text: /*
   * 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 "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(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
  
  typedef void (*ff_reorder_func)(const void *, void *, int);
  
  #define ALSA_BUFFER_SIZE_MAX 65536
  
  typedef struct AlsaData {
      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
      int last_period;
      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 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 AVCodecID *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 */
  