doc-rst: auto-generate audio.h.rst
This is an auto-generated header. Remove the hardcoded one and do the right thing here. NOTE: this is a deprecated API. So, we won't make any effort to try identifying the meaning of this obscure API that is used only on a legacy driver. Signed-off-by: Mauro Carvalho Chehab <mchehab@s-opensource.com>
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PARSER = ../sphinx/parse-headers.pl
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UAPI = ../../include/uapi/linux
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htmldocs: frontend.h.rst dmx.h.rst
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htmldocs: audio.h.rst dmx.h.rst frontend.h.rst
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frontend.h.rst: ${PARSER} ${UAPI}/dvb/frontend.h frontend.h.rst.exceptions
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${PARSER} ${UAPI}/dvb/frontend.h $@ frontend.h.rst.exceptions
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audio.h.rst: ${PARSER} ${UAPI}/dvb/audio.h audio.h.rst.exceptions
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${PARSER} ${UAPI}/dvb/audio.h $@ audio.h.rst.exceptions
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dmx.h.rst: ${PARSER} ${UAPI}/dvb/dmx.h dmx.h.rst.exceptions
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${PARSER} ${UAPI}/dvb/dmx.h $@ dmx.h.rst.exceptions
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frontend.h.rst: ${PARSER} ${UAPI}/dvb/frontend.h frontend.h.rst.exceptions
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${PARSER} ${UAPI}/dvb/frontend.h $@ frontend.h.rst.exceptions
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clean:
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-rm frontend.h.rst
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@ -1,142 +0,0 @@
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.. -*- coding: utf-8; mode: rst -*-
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file: audio.h
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=============
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.. code-block:: c
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/*
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* audio.h
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*
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* Copyright (C) 2000 Ralph Metzler <ralph@convergence.de>
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* & Marcus Metzler <marcus@convergence.de>
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* for convergence integrated media GmbH
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Lesser Public License
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* as published by the Free Software Foundation; either version 2.1
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* of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*
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*/
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#ifndef _DVBAUDIO_H_
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#define _DVBAUDIO_H_
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#include <linux/types.h>
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typedef enum {
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AUDIO_SOURCE_DEMUX, /* Select the demux as the main source */
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AUDIO_SOURCE_MEMORY /* Select internal memory as the main source */
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} audio_stream_source_t;
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typedef enum {
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AUDIO_STOPPED, /* Device is stopped */
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AUDIO_PLAYING, /* Device is currently playing */
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AUDIO_PAUSED /* Device is paused */
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} audio_play_state_t;
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typedef enum {
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AUDIO_STEREO,
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AUDIO_MONO_LEFT,
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AUDIO_MONO_RIGHT,
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AUDIO_MONO,
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AUDIO_STEREO_SWAPPED
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} audio_channel_select_t;
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typedef struct audio_mixer {
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unsigned int volume_left;
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unsigned int volume_right;
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// what else do we need? bass, pass-through, ...
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} audio_mixer_t;
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typedef struct audio_status {
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int AV_sync_state; /* sync audio and video? */
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int mute_state; /* audio is muted */
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audio_play_state_t play_state; /* current playback state */
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audio_stream_source_t stream_source; /* current stream source */
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audio_channel_select_t channel_select; /* currently selected channel */
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int bypass_mode; /* pass on audio data to */
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audio_mixer_t mixer_state; /* current mixer state */
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} audio_status_t; /* separate decoder hardware */
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typedef
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struct audio_karaoke { /* if Vocal1 or Vocal2 are non-zero, they get mixed */
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int vocal1; /* into left and right t at 70% each */
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int vocal2; /* if both, Vocal1 and Vocal2 are non-zero, Vocal1 gets*/
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int melody; /* mixed into the left channel and */
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/* Vocal2 into the right channel at 100% each. */
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/* if Melody is non-zero, the melody channel gets mixed*/
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} audio_karaoke_t; /* into left and right */
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typedef __u16 audio_attributes_t;
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/* bits: descr. */
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/* 15-13 audio coding mode (0=ac3, 2=mpeg1, 3=mpeg2ext, 4=LPCM, 6=DTS, */
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/* 12 multichannel extension */
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/* 11-10 audio type (0=not spec, 1=language included) */
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/* 9- 8 audio application mode (0=not spec, 1=karaoke, 2=surround) */
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/* 7- 6 Quantization / DRC (mpeg audio: 1=DRC exists)(lpcm: 0=16bit, */
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/* 5- 4 Sample frequency fs (0=48kHz, 1=96kHz) */
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/* 2- 0 number of audio channels (n+1 channels) */
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/* for GET_CAPABILITIES and SET_FORMAT, the latter should only set one bit */
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#define AUDIO_CAP_DTS 1
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#define AUDIO_CAP_LPCM 2
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#define AUDIO_CAP_MP1 4
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#define AUDIO_CAP_MP2 8
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#define AUDIO_CAP_MP3 16
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#define AUDIO_CAP_AAC 32
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#define AUDIO_CAP_OGG 64
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#define AUDIO_CAP_SDDS 128
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#define AUDIO_CAP_AC3 256
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#define AUDIO_STOP _IO('o', 1)
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#define AUDIO_PLAY _IO('o', 2)
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#define AUDIO_PAUSE _IO('o', 3)
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#define AUDIO_CONTINUE _IO('o', 4)
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#define AUDIO_SELECT_SOURCE _IO('o', 5)
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#define AUDIO_SET_MUTE _IO('o', 6)
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#define AUDIO_SET_AV_SYNC _IO('o', 7)
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#define AUDIO_SET_BYPASS_MODE _IO('o', 8)
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#define AUDIO_CHANNEL_SELECT _IO('o', 9)
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#define AUDIO_GET_STATUS _IOR('o', 10, audio_status_t)
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#define AUDIO_GET_CAPABILITIES _IOR('o', 11, unsigned int)
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#define AUDIO_CLEAR_BUFFER _IO('o', 12)
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#define AUDIO_SET_ID _IO('o', 13)
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#define AUDIO_SET_MIXER _IOW('o', 14, audio_mixer_t)
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#define AUDIO_SET_STREAMTYPE _IO('o', 15)
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#define AUDIO_SET_EXT_ID _IO('o', 16)
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#define AUDIO_SET_ATTRIBUTES _IOW('o', 17, audio_attributes_t)
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#define AUDIO_SET_KARAOKE _IOW('o', 18, audio_karaoke_t)
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/**
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* AUDIO_GET_PTS
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*
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* Read the 33 bit presentation time stamp as defined
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* in ITU T-REC-H.222.0 / ISO/IEC 13818-1.
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*
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* The PTS should belong to the currently played
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* frame if possible, but may also be a value close to it
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* like the PTS of the last decoded frame or the last PTS
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* extracted by the PES parser.
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*/
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#define AUDIO_GET_PTS _IOR('o', 19, __u64)
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#define AUDIO_BILINGUAL_CHANNEL_SELECT _IO('o', 20)
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#endif /* _DVBAUDIO_H_ */
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@ -0,0 +1,20 @@
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# Ignore header name
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ignore define _DVBAUDIO_H_
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# Typedef pointing to structs
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replace typedef audio_karaoke_t audio-karaoke
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# Undocumented audio caps, as this is a deprecated API anyway
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ignore define AUDIO_CAP_DTS
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ignore define AUDIO_CAP_LPCM
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ignore define AUDIO_CAP_MP1
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ignore define AUDIO_CAP_MP2
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ignore define AUDIO_CAP_MP3
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ignore define AUDIO_CAP_AAC
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ignore define AUDIO_CAP_OGG
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ignore define AUDIO_CAP_SDDS
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ignore define AUDIO_CAP_AC3
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# some typedefs should point to struct/enums
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replace typedef audio_mixer_t audio-mixer
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replace typedef audio_status_t audio-status
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