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author | Nicolas George <nicola.george@normalesup.org> | 2009-01-26 09:16:09 +0000 |
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committer | Benoit Fouet <benoit.fouet@free.fr> | 2009-01-26 09:16:09 +0000 |
commit | 35fd81224aa8a80ffa4b17bd143fd6911924e8f6 (patch) | |
tree | ed0213baeb236386ab3d896922966d80ea271dc2 /libavdevice/alsa-audio.h | |
parent | 1db2c5c9efceef9fb250b772e69b58c5259fcb42 (diff) | |
download | ffmpeg-35fd81224aa8a80ffa4b17bd143fd6911924e8f6.tar.gz |
Add ALSA support in libavdevice.
Patch by Nicolas George: name surname normalesup org
Original thread: [FFmpeg-devel] [PATCH] ALSA for libavdevice
Date: 12/09/2008 07:17 PM
Originally committed as revision 16800 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavdevice/alsa-audio.h')
-rw-r--r-- | libavdevice/alsa-audio.h | 84 |
1 files changed, 84 insertions, 0 deletions
diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h new file mode 100644 index 0000000000..9547f790af --- /dev/null +++ b/libavdevice/alsa-audio.h @@ -0,0 +1,84 @@ +/* + * 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-audio.h + * 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 + +/* XXX: we make the assumption that the soundcard accepts this format */ +/* XXX: find better solution with "preinit" method, needed also in + other formats */ +#ifdef WORDS_BIGENDIAN +#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16BE +#else +#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE +#endif + +typedef struct { + snd_pcm_t *h; + int frame_size; ///< preferred size for reads and writes + int period_size; ///< bytes per sample * channels +} AlsaData; + +/** + * Opens 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, int mode, unsigned int *sample_rate, + int channels, int *codec_id); + +/** + * Closes the ALSA PCM. + * + * @param s1 media file handle + * + * @return 0 + */ +int ff_alsa_close(AVFormatContext *s1); + +/** + * Tries 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 */ |