Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...

1 files changed, 37 insertions, 5 deletions

diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index a4a9fc7e8c76..c7e1c28528a4 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -25,7 +25,7 @@ static int speyside_set_bias_level(struct snd_soc_card *card,
 				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level)
 {
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai;
 	int ret;
 
 	if (dapm->dev != codec_dai->dev)
@@ -57,7 +57,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card,
 					struct snd_soc_dapm_context *dapm,
 					enum snd_soc_bias_level level)
 {
-	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai;
 	int ret;
 
 	if (dapm->dev != codec_dai->dev)
@@ -126,6 +126,18 @@ static void speyside_set_polarity(struct snd_soc_codec *codec,
 	snd_soc_dapm_sync(&codec->dapm);
 }
 
+static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *dai = rtd->codec_dai;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
 static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *dai = rtd->codec_dai;
@@ -172,17 +184,37 @@ static int speyside_late_probe(struct snd_soc_card *card)
 	return 0;
 }
 
+static const struct snd_soc_pcm_stream dsp_codec_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 48000,
+	.rate_max = 48000,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
 static struct snd_soc_dai_link speyside_dai[] = {
 	{
-		.name = "CPU",
-		.stream_name = "CPU",
+		.name = "CPU-DSP",
+		.stream_name = "CPU-DSP",
 		.cpu_dai_name = "samsung-i2s.0",
-		.codec_dai_name = "wm8996-aif1",
+		.codec_dai_name = "wm0010-sdi1",
 		.platform_name = "samsung-audio",
+		.codec_name = "spi0.0",
+		.init = speyside_wm0010_init,
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+	},
+	{
+		.name = "DSP-CODEC",
+		.stream_name = "DSP-CODEC",
+		.cpu_dai_name = "wm0010-sdi2",
+		.codec_dai_name = "wm8996-aif1",
 		.codec_name = "wm8996.1-001a",
 		.init = speyside_wm8996_init,
 		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
 				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dsp_codec_params,
+		.ignore_suspend = 1,
 	},
 	{
 		.name = "Baseband",