Merge remote-tracking branch 'asoc/for-5.14' into asoc-linus

broonie · broonie · commit 85666456ceab · 2021-07-19T16:27:43.000+01:00
diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml
@@ -114,7 +114,7 @@ properties:
 
   ports:
     $ref: /schemas/graph.yaml#/properties/ports
-    properties:
+    patternProperties:
       port(@[0-9a-f]+)?:
         $ref: audio-graph-port.yaml#
         unevaluatedProperties: false
diff --git a/include/sound/soc.h b/include/sound/soc.h
@@ -712,6 +712,12 @@ struct snd_soc_dai_link {
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int ignore:1;
 
+	/* This flag will reorder stop sequence. By enabling this flag
+	 * DMA controller stop sequence will be invoked first followed by
+	 * CPU DAI driver stop sequence
+	 */
+	unsigned int stop_dma_first:1;
+
 #ifdef CONFIG_SND_SOC_TOPOLOGY
 	struct snd_soc_dobj dobj; /* For topology */
 #endif
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
@@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.init = cz_rt5682_init,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_play_ops,
 		SND_SOC_DAILINK_REG(designware1, rt5682, platform),
 	},
@@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, rt5682, platform),
 	},
@@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_playback = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_max_play_ops,
 		SND_SOC_DAILINK_REG(designware3, mx, platform),
 	},
@@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_dmic0_cap_ops,
 		SND_SOC_DAILINK_REG(designware3, adau, platform),
 	},
@@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = {
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
 		.dpcm_capture = 1,
+		.stop_dma_first = 1,
 		.ops = &cz_rt5682_dmic1_cap_ops,
 		SND_SOC_DAILINK_REG(designware2, adau, platform),
 	},
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
@@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305
 	  high-efficiency mono Class-D audio power amplifiers.
 
 config SND_SOC_SSM2518
-	tristate
+	tristate "Analog Devices SSM2518 Class-D Amplifier"
 	depends on I2C
 
 config SND_SOC_SSM2602
@@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X
 	  Qualcomm SoCs like SDM845.
 
 config SND_SOC_WCD938X
+	depends on SND_SOC_WCD938X_SDW
 	tristate
 
 config SND_SOC_WCD938X_SDW
@@ -1813,11 +1814,6 @@ config SND_SOC_ZL38060
 	  which consists of a Digital Signal Processor (DSP), several Digital
 	  Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs.
 
-config SND_SOC_ZX_AUD96P22
-	tristate "ZTE ZX AUD96P22 CODEC"
-	depends on I2C
-	select REGMAP_I2C
-
 # Amp
 config SND_SOC_LM4857
 	tristate
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
@@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = {
 	.reg_defaults = rt5631_reg,
 	.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
 	.cache_type = REGCACHE_RBTREE,
+	.use_single_read = true,
+	.use_single_write = true,
 };
 
 static int rt5631_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
@@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
-		if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+		if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
 			snd_soc_component_update_bits(component,
 				RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
-		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
+		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL1") &&
+			!snd_soc_dapm_get_pin_status(dapm, "PLL2B"))
 			snd_soc_component_update_bits(component,
 				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
@@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 			ret);
 		return ret;
 	}
+	regcache_cache_only(aic31xx->regmap, true);
+
 	aic31xx->dev = &i2c->dev;
 	aic31xx->irq = i2c->irq;
 
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
@@ -151,8 +151,8 @@ struct aic31xx_pdata {
 #define AIC31XX_WORD_LEN_24BITS		0x02
 #define AIC31XX_WORD_LEN_32BITS		0x03
 #define AIC31XX_IFACE1_MASTER_MASK	GENMASK(3, 2)
-#define AIC31XX_BCLK_MASTER		BIT(2)
-#define AIC31XX_WCLK_MASTER		BIT(3)
+#define AIC31XX_BCLK_MASTER		BIT(3)
+#define AIC31XX_WCLK_MASTER		BIT(2)
 
 /* AIC31XX_DATA_OFFSET */
 #define AIC31XX_DATA_OFFSET_MASK	GENMASK(7, 0)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
@@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
 static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
 /* -12dB min, 0.5dB steps */
 static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
-
-static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0);
 static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1);
 
 static const char * const lo_cm_text[] = {
@@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = {
 };
 
 static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = {
-	SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
-			AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+	SOC_SINGLE_S8_TLV("PCM Playback Volume",
+			  AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm),
 	SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum),
-	SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN,
-			AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0,
-			tlv_driver_gain),
-	SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
-			AIC32X4_HPLGAIN, 6, 0x01, 1),
 
-	SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+	SOC_SINGLE_TLV("HP Driver Gain Volume",
+			AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain),
+	SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1),
 
-	SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1,
-			0, 0, 117, 1, tlv_spk_vol),
-	SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2,
-			4, 5, 0, tlv_amp_vol),
+	SOC_SINGLE_TLV("Speaker Driver Playback Volume",
+			TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain),
+	SOC_SINGLE_TLV("Speaker Amplifier Playback Volume",
+			TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol),
+
+	SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
 };
 
 static const struct snd_kcontrol_new hp_output_mixer_controls[] = {
diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c
@@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component)
 			     (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0);
 	}
 
-	ret = wcd938x_irq_init(wcd938x, component->dev);
-	if (ret) {
-		dev_err(component->dev, "%s: IRQ init failed: %d\n",
-			__func__, ret);
-		return ret;
-	}
-
 	wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
 						       WCD938X_IRQ_HPHR_PDM_WD_INT);
 	wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip,
@@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev)
 	}
 	wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev);
 	wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x;
-	wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
 
 	wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode);
 	if (!wcd938x->txdev) {
@@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev)
 	}
 	wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev);
 	wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x;
-	wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
 	wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev);
 	if (!wcd938x->tx_sdw_dev) {
 		dev_err(dev, "could not get txslave with matching of dev\n");
@@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev)
 		return PTR_ERR(wcd938x->regmap);
 	}
 
+	ret = wcd938x_irq_init(wcd938x, dev);
+	if (ret) {
+		dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret);
+		return ret;
+	}
+
+	wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq;
+	wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq;
+
 	ret = wcd938x_set_micbias_data(wcd938x);
 	if (ret < 0) {
 		dev_err(dev, "%s: bad micbias pdata\n", __func__);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
@@ -282,6 +282,7 @@
 /*
  * HALO_CCM_CORE_CONTROL
  */
+#define HALO_CORE_RESET                     0x00000200
 #define HALO_CORE_EN                        0x00000001
 
 /*
@@ -1213,7 +1214,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl,
 
 	mutex_lock(&ctl->dsp->pwr_lock);
 
-	ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size);
+	ret = wm_coeff_read_ctrl(ctl, ctl->cache, size);
 
 	if (!ret && copy_to_user(bytes, ctl->cache, size))
 		ret = -EFAULT;
@@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp)
 {
 	return regmap_update_bits(dsp->regmap,
 				  dsp->base + HALO_CCM_CORE_CONTROL,
-				  HALO_CORE_EN, HALO_CORE_EN);
+				  HALO_CORE_RESET | HALO_CORE_EN,
+				  HALO_CORE_RESET | HALO_CORE_EN);
 }
 
 static void wm_halo_stop_core(struct wm_adsp *dsp)
diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c
@@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
-static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd)
+static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable)
 {
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *cpu_dai;
 	int ret;
+	int j;
 
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		/* enable max98373 first */
-		ret = max_98373_trigger(substream, cmd);
-		if (ret < 0)
-			break;
-
-		ret = sdw_trigger(substream, cmd);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ret = sdw_trigger(substream, cmd);
-		if (ret < 0)
-			break;
-
-		ret = max_98373_trigger(substream, cmd);
-		break;
-	default:
-		ret = -EINVAL;
-		break;
+	/* set spk pin by playback only */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		return 0;
+
+	cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	for_each_rtd_codec_dais(rtd, j, codec_dai) {
+		struct snd_soc_dapm_context *dapm =
+				snd_soc_component_get_dapm(cpu_dai->component);
+		char pin_name[16];
+
+		snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk",
+			 codec_dai->component->name_prefix);
+
+		if (enable)
+			ret = snd_soc_dapm_enable_pin(dapm, pin_name);
+		else
+			ret = snd_soc_dapm_disable_pin(dapm, pin_name);
+
+		if (!ret)
+			snd_soc_dapm_sync(dapm);
 	}
 
-	return ret;
+	return 0;
+}
+
+static int mx8373_sdw_prepare(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	/* according to soc_pcm_prepare dai link prepare is called first */
+	ret = sdw_prepare(substream);
+	if (ret < 0)
+		return ret;
+
+	return mx8373_enable_spk_pin(substream, true);
+}
+
+static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+
+	/* according to soc_pcm_hw_free dai link free is called first */
+	ret = sdw_hw_free(substream);
+	if (ret < 0)
+		return ret;
+
+	return mx8373_enable_spk_pin(substream, false);
 }
 
 static const struct snd_soc_ops max_98373_sdw_ops = {
 	.startup = sdw_startup,
-	.prepare = sdw_prepare,
-	.trigger = max98373_sdw_trigger,
-	.hw_free = sdw_hw_free,
+	.prepare = mx8373_sdw_prepare,
+	.trigger = sdw_trigger,
+	.hw_free = mx8373_sdw_hw_free,
 	.shutdown = sdw_shutdown,
 };
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
@@ -1015,6 +1015,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 
 static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	int ret = -EINVAL, _ret = 0;
 	int rollback = 0;
 
@@ -1055,14 +1056,23 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
-		if (ret < 0)
-			break;
+		if (rtd->dai_link->stop_dma_first) {
+			ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
 
-		ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
-		if (ret < 0)
-			break;
+			ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
+		} else {
+			ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
 
+			ret = snd_soc_pcm_component_trigger(substream, cmd, rollback);
+			if (ret < 0)
+				break;
+		}
 		ret = snd_soc_link_trigger(substream, cmd, rollback);
 		break;
 	}
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
@@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = {
 static const struct sof_dev_desc adl_desc = {
 	.machines               = snd_soc_acpi_intel_adl_machines,
 	.alt_machines           = snd_soc_acpi_intel_adl_sdw_machines,
+	.use_acpi_target_states = true,
 	.resindex_lpe_base      = 0,
 	.resindex_pcicfg_base   = -1,
 	.resindex_imr_base      = -1,
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c

Original file line number	Diff line number	Diff line change
`@@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,`
`1604`	`1604`	`ret);`
`1605`	`1605`	`return ret;`
`1606`	`1606`	`}`
	`1607`	`+ regcache_cache_only(aic31xx->regmap, true);`
	`1608`	`+`
`1607`	`1609`	`aic31xx->dev = &i2c->dev;`
`1608`	`1610`	`aic31xx->irq = i2c->irq;`
`1609`	`1611`