/*
  * 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 can 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 "libavutil/internal.h"

 #include "libavutil/time.h"

 #include "libavformat/internal.h"

 #include "avdevice.h"

 #include "alsa.h"

 static av_cold int audio_write_header(AVFormatContext *s1)

 {
     AlsaData *s = s1->priv_data;

     AVStream *st = NULL;

     unsigned int sample_rate;

     enum AVCodecID codec_id;

     int res;
 

     if (s1->nb_streams != 1 || s1->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
         av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
         return AVERROR(EINVAL);
     }

     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;
     }

     avpriv_set_pts_info(st, 64, 1, sample_rate);

 
     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 (pkt->dts != AV_NOPTS_VALUE)
         s->timestamp = pkt->dts;
     s->timestamp += pkt->duration ? pkt->duration : 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;
 }
 

 static int audio_write_frame(AVFormatContext *s1, int stream_index,
                              AVFrame **frame, unsigned flags)
 {
     AlsaData *s = s1->priv_data;
     AVPacket pkt;
 
     /* ff_alsa_open() should have accepted only supported formats */
     if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
         return av_sample_fmt_is_planar(s1->streams[stream_index]->codec->sample_fmt) ?
                AVERROR(EINVAL) : 0;
     /* set only used fields */
     pkt.data     = (*frame)->data[0];
     pkt.size     = (*frame)->nb_samples * s->frame_size;
     pkt.dts      = (*frame)->pkt_dts;
     pkt.duration = av_frame_get_pkt_duration(*frame);
     return audio_write_packet(s1, &pkt);
 }
 

 static void
 audio_get_output_timestamp(AVFormatContext *s1, int stream,
     int64_t *dts, int64_t *wall)
 {
     AlsaData *s  = s1->priv_data;
     snd_pcm_sframes_t delay = 0;
     *wall = av_gettime();
     snd_pcm_delay(s->h, &delay);

     *dts = s->timestamp - delay;

 }
 

 static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
 {
     return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
 }
 

 static const AVClass alsa_muxer_class = {
     .class_name     = "ALSA muxer",
     .item_name      = av_default_item_name,
     .version        = LIBAVUTIL_VERSION_INT,
     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
 };
 

 AVOutputFormat ff_alsa_muxer = {

     .name           = "alsa",
     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
     .priv_data_size = sizeof(AlsaData),
     .audio_codec    = DEFAULT_CODEC_ID,

     .video_codec    = AV_CODEC_ID_NONE,

     .write_header   = audio_write_header,
     .write_packet   = audio_write_packet,
     .write_trailer  = ff_alsa_close,

     .write_uncoded_frame = audio_write_frame,

     .get_device_list = audio_get_device_list,

     .get_output_timestamp = audio_get_output_timestamp,

     .flags          = AVFMT_NOFILE,

     .priv_class     = &alsa_muxer_class,

 };