/*
  * 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 "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;  ///< preferred size for reads and writes
     int period_size; ///< bytes per sample * channels

     ff_reorder_func reorder_func;
     void *reorder_buf;
     int reorder_buf_size; ///< in frames

     int sample_rate; ///< sample rate set by user
     int channels;    ///< number of channels set by user

 } 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 */