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authorLinus Torvalds <torvalds@linux-foundation.org>2008-12-28 22:41:32 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2008-12-28 22:41:32 +0300
commitcb10ea549fdc0ab2dd8988adab5bf40b4fa642f3 (patch)
tree6bc11e0af9f0639a5eedd055401086c8c771f21e /sound/pci/ca0106/ca0106.h
parent81d6e59dabb1ae0c782e9eb7e3d88f699d25b314 (diff)
parent5ce442fe2c9423ec5451222aee6f9b2127bb8311 (diff)
downloadlinux-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.h30
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