ASoC: Merge up fixes

For the benefit of CI.

4 files changed, 24 insertions, 7 deletions

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5b31c12a56f9..0957ff7c55c2 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -52,8 +52,8 @@ struct codec_priv {
 	unsigned long mclk_freq;
 	unsigned long free_freq;
 	u32 mclk_id;
-	u32 fll_id;
-	u32 pll_id;
+	int fll_id;
+	int pll_id;
 };
 
 /**
@@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* Specific configuration for PLL */
-	if (codec_priv->pll_id && codec_priv->fll_id) {
+	if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
 		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
 			pll_out = priv->sample_rate * 384;
 		else
@@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
 
 	priv->streams &= ~BIT(substream->stream);
 
-	if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
+	if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
 		/* Force freq to be free_freq to avoid error message in codec */
 		ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0),
 					     codec_priv->mclk_id,
@@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	priv->card.dapm_routes = audio_map;
 	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	priv->card.driver_name = DRIVER_NAME;
+
+	priv->codec_priv.fll_id = -1;
+	priv->codec_priv.pll_id = -1;
+
 	/* Diversify the card configurations */
 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
 		codec_dai_name = "cs42888";
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 1e4020fae05a..8a9a30dd31e2 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -710,10 +710,15 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir)
 {
 	unsigned int ofs = sai->soc_data->reg_offset;
 	bool tx = dir == TX;
-	u32 xcsr, count = 100;
+	u32 xcsr, count = 100, mask;
+
+	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output)
+		mask = FSL_SAI_CSR_TERE;
+	else
+		mask = FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE;
 
 	regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs),
-			   FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0);
+			   mask, 0);
 
 	/* TERE will remain set till the end of current frame */
 	do {
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index b2c12e4ed5bf..2aeb18397bcb 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -315,7 +315,7 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	if (IS_ERR(priv->cpu_mclk)) {
 		ret = PTR_ERR(priv->cpu_mclk);
 		dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret);
-		return -EINVAL;
+		return ret;
 	}
 
 	priv->audmix_pdev = audmix_pdev;
diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c
index e0c416a2eff8..a9324712e3fa 100644
--- a/sound/soc/fsl/imx-rpmsg.c
+++ b/sound/soc/fsl/imx-rpmsg.c
@@ -89,6 +89,14 @@ static int imx_rpmsg_probe(struct platform_device *pdev)
 			    SND_SOC_DAIFMT_NB_NF |
 			    SND_SOC_DAIFMT_CBC_CFC;
 
+	/*
+	 * i.MX rpmsg sound cards work on codec slave mode. MCLK will be
+	 * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK
+	 * present at power up/down sequence. So need to set ignore_pmdown_time
+	 * to power down codec immediately before MCLK is turned off.
+	 */
+	data->dai.ignore_pmdown_time = 1;
+
 	/* Optional codec node */
 	ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args);
 	if (ret) {