From 8a309d71ed9d17ff251b4b891fcef1c72bf625d1 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 30 Aug 2013 17:38:08 +0800 Subject: ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c Add wrapping '\n' for dev_dbg() in fsl_spdif.c Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0dfb0d9..98741e96f77a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -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; -- cgit v1.2.3 From 8626bdf05e93ae9d199cb2ad77b58832b98141f5 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 30 Aug 2013 17:39:00 +0800 Subject: ASoC: fsl: Add one blank space after ':=' in Makefile There is a blank space missing between ':=' and 'imx-spdif.o', thus add it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index e2aaff717f8a..8db705b0fdf9 100644 --- a/sound/soc/fsl/Makefile +++ b/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 -- cgit v1.2.3 From 6b4c80f947df9d92b97eb62afc29dda6d7220c7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 16:40:51 +0100 Subject: ASoC: fsl_spdif: Staticse non-exported symbols Signed-off-by: Mark Brown Acked-by: Nicolin Chen --- sound/soc/fsl/fsl_spdif.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 98741e96f77a..3920c3e849ce 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/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, -- cgit v1.2.3 From d6bead020d8f8bcaca5cdcb035250c44b21c93e7 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 29 Aug 2013 10:32:13 -0300 Subject: ASoC: soc-pcm: Allow to specify unidirectional dai_link Add 'playback_only' and 'capture_only' fields that can be used for specifying that a dai_link has a unidirectional capability. The motivation for this is for the cases of systems, such as Freescale MX28, that has two unidirectional DAIs. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/soc-pcm.c | 10 ++++++++++ 2 files changed, 14 insertions(+) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index c0ac3bc7b7f4..65414e82c762 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -936,6 +936,10 @@ 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 { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c640332a17..9abaa52efb2e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2027,6 +2027,16 @@ 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)", -- cgit v1.2.3 From a90e6053baa61feed8b19a9f4cbec6b56479d1ba Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 29 Aug 2013 10:32:14 -0300 Subject: ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional On a mx28 board, running "aplay -l" and "arecord -l" results in the following: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ,which is not correct because we got a capture device listed in aplay and a playback device listed in arecord. On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Allow to specify a dai link as 'playback_only' or 'capture_only', which suits well for this case. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Also tested playback and capture on the mx28evk board. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 1b134d72f120..ed8a51945306 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -91,11 +91,13 @@ 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, }, }; -- cgit v1.2.3