Merge branch 'asoc-4.17' into asoc-4.18 to get adau17x1 changes so

further patches can be applied.

sound/soc/codecs/adau17x1.c

@@ -502,7 +502,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (adau->sigmadsp) {
-		ret = adau17x1_setup_firmware(adau, params_rate(params));
+		ret = adau17x1_setup_firmware(component, params_rate(params));
 		if (ret < 0)
 			return ret;
 	}
@@ -835,26 +835,40 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct adau *adau, unsigned int rate)
+int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate)
 {
 	int ret;
-	int dspsr;
+	int dspsr, dsp_run;
+	struct adau *adau = snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+
+	snd_soc_dapm_mutex_lock(dapm);
 
 	ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr);
 	if (ret)
-		return ret;
+		goto err;
+
+	ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run);
+	if (ret)
+		goto err;
 
 	regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1);
 	regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf);
+	regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0);
 
 	ret = sigmadsp_setup(adau->sigmadsp, rate);
 	if (ret) {
 		regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0);
-		return ret;
+		goto err;
 	}
 	regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr);
+	regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run);
 
-	return 0;
+err:
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 

sound/soc/codecs/adau17x1.h

@@ -68,7 +68,8 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct adau *adau, unsigned int rate);
+int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate);
 bool adau17x1_has_dsp(struct adau *adau);
 
 #define ADAU17X1_CLOCK_CONTROL			0x4000

sound/soc/fsl/fsl_esai.c

@@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio,
 
 	psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8;
 
+	/* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */
+	if (ratio <= 256) {
+		pm = ratio;
+		fp = 1;
+		goto out;
+	}
+
 	/* Set the max fluctuation -- 0.1% of the max devisor */
 	savesub = (psr ? 1 : 8)  * 256 * maxfp / 1000;
 

sound/soc/fsl/fsl_ssi.c

@@ -217,6 +217,7 @@ struct fsl_ssi_soc_data {
  * @dai_fmt: DAI configuration this device is currently used with
  * @streams: Mask of current active streams: BIT(TX) and BIT(RX)
  * @i2s_net: I2S and Network mode configurations of SCR register
+ *           (this is the initial settings based on the DAI format)
  * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK
  * @use_dma: DMA is used or FIQ with stream filter
  * @use_dual_fifo: DMA with support for dual FIFO mode
@@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (!fsl_ssi_is_ac97(ssi)) {
+		/*
+		 * Keep the ssi->i2s_net intact while having a local variable
+		 * to override settings for special use cases. Otherwise, the
+		 * ssi->i2s_net will lose the settings for regular use cases.
+		 */
+		u8 i2s_net = ssi->i2s_net;
+
 		/* Normal + Network mode to send 16-bit data in 32-bit frames */
 		if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16)
-			ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET;
+			i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET;
 
 		/* Use Normal mode to send mono data at 1st slot of 2 slots */
 		if (channels == 1)
-			ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL;
+			i2s_net = SSI_SCR_I2S_MODE_NORMAL;
 
 		regmap_update_bits(regs, REG_SSI_SCR,
-				   SSI_SCR_I2S_NET_MASK, ssi->i2s_net);
+				   SSI_SCR_I2S_NET_MASK, i2s_net);
 	}
 
 	/* In synchronous mode, the SSI uses STCCR for capture */

sound/soc/intel/Kconfig

@@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL
 	  for Baytrail Chromebooks but this option is now deprecated and is
 	  not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead.
 
+config SND_SST_ATOM_HIFI2_PLATFORM
+	tristate
+	select SND_SOC_COMPRESS
+
 config SND_SST_ATOM_HIFI2_PLATFORM_PCI
-	tristate "PCI HiFi2 (Medfield, Merrifield) Platforms"
+	tristate "PCI HiFi2 (Merrifield) Platforms"
 	depends on X86 && PCI
 	select SND_SST_IPC_PCI
-	select SND_SOC_COMPRESS
+	select SND_SST_ATOM_HIFI2_PLATFORM
 	help
-	  If you have a Intel Medfield or Merrifield/Edison platform, then
+	  If you have a Intel Merrifield/Edison platform, then
 	  enable this option by saying Y or m. Distros will typically not
-	  enable this option: Medfield devices are not available to
+	  enable this option: while Merrifield/Edison can run a mainline
-	  developers and while Merrifield/Edison can run a mainline kernel with
+	  kernel with limited functionality it will require a firmware file
-	  limited functionality it will require a firmware file which
+	  which is not in the standard firmware tree
-	  is not in the standard firmware tree
 
-config SND_SST_ATOM_HIFI2_PLATFORM
+config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
+	default ACPI
 	depends on X86 && ACPI
 	select SND_SST_IPC_ACPI
-	select SND_SOC_COMPRESS
+	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH
 	select IOSF_MBI
 	help

sound/soc/omap/omap-dmic.c

@@ -281,7 +281,7 @@ static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream,
 static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
 				 unsigned int freq)
 {
-	struct clk *parent_clk;
+	struct clk *parent_clk, *mux;
 	char *parent_clk_name;
 	int ret = 0;
 
@@ -329,14 +329,21 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
 		return -ENODEV;
 	}
 
+	mux = clk_get_parent(dmic->fclk);
+	if (IS_ERR(mux)) {
+		dev_err(dmic->dev, "can't get fck mux parent\n");
+		clk_put(parent_clk);
+		return -ENODEV;
+	}
+
 	mutex_lock(&dmic->mutex);
 	if (dmic->active) {
 		/* disable clock while reparenting */
 		pm_runtime_put_sync(dmic->dev);
-		ret = clk_set_parent(dmic->fclk, parent_clk);
+		ret = clk_set_parent(mux, parent_clk);
 		pm_runtime_get_sync(dmic->dev);
 	} else {
-		ret = clk_set_parent(dmic->fclk, parent_clk);
+		ret = clk_set_parent(mux, parent_clk);
 	}
 	mutex_unlock(&dmic->mutex);
 
@@ -349,6 +356,7 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
 	dmic->fclk_freq = freq;
 
 err_busy:
+	clk_put(mux);
 	clk_put(parent_clk);
 
 	return ret;

sound/soc/sh/rcar/core.c

@@ -1573,7 +1573,7 @@ static int rsnd_remove(struct platform_device *pdev)
 	return ret;
 }
 
-static int rsnd_suspend(struct device *dev)
+static int __maybe_unused rsnd_suspend(struct device *dev)
 {
 	struct rsnd_priv *priv = dev_get_drvdata(dev);
 
@@ -1582,7 +1582,7 @@ static int rsnd_suspend(struct device *dev)
 	return 0;
 }
 
-static int rsnd_resume(struct device *dev)
+static int __maybe_unused rsnd_resume(struct device *dev)
 {
 	struct rsnd_priv *priv = dev_get_drvdata(dev);
 

sound/soc/soc-topology.c

@@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
 			ec->hdr.name);
 
 		kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL);
-		if (kc[i].name == NULL)
+		if (kc[i].name == NULL) {
+			kfree(se);
 			goto err_se;
+		}
 		kc[i].private_value = (long)se;
 		kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 		kc[i].access = ec->hdr.access;
@@ -1442,8 +1444,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create(
 			be->hdr.name, be->hdr.access);
 
 		kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL);
-		if (kc[i].name == NULL)
+		if (kc[i].name == NULL) {
+			kfree(sbe);
 			goto err;
+		}
 		kc[i].private_value = (long)sbe;
 		kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 		kc[i].access = be->hdr.access;