ASoC: Merge fixes due to dependencies

So we can apply the tlv320aic3xxx DT conversion.

Documentation/devicetree/bindings/sound/tlv320aic32x4.txt

@@ -8,7 +8,7 @@ Required properties:
 	"ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256
 	"ti,tas2505" TAS2505, TAS2521
  - reg: I2C slave address
- - supply-*: Required supply regulators are:
+ - *-supply: Required supply regulators are:
     "iov" - digital IO power supply
     "ldoin" - LDO power supply
     "dv" - Digital core power supply

drivers/firmware/cirrus/cs_dsp.c

@@ -2124,6 +2124,7 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 				   file, blocks, le32_to_cpu(blk->len),
 				   type, le32_to_cpu(blk->id));
 
+			region_name = cs_dsp_mem_region_name(type);
 			mem = cs_dsp_find_region(dsp, type);
 			if (!mem) {
 				cs_dsp_err(dsp, "No base for region %x\n", type);
@@ -2147,8 +2148,8 @@ static int cs_dsp_load_coeff(struct cs_dsp *dsp, const struct firmware *firmware
 				reg = dsp->ops->region_to_reg(mem, reg);
 				reg += offset;
 			} else {
-				cs_dsp_err(dsp, "No %x for algorithm %x\n",
+				cs_dsp_err(dsp, "No %s for algorithm %x\n",
-					   type, le32_to_cpu(blk->id));
+					   region_name, le32_to_cpu(blk->id));
 			}
 			break;
 

include/sound/soc-acpi.h

@@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr {
 /* Descriptor for SST ASoC machine driver */
 struct snd_soc_acpi_mach {
 	u8 id[ACPI_ID_LEN];
+	const char *uid;
 	const struct snd_soc_acpi_codecs *comp_ids;
 	const u32 link_mask;
 	const struct snd_soc_acpi_link_adr *links;

include/sound/soc-dpcm.h

@@ -122,6 +122,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
 int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 		struct snd_soc_pcm_runtime *be, int stream);
 
+/* can this BE perform prepare */
+int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe,
+				 struct snd_soc_pcm_runtime *be, int stream);
+
 /* is the current PCM operation for this FE ? */
 int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);
 

include/uapi/sound/skl-tplg-interface.h

@@ -66,7 +66,8 @@ enum skl_ch_cfg {
 	SKL_CH_CFG_DUAL_MONO = 9,
 	SKL_CH_CFG_I2S_DUAL_STEREO_0 = 10,
 	SKL_CH_CFG_I2S_DUAL_STEREO_1 = 11,
-	SKL_CH_CFG_4_CHANNEL = 12,
+	SKL_CH_CFG_7_1 = 12,
+	SKL_CH_CFG_4_CHANNEL = SKL_CH_CFG_7_1,
 	SKL_CH_CFG_INVALID
 };
 

sound/soc/amd/ps/pci-ps.c

@@ -198,8 +198,7 @@ static int create_acp63_platform_devs(struct pci_dev *pci, struct acp63_dev_data
 	case ACP63_PDM_DEV_MASK:
 		adata->pdm_dev_index  = 0;
 		acp63_fill_platform_dev_info(&pdevinfo[0], parent, NULL, "acp_ps_pdm_dma",
-					     0, adata->res, 1, &adata->acp_lock,
+					     0, adata->res, 1, NULL, 0);
-					     sizeof(adata->acp_lock));
 		acp63_fill_platform_dev_info(&pdevinfo[1], parent, NULL, "dmic-codec",
 					     0, NULL, 0, NULL, 0);
 		acp63_fill_platform_dev_info(&pdevinfo[2], parent, NULL, "acp_ps_mach",

sound/soc/amd/ps/ps-pdm-dma.c

@@ -358,12 +358,12 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev)
 {
 	struct resource *res;
 	struct pdm_dev_data *adata;
+	struct acp63_dev_data *acp_data;
+	struct device *parent;
 	int status;
 
-	if (!pdev->dev.platform_data) {
+	parent = pdev->dev.parent;
-		dev_err(&pdev->dev, "platform_data not retrieved\n");
+	acp_data = dev_get_drvdata(parent);
-		return -ENODEV;
-	}
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "IORESOURCE_MEM FAILED\n");
@@ -379,7 +379,7 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	adata->capture_stream = NULL;
-	adata->acp_lock = pdev->dev.platform_data;
+	adata->acp_lock = &acp_data->acp_lock;
 	dev_set_drvdata(&pdev->dev, adata);
 	status = devm_snd_soc_register_component(&pdev->dev,
 						 &acp63_pdm_component,

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

@@ -171,6 +171,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "21CL"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21EF"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

sound/soc/codecs/cs35l41-lib.c

@@ -46,7 +46,7 @@ static const struct reg_default cs35l41_reg[] = {
 	{ CS35L41_DSP1_RX5_SRC,			0x00000020 },
 	{ CS35L41_DSP1_RX6_SRC,			0x00000021 },
 	{ CS35L41_DSP1_RX7_SRC,			0x0000003A },
-	{ CS35L41_DSP1_RX8_SRC,			0x00000001 },
+	{ CS35L41_DSP1_RX8_SRC,			0x0000003B },
 	{ CS35L41_NGATE1_SRC,			0x00000008 },
 	{ CS35L41_NGATE2_SRC,			0x00000009 },
 	{ CS35L41_AMP_DIG_VOL_CTRL,		0x00008000 },
@@ -58,8 +58,8 @@ static const struct reg_default cs35l41_reg[] = {
 	{ CS35L41_IRQ1_MASK2,			0xFFFFFFFF },
 	{ CS35L41_IRQ1_MASK3,			0xFFFF87FF },
 	{ CS35L41_IRQ1_MASK4,			0xFEFFFFFF },
-	{ CS35L41_GPIO1_CTRL1,			0xE1000001 },
+	{ CS35L41_GPIO1_CTRL1,			0x81000001 },
-	{ CS35L41_GPIO2_CTRL1,			0xE1000001 },
+	{ CS35L41_GPIO2_CTRL1,			0x81000001 },
 	{ CS35L41_MIXER_NGATE_CFG,		0x00000000 },
 	{ CS35L41_MIXER_NGATE_CH1_CFG,		0x00000303 },
 	{ CS35L41_MIXER_NGATE_CH2_CFG,		0x00000303 },

sound/soc/codecs/cs35l56.c

@@ -704,9 +704,6 @@ static int cs35l56_sdw_dai_hw_free(struct snd_pcm_substream *substream,
 static int cs35l56_sdw_dai_set_stream(struct snd_soc_dai *dai,
 				      void *sdw_stream, int direction)
 {
-	if (!sdw_stream)
-		return 0;
-
 	snd_soc_dai_dma_data_set(dai, direction, sdw_stream);
 
 	return 0;

sound/soc/codecs/lpass-tx-macro.c

@@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol,
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
 
 	val = ucontrol->value.enumerated.item[0];
+	if (val >= e->items)
+		return -EINVAL;
 
 	switch (e->reg) {
 	case CDC_TX_INP_MUX_ADC_MUX0_CFG0:
@@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol,
 	case CDC_TX_INP_MUX_ADC_MUX7_CFG0:
 		mic_sel_reg = CDC_TX7_TX_PATH_CFG0;
 		break;
+	default:
+		dev_err(component->dev, "Error in configuration!!\n");
+		return -EINVAL;
 	}
 
 	if (val != 0) {

sound/soc/codecs/max98363.c

@@ -211,7 +211,7 @@ static int max98363_io_init(struct sdw_slave *slave)
 }
 
 #define MAX98363_RATES SNDRV_PCM_RATE_8000_192000
-#define MAX98363_FORMATS (SNDRV_PCM_FMTBIT_S32_LE)
+#define MAX98363_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream,
 				      struct snd_pcm_hw_params *params,
@@ -246,7 +246,7 @@ static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream,
 	stream_config.frame_rate = params_rate(params);
 	stream_config.bps = snd_pcm_format_width(params_format(params));
 	stream_config.direction = direction;
-	stream_config.ch_count = params_channels(params);
+	stream_config.ch_count = 1;
 
 	if (stream_config.ch_count > runtime->hw.channels_max) {
 		stream_config.ch_count = runtime->hw.channels_max;

sound/soc/codecs/nau8824.c

@@ -1903,6 +1903,30 @@ static const struct dmi_system_id nau8824_quirk_table[] = {
 		},
 		.driver_data = (void *)(NAU8824_MONO_SPEAKER),
 	},
+	{
+		/* Positivo CW14Q01P */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"),
+			DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P"),
+		},
+		.driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+	},
+	{
+		/* Positivo K1424G */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"),
+			DMI_MATCH(DMI_BOARD_NAME, "K1424G"),
+		},
+		.driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+	},
+	{
+		/* Positivo N14ZP74G */
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"),
+			DMI_MATCH(DMI_BOARD_NAME, "N14ZP74G"),
+		},
+		.driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+	},
 	{}
 };
 

sound/soc/codecs/rt5682-i2c.c

@@ -266,7 +266,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c)
 		ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL,
 			rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING
 			| IRQF_ONESHOT, "rt5682", rt5682);
-		if (ret)
+		if (!ret)
+			rt5682->irq = i2c->irq;
+		else
 			dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret);
 	}
 

sound/soc/codecs/rt5682.c

@@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component)
 	if (rt5682->is_sdw)
 		return 0;
 
+	if (rt5682->irq)
+		disable_irq(rt5682->irq);
+
 	cancel_delayed_work_sync(&rt5682->jack_detect_work);
 	cancel_delayed_work_sync(&rt5682->jd_check_work);
 	if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) {
@@ -3027,6 +3030,9 @@ static int rt5682_resume(struct snd_soc_component *component)
 	mod_delayed_work(system_power_efficient_wq,
 		&rt5682->jack_detect_work, msecs_to_jiffies(0));
 
+	if (rt5682->irq)
+		enable_irq(rt5682->irq);
+
 	return 0;
 }
 #else

sound/soc/codecs/rt5682.h

@@ -1461,6 +1461,7 @@ struct rt5682_priv {
 	int pll_out[RT5682_PLLS];
 
 	int jack_type;
+	int irq;
 	int irq_work_delay_time;
 };
 

sound/soc/codecs/wcd938x-sdw.c

@@ -1190,7 +1190,6 @@ static const struct regmap_config wcd938x_regmap_config = {
 	.readable_reg = wcd938x_readable_register,
 	.writeable_reg = wcd938x_writeable_register,
 	.volatile_reg = wcd938x_volatile_register,
-	.can_multi_write = true,
 };
 
 static const struct sdw_slave_ops wcd9380_slave_ops = {

sound/soc/codecs/wsa881x.c

@@ -645,7 +645,6 @@ static struct regmap_config wsa881x_regmap_config = {
 	.readable_reg = wsa881x_readable_register,
 	.reg_format_endian = REGMAP_ENDIAN_NATIVE,
 	.val_format_endian = REGMAP_ENDIAN_NATIVE,
-	.can_multi_write = true,
 };
 
 enum {

sound/soc/codecs/wsa883x.c

@@ -946,7 +946,6 @@ static struct regmap_config wsa883x_regmap_config = {
 	.writeable_reg = wsa883x_writeable_register,
 	.reg_format_endian = REGMAP_ENDIAN_NATIVE,
 	.val_format_endian = REGMAP_ENDIAN_NATIVE,
-	.can_multi_write = true,
 	.use_single_read = true,
 };
 

sound/soc/dwc/dwc-i2s.c

@@ -184,30 +184,6 @@ static void i2s_stop(struct dw_i2s_dev *dev,
 	}
 }
 
-static int dw_i2s_startup(struct snd_pcm_substream *substream,
-		struct snd_soc_dai *cpu_dai)
-{
-	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
-	union dw_i2s_snd_dma_data *dma_data = NULL;
-
-	if (!(dev->capability & DWC_I2S_RECORD) &&
-			(substream->stream == SNDRV_PCM_STREAM_CAPTURE))
-		return -EINVAL;
-
-	if (!(dev->capability & DWC_I2S_PLAY) &&
-			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
-		return -EINVAL;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		dma_data = &dev->play_dma_data;
-	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-		dma_data = &dev->capture_dma_data;
-
-	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data);
-
-	return 0;
-}
-
 static void dw_i2s_config(struct dw_i2s_dev *dev, int stream)
 {
 	u32 ch_reg;
@@ -306,12 +282,6 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
-		struct snd_soc_dai *dai)
-{
-	snd_soc_dai_set_dma_data(dai, substream, NULL);
-}
-
 static int dw_i2s_prepare(struct snd_pcm_substream *substream,
 			  struct snd_soc_dai *dai)
 {
@@ -383,8 +353,6 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 }
 
 static const struct snd_soc_dai_ops dw_i2s_dai_ops = {
-	.startup	= dw_i2s_startup,
-	.shutdown	= dw_i2s_shutdown,
 	.hw_params	= dw_i2s_hw_params,
 	.prepare	= dw_i2s_prepare,
 	.trigger	= dw_i2s_trigger,
@@ -626,6 +594,14 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev,
 
 }
 
+static int dw_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_init_dma_data(dai, &dev->play_dma_data, &dev->capture_dma_data);
+	return 0;
+}
+
 static int dw_i2s_probe(struct platform_device *pdev)
 {
 	const struct i2s_platform_data *pdata = pdev->dev.platform_data;
@@ -644,6 +620,7 @@ static int dw_i2s_probe(struct platform_device *pdev)
 		return -ENOMEM;
 
 	dw_i2s_dai->ops = &dw_i2s_dai_ops;
+	dw_i2s_dai->probe = dw_i2s_dai_probe;
 
 	dev->i2s_base = devm_platform_get_and_ioremap_resource(pdev, 0, &res);
 	if (IS_ERR(dev->i2s_base))

sound/soc/fsl/fsl_sai.c

@@ -491,14 +491,21 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
 	regmap_update_bits(sai->regmap, reg, FSL_SAI_CR2_MSEL_MASK,
 			   FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
 
-	if (savediv == 1)
+	if (savediv == 1) {
 		regmap_update_bits(sai->regmap, reg,
 				   FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP,
 				   FSL_SAI_CR2_BYP);
-	else
+		if (fsl_sai_dir_is_synced(sai, adir))
+			regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs),
+					   FSL_SAI_CR2_BCI, FSL_SAI_CR2_BCI);
+		else
+			regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx, ofs),
+					   FSL_SAI_CR2_BCI, 0);
+	} else {
 		regmap_update_bits(sai->regmap, reg,
 				   FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP,
 				   savediv / 2 - 1);
+	}
 
 	if (sai->soc_data->max_register >= FSL_SAI_MCTL) {
 		/* SAI is in master mode at this point, so enable MCLK */

sound/soc/fsl/fsl_sai.h

@@ -116,6 +116,7 @@
 
 /* SAI Transmit and Receive Configuration 2 Register */
 #define FSL_SAI_CR2_SYNC	BIT(30)
+#define FSL_SAI_CR2_BCI		BIT(28)
 #define FSL_SAI_CR2_MSEL_MASK	(0x3 << 26)
 #define FSL_SAI_CR2_MSEL_BUS	0
 #define FSL_SAI_CR2_MSEL_MCLK1	BIT(26)

sound/soc/generic/simple-card-utils.c

@@ -314,7 +314,7 @@ int asoc_simple_startup(struct snd_pcm_substream *substream)
 		}
 		ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE,
 			fixed_rate, fixed_rate);
-		if (ret)
+		if (ret < 0)
 			goto codec_err;
 	}
 

sound/soc/generic/simple-card.c

@@ -416,6 +416,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv,
 
 			if (ret < 0) {
 				of_node_put(codec);
+				of_node_put(plat);
 				of_node_put(np);
 				goto error;
 			}

sound/soc/intel/avs/apl.c

@@ -169,6 +169,7 @@ static bool apl_lp_streaming(struct avs_dev *adev)
 {
 	struct avs_path *path;
 
+	spin_lock(&adev->path_list_lock);
 	/* Any gateway without buffer allocated in LP area disqualifies D0IX. */
 	list_for_each_entry(path, &adev->path_list, node) {
 		struct avs_path_pipeline *ppl;
@@ -188,11 +189,14 @@ static bool apl_lp_streaming(struct avs_dev *adev)
 				if (cfg->copier.dma_type == INVALID_OBJECT_ID)
 					continue;
 
-				if (!mod->gtw_attrs.lp_buffer_alloc)
+				if (!mod->gtw_attrs.lp_buffer_alloc) {
+					spin_unlock(&adev->path_list_lock);
 					return false;
+				}
 			}
 		}
 	}
+	spin_unlock(&adev->path_list_lock);
 
 	return true;
 }

sound/soc/intel/avs/avs.h

@@ -283,8 +283,8 @@ void avs_release_firmwares(struct avs_dev *adev);
 
 int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id,
 			u8 core_id, u8 domain, void *param, u32 param_size,
-			u16 *instance_id);
+			u8 *instance_id);
-void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id,
+void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id,
 			   u8 ppl_instance_id, u8 core_id);
 int avs_dsp_create_pipeline(struct avs_dev *adev, u16 req_size, u8 priority,
 			    bool lp, u16 attributes, u8 *instance_id);

sound/soc/intel/avs/board_selection.c

@@ -443,7 +443,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev)
 	}
 
 	for (mach = boards->machs; mach->id[0]; mach++) {
-		if (!acpi_dev_present(mach->id, NULL, -1))
+		if (!acpi_dev_present(mach->id, mach->uid, -1))
 			continue;
 
 		if (mach->machine_quirk)

sound/soc/intel/avs/control.c

@@ -21,17 +21,25 @@ static struct avs_dev *avs_get_kcontrol_adev(struct snd_kcontrol *kcontrol)
 	return to_avs_dev(w->dapm->component->dev);
 }
 
-static struct avs_path_module *avs_get_kcontrol_module(struct avs_dev *adev, u32 id)
+static struct avs_path_module *avs_get_volume_module(struct avs_dev *adev, u32 id)
 {
 	struct avs_path *path;
 	struct avs_path_pipeline *ppl;
 	struct avs_path_module *mod;
 
-	list_for_each_entry(path, &adev->path_list, node)
+	spin_lock(&adev->path_list_lock);
-		list_for_each_entry(ppl, &path->ppl_list, node)
+	list_for_each_entry(path, &adev->path_list, node) {
-			list_for_each_entry(mod, &ppl->mod_list, node)
+		list_for_each_entry(ppl, &path->ppl_list, node) {
-				if (mod->template->ctl_id && mod->template->ctl_id == id)
+			list_for_each_entry(mod, &ppl->mod_list, node) {
+				if (guid_equal(&mod->template->cfg_ext->type, &AVS_PEAKVOL_MOD_UUID)
+				    && mod->template->ctl_id == id) {
+					spin_unlock(&adev->path_list_lock);
 					return mod;
+				}
+			}
+		}
+	}
+	spin_unlock(&adev->path_list_lock);
 
 	return NULL;
 }
@@ -49,7 +57,7 @@ int avs_control_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
 	/* prevent access to modules while path is being constructed */
 	mutex_lock(&adev->path_mutex);
 
-	active_module = avs_get_kcontrol_module(adev, ctl_data->id);
+	active_module = avs_get_volume_module(adev, ctl_data->id);
 	if (active_module) {
 		ret = avs_ipc_peakvol_get_volume(adev, active_module->module_id,
 						 active_module->instance_id, &dspvols,
@@ -89,7 +97,7 @@ int avs_control_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
 		changed = 1;
 	}
 
-	active_module = avs_get_kcontrol_module(adev, ctl_data->id);
+	active_module = avs_get_volume_module(adev, ctl_data->id);
 	if (active_module) {
 		dspvol.channel_id = AVS_ALL_CHANNELS_MASK;
 		dspvol.target_volume = *volume;

sound/soc/intel/avs/dsp.c

@@ -225,7 +225,7 @@ err:
 
 int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id,
 			u8 core_id, u8 domain, void *param, u32 param_size,
-			u16 *instance_id)
+			u8 *instance_id)
 {
 	struct avs_module_entry mentry;
 	bool was_loaded = false;
@@ -272,7 +272,7 @@ err_mod_entry:
 	return ret;
 }
 
-void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id,
+void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id,
 			   u8 ppl_instance_id, u8 core_id)
 {
 	struct avs_module_entry mentry;

sound/soc/intel/avs/messages.h

@@ -619,7 +619,7 @@ enum avs_channel_config {
 	AVS_CHANNEL_CONFIG_DUAL_MONO = 9,
 	AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_0 = 10,
 	AVS_CHANNEL_CONFIG_I2S_DUAL_STEREO_1 = 11,
-	AVS_CHANNEL_CONFIG_4_CHANNEL = 12,
+	AVS_CHANNEL_CONFIG_7_1 = 12,
 	AVS_CHANNEL_CONFIG_INVALID
 };
 

sound/soc/intel/avs/path.h

@@ -37,7 +37,7 @@ struct avs_path_pipeline {
 
 struct avs_path_module {
 	u16 module_id;
-	u16 instance_id;
+	u8 instance_id;
 	union avs_gtw_attributes gtw_attrs;
 
 	struct avs_tplg_module *template;

sound/soc/intel/avs/pcm.c

@@ -468,21 +468,34 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so
 
 	host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST);
 	if (!host_stream) {
-		kfree(data);
+		ret = -EBUSY;
-		return -EBUSY;
+		goto err;
 	}
 
 	data->host_stream = host_stream;
-	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto err;
+
 	/* avoid wrap-around with wall-clock */
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
+	ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000);
-	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+	if (ret < 0)
+		goto err;
+
+	ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates);
+	if (ret < 0)
+		goto err;
+
 	snd_pcm_set_sync(substream);
 
 	dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p",
 		__func__, hdac_stream(host_stream)->stream_tag, substream);
 
 	return 0;
+
+err:
+	kfree(data);
+	return ret;
 }
 
 static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)

sound/soc/intel/avs/probes.c

@@ -18,7 +18,7 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
 {
 	struct avs_probe_cfg cfg = {{0}};
 	struct avs_module_entry mentry;
-	u16 dummy;
+	u8 dummy;
 
 	avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry);
 

sound/soc/mediatek/mt8188/mt8188-afe-clk.c

@@ -436,13 +436,6 @@ int mt8188_afe_init_clock(struct mtk_base_afe *afe)
 	return 0;
 }
 
-void mt8188_afe_deinit_clock(void *priv)
-{
-	struct mtk_base_afe *afe = priv;
-
-	mt8188_audsys_clk_unregister(afe);
-}
-
 int mt8188_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk)
 {
 	int ret;

sound/soc/mediatek/mt8188/mt8188-afe-clk.h

@@ -111,7 +111,6 @@ int mt8188_afe_get_default_mclk_source_by_rate(int rate);
 int mt8188_get_apll_by_rate(struct mtk_base_afe *afe, int rate);
 int mt8188_get_apll_by_name(struct mtk_base_afe *afe, const char *name);
 int mt8188_afe_init_clock(struct mtk_base_afe *afe);
-void mt8188_afe_deinit_clock(void *priv);
 int mt8188_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk);
 void mt8188_afe_disable_clk(struct mtk_base_afe *afe, struct clk *clk);
 int mt8188_afe_set_clk_rate(struct mtk_base_afe *afe, struct clk *clk,

sound/soc/mediatek/mt8188/mt8188-afe-pcm.c

@@ -3256,10 +3256,6 @@ static int mt8188_afe_pcm_dev_probe(struct platform_device *pdev)
 	if (ret)
 		return dev_err_probe(dev, ret, "init clock error");
 
-	ret = devm_add_action_or_reset(dev, mt8188_afe_deinit_clock, (void *)afe);
-	if (ret)
-		return ret;
-
 	spin_lock_init(&afe_priv->afe_ctrl_lock);
 
 	mutex_init(&afe->irq_alloc_lock);

sound/soc/mediatek/mt8188/mt8188-audsys-clk.c

@@ -138,6 +138,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
 	GATE_AUD6(CLK_AUD_GASRC11, "aud_gasrc11", "top_asm_h", 11),
 };
 
+static void mt8188_audsys_clk_unregister(void *data)
+{
+	struct mtk_base_afe *afe = data;
+	struct mt8188_afe_private *afe_priv = afe->platform_priv;
+	struct clk *clk;
+	struct clk_lookup *cl;
+	int i;
+
+	if (!afe_priv)
+		return;
+
+	for (i = 0; i < CLK_AUD_NR_CLK; i++) {
+		cl = afe_priv->lookup[i];
+		if (!cl)
+			continue;
+
+		clk = cl->clk;
+		clk_unregister_gate(clk);
+
+		clkdev_drop(cl);
+	}
+}
+
 int mt8188_audsys_clk_register(struct mtk_base_afe *afe)
 {
 	struct mt8188_afe_private *afe_priv = afe->platform_priv;
@@ -179,27 +202,5 @@ int mt8188_audsys_clk_register(struct mtk_base_afe *afe)
 		afe_priv->lookup[i] = cl;
 	}
 
-	return 0;
+	return devm_add_action_or_reset(afe->dev, mt8188_audsys_clk_unregister, afe);
-}
-
-void mt8188_audsys_clk_unregister(struct mtk_base_afe *afe)
-{
-	struct mt8188_afe_private *afe_priv = afe->platform_priv;
-	struct clk *clk;
-	struct clk_lookup *cl;
-	int i;
-
-	if (!afe_priv)
-		return;
-
-	for (i = 0; i < CLK_AUD_NR_CLK; i++) {
-		cl = afe_priv->lookup[i];
-		if (!cl)
-			continue;
-
-		clk = cl->clk;
-		clk_unregister_gate(clk);
-
-		clkdev_drop(cl);
-	}
 }

sound/soc/mediatek/mt8188/mt8188-audsys-clk.h

@@ -10,6 +10,5 @@
 #define _MT8188_AUDSYS_CLK_H_
 
 int mt8188_audsys_clk_register(struct mtk_base_afe *afe);
-void mt8188_audsys_clk_unregister(struct mtk_base_afe *afe);
 
 #endif

sound/soc/mediatek/mt8195/mt8195-afe-clk.c

@@ -410,11 +410,6 @@ int mt8195_afe_init_clock(struct mtk_base_afe *afe)
 	return 0;
 }
 
-void mt8195_afe_deinit_clock(struct mtk_base_afe *afe)
-{
-	mt8195_audsys_clk_unregister(afe);
-}
-
 int mt8195_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk)
 {
 	int ret;

sound/soc/mediatek/mt8195/mt8195-afe-clk.h

@@ -101,7 +101,6 @@ int mt8195_afe_get_mclk_source_clk_id(int sel);
 int mt8195_afe_get_mclk_source_rate(struct mtk_base_afe *afe, int apll);
 int mt8195_afe_get_default_mclk_source_by_rate(int rate);
 int mt8195_afe_init_clock(struct mtk_base_afe *afe);
-void mt8195_afe_deinit_clock(struct mtk_base_afe *afe);
 int mt8195_afe_enable_clk(struct mtk_base_afe *afe, struct clk *clk);
 void mt8195_afe_disable_clk(struct mtk_base_afe *afe, struct clk *clk);
 int mt8195_afe_prepare_clk(struct mtk_base_afe *afe, struct clk *clk);

sound/soc/mediatek/mt8195/mt8195-afe-pcm.c

@@ -3224,15 +3224,11 @@ err_pm_put:
 
 static void mt8195_afe_pcm_dev_remove(struct platform_device *pdev)
 {
-	struct mtk_base_afe *afe = platform_get_drvdata(pdev);
-
 	snd_soc_unregister_component(&pdev->dev);
 
 	pm_runtime_disable(&pdev->dev);
 	if (!pm_runtime_status_suspended(&pdev->dev))
 		mt8195_afe_runtime_suspend(&pdev->dev);
-
-	mt8195_afe_deinit_clock(afe);
 }
 
 static const struct of_device_id mt8195_afe_pcm_dt_match[] = {

sound/soc/mediatek/mt8195/mt8195-audsys-clk.c

@@ -148,6 +148,29 @@ static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = {
 	GATE_AUD6(CLK_AUD_GASRC19, "aud_gasrc19", "top_asm_h", 19),
 };
 
+static void mt8195_audsys_clk_unregister(void *data)
+{
+	struct mtk_base_afe *afe = data;
+	struct mt8195_afe_private *afe_priv = afe->platform_priv;
+	struct clk *clk;
+	struct clk_lookup *cl;
+	int i;
+
+	if (!afe_priv)
+		return;
+
+	for (i = 0; i < CLK_AUD_NR_CLK; i++) {
+		cl = afe_priv->lookup[i];
+		if (!cl)
+			continue;
+
+		clk = cl->clk;
+		clk_unregister_gate(clk);
+
+		clkdev_drop(cl);
+	}
+}
+
 int mt8195_audsys_clk_register(struct mtk_base_afe *afe)
 {
 	struct mt8195_afe_private *afe_priv = afe->platform_priv;
@@ -188,27 +211,5 @@ int mt8195_audsys_clk_register(struct mtk_base_afe *afe)
 		afe_priv->lookup[i] = cl;
 	}
 
-	return 0;
+	return devm_add_action_or_reset(afe->dev, mt8195_audsys_clk_unregister, afe);
-}
-
-void mt8195_audsys_clk_unregister(struct mtk_base_afe *afe)
-{
-	struct mt8195_afe_private *afe_priv = afe->platform_priv;
-	struct clk *clk;
-	struct clk_lookup *cl;
-	int i;
-
-	if (!afe_priv)
-		return;
-
-	for (i = 0; i < CLK_AUD_NR_CLK; i++) {
-		cl = afe_priv->lookup[i];
-		if (!cl)
-			continue;
-
-		clk = cl->clk;
-		clk_unregister_gate(clk);
-
-		clkdev_drop(cl);
-	}
 }

sound/soc/mediatek/mt8195/mt8195-audsys-clk.h

@@ -10,6 +10,5 @@
 #define _MT8195_AUDSYS_CLK_H_
 
 int mt8195_audsys_clk_register(struct mtk_base_afe *afe);
-void mt8195_audsys_clk_unregister(struct mtk_base_afe *afe);
 
 #endif

sound/soc/soc-pcm.c

@@ -2431,6 +2431,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
 		if (!snd_soc_dpcm_be_can_update(fe, be, stream))
 			continue;
 
+		if (!snd_soc_dpcm_can_be_prepared(fe, be, stream))
+			continue;
+
 		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) &&
@@ -3087,3 +3090,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 	return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
+
+/*
+ * We can only prepare a BE DAI if any of it's FE are not prepared,
+ * running or paused for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe,
+				 struct snd_soc_pcm_runtime *be, int stream)
+{
+	const enum snd_soc_dpcm_state state[] = {
+		SND_SOC_DPCM_STATE_START,
+		SND_SOC_DPCM_STATE_PAUSED,
+		SND_SOC_DPCM_STATE_PREPARE,
+	};
+
+	return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared);

sound/soc/tegra/tegra_pcm.c

@@ -117,6 +117,9 @@ int tegra_pcm_open(struct snd_soc_component *component,
 		return ret;
 	}
 
+	/* Set wait time to 500ms by default */
+	substream->wait_time = 500;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(tegra_pcm_open);