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author | Mark Brown | 2012-10-25 17:36:02 +0100 |
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committer | Mark Brown | 2012-10-25 17:36:02 +0100 |
commit | 456ba5a7802e58eccb5aa9751b3ab515ef99b9ca (patch) | |
tree | 4ca4dd3726b34dead51af13b67475af7bf857893 /include/uapi/linux/dvb/audio.h | |
parent | 05304949332c6d2c7b50f2d0f666a52369f09ced (diff) | |
parent | 79748cdb39dbf914bc5f26c75cfd5f91d84d82c9 (diff) |
Merge remote-tracking branches 'asoc/fix/ux500' and 'asoc/fix/wm8994' into for-3.7
Diffstat (limited to 'include/uapi/linux/dvb/audio.h')
-rw-r--r-- | include/uapi/linux/dvb/audio.h | 135 |
1 files changed, 135 insertions, 0 deletions
diff --git a/include/uapi/linux/dvb/audio.h b/include/uapi/linux/dvb/audio.h new file mode 100644 index 000000000000..d47bccd604e4 --- /dev/null +++ b/include/uapi/linux/dvb/audio.h @@ -0,0 +1,135 @@ +/* + * audio.h + * + * Copyright (C) 2000 Ralph Metzler <ralph@convergence.de> + * & Marcus Metzler <marcus@convergence.de> + * for convergence integrated media GmbH + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Lesser Public License + * as published by the Free Software Foundation; either version 2.1 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + * + */ + +#ifndef _DVBAUDIO_H_ +#define _DVBAUDIO_H_ + +#include <linux/types.h> + +typedef enum { + AUDIO_SOURCE_DEMUX, /* Select the demux as the main source */ + AUDIO_SOURCE_MEMORY /* Select internal memory as the main source */ +} audio_stream_source_t; + + +typedef enum { + AUDIO_STOPPED, /* Device is stopped */ + AUDIO_PLAYING, /* Device is currently playing */ + AUDIO_PAUSED /* Device is paused */ +} audio_play_state_t; + + +typedef enum { + AUDIO_STEREO, + AUDIO_MONO_LEFT, + AUDIO_MONO_RIGHT, + AUDIO_MONO, + AUDIO_STEREO_SWAPPED +} audio_channel_select_t; + + +typedef struct audio_mixer { + unsigned int volume_left; + unsigned int volume_right; + // what else do we need? bass, pass-through, ... +} audio_mixer_t; + + +typedef struct audio_status { + int AV_sync_state; /* sync audio and video? */ + int mute_state; /* audio is muted */ + audio_play_state_t play_state; /* current playback state */ + audio_stream_source_t stream_source; /* current stream source */ + audio_channel_select_t channel_select; /* currently selected channel */ + int bypass_mode; /* pass on audio data to */ + audio_mixer_t mixer_state; /* current mixer state */ +} audio_status_t; /* separate decoder hardware */ + + +typedef +struct audio_karaoke { /* if Vocal1 or Vocal2 are non-zero, they get mixed */ + int vocal1; /* into left and right t at 70% each */ + int vocal2; /* if both, Vocal1 and Vocal2 are non-zero, Vocal1 gets*/ + int melody; /* mixed into the left channel and */ + /* Vocal2 into the right channel at 100% each. */ + /* if Melody is non-zero, the melody channel gets mixed*/ +} audio_karaoke_t; /* into left and right */ + + +typedef __u16 audio_attributes_t; +/* bits: descr. */ +/* 15-13 audio coding mode (0=ac3, 2=mpeg1, 3=mpeg2ext, 4=LPCM, 6=DTS, */ +/* 12 multichannel extension */ +/* 11-10 audio type (0=not spec, 1=language included) */ +/* 9- 8 audio application mode (0=not spec, 1=karaoke, 2=surround) */ +/* 7- 6 Quantization / DRC (mpeg audio: 1=DRC exists)(lpcm: 0=16bit, */ +/* 5- 4 Sample frequency fs (0=48kHz, 1=96kHz) */ +/* 2- 0 number of audio channels (n+1 channels) */ + + +/* for GET_CAPABILITIES and SET_FORMAT, the latter should only set one bit */ +#define AUDIO_CAP_DTS 1 +#define AUDIO_CAP_LPCM 2 +#define AUDIO_CAP_MP1 4 +#define AUDIO_CAP_MP2 8 +#define AUDIO_CAP_MP3 16 +#define AUDIO_CAP_AAC 32 +#define AUDIO_CAP_OGG 64 +#define AUDIO_CAP_SDDS 128 +#define AUDIO_CAP_AC3 256 + +#define AUDIO_STOP _IO('o', 1) +#define AUDIO_PLAY _IO('o', 2) +#define AUDIO_PAUSE _IO('o', 3) +#define AUDIO_CONTINUE _IO('o', 4) +#define AUDIO_SELECT_SOURCE _IO('o', 5) +#define AUDIO_SET_MUTE _IO('o', 6) +#define AUDIO_SET_AV_SYNC _IO('o', 7) +#define AUDIO_SET_BYPASS_MODE _IO('o', 8) +#define AUDIO_CHANNEL_SELECT _IO('o', 9) +#define AUDIO_GET_STATUS _IOR('o', 10, audio_status_t) + +#define AUDIO_GET_CAPABILITIES _IOR('o', 11, unsigned int) +#define AUDIO_CLEAR_BUFFER _IO('o', 12) +#define AUDIO_SET_ID _IO('o', 13) +#define AUDIO_SET_MIXER _IOW('o', 14, audio_mixer_t) +#define AUDIO_SET_STREAMTYPE _IO('o', 15) +#define AUDIO_SET_EXT_ID _IO('o', 16) +#define AUDIO_SET_ATTRIBUTES _IOW('o', 17, audio_attributes_t) +#define AUDIO_SET_KARAOKE _IOW('o', 18, audio_karaoke_t) + +/** + * AUDIO_GET_PTS + * + * Read the 33 bit presentation time stamp as defined + * in ITU T-REC-H.222.0 / ISO/IEC 13818-1. + * + * The PTS should belong to the currently played + * frame if possible, but may also be a value close to it + * like the PTS of the last decoded frame or the last PTS + * extracted by the PES parser. + */ +#define AUDIO_GET_PTS _IOR('o', 19, __u64) +#define AUDIO_BILINGUAL_CHANNEL_SELECT _IO('o', 20) + +#endif /* _DVBAUDIO_H_ */ |