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author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-08-07 17:07:31 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-08-07 17:07:31 -0700 |
commit | faa38b5e0e092914764cdba9f83d31a3f794d182 (patch) | |
tree | b3e5921bdc36378033b4910eb4f29cb0dfc486e0 /include/sound | |
parent | 78417334b5cb6e1f915b8fdcc4fce3f1a1b4420c (diff) | |
parent | 74bf40f0793fed9e01eb6164c2ce63e8c27ca205 (diff) | |
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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: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
Diffstat (limited to 'include/sound')
-rw-r--r-- | include/sound/asound.h | 6 | ||||
-rw-r--r-- | include/sound/pcm.h | 6 | ||||
-rw-r--r-- | include/sound/sh_fsi.h | 49 | ||||
-rw-r--r-- | include/sound/soc-dapm.h | 2 | ||||
-rw-r--r-- | include/sound/soc.h | 21 | ||||
-rw-r--r-- | include/sound/tlv320dac33-plat.h | 2 | ||||
-rw-r--r-- | include/sound/uda134x.h | 12 |
7 files changed, 90 insertions, 8 deletions
diff --git a/include/sound/asound.h b/include/sound/asound.h index 9f1eecf..a1803ec 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -212,7 +212,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43) /* in three bytes */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE +#define SNDRV_PCM_FORMAT_G723_24 ((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */ +#define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ +#define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6e3a297..85f1c6b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -174,6 +174,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) #define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) #define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) +#define SNDRV_PCM_FMTBIT_G723_40 (1ULL << SNDRV_PCM_FORMAT_G723_40) +#define SNDRV_PCM_FMTBIT_G723_40_1B (1ULL << SNDRV_PCM_FORMAT_G723_40_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE @@ -313,7 +317,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int twake: 1; /* do transfer (!poll) wakeup */ + snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index c022736..9d51d6f 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -12,6 +12,9 @@ * published by the Free Software Foundation. */ +#define FSI_PORT_A 0 +#define FSI_PORT_B 1 + /* flags format * 0xABCDEEFF @@ -55,12 +58,14 @@ #define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) #define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) -#define SH_FSI_FMT_MONO (1 << 0) -#define SH_FSI_FMT_MONO_DELAY (1 << 1) -#define SH_FSI_FMT_PCM (1 << 2) -#define SH_FSI_FMT_I2S (1 << 3) -#define SH_FSI_FMT_TDM (1 << 4) -#define SH_FSI_FMT_TDM_DELAY (1 << 5) +#define SH_FSI_FMT_MONO 0 +#define SH_FSI_FMT_MONO_DELAY 1 +#define SH_FSI_FMT_PCM 2 +#define SH_FSI_FMT_I2S 3 +#define SH_FSI_FMT_TDM 4 +#define SH_FSI_FMT_TDM_DELAY 5 +#define SH_FSI_FMT_SPDIF 6 + #define SH_FSI_IFMT_TDM_CH(x) \ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) @@ -72,9 +77,41 @@ #define SH_FSI_OFMT_TDM_DELAY_CH(x) \ (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +/* + * set_rate return value + * + * see ACKMD/BPFMD on + * ACK_MD (FSI2) + * CKG1 (FSI) + * + * err: return value < 0 + * + * 0x-00000AB + * + * A: ACKMD value + * B: BPFMD value + */ + +#define SH_FSI_ACKMD_MASK (0xF << 0) +#define SH_FSI_ACKMD_512 (1 << 0) +#define SH_FSI_ACKMD_256 (2 << 0) +#define SH_FSI_ACKMD_128 (3 << 0) +#define SH_FSI_ACKMD_64 (4 << 0) +#define SH_FSI_ACKMD_32 (5 << 0) + +#define SH_FSI_BPFMD_MASK (0xF << 4) +#define SH_FSI_BPFMD_512 (1 << 4) +#define SH_FSI_BPFMD_256 (2 << 4) +#define SH_FSI_BPFMD_128 (3 << 4) +#define SH_FSI_BPFMD_64 (4 << 4) +#define SH_FSI_BPFMD_32 (5 << 4) +#define SH_FSI_BPFMD_16 (6 << 4) + struct sh_fsi_platform_info { unsigned long porta_flags; unsigned long portb_flags; + int (*set_rate)(int is_porta, int rate); /* for master mode */ }; extern struct snd_soc_dai fsi_soc_dai[2]; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 66ff4c1..c5d9987 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -273,6 +273,8 @@ #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ +#define SND_SOC_DAPM_PRE_POST_PMD \ + (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) /* convenience event type detection */ #define SND_SOC_DAPM_EVENT_ON(e) \ diff --git a/include/sound/soc.h b/include/sound/soc.h index 697e7ff..65e9d03 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -170,6 +170,21 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\ + xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r_sx, \ + .get = snd_soc_get_volsw_2r_sx, \ + .put = snd_soc_put_volsw_2r_sx, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg_left, \ + .rreg = xreg_right, .shift = xshift, \ + .min = xmin, .max = xmax} } + + /* * Simplified versions of above macros, declaring a struct and calculating * ARRAY_SIZE internally @@ -329,6 +344,12 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); +int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h index 3f428d5..6c66496 100644 --- a/include/sound/tlv320dac33-plat.h +++ b/include/sound/tlv320dac33-plat.h @@ -15,6 +15,8 @@ struct tlv320dac33_platform_data { int power_gpio; + int mode1_latency; /* latency caused by the i2c writes in us */ + int auto_fifo_config; /* FIFO config based on the period size */ int keep_bclk; /* Keep the BCLK running in FIFO modes */ u8 burst_bclkdiv; }; diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 509efb0..e475659 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,6 +18,18 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; + /* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you unset is_powered_on_standby the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + set is_powered_on_standby, you will have slightly higher power + consumption. Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 |