1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavdevice/alsa-audio.h
Michael Niedermayer aedc908601 Merge remote-tracking branch 'qatar/master'
* qatar/master: (35 commits)
  flvdec: Do not call parse_keyframes_index with a NULL stream
  libspeexdec: include system headers before local headers
  libspeexdec: return meaningful error codes
  libspeexdec: cosmetics: reindent
  libspeexdec: decode one frame at a time.
  swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
  Move timefilter code from lavf to lavd.
  mov: add support for hdvd and pgapmetadata atoms
  mov: rename function _stik, some indentation cosmetics
  mov: rename function _int8 to remove ambiguity, some indentation cosmetics
  mov: parse the gnre atom
  mp3on4: check for allocation failures in decode_init_mp3on4()
  mp3on4: create a separate flush function for MP3onMP4.
  mp3on4: ensure that the frame channel count does not exceed the codec channel count.
  mp3on4: set channel layout
  mp3on4: fix the output channel order
  mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
  mp3on4: copy MPADSPContext from first context to all contexts.
  fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
  fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
  ...

Conflicts:
	libavcodec/arm/h264dsp_init_arm.c
	libavcodec/h264.c
	libavcodec/h264.h
	libavcodec/h264_cabac.c
	libavcodec/h264_cavlc.c
	libavcodec/h264_ps.c
	libavcodec/h264dsp_template.c
	libavcodec/h264idct_template.c
	libavcodec/h264pred.c
	libavcodec/h264pred_template.c
	libavcodec/x86/h264dsp_mmx.c
	libavdevice/Makefile
	libavdevice/jack_audio.c
	libavformat/Makefile
	libavformat/flvdec.c
	libavformat/flvenc.c
	libavutil/pixfmt.h
	libswscale/utils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-22 01:16:41 +02:00

101 lines
3.1 KiB
C

/*
* ALSA input and output
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALSA input and output: definitions and structures
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
*/
#ifndef AVDEVICE_ALSA_AUDIO_H
#define AVDEVICE_ALSA_AUDIO_H
#include <alsa/asoundlib.h>
#include "config.h"
#include "libavutil/log.h"
#include "timefilter.h"
#include "avdevice.h"
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
typedef void (*ff_reorder_func)(const void *, void *, int);
#define ALSA_BUFFER_SIZE_MAX 65536
typedef struct {
AVClass *class;
snd_pcm_t *h;
int frame_size; ///< bytes per sample * channels
int period_size; ///< preferred size for reads and writes, in frames
int sample_rate; ///< sample rate set by user
int channels; ///< number of channels set by user
TimeFilter *timefilter;
void (*reorder_func)(const void *, void *, int);
void *reorder_buf;
int reorder_buf_size; ///< in frames
} AlsaData;
/**
* Open an ALSA PCM.
*
* @param s media file handle
* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
* @param sample_rate in: requested sample rate;
* out: actually selected sample rate
* @param channels number of channels
* @param codec_id in: requested CodecID or CODEC_ID_NONE;
* out: actually selected CodecID, changed only if
* CODEC_ID_NONE was requested
*
* @return 0 if OK, AVERROR_xxx on error
*/
int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum CodecID *codec_id);
/**
* Close the ALSA PCM.
*
* @param s1 media file handle
*
* @return 0
*/
int ff_alsa_close(AVFormatContext *s1);
/**
* Try to recover from ALSA buffer underrun.
*
* @param s1 media file handle
* @param err error code reported by the previous ALSA call
*
* @return 0 if OK, AVERROR_xxx on error
*/
int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
#endif /* AVDEVICE_ALSA_AUDIO_H */