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author | Linus Torvalds <torvalds@linux-foundation.org> | 2008-12-28 22:41:32 +0300 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2008-12-28 22:41:32 +0300 |
commit | cb10ea549fdc0ab2dd8988adab5bf40b4fa642f3 (patch) | |
tree | 6bc11e0af9f0639a5eedd055401086c8c771f21e /sound/pci/ca0106/ca0106.h | |
parent | 81d6e59dabb1ae0c782e9eb7e3d88f699d25b314 (diff) | |
parent | 5ce442fe2c9423ec5451222aee6f9b2127bb8311 (diff) | |
download | linux-cb10ea549fdc0ab2dd8988adab5bf40b4fa642f3.tar.xz |
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: (367 commits)
ALSA: ASoC: fix a typo in omp-pcm.c
ASoC: Fix DSP formats in SSM2602 audio codec
ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers
ALSA: hda: fix incorrect mixer index values for 92hd83xx
ALSA: hda: dinput_mux check
ALSA: hda - Add quirk for another HP dv7
ALSA: ASoC - Add missing __devexit annotation to wm8350.c
ALSA: ASoc: DaVinci: davinci-evm use dsp_b mode
ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_dai
ALSA: ASoC: tlv320aic3x add dsp_a
ALSA: ASoC: DaVinci: document I2S limitations
ALSA: ASoC: DaVinci: davinci-i2s clean up
ALSA: ASoC: DaVinci: davinci-i2s clean up
ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarity
ALSA: ASoC: DaVinci: davinvi-evm, make requests explicit
ALSA: ca0106 - disable 44.1kHz capture
ALSA: ca0106 - Add missing card->private_data initialization
ALSA: ca0106 - Check ac97 availability at PM
ALSA: hda - Power up always when no jack detection is available
ALSA: hda - Fix unused variable warnings in patch_sigmatel.c
...
Diffstat (limited to 'sound/pci/ca0106/ca0106.h')
-rw-r--r-- | sound/pci/ca0106/ca0106.h | 30 |
1 files changed, 24 insertions, 6 deletions
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 74175fc80c7f..14b8d9a91aae 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -664,10 +664,14 @@ struct snd_ca0106_pcm { struct snd_ca0106_details { u32 serial; char * name; - int ac97; - int gpio_type; - int i2c_adc; - int spi_dac; + int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. + ac97 = 1 -> Default to AC97 in. */ + int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in + gpio_type = 2 -> shared side-out/line-in. */ + int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume + controls, phone, mic, line-in and aux. */ + int spi_dac; /* spi_dac=1 adds the mute switch for each analog + output, front, rear, etc. */ }; // definition of the chip-specific record @@ -686,11 +690,12 @@ struct snd_ca0106 { spinlock_t emu_lock; struct snd_ac97 *ac97; - struct snd_pcm *pcm; + struct snd_pcm *pcm[4]; struct snd_ca0106_channel playback_channels[4]; struct snd_ca0106_channel capture_channels[4]; - u32 spdif_bits[4]; /* s/pdif out setup */ + u32 spdif_bits[4]; /* s/pdif out default setup */ + u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */ int spdif_enable; int capture_source; int i2c_capture_source; @@ -703,6 +708,11 @@ struct snd_ca0106 { struct snd_ca_midi midi2; u16 spi_dac_reg[16]; + +#ifdef CONFIG_PM +#define NUM_SAVED_VOLUMES 9 + unsigned int saved_vol[NUM_SAVED_VOLUMES]; +#endif }; int snd_ca0106_mixer(struct snd_ca0106 *emu); @@ -721,3 +731,11 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); int snd_ca0106_spi_write(struct snd_ca0106 * emu, unsigned int data); + +#ifdef CONFIG_PM +void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip); +void snd_ca0106_mixer_resume(struct snd_ca0106 *chip); +#else +#define snd_ca0106_mixer_suspend(chip) do { } while (0) +#define snd_ca0106_mixer_resume(chip) do { } while (0) +#endif |