diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-10-09 02:07:14 +0400 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-10-09 02:07:14 +0400 |
commit | f5a246eab9a268f51ba8189ea5b098a1bfff200e (patch) | |
tree | a6ff7169e0bcaca498d9aec8b0624de1b74eaecb /sound/soc/fsl/mpc5200_psc_ac97.c | |
parent | d5bbd43d5f450c3fca058f5b85f3dfb4e8cc88c9 (diff) | |
parent | 7ff34ad80b7080fafaac8efa9ef0061708eddd51 (diff) | |
download | linux-f5a246eab9a268f51ba8189ea5b098a1bfff200e.tar.xz |
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
...
Diffstat (limited to 'sound/soc/fsl/mpc5200_psc_ac97.c')
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_ac97.c | 10 |
1 files changed, 10 insertions, 0 deletions
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index ffa00a2eb770..a313c0ae36db 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -237,15 +237,18 @@ static const struct snd_soc_dai_ops psc_ac97_digital_ops = { static struct snd_soc_dai_driver psc_ac97_dai[] = { { + .name = "mpc5200-psc-ac97.0", .ac97_control = 1, .probe = psc_ac97_probe, .playback = { + .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 6, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { + .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, @@ -254,8 +257,10 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { .ops = &psc_ac97_analog_ops, }, { + .name = "mpc5200-psc-ac97.1", .ac97_control = 1, .playback = { + .stream_name = "AC97 SPDIF", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ @@ -278,6 +283,10 @@ static int __devinit psc_ac97_of_probe(struct platform_device *op) struct snd_ac97 ac97; struct mpc52xx_psc __iomem *regs; + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + rc = snd_soc_register_dais(&op->dev, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { dev_err(&op->dev, "Failed to register DAI\n"); @@ -303,6 +312,7 @@ static int __devinit psc_ac97_of_probe(struct platform_device *op) static int __devexit psc_ac97_of_remove(struct platform_device *op) { + mpc5200_audio_dma_destroy(op); snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_ac97_dai)); return 0; } |