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  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
  25.  * ALSA input and output: output
  26.  * @author Luca Abeni ( lucabe72 email it )
  27.  * @author Benoit Fouet ( benoit fouet free fr )
  28.  *
  29.  * This avdevice encoder allows to play audio to an ALSA (Advanced Linux
  30.  * Sound Architecture) device.
  31.  *
  32.  * The filename parameter is the name of an ALSA PCM device capable of
  33.  * capture, for example "default" or "plughw:1"; see the ALSA documentation
  34.  * for naming conventions. The empty string is equivalent to "default".
  35.  *
  36.  * The playback period is set to the lower value available for the device,
  37.  * which gives a low latency suitable for real-time playback.
  38.  */
  39.  
  40. #include <alsa/asoundlib.h>
  41.  
  42. #include "libavutil/time.h"
  43. #include "libavformat/internal.h"
  44. #include "avdevice.h"
  45. #include "alsa-audio.h"
  46.  
  47. static av_cold int audio_write_header(AVFormatContext *s1)
  48. {
  49.     AlsaData *s = s1->priv_data;
  50.     AVStream *st;
  51.     unsigned int sample_rate;
  52.     enum AVCodecID codec_id;
  53.     int res;
  54.  
  55.     st = s1->streams[0];
  56.     sample_rate = st->codec->sample_rate;
  57.     codec_id    = st->codec->codec_id;
  58.     res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
  59.         st->codec->channels, &codec_id);
  60.     if (sample_rate != st->codec->sample_rate) {
  61.         av_log(s1, AV_LOG_ERROR,
  62.                "sample rate %d not available, nearest is %d\n",
  63.                st->codec->sample_rate, sample_rate);
  64.         goto fail;
  65.     }
  66.     avpriv_set_pts_info(st, 64, 1, sample_rate);
  67.  
  68.     return res;
  69.  
  70. fail:
  71.     snd_pcm_close(s->h);
  72.     return AVERROR(EIO);
  73. }
  74.  
  75. static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
  76. {
  77.     AlsaData *s = s1->priv_data;
  78.     int res;
  79.     int size     = pkt->size;
  80.     uint8_t *buf = pkt->data;
  81.  
  82.     size /= s->frame_size;
  83.     if (s->reorder_func) {
  84.         if (size > s->reorder_buf_size)
  85.             if (ff_alsa_extend_reorder_buf(s, size))
  86.                 return AVERROR(ENOMEM);
  87.         s->reorder_func(buf, s->reorder_buf, size);
  88.         buf = s->reorder_buf;
  89.     }
  90.     while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
  91.         if (res == -EAGAIN) {
  92.  
  93.             return AVERROR(EAGAIN);
  94.         }
  95.  
  96.         if (ff_alsa_xrun_recover(s1, res) < 0) {
  97.             av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
  98.                    snd_strerror(res));
  99.  
  100.             return AVERROR(EIO);
  101.         }
  102.     }
  103.  
  104.     return 0;
  105. }
  106.  
  107. static void
  108. audio_get_output_timestamp(AVFormatContext *s1, int stream,
  109.     int64_t *dts, int64_t *wall)
  110. {
  111.     AlsaData *s  = s1->priv_data;
  112.     snd_pcm_sframes_t delay = 0;
  113.     *wall = av_gettime();
  114.     snd_pcm_delay(s->h, &delay);
  115.     *dts = s1->streams[0]->cur_dts - delay;
  116. }
  117.  
  118. AVOutputFormat ff_alsa_muxer = {
  119.     .name           = "alsa",
  120.     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
  121.     .priv_data_size = sizeof(AlsaData),
  122.     .audio_codec    = DEFAULT_CODEC_ID,
  123.     .video_codec    = AV_CODEC_ID_NONE,
  124.     .write_header   = audio_write_header,
  125.     .write_packet   = audio_write_packet,
  126.     .write_trailer  = ff_alsa_close,
  127.     .get_output_timestamp = audio_get_output_timestamp,
  128.     .flags          = AVFMT_NOFILE,
  129. };
  130.