mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
7a72695c05
* commit '36ef5369ee9b336febc2c270f8718cec4476cb85': Replace all CODEC_ID_* with AV_CODEC_ID_* lavc: add AV prefix to codec ids. Conflicts: doc/APIchanges doc/examples/decoding_encoding.c doc/examples/muxing.c ffmpeg.c ffprobe.c ffserver.c libavcodec/8svx.c libavcodec/avcodec.h libavcodec/dnxhd_parser.c libavcodec/dvdsubdec.c libavcodec/error_resilience.c libavcodec/h263dec.c libavcodec/libvorbisenc.c libavcodec/mjpeg_parser.c libavcodec/mjpegenc.c libavcodec/mpeg12.c libavcodec/mpeg4videodec.c libavcodec/mpegvideo.c libavcodec/mpegvideo_enc.c libavcodec/pcm.c libavcodec/r210dec.c libavcodec/utils.c libavcodec/v210dec.c libavcodec/version.h libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/v4l2.c libavformat/asfdec.c libavformat/asfenc.c libavformat/avformat.h libavformat/avidec.c libavformat/caf.c libavformat/electronicarts.c libavformat/flacdec.c libavformat/flvdec.c libavformat/flvenc.c libavformat/framecrcenc.c libavformat/img2.c libavformat/img2dec.c libavformat/img2enc.c libavformat/ipmovie.c libavformat/isom.c libavformat/matroska.c libavformat/matroskadec.c libavformat/matroskaenc.c libavformat/mov.c libavformat/movenc.c libavformat/mp3dec.c libavformat/mpeg.c libavformat/mpegts.c libavformat/mxf.c libavformat/mxfdec.c libavformat/mxfenc.c libavformat/nsvdec.c libavformat/nut.c libavformat/oggenc.c libavformat/pmpdec.c libavformat/rawdec.c libavformat/rawenc.c libavformat/riff.c libavformat/sdp.c libavformat/utils.c libavformat/vocenc.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
102 lines
3.1 KiB
C
102 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(AV_CODEC_ID_PCM_S16BE, AV_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
|
|
int last_period;
|
|
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 AVCodecID or AV_CODEC_ID_NONE;
|
|
* out: actually selected AVCodecID, changed only if
|
|
* AV_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 AVCodecID *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 */
|