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| author | Linus Torvalds <torvalds@g5.osdl.org> | 2006-03-22 10:59:20 -0800 |
|---|---|---|
| committer | Linus Torvalds <torvalds@g5.osdl.org> | 2006-03-22 10:59:20 -0800 |
| commit | 1c2e02750b992703a8a18634e08b04353face243 (patch) | |
| tree | 5dc2d10bad329eeb73b9e219e237662a8383f971 /arch | |
| parent | 8b4b6707ee32f929846d947d18b1b9bf42e988aa (diff) | |
| parent | a3c44854a59f7e983c867060aa906bbf5befb1ef (diff) | |
| download | op-kernel-dev-1c2e02750b992703a8a18634e08b04353face243.zip op-kernel-dev-1c2e02750b992703a8a18634e08b04353face243.tar.gz | |
Merge git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa
* git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa: (124 commits)
[ALSA] version 1.0.11rc4
[PATCH] Intruduce DMA_28BIT_MASK
[ALSA] hda-codec - Add support for ASUS P4GPL-X
[ALSA] hda-codec - Add support for HP nx9420 laptop
[ALSA] Fix memory leaks in error path of control.c
[ALSA] AMD Au1x00: AC'97 controller is memory mapped
[ALSA] AMD Au1x00: fix DMA init/cleanup
[ALSA] hda-codec - Fix generic auto-configurator
[ALSA] hda-codec - Fix BIOS auto-configuration
[ALSA] Fixes typos in Audiophile-USB.txt
[ALSA] ice1712 - typo fixes for dxr_enable module option
[ALSA] AMD Au1x00: make driver build after cleanup
[ALSA] ice1712 - Fix wrong value types for enum items
[ALSA] fix resource leak in usbmixer
[ALSA] Fix gus_pcm dereference before NULL
[ALSA] Fix seq_clientmgr dereferences before NULL check
[ALSA] hda-codec - Fix for Samsung R65 and ASUS A6J
[ALSA] hda-codec - Add support for VAIO FE550G and SZ110
[ALSA] usb-audio: add Maya44 mixer control names
[ALSA] usb-audio: add Casio PL-40R support
...
Diffstat (limited to 'arch')
| -rw-r--r-- | arch/arm/mach-pxa/mainstone.c | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/arch/arm/mach-pxa/mainstone.c b/arch/arm/mach-pxa/mainstone.c index d5bda60..98356f8 100644 --- a/arch/arm/mach-pxa/mainstone.c +++ b/arch/arm/mach-pxa/mainstone.c @@ -157,14 +157,14 @@ static struct platform_device smc91x_device = { .resource = smc91x_resources, }; -static int mst_audio_startup(snd_pcm_substream_t *substream, void *priv) +static int mst_audio_startup(struct snd_pcm_substream *substream, void *priv) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; return 0; } -static void mst_audio_shutdown(snd_pcm_substream_t *substream, void *priv) +static void mst_audio_shutdown(struct snd_pcm_substream *substream, void *priv) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; |
