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96 lines
3.0 KiB
96 lines
3.0 KiB
/* |
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* ALSA input and output |
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) |
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) |
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* |
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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 |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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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 |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* 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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*/ |
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/** |
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* @file |
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* ALSA input and output: definitions and structures |
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* @author Luca Abeni ( lucabe72 email it ) |
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* @author Benoit Fouet ( benoit fouet free fr ) |
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*/ |
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#ifndef AVDEVICE_ALSA_AUDIO_H |
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#define AVDEVICE_ALSA_AUDIO_H |
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#include <alsa/asoundlib.h> |
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#include "config.h" |
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#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 */ |
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/* XXX: find better solution with "preinit" method, needed also in |
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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; |
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int frame_size; ///< preferred size for reads and writes |
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int period_size; ///< bytes per sample * channels |
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int sample_rate; ///< sample rate set by user |
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int channels; ///< number of channels set by user |
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void (*reorder_func)(const void *, void *, int); |
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void *reorder_buf; |
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int reorder_buf_size; ///< in frames |
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} AlsaData; |
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/** |
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* Open an ALSA PCM. |
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* |
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* @param s media file handle |
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* @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK |
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* @param sample_rate in: requested sample rate; |
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* out: actually selected sample rate |
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* @param channels number of channels |
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* @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE; |
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* out: actually selected AVCodecID, changed only if |
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* AV_CODEC_ID_NONE was requested |
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* |
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* @return 0 if OK, AVERROR_xxx on error |
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*/ |
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int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode, |
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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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* |
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* @param s1 media file handle |
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* |
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* @return 0 |
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*/ |
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int ff_alsa_close(AVFormatContext *s1); |
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/** |
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* Try to recover from ALSA buffer underrun. |
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* |
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* @param s1 media file handle |
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* @param err error code reported by the previous ALSA call |
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* |
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* @return 0 if OK, AVERROR_xxx on error |
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*/ |
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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 */
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