5 files changed, 25 insertions, 18 deletions

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 3f6959c..de43887 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	memcpy(priv->dai_link, fsl_asoc_card_dai,
 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
 
+	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+		goto asrc_fail;
+	}
+
 	/* Normal DAI Link */
 	priv->dai_link[0].cpu_of_node = cpu_np;
 	priv->dai_link[0].codec_of_node = codec_np;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 75870c0..91eb3ae 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1049,7 +1049,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 				enum spdif_txrate index, bool round)
 {
 	const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 };
-	bool is_sysclk = clk == spdif_priv->sysclk;
+	bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk);
 	u64 rate_ideal, rate_actual, sub;
 	u32 sysclk_dfmin, sysclk_dfmax;
 	u32 txclk_df, sysclk_df, arate;
@@ -1143,7 +1143,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			spdif_priv->txclk_src[index], rate[index]);
 	dev_dbg(&pdev->dev, "use txclk df %d for %dHz sample rate\n",
 			spdif_priv->txclk_df[index], rate[index]);
-	if (spdif_priv->txclk[index] == spdif_priv->sysclk)
+	if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk))
 		dev_dbg(&pdev->dev, "use sysclk df %d for %dHz sample rate\n",
 				spdif_priv->sysclk_df[index], rate[index]);
 	dev_dbg(&pdev->dev, "the best rate for %dHz sample rate is %dHz\n",
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2595611..6b0c8f7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,17 +603,20 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
 	factor = (div2 + 1) * (7 * psr + 1) * 2;
 
 	for (i = 0; i < 255; i++) {
-		/* The bclk rate must be smaller than 1/5 sysclk rate */
-		if (factor * (i + 1) < 5)
-			continue;
-
-		tmprate = freq * factor * (i + 2);
+		tmprate = freq * factor * (i + 1);
 
 		if (baudclk_is_used)
 			clkrate = clk_get_rate(ssi_private->baudclk);
 		else
 			clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
 
+		/*
+		 * Hardware limitation: The bclk rate must be
+		 * never greater than 1/5 IPG clock rate
+		 */
+		if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+			continue;
+
 		clkrate /= factor;
 		afreq = clkrate / (i + 1);
 
@@ -1224,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev,
 	ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0;
 	ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0;
 
-	ret = !of_property_read_u32_array(np, "dmas", dmas, 4);
+	ret = of_property_read_u32_array(np, "dmas", dmas, 4);
 	if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) {
 		ssi_private->use_dual_fifo = true;
 		/* When using dual fifo mode, we need to keep watermark
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index f8cf10e..20e7400 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
 
 	/* Headphone jack detection */
 	if (gpio_is_valid(data->jack_gpio)) {
-		ret = snd_soc_jack_new(rtd->codec, "Headphone",
-				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
-				       &headset_jack);
+		ret = snd_soc_card_jack_new(rtd->card, "Headphone",
+					    SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+					    &headset_jack, NULL, 0);
 		if (ret)
 			return ret;
 
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index a958937..0653aa8 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -205,16 +205,14 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Headphone jack detection */
-	snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
-	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
-			      hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
+			      &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
 	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
 
 	/* Microphone jack detection */
-	snd_soc_jack_new(codec, "Microphone",
-			 SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
-	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
-			      mic_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Microphone",
+			      SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
+			      mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
 	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
 			       SND_JACK_BTN_0);