alsa-audio.h 2.99 KB
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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 )
 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * 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.
 *
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 * Libav is distributed in the hope that it will be useful,
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 * 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
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
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 * @file
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 * 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

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#include <alsa/asoundlib.h>
#include "config.h"
#include "libavformat/avformat.h"
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#include "libavutil/log.h"
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/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
        other formats */
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#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
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#define ALSA_BUFFER_SIZE_MAX 32768

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typedef struct AlsaData {
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    AVClass *class;
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    snd_pcm_t *h;
    int frame_size;  ///< preferred size for reads and writes
    int period_size; ///< bytes per sample * channels
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    int sample_rate; ///< sample rate set by user
    int channels;    ///< number of channels set by user
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    void (*reorder_func)(const void *, void *, int);
    void *reorder_buf;
    int reorder_buf_size; ///< in frames
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} AlsaData;

/**
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 * Open an ALSA PCM.
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 *
 * @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
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 * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
 *                 out: actually selected AVCodecID, changed only if
 *                 AV_CODEC_ID_NONE was requested
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 *
 * @return 0 if OK, AVERROR_xxx on error
 */
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int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
                 unsigned int *sample_rate,
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                 int channels, enum AVCodecID *codec_id);
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/**
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 * Close the ALSA PCM.
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 *
 * @param s1 media file handle
 *
 * @return 0
 */
int ff_alsa_close(AVFormatContext *s1);

/**
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 * Try to recover from ALSA buffer underrun.
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 *
 * @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);

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int ff_alsa_extend_reorder_buf(AlsaData *s, int size);

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#endif /* AVDEVICE_ALSA_AUDIO_H */