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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 08:32:05 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 08:32:05 -0700 |
| commit | 33081adf8b89d5a716d7e1c60171768d39795b39 (patch) | |
| tree | 275de58bbbb5f7ddffcdc087844cfc7fbe4315be /sound/soc/s3c24xx/s3c24xx_simtec.c | |
| parent | c55960499f810357a29659b32d6ea594abee9237 (diff) | |
| parent | 506ecbca71d07fa327dd986be1682e90885678ee (diff) | |
| download | kernel_samsung_smdk4412-33081adf8b89d5a716d7e1c60171768d39795b39.tar.gz kernel_samsung_smdk4412-33081adf8b89d5a716d7e1c60171768d39795b39.tar.bz2 kernel_samsung_smdk4412-33081adf8b89d5a716d7e1c60171768d39795b39.zip | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
ALSA: hda - Disable sticky PCM stream assignment for AD codecs
ALSA: usb - Creative USB X-Fi volume knob support
ALSA: ca0106: Use card specific dac id for mute controls.
ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
ALSA: ca0106: Create a nice spot for mapping channels to dacs.
ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
ALSA: ca0106: Pull out dac powering routine into separate function.
ALSA: ca0106 - add Sound Blaster 5.1vx info.
ASoC: tlv320dac33: Use usleep_range for delays
ALSA: usb-audio: add Novation Launchpad support
ALSA: hda - Add workarounds for CT-IBG controllers
ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
ASoC: tpa6130a2: Error handling for broken chip
ASoC: max98088: Staticise m98088_eq_band
ASoC: soc-core: Fix codec->name memory leak
ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
ALSA: hda - Add some workarounds for Creative IBG
ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
ALSA: hda - Add alc_init_jacks() call to other codecs
...
Diffstat (limited to 'sound/soc/s3c24xx/s3c24xx_simtec.c')
| -rw-r--r-- | sound/soc/s3c24xx/s3c24xx_simtec.c | 15 |
1 files changed, 8 insertions, 7 deletions
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 4984754f329..c4c11144201 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -139,8 +139,10 @@ static const struct snd_kcontrol_new amp_unmute_controls[] = { speaker_unmute_get, speaker_unmute_put), }; -void simtec_audio_init(struct snd_soc_codec *codec) +void simtec_audio_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_codec *codec = rtd->codec; + if (pdata->amp_gpio > 0) { pr_debug("%s: adding amp routes\n", __func__); @@ -170,8 +172,8 @@ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; /* Set the CODEC as the bus clock master, I2S */ @@ -319,12 +321,12 @@ EXPORT_SYMBOL_GPL(simtec_audio_pmops); #endif int __devinit simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev) + struct snd_soc_card *card) { struct platform_device *snd_dev; int ret; - socdev->card->dai_link->ops = &simtec_snd_ops; + card->dai_link->ops = &simtec_snd_ops; pdata = pdev->dev.platform_data; if (!pdata) { @@ -353,8 +355,7 @@ int __devinit simtec_audio_core_probe(struct platform_device *pdev, goto err_gpio; } - platform_set_drvdata(snd_dev, socdev); - socdev->dev = &snd_dev->dev; + platform_set_drvdata(snd_dev, card); ret = platform_device_add(snd_dev); if (ret) { |