Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
  ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
  ALSA: hda - Keep EAPD turned on for old Conexant chips
  ALSA: hda/realtek - Fix missing volume controls with ALC260
  ASoC: wm8940: Properly set codec->dapm.bias_level
  ALSA: hda - Fix pin-config for ASUS W90V
  ALSA: hda - Fix surround/CLFE headphone and speaker pins order
  ALSA: hda - Fix typo
  ALSA: Update the sound git tree URL
  ALSA: HDA: Add new revision for ALC662
  ASoC: max98095: Convert codec->hw_write to snd_soc_write
  ASoC: keep pointer to resource so it can be freed
  ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
  ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
  ASoC: da7210: Add support for line out and DAC
  ASoC: da7210: Add support for DAPM
  ALSA: hda/realtek - Fix DAC assignments of multiple speakers
  ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
  ASoC: Set sgtl5000->ldo in ldo_regulator_register
  ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
  ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
  ...

1 files changed, 7 insertions, 43 deletions

diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 7e75e775fb4..db91ccaf6c9 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -55,29 +55,8 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int err;
 
-	/* Set codec DAI configuration */
-	err = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (err < 0) {
-		printk(KERN_ERR "can't set codec DAI configuration\n");
-		return err;
-	}
-
-	/* Set cpu DAI configuration */
-	err = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_NB_NF |
-				  SND_SOC_DAIFMT_CBM_CFM);
-	if (err < 0) {
-		printk(KERN_ERR "can't set cpu DAI configuration\n");
-		return err;
-	}
-
 	/* Set the codec system clock for DAC and ADC */
 	err =
 	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
@@ -112,27 +91,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"MICIN", NULL, "Mic Jack"},
 };
 
-static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	/* Add osk5912 specific widgets */
-	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
-				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
-
-	/* Set up osk5912 specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
-	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
-	snd_soc_dapm_enable_pin(dapm, "Line In");
-	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
-
-	snd_soc_dapm_sync(dapm);
-
-	return 0;
-}
-
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link osk_dai = {
 	.name = "TLV320AIC23",
@@ -141,7 +99,8 @@ static struct snd_soc_dai_link osk_dai = {
 	.codec_dai_name = "tlv320aic23-hifi",
 	.platform_name = "omap-pcm-audio",
 	.codec_name = "tlv320aic23-codec",
-	.init = osk_tlv320aic23_init,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
 	.ops = &osk_ops,
 };
 
@@ -150,6 +109,11 @@ static struct snd_soc_card snd_soc_card_osk = {
 	.name = "OSK5912",
 	.dai_link = &osk_dai,
 	.num_links = 1,
+
+	.dapm_widgets = tlv320aic23_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };
 
 static struct platform_device *osk_snd_device;