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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 "libavformat/avformat.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 struct {
- snd_pcm_t *h;
- int frame_size; ///< preferred size for reads and writes
- int period_size; ///< bytes per sample * channels
- } 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);
- #endif /* AVDEVICE_ALSA_AUDIO_H */
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