ASoC: Merge up fixes for CI

Avoid tripping over fixed issues in CI.

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>;

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>;

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>;

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"),
+		}
+	},
 	{}
 };
 

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);

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;
 		}
 	}

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,

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);

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;