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-rw-r--r--Documentation/devicetree/bindings/sound/tas2562.yaml6
-rw-r--r--Documentation/devicetree/bindings/sound/tas2770.yaml6
-rw-r--r--Documentation/devicetree/bindings/sound/tas27xx.yaml6
-rw-r--r--sound/soc/amd/yc/acp6x-mach.c7
-rw-r--r--sound/soc/codecs/ssm2602.c15
-rw-r--r--sound/soc/dwc/dwc-i2s.c4
-rw-r--r--sound/soc/fsl/fsl_micfil.c14
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c54
-rw-r--r--sound/soc/sof/amd/acp-ipc.c7
9 files changed, 105 insertions, 14 deletions
diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml
index 41489a3ac79f..f01c0dde0cf7 100644
--- a/Documentation/devicetree/bindings/sound/tas2562.yaml
+++ b/Documentation/devicetree/bindings/sound/tas2562.yaml
@@ -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>;
diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml
index 930bd111b072..be2536e8c440 100644
--- a/Documentation/devicetree/bindings/sound/tas2770.yaml
+++ b/Documentation/devicetree/bindings/sound/tas2770.yaml
@@ -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>;
diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml
index bda26b246634..f2d878f6f495 100644
--- a/Documentation/devicetree/bindings/sound/tas27xx.yaml
+++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml
@@ -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>;
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index a134eba4d59a..31e466917c3d 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -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"),
+ }
+ },
{}
};
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 00b60369b029..c29324403e9d 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -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);
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index c5ba88e050e7..dd5d8d77bdc9 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -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;
}
}
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 94341e4352b3..3f08082a55be 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -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,
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 7cb563bb8b09..578af21769c9 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -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);
diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c
index 4e0c48a36159..749e856dc601 100644
--- a/sound/soc/sof/amd/acp-ipc.c
+++ b/sound/soc/sof/amd/acp-ipc.c
@@ -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;