Unverified Commit 6cbff4b3 authored by Mark Brown's avatar Mark Brown
Browse files

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

parents e783362e 579b2c8f

sound/soc/codecs/cpcap.c

+2 −0
Original line number Diff line number Diff line
@@ -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

+2 −1
Original line number Diff line number Diff line
@@ -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

+25 −1
Original line number Diff line number Diff line
@@ -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

+1 −1
Original line number Diff line number Diff line
@@ -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

+24 −3
Original line number Diff line number Diff line
@@ -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;

	}