mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-11-24 12:09:55 +00:00
97 lines
3.0 KiB
C
97 lines
3.0 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 Libav.
|
|
*
|
|
* Libav 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.
|
|
*
|
|
* Libav 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 Libav; 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 "libavformat/avformat.h"
|
|
#include "libavutil/log.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)
|
|
|
|
#define ALSA_BUFFER_SIZE_MAX 32768
|
|
|
|
typedef struct AlsaData {
|
|
AVClass *class;
|
|
snd_pcm_t *h;
|
|
int frame_size; ///< preferred size for reads and writes
|
|
int period_size; ///< bytes per sample * channels
|
|
int sample_rate; ///< sample rate set by user
|
|
int channels; ///< number of channels set by user
|
|
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 */
|