/* * 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); #define ALSA_BUFFER_SIZE_MAX 65536 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 */