diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 08:32:05 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 08:32:05 -0700 |
commit | 33081adf8b89d5a716d7e1c60171768d39795b39 (patch) | |
tree | 275de58bbbb5f7ddffcdc087844cfc7fbe4315be /sound/soc/atmel/snd-soc-afeb9260.c | |
parent | c55960499f810357a29659b32d6ea594abee9237 (diff) | |
parent | 506ecbca71d07fa327dd986be1682e90885678ee (diff) |
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/atmel/snd-soc-afeb9260.c')
-rw-r--r-- | sound/soc/atmel/snd-soc-afeb9260.c | 35 |
1 files changed, 9 insertions, 26 deletions
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 23349de2731..e3d283561c1 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -46,8 +46,8 @@ static int afeb9260_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 err; /* Set codec DAI configuration */ @@ -102,8 +102,9 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec) +static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_codec *codec = rtd->codec; /* Add afeb9260 specific widgets */ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, @@ -125,8 +126,10 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec) static struct snd_soc_dai_link afeb9260_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", - .cpu_dai = &atmel_ssc_dai[0], - .codec_dai = &tlv320aic23_dai, + .cpu_dai_name = "atmel-ssc-dai.0", + .codec_dai_name = "tlv320aic23-hifi", + .platform_name = "atmel_pcm-audio", + .codec_name = "tlv320aic23-codec.0-0x1a", .init = afeb9260_tlv320aic23_init, .ops = &afeb9260_ops, }; @@ -134,37 +137,20 @@ static struct snd_soc_dai_link afeb9260_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_machine_afeb9260 = { .name = "AFEB9260", - .platform = &atmel_soc_platform, .dai_link = &afeb9260_dai, .num_links = 1, }; -/* Audio subsystem */ -static struct snd_soc_device afeb9260_snd_devdata = { - .card = &snd_soc_machine_afeb9260, - .codec_dev = &soc_codec_dev_tlv320aic23, -}; - static struct platform_device *afeb9260_snd_device; static int __init afeb9260_soc_init(void) { int err; struct device *dev; - struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data; - struct ssc_device *ssc = NULL; if (!(machine_is_afeb9260())) return -ENODEV; - ssc = ssc_request(0); - if (IS_ERR(ssc)) { - printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); - err = PTR_ERR(ssc); - ssc = NULL; - goto err_ssc; - } - ssc_p->ssc = ssc; afeb9260_snd_device = platform_device_alloc("soc-audio", -1); if (!afeb9260_snd_device) { @@ -172,8 +158,7 @@ static int __init afeb9260_soc_init(void) return -ENOMEM; } - platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata); - afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev; + platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260); err = platform_device_add(afeb9260_snd_device); if (err) goto err1; @@ -184,9 +169,7 @@ static int __init afeb9260_soc_init(void) err1: platform_device_del(afeb9260_snd_device); platform_device_put(afeb9260_snd_device); -err_ssc: return err; - } static void __exit afeb9260_soc_exit(void) |