pcm_set_dai_params(codec_dai, &tmp_params);
-		snd_soc_dapm_update_dai(substream, &codec_params, codec_dai);
+		snd_soc_dapm_update_dai(substream, &tmp_params, codec_dai);
 	}
 
 	for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
-		struct snd_pcm_hw_params cpu_params;
 		unsigned int ch_mask = 0;
 		int j;
 
@@ -1047,7 +1046,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 			continue;
 
 		/* copy params for each cpu */
-		cpu_params = *params;
+		tmp_params = *params;
 
 		if (!rtd->dai_link->codec_ch_maps)
 			goto hw_params;
@@ -1062,16 +1061,16 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd,
 
 		/* fixup cpu channel number */
 		if (ch_mask)
-			soc_pcm_codec_params_fixup(&cpu_params, ch_mask);
+			soc_pcm_codec_params_fixup(&tmp_params, ch_mask);
 
 hw_params:
-		ret = snd_soc_dai_hw_params(cpu_dai, substream, &cpu_params);
+		ret = snd_soc_dai_hw_params(cpu_dai, substream, &tmp_params);
 		if (ret < 0)
 			goto out;
 
 		/* store the parameters for each DAI */
-		soc_pcm_set_dai_params(cpu_dai, &cpu_params);
+		soc_pcm_set_dai_params(cpu_dai, &tmp_params);
-		snd_soc_dapm_update_dai(substream, &cpu_params, cpu_dai);
+		snd_soc_dapm_update_dai(substream, &tmp_params, cpu_dai);
 	}
 
 	ret = snd_soc_pcm_component_hw_params(substream, params);

sound/soc/soc-utils.c

@@ -217,6 +217,7 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)
 		return 1;
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy);
 
 int snd_soc_component_is_dummy(struct snd_soc_component *component)
 {

sound/soc/sof/core.c

@@ -486,10 +486,9 @@ int snd_sof_device_remove(struct device *dev)
 		snd_sof_ipc_free(sdev);
 		snd_sof_free_debug(sdev);
 		snd_sof_remove(sdev);
+		sof_ops_free(sdev);
 	}
 
-	sof_ops_free(sdev);
-
 	/* release firmware */
 	snd_sof_fw_unload(sdev);
 

sound/soc/sof/intel/mtl.c

@@ -460,7 +460,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot)
 	/* step 3: wait for IPC DONE bit from ROM */
 	ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->ipc_ack, status,
 					    ((status & chip->ipc_ack_mask) == chip->ipc_ack_mask),
-					    HDA_DSP_REG_POLL_INTERVAL_US, MTL_DSP_PURGE_TIMEOUT_US);
+					    HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_INIT_TIMEOUT_US);
 	if (ret < 0) {
 		if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS)
 			dev_err(sdev->dev, "timeout waiting for purge IPC done\n");

sound/soc/sof/intel/mtl.h

@@ -62,7 +62,6 @@
 #define MTL_DSP_IRQSTS_IPC		BIT(0)
 #define MTL_DSP_IRQSTS_SDW		BIT(6)
 
-#define MTL_DSP_PURGE_TIMEOUT_US	20000000 /* 20s */
 #define MTL_DSP_REG_POLL_INTERVAL_US	10	/* 10 us */
 
 /* Memory windows */

sound/soc/sof/ipc4-topology.c

@@ -231,7 +231,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp,
 
 	ret = sof_update_ipc_object(scomp, available_fmt,
 				    SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples,
-				    swidget->num_tuples, sizeof(available_fmt), 1);
+				    swidget->num_tuples, sizeof(*available_fmt), 1);
 	if (ret) {
 		dev_err(scomp->dev, "Failed to parse audio format token count\n");
 		return ret;

sound/soc/sof/sof-audio.c

@@ -212,7 +212,8 @@ widget_free:
 	sof_widget_free_unlocked(sdev, swidget);
 	use_count_decremented = true;
 core_put:
-	snd_sof_dsp_core_put(sdev, swidget->core);
+	if (!use_count_decremented)
+		snd_sof_dsp_core_put(sdev, swidget->core);
 pipe_widget_free:
 	if (swidget->id != snd_soc_dapm_scheduler)
 		sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget);

sound/soc/tegra/tegra_audio_graph_card.c

@@ -10,6 +10,7 @@
 #include <linux/platform_device.h>
 #include <sound/graph_card.h>
 #include <sound/pcm_params.h>
+#include <sound/soc-dai.h>
 
 #define MAX_PLLA_OUT0_DIV 128
 
@@ -44,6 +45,21 @@ struct tegra_audio_cdata {
 	unsigned int plla_out0_rates[NUM_RATE_TYPE];
 };
 
+static bool need_clk_update(struct snd_soc_dai *dai)
+{
+	if (snd_soc_dai_is_dummy(dai) ||
+	    !dai->driver->ops ||
+	    !dai->driver->name)
+		return false;
+
+	if (strstr(dai->driver->name, "I2S") ||
+	    strstr(dai->driver->name, "DMIC") ||
+	    strstr(dai->driver->name, "DSPK"))
+		return true;
+
+	return false;
+}
+
 /* Setup PLL clock as per the given sample rate */
 static int tegra_audio_graph_update_pll(struct snd_pcm_substream *substream,
 					struct snd_pcm_hw_params *params)
@@ -140,19 +156,7 @@ static int tegra_audio_graph_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
 	int err;
 
-	/*
+	if (need_clk_update(cpu_dai)) {
-	 * This gets called for each DAI link (FE or BE) when DPCM is used.
-	 * We may not want to update PLLA rate for each call. So PLLA update
-	 * must be restricted to external I/O links (I2S, DMIC or DSPK) since
-	 * they actually depend on it. I/O modules update their clocks in
-	 * hw_param() of their respective component driver and PLLA rate
-	 * update here helps them to derive appropriate rates.
-	 *
-	 * TODO: When more HW accelerators get added (like sample rate
-	 * converter, volume gain controller etc., which don't really
-	 * depend on PLLA) we need a better way to filter here.
-	 */
-	if (cpu_dai->driver->ops && rtd->dai_link->no_pcm) {
 		err = tegra_audio_graph_update_pll(substream, params);
 		if (err)
 			return err;