libavdevice/alsa-audio.h

   1 /*
   2  * ALSA input and output
   3  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
   4  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 /**
  24  * @file libavdevice/alsa-audio.h
  25  * ALSA input and output: definitions and structures
  26  * @author Luca Abeni ( lucabe72 email it )
  27  * @author Benoit Fouet ( benoit fouet free fr )
  28  */
  29
  30 #ifndef AVDEVICE_ALSA_AUDIO_H
  31 #define AVDEVICE_ALSA_AUDIO_H
  32
  33 #include <alsa/asoundlib.h>
  34 #include "config.h"
  35 #include "libavformat/avformat.h"
  36
  37 /* XXX: we make the assumption that the soundcard accepts this format */
  38 /* XXX: find better solution with "preinit" method, needed also in
  39         other formats */
  40 #if HAVE_BIGENDIAN
  41 #define DEFAULT_CODEC_ID CODEC_ID_PCM_S16BE
  42 #else
  43 #define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE
  44 #endif
  45
  46 typedef struct {
  47     snd_pcm_t *h;
  48     int frame_size;  ///< preferred size for reads and writes
  49     int period_size; ///< bytes per sample * channels
  50 } AlsaData;
  51
  52 /**
  53  * Opens an ALSA PCM.
  54  *
  55  * @param s media file handle
  56  * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
  57  * @param sample_rate in: requested sample rate;
  58  *                    out: actually selected sample rate
  59  * @param channels number of channels
  60  * @param codec_id in: requested CodecID or CODEC_ID_NONE;
  61  *                 out: actually selected CodecID, changed only if
  62  *                 CODEC_ID_NONE was requested
  63  *
  64  * @return 0 if OK, AVERROR_xxx on error
  65  */
  66 int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
  67                  unsigned int *sample_rate,
  68                  int channels, enum CodecID *codec_id);
  69
  70 /**
  71  * Closes the ALSA PCM.
  72  *
  73  * @param s1 media file handle
  74  *
  75  * @return 0
  76  */
  77 int ff_alsa_close(AVFormatContext *s1);
  78
  79 /**
  80  * Tries to recover from ALSA buffer underrun.
  81  *
  82  * @param s1 media file handle
  83  * @param err error code reported by the previous ALSA call
  84  *
  85  * @return 0 if OK, AVERROR_xxx on error
  86  */
  87 int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
  88
  89 #endif /* AVDEVICE_ALSA_AUDIO_H */