ASoC: Merge up fixes

For the benefit of CI.

Documentation/devicetree/bindings/sound/rockchip-spdif.yaml

@@ -26,6 +26,7 @@ properties:
       - const: rockchip,rk3568-spdif
       - items:
           - enum:
+              - rockchip,rk3128-spdif
               - rockchip,rk3188-spdif
               - rockchip,rk3288-spdif
               - rockchip,rk3308-spdif

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,
-			    "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n",
-			    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) & 0xff00) >> 8,
-			    be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff,
-			    be32_to_cpu(adsp2_alg[i].xm),
-			    be32_to_cpu(adsp2_alg[i].ym),
-			    be32_to_cpu(adsp2_alg[i].zm));
+		cs_dsp_dbg(dsp,
+			   "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n",
+			   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) & 0xff00) >> 8,
+			   be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff,
+			   be32_to_cpu(adsp2_alg[i].xm),
+			   be32_to_cpu(adsp2_alg[i].ym),
+			   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,
-			    "%d: ID %x v%d.%d.%d XM@%x YM@%x\n",
-			    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) & 0xff00) >> 8,
-			    be32_to_cpu(halo_alg[i].alg.ver) & 0xff,
-			    be32_to_cpu(halo_alg[i].xm_base),
-			    be32_to_cpu(halo_alg[i].ym_base));
+		cs_dsp_dbg(dsp,
+			   "%d: ID %x v%d.%d.%d XM@%x YM@%x\n",
+			   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) & 0xff00) >> 8,
+			   be32_to_cpu(halo_alg[i].alg.ver) & 0xff,
+			   be32_to_cpu(halo_alg[i].xm_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/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/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,
-					name, priv);
+	ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler,
+					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) {
-		cancel_delayed_work_sync(&rt5640->jack_work);
+	if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER)
 		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,
-					NULL, rt5640_irq,
-					IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
-					"rt5640", rt5640);
+	ret = request_irq(rt5640->irq, rt5640_irq,
+			  IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
+			  "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,
-					NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT,
-					"rt5640", rt5640);
+	ret = request_irq(rt5640->irq, rt5640_irq,
+			  IRQF_TRIGGER_RISING | IRQF_ONESHOT, "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/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),
-				      0, buf, len);
+	int ret;
+
+	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/fsl-asoc-card.c

@@ -52,8 +52,8 @@ struct codec_priv {
 	unsigned long mclk_freq;
 	unsigned long free_freq;
 	u32 mclk_id;
-	u32 fll_id;
-	u32 pll_id;
+	int fll_id;
+	int pll_id;
 };
 
 /**
@@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Specific configuration for PLL */
-	if (codec_priv->pll_id && codec_priv->fll_id) {
+	if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
 		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
 			pll_out = priv->sample_rate * 384;
 		else
@@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
 
 	priv->streams &= ~BIT(substream->stream);
 
-	if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
+	if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
 		/* Force freq to be free_freq to avoid error message in codec */
 		ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0),
 					     codec_priv->mclk_id,
@@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	priv->card.dapm_routes = audio_map;
 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	priv->card.driver_name = DRIVER_NAME;
+
+	priv->codec_priv.fll_id = -1;
+	priv->codec_priv.pll_id = -1;
+
 	/* Diversify the card configurations */
 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
 		codec_dai_name = "cs42888";

sound/soc/fsl/fsl_sai.c

@@ -710,10 +710,15 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir)
 {
 	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = dir == TX;
-	u32 xcsr, count = 100;
+	u32 xcsr, count = 100, mask;
+
+	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output)
+		mask = FSL_SAI_CSR_TERE;
+	else
+		mask = FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE;
 
 	regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
-			   FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0);
+			   mask, 0);
 
 	/* TERE will remain set till the end of current frame */
 	do {

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-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/generic/simple-card-utils.c

@@ -310,7 +310,8 @@ int simple_util_startup(struct snd_pcm_substream *substream)
 		if (fixed_sysclk % props->mclk_fs) {
 			dev_err(rtd->dev, "fixed sysclk %u not divisible by mclk_fs %u\n",
 				fixed_sysclk, props->mclk_fs);
-			return -EINVAL;
+			ret = -EINVAL;
+			goto codec_err;
 		}
 		ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE,
 			fixed_rate, fixed_rate);

sound/soc/generic/simple-card.c

@@ -758,10 +758,12 @@ static int simple_probe(struct platform_device *pdev)
 		struct snd_soc_dai_link *dai_link = priv->dai_link;
 		struct simple_dai_props *dai_props = priv->dai_props;
 
+		ret = -EINVAL;
+
 		cinfo = dev->platform_data;
 		if (!cinfo) {
 			dev_err(dev, "no info for asoc-simple-card\n");
-			return -EINVAL;
+			goto err;
 		}
 
 		if (!cinfo->name ||
@@ -770,7 +772,7 @@ static int simple_probe(struct platform_device *pdev)
 		    !cinfo->platform ||
 		    !cinfo->cpu_dai.name) {
 			dev_err(dev, "insufficient simple_util_info settings\n");
-			return -EINVAL;
+			goto err;
 		}
 
 		cpus			= dai_link->cpus;

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/intel/boards/sof_es8336.c

@@ -808,6 +808,16 @@ static const struct platform_device_id board_ids[] = {
 					SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK |
 					SOF_ES8336_JD_INVERTED),
 	},
+	{
+		.name = "mtl_es83x6_c1_h02",
+		.driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) |
+					SOF_NO_OF_HDMI_CAPTURE_SSP(2) |
+					SOF_HDMI_CAPTURE_1_SSP(0) |
+					SOF_HDMI_CAPTURE_2_SSP(2) |
+					SOF_SSP_HDMI_CAPTURE_PRESENT |
+					SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK |
+					SOF_ES8336_JD_INVERTED),
+	},
 	{ }
 };
 MODULE_DEVICE_TABLE(platform, board_ids);

sound/soc/intel/boards/sof_sdw.c

@@ -369,6 +369,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 		/* No Jack */
 		.driver_data = (void *)SOF_SDW_TGL_HDMI,
 	},
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"),
+		},
+		/* No Jack */
+		.driver_data = (void *)SOF_SDW_TGL_HDMI,
+	},
+
 	{
 		.callback = sof_sdw_quirk_cb,
 		.matches = {

sound/soc/intel/common/soc-acpi-intel-adl-match.c

@@ -674,18 +674,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = {
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l3.tplg",
 	},
-	{
-		.link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */
-		.links = adl_sdw_rt1316_link1_rt714_link0,
-		.drv_name = "sof_sdw",
-		.sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg",
-	},
 	{
 		.link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */
 		.links = adl_sdw_rt1316_link12_rt714_link0,
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-adl-rt1316-l12-rt714-l0.tplg",
 	},
+	{
+		.link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */
+		.links = adl_sdw_rt1316_link1_rt714_link0,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg",
+	},
 	{
 		.link_mask = 0x5, /* 2 active links required */
 		.links = adl_sdw_rt1316_link2_rt714_link0,

sound/soc/intel/common/soc-acpi-intel-mtl-match.c

@@ -30,6 +30,16 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = {
 	.codecs = {"10EC5682", "RTL5682"},
 };
 
+static const struct snd_soc_acpi_codecs mtl_essx_83x6 = {
+	.num_codecs = 3,
+	.codecs = { "ESSX8316", "ESSX8326", "ESSX8336"},
+};
+
+static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = {
+	.num_codecs = 1,
+	.codecs = {"INTC10B0"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
 	{
 		.comp_ids = &mtl_rt5682_rt5682s_hp,
@@ -52,6 +62,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = {
 		.quirk_data = &mtl_rt1019p_amp,
 		.sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg",
 	},
+	{
+		.comp_ids = &mtl_essx_83x6,
+		.drv_name = "mtl_es83x6_c1_h02",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &mtl_lt6911_hdmi,
+		.sof_tplg_filename = "sof-mtl-es83x6-ssp1-hdmi-ssp02.tplg",
+	},
+	{
+		.comp_ids = &mtl_essx_83x6,
+		.drv_name = "sof-essx8336",
+		.sof_tplg_filename = "sof-mtl-es8336", /* the tplg suffix is added at run time */
+		.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER |
+					SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
+					SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines);

sound/soc/soc-generic-dmaengine-pcm.c

@@ -44,8 +44,8 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm,
  * platforms which make use of the snd_dmaengine_dai_dma_data struct for their
  * DAI DMA data. Internally the function will first call
  * snd_hwparams_to_dma_slave_config to fill in the slave config based on the
- * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the
- * remaining fields based on the DAI DMA data.
+ * hw_params, followed by snd_dmaengine_pcm_set_config_from_dai_data to fill in
+ * the remaining fields based on the DAI DMA data.
  */
 int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config)

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

@@ -463,7 +463,7 @@ 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 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev,
 	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);