ASoC: Merge up workaround for CODECs that play noise on stopped stream

This was sent too late to actually make it for v6.6 but was sent against
v6.6 so merge it up here.

7 files changed, 34 insertions, 11 deletions

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 5fcfba47d98c..adcd8719d343 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -370,6 +370,7 @@ struct snd_soc_dai_ops {
 
 	/* bit field */
 	unsigned int no_capture_mute:1;
+	unsigned int mute_unmute_on_trigger:1;
 };
 
 struct snd_soc_cdai_ops {
diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c
index 197fae23762f..cb83c569e18d 100644
--- a/sound/soc/codecs/wsa883x.c
+++ b/sound/soc/codecs/wsa883x.c
@@ -1203,9 +1203,6 @@ static int wsa883x_spkr_event(struct snd_soc_dapm_widget *w,
 			break;
 		}
 
-		snd_soc_component_write_field(component, WSA883X_DRE_CTL_1,
-					      WSA883X_DRE_GAIN_EN_MASK,
-					      WSA883X_DRE_GAIN_FROM_CSR);
 		if (wsa883x->port_enable[WSA883X_PORT_COMP])
 			snd_soc_component_write_field(component, WSA883X_DRE_CTL_0,
 						      WSA883X_DRE_OFFSET_MASK,
@@ -1218,9 +1215,6 @@ static int wsa883x_spkr_event(struct snd_soc_dapm_widget *w,
 		snd_soc_component_write_field(component, WSA883X_PDM_WD_CTL,
 					      WSA883X_PDM_EN_MASK,
 					      WSA883X_PDM_ENABLE);
-		snd_soc_component_write_field(component, WSA883X_PA_FSM_CTL,
-					      WSA883X_GLOBAL_PA_EN_MASK,
-					      WSA883X_GLOBAL_PA_ENABLE);
 
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
@@ -1346,6 +1340,7 @@ static const struct snd_soc_dai_ops wsa883x_dai_ops = {
 	.hw_free = wsa883x_hw_free,
 	.mute_stream = wsa883x_digital_mute,
 	.set_stream = wsa883x_set_sdw_stream,
+	.mute_unmute_on_trigger = true,
 };
 
 static struct snd_soc_dai_driver wsa883x_dais[] = {
diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c
index ba62995c909a..ec53bda46a46 100644
--- a/sound/soc/fsl/fsl_easrc.c
+++ b/sound/soc/fsl/fsl_easrc.c
@@ -1966,17 +1966,21 @@ static int fsl_easrc_probe(struct platform_device *pdev)
 					      &fsl_easrc_dai, 1);
 	if (ret) {
 		dev_err(dev, "failed to register ASoC DAI\n");
-		return ret;
+		goto err_pm_disable;
 	}
 
 	ret = devm_snd_soc_register_component(dev, &fsl_asrc_component,
 					      NULL, 0);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register ASoC platform\n");
-		return ret;
+		goto err_pm_disable;
 	}
 
 	return 0;
+
+err_pm_disable:
+	pm_runtime_disable(&pdev->dev);
+	return ret;
 }
 
 static void fsl_easrc_remove(struct platform_device *pdev)
diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c
index 57ea815d3f04..b776c58dcf47 100644
--- a/sound/soc/intel/skylake/skl-sst-utils.c
+++ b/sound/soc/intel/skylake/skl-sst-utils.c
@@ -299,6 +299,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw,
 		module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL);
 		if (!module->instance_id) {
 			ret = -ENOMEM;
+			kfree(module);
 			goto free_uuid_list;
 		}
 
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 3fe1271204fc..6f8773a8fc05 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -658,6 +658,10 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream,
 			ret = soc_dai_trigger(dai, substream, cmd);
 			if (ret < 0)
 				break;
+
+			if (dai->driver->ops && dai->driver->ops->mute_unmute_on_trigger)
+				snd_soc_dai_digital_mute(dai, 0, substream->stream);
+
 			soc_dai_mark_push(dai, substream, trigger);
 		}
 		break;
@@ -668,6 +672,9 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream,
 			if (rollback && !soc_dai_mark_match(dai, substream, trigger))
 				continue;
 
+			if (dai->driver->ops && dai->driver->ops->mute_unmute_on_trigger)
+				snd_soc_dai_digital_mute(dai, 1, substream->stream);
+
 			r = soc_dai_trigger(dai, substream, cmd);
 			if (r < 0)
 				ret = r; /* use last ret */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 8c168dc553f6..323e4d7b6adf 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -700,10 +700,16 @@ static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd,
 		snd_soc_runtime_deactivate(rtd, substream->stream);
 
 		/* Make sure DAI parameters cleared if the DAI becomes inactive */
-		for_each_rtd_dais(rtd, i, dai)
+		for_each_rtd_dais(rtd, i, dai) {
 			if (snd_soc_dai_active(dai) == 0 &&
 			    (dai->rate || dai->channels || dai->sample_bits))
 				soc_pcm_set_dai_params(dai, NULL);
+
+			if (snd_soc_dai_stream_active(dai, substream->stream) ==  0) {
+				if (dai->driver->ops && !dai->driver->ops->mute_unmute_on_trigger)
+					snd_soc_dai_digital_mute(dai, 1, substream->stream);
+			}
+		}
 	}
 
 	for_each_rtd_dais(rtd, i, dai)
@@ -896,8 +902,10 @@ static int __soc_pcm_prepare(struct snd_soc_pcm_runtime *rtd,
 	snd_soc_dapm_stream_event(rtd, substream->stream,
 			SND_SOC_DAPM_STREAM_START);
 
-	for_each_rtd_dais(rtd, i, dai)
-		snd_soc_dai_digital_mute(dai, 0, substream->stream);
+	for_each_rtd_dais(rtd, i, dai) {
+		if (dai->driver->ops && !dai->driver->ops->mute_unmute_on_trigger)
+			snd_soc_dai_digital_mute(dai, 0, substream->stream);
+	}
 
 out:
 	return soc_pcm_ret(rtd, ret);
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 1d706490588e..64b326e3ef85 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -145,6 +145,13 @@ static const struct dmi_system_id community_key_platforms[] = {
 			DMI_MATCH(DMI_PRODUCT_FAMILY, "Google"),
 		}
 	},
+	{
+		.ident = "Google firmware",
+		.callback = chromebook_use_community_key,
+		.matches = {
+			DMI_MATCH(DMI_BIOS_VERSION, "Google"),
+		}
+	},
 	{},
 };