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ASoC: Merge up fixes for CI

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Avoid tripping over fixed issues in CI.
This commit is contained in:
Mark Brown 2023-05-12 13:14:20 +09:00
commit 1946dda47c
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GPG Key ID: 24D68B725D5487D0
9 changed files with 105 additions and 14 deletions
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6
Documentation/devicetree/bindings/sound/tas2562.yaml
View File

@ -55,7 +55,9 @@ properties:
description: TDM TX current sense time slot.
'#sound-dai-cells':
const: 1
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
# compatibility but is deprecated.
enum: [0, 1]
required:
- compatible
@ -72,7 +74,7 @@ examples:
codec: codec@4c {
compatible = "ti,tas2562";
reg = <0x4c>;
#sound-dai-cells = <1>;
#sound-dai-cells = <0>;
interrupt-parent = <&gpio1>;
interrupts = <14>;
shutdown-gpios = <&gpio1 15 0>;

6
Documentation/devicetree/bindings/sound/tas2770.yaml
View File

@ -57,7 +57,9 @@ properties:
- 1 # Falling edge
'#sound-dai-cells':
const: 1
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
# compatibility but is deprecated.
enum: [0, 1]
required:
- compatible
@ -74,7 +76,7 @@ examples:
codec: codec@41 {
compatible = "ti,tas2770";
reg = <0x41>;
#sound-dai-cells = <1>;
#sound-dai-cells = <0>;
interrupt-parent = <&gpio1>;
interrupts = <14>;
reset-gpio = <&gpio1 15 0>;

6
Documentation/devicetree/bindings/sound/tas27xx.yaml
View File

@ -50,7 +50,9 @@ properties:
description: TDM TX voltage sense time slot.
'#sound-dai-cells':
const: 1
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
# compatibility but is deprecated.
enum: [0, 1]
required:
- compatible
@ -67,7 +69,7 @@ examples:
codec: codec@38 {
compatible = "ti,tas2764";
reg = <0x38>;
#sound-dai-cells = <1>;
#sound-dai-cells = <0>;
interrupt-parent = <&gpio1>;
interrupts = <14>;
reset-gpios = <&gpio1 15 0>;

7
sound/soc/amd/yc/acp6x-mach.c
View File

@ -318,6 +318,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
DMI_MATCH(DMI_BOARD_NAME, "MRID6"),
}
},
{
.driver_data = &acp6x_card,
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "System76"),
DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"),
}
},
{}
};

15
sound/soc/codecs/ssm2602.c
View File

@ -53,6 +53,18 @@ static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = {
{ .reg = 0x09, .def = 0x0000 }
};
/*
* ssm2602 register patch
* Workaround for playback distortions after power up: activates digital
* core, and then powers on output, DAC, and whole chip at the same time
*/
static const struct reg_sequence ssm2602_patch[] = {
{ SSM2602_ACTIVE, 0x01 },
{ SSM2602_PWR, 0x07 },
{ SSM2602_RESET, 0x00 },
};
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
@ -598,6 +610,9 @@ static int ssm260x_component_probe(struct snd_soc_component *component)
return ret;
}
regmap_register_patch(ssm2602->regmap, ssm2602_patch,
ARRAY_SIZE(ssm2602_patch));
/* set the update bits */
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);

4
sound/soc/dwc/dwc-i2s.c
View File

@ -133,13 +133,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
/* Error Handling: TX */
if (isr[i] & ISR_TXFO) {
dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i);
dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i);
irq_valid = true;
}
/* Error Handling: TX */
if (isr[i] & ISR_RXFO) {
dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i);
dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i);
irq_valid = true;
}
}

14
sound/soc/fsl/fsl_micfil.c
View File

@ -1159,7 +1159,7 @@ static int fsl_micfil_probe(struct platform_device *pdev)
ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
if (ret) {
dev_err(&pdev->dev, "failed to pcm register\n");
return ret;
goto err_pm_disable;
}
fsl_micfil_dai.capture.formats = micfil->soc->formats;
@ -1169,9 +1169,20 @@ static int fsl_micfil_probe(struct platform_device *pdev)
if (ret) {
dev_err(&pdev->dev, "failed to register component %s\n",
fsl_micfil_component.name);
goto err_pm_disable;
}
return ret;
err_pm_disable:
pm_runtime_disable(&pdev->dev);
return ret;
}
static void fsl_micfil_remove(struct platform_device *pdev)
{
pm_runtime_disable(&pdev->dev);
}
static int __maybe_unused fsl_micfil_runtime_suspend(struct device *dev)
@ -1232,6 +1243,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = {
static struct platform_driver fsl_micfil_driver = {
.probe = fsl_micfil_probe,
.remove_new = fsl_micfil_remove,
.driver = {
.name = "fsl-micfil-dai",
.pm = &fsl_micfil_pm_ops,

54
sound/soc/jz4740/jz4740-i2s.c
View File

@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate,
unsigned long i2sdiv_max)
{
unsigned long div, rate1, rate2, err1, err2;
div = mclk / (64 * rate);
if (div == 0)
div = 1;
rate1 = mclk / (64 * div);
rate2 = mclk / (64 * (div + 1));
err1 = abs(rate1 - rate);
err2 = abs(rate2 - rate);
/*
* Choose the divider that produces the smallest error in the
* output rate and reject dividers with a 5% or higher error.
* In the event that both dividers are outside the acceptable
* error margin, reject the rate to prevent distorted audio.
* (The number 5% is arbitrary.)
*/
if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20)
return div;
if (div < i2sdiv_max && err2 < rate/20)
return div + 1;
return -EINVAL;
}
static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
struct regmap_field *div_field;
unsigned long i2sdiv_max;
unsigned int sample_size;
uint32_t ctrl;
int div;
uint32_t ctrl, conf;
int div = 1;
regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl);
div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
div_field = i2s->field_i2sdiv_playback;
i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb,
i2s->soc_info->field_i2sdiv_playback.lsb);
} else {
ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE;
ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size);
div_field = i2s->field_i2sdiv_capture;
i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb,
i2s->soc_info->field_i2sdiv_capture.lsb);
}
/*
* Only calculate I2SDIV if we're supplying the bit or frame clock.
* If the codec is supplying both clocks then the divider output is
* unused, and we don't want it to limit the allowed sample rates.
*/
if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) {
div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s),
params_rate(params), i2sdiv_max);
if (div < 0)
return div;
}
regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl);

7
sound/soc/sof/amd/acp-ipc.c
View File

@ -209,7 +209,12 @@ int acp_sof_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sp
acp_mailbox_read(sdev, offset, p, sz);
} else {
struct snd_pcm_substream *substream = sps->substream;
struct acp_dsp_stream *stream = substream->runtime->private_data;
struct acp_dsp_stream *stream;
if (!substream || !substream->runtime)
return -ESTRPIPE;
stream = substream->runtime->private_data;
if (!stream)
return -ESTRPIPE;