Merge existing fixes from asoc/for-5.17 into new branch

sound/soc/codecs/cpcap.c

@@ -1667,6 +1667,8 @@ static int cpcap_codec_probe(struct platform_device *pdev)
 {
 	struct device_node *codec_node =
 		of_get_child_by_name(pdev->dev.parent->of_node, "audio-codec");
+	if (!codec_node)
+		return -ENODEV;
 
 	pdev->dev.of_node = codec_node;
 

sound/soc/codecs/max9759.c

@@ -64,7 +64,8 @@ static int speaker_gain_control_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol);
 	struct max9759 *priv = snd_soc_component_get_drvdata(c);
 
-	if (ucontrol->value.integer.value[0] > 3)
+	if (ucontrol->value.integer.value[0] < 0 ||
+	    ucontrol->value.integer.value[0] > 3)
 		return -EINVAL;
 
 	priv->gain = ucontrol->value.integer.value[0];

sound/soc/generic/simple-card.c

@@ -28,6 +28,30 @@ static const struct snd_soc_ops simple_ops = {
 	.hw_params	= asoc_simple_hw_params,
 };
 
+static int asoc_simple_parse_platform(struct device_node *node,
+				      struct snd_soc_dai_link_component *dlc)
+{
+	struct of_phandle_args args;
+	int ret;
+
+	if (!node)
+		return 0;
+
+	/*
+	 * Get node via "sound-dai = <&phandle port>"
+	 * it will be used as xxx_of_node on soc_bind_dai_link()
+	 */
+	ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args);
+	if (ret)
+		return ret;
+
+	/* dai_name is not required and may not exist for plat component */
+
+	dlc->of_node = args.np;
+
+	return 0;
+}
+
 static int asoc_simple_parse_dai(struct device_node *node,
 				 struct snd_soc_dai_link_component *dlc,
 				 int *is_single_link)
@@ -289,7 +313,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv,
 	if (ret < 0)
 		goto dai_link_of_err;
 
-	ret = asoc_simple_parse_dai(plat, platforms, NULL);
+	ret = asoc_simple_parse_platform(plat, platforms);
 	if (ret < 0)
 		goto dai_link_of_err;
 

sound/soc/mediatek/Kconfig

@@ -216,7 +216,7 @@ config SND_SOC_MT8195_MT6359_RT1019_RT5682
 
 config SND_SOC_MT8195_MT6359_RT1011_RT5682
 	tristate "ASoC Audio driver for MT8195 with MT6359 RT1011 RT5682 codec"
-	depends on I2C
+	depends on I2C && GPIOLIB
 	depends on SND_SOC_MT8195 && MTK_PMIC_WRAP
 	select SND_SOC_MT6359
 	select SND_SOC_RT1011

sound/soc/xilinx/xlnx_formatter_pcm.c

@@ -37,6 +37,7 @@
 #define XLNX_AUD_XFER_COUNT	0x28
 #define XLNX_AUD_CH_STS_START	0x2C
 #define XLNX_BYTES_PER_CH	0x44
+#define XLNX_AUD_ALIGN_BYTES	64
 
 #define AUD_STS_IOC_IRQ_MASK	BIT(31)
 #define AUD_STS_CH_STS_MASK	BIT(29)
@@ -368,12 +369,32 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component,
 	snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware);
 	runtime->private_data = stream_data;
 
-	/* Resize the period size divisible by 64 */
+	/* Resize the period bytes as divisible by 64 */
 	err = snd_pcm_hw_constraint_step(runtime, 0,
-					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					 XLNX_AUD_ALIGN_BYTES);
 	if (err) {
 		dev_err(component->dev,
-			"unable to set constraint on period bytes\n");
+			"Unable to set constraint on period bytes\n");
+		return err;
+	}
+
+	/* Resize the buffer bytes as divisible by 64 */
+	err = snd_pcm_hw_constraint_step(runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+					 XLNX_AUD_ALIGN_BYTES);
+	if (err) {
+		dev_err(component->dev,
+			"Unable to set constraint on buffer bytes\n");
+		return err;
+	}
+
+	/* Set periods as integer multiple */
+	err = snd_pcm_hw_constraint_integer(runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0) {
+		dev_err(component->dev,
+			"Unable to set constraint on periods to be integer\n");
 		return err;
 	}