X-Git-Url: http://git.ffmpeg.org/gitweb/ffmpeg.git/blobdiff_plain/63613fe615160671b394a232c1a3736319a6a8ec..e90a6846c2c006fbebd00e1f2789f4a86fafacef:/libavdevice/alsa-audio.h diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h index 6adde77..26eaee6 100644 --- a/libavdevice/alsa-audio.h +++ b/libavdevice/alsa-audio.h @@ -3,25 +3,25 @@ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file libavdevice/alsa-audio.h + * @file * ALSA input and output: definitions and structures * @author Luca Abeni ( lucabe72 email it ) * @author Benoit Fouet ( benoit fouet free fr ) @@ -33,42 +33,47 @@ #include #include "config.h" #include "libavformat/avformat.h" +#include "libavutil/log.h" /* XXX: we make the assumption that the soundcard accepts this format */ /* XXX: find better solution with "preinit" method, needed also in other formats */ -#if HAVE_BIGENDIAN -#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16BE -#else -#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE -#endif +#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) -typedef struct { +#define ALSA_BUFFER_SIZE_MAX 32768 + +typedef struct AlsaData { + AVClass *class; snd_pcm_t *h; int frame_size; ///< preferred size for reads and writes int period_size; ///< bytes per sample * channels + int sample_rate; ///< sample rate set by user + int channels; ///< number of channels set by user + void (*reorder_func)(const void *, void *, int); + void *reorder_buf; + int reorder_buf_size; ///< in frames } AlsaData; /** - * Opens an ALSA PCM. + * 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 + * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE; + * out: actually selected AVCodecID, changed only if + * AV_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); + int channels, enum AVCodecID *codec_id); /** - * Closes the ALSA PCM. + * Close the ALSA PCM. * * @param s1 media file handle * @@ -77,7 +82,7 @@ int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode, int ff_alsa_close(AVFormatContext *s1); /** - * Tries to recover from ALSA buffer underrun. + * Try to recover from ALSA buffer underrun. * * @param s1 media file handle * @param err error code reported by the previous ALSA call @@ -86,4 +91,6 @@ int ff_alsa_close(AVFormatContext *s1); */ int ff_alsa_xrun_recover(AVFormatContext *s1, int err); +int ff_alsa_extend_reorder_buf(AlsaData *s, int size); + #endif /* AVDEVICE_ALSA_AUDIO_H */