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/*
* 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 "libavformat/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);
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 */
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