/*
  * 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: output
  * @author Luca Abeni ( lucabe72 email it )
  * @author Benoit Fouet ( benoit fouet free fr )
  *
  * This avdevice encoder allows to play audio to an ALSA (Advanced Linux
  * Sound Architecture) device.
  *
  * The filename parameter is the name of an ALSA PCM device capable of
  * capture, for example "default" or "plughw:1"; see the ALSA documentation
  * for naming conventions. The empty string is equivalent to "default".
  *
  * The playback period is set to the lower value available for the device,
  * which gives a low latency suitable for real-time playback.
  */
 
 #include <alsa/asoundlib.h>
 

 #include "avdevice.h"

 #include "alsa-audio.h"
 

 static av_cold int audio_write_header(AVFormatContext *s1)

 {
     AlsaData *s = s1->priv_data;
     AVStream *st;
     unsigned int sample_rate;

     enum CodecID codec_id;

     int res;
 
     st = s1->streams[0];
     sample_rate = st->codec->sample_rate;
     codec_id    = st->codec->codec_id;
     res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
         st->codec->channels, &codec_id);
     if (sample_rate != st->codec->sample_rate) {
         av_log(s1, AV_LOG_ERROR,
                "sample rate %d not available, nearest is %d\n",
                st->codec->sample_rate, sample_rate);
         goto fail;
     }
 
     return res;
 
 fail:
     snd_pcm_close(s->h);
     return AVERROR(EIO);
 }
 
 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     AlsaData *s = s1->priv_data;
     int res;
     int size     = pkt->size;
     uint8_t *buf = pkt->data;
 

     size /= s->frame_size;
     if (s->reorder_func) {
         if (size > s->reorder_buf_size)
             if (ff_alsa_extend_reorder_buf(s, size))
                 return AVERROR(ENOMEM);
         s->reorder_func(buf, s->reorder_buf, size);
         buf = s->reorder_buf;
     }
     while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {

         if (res == -EAGAIN) {
 
             return AVERROR(EAGAIN);
         }
 
         if (ff_alsa_xrun_recover(s1, res) < 0) {
             av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
                    snd_strerror(res));
 
             return AVERROR(EIO);
         }
     }
 
     return 0;
 }
 

 AVOutputFormat ff_alsa_muxer = {

     "alsa",
     NULL_IF_CONFIG_SMALL("ALSA audio output"),
     "",
     "",
     sizeof(AlsaData),
     DEFAULT_CODEC_ID,
     CODEC_ID_NONE,
     audio_write_header,
     audio_write_packet,
     ff_alsa_close,
     .flags = AVFMT_NOFILE,
 };