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Merge remote-tracking branch 'asoc/topic/fsl' into tmp

Mark Brown 11 years ago
parent
commit
90d561bed9
5 changed files with 23 additions and 7 deletions
  1. 4 0
      include/sound/soc.h
  2. 1 1
      sound/soc/fsl/Makefile
  3. 6 6
      sound/soc/fsl/fsl_spdif.c
  4. 2 0
      sound/soc/mxs/mxs-sgtl5000.c
  5. 10 0
      sound/soc/soc-pcm.c

+ 4 - 0
include/sound/soc.h

@@ -930,6 +930,10 @@ struct snd_soc_dai_link {
 	/* machine stream operations */
 	const struct snd_soc_ops *ops;
 	const struct snd_soc_compr_ops *compr_ops;
+
+	/* For unidirectional dai links */
+	bool playback_only;
+	bool capture_only;
 };
 
 struct snd_soc_codec_conf {

+ 1 - 1
sound/soc/fsl/Makefile

@@ -45,7 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
-snd-soc-imx-spdif-objs :=imx-spdif.o
+snd-soc-imx-spdif-objs := imx-spdif.o
 snd-soc-imx-mc13783-objs := imx-mc13783.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o

+ 6 - 6
sound/soc/fsl/fsl_spdif.c

@@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-int fsl_spdif_startup(struct snd_pcm_substream *substream,
-			struct snd_soc_dai *cpu_dai)
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
@@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 	.startup = fsl_spdif_startup,
 	.hw_params = fsl_spdif_hw_params,
 	.trigger = fsl_spdif_trigger,
@@ -919,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-struct snd_soc_dai_driver fsl_spdif_dai = {
+static struct snd_soc_dai_driver fsl_spdif_dai = {
 	.probe = &fsl_spdif_dai_probe,
 	.playback = {
 		.channels_min = 2,
@@ -1071,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			break;
 	}
 
-	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
 			spdif_priv->txclk_src[index], rate[index]);
-	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
 			spdif_priv->txclk_div[index], rate[index]);
 
 	return 0;

+ 2 - 0
sound/soc/mxs/mxs-sgtl5000.c

@@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.stream_name	= "HiFi Playback",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.playback_only	= true,
 	}, {
 		.name		= "HiFi Rx",
 		.stream_name	= "HiFi Capture",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.capture_only	= true,
 	},
 };
 

+ 10 - 0
sound/soc/soc-pcm.c

@@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			capture = 1;
 	}
 
+	if (rtd->dai_link->playback_only) {
+		playback = 1;
+		capture = 0;
+	}
+
+	if (rtd->dai_link->capture_only) {
+		playback = 0;
+		capture = 1;
+	}
+
 	/* create the PCM */
 	if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",