libavdevice/alsa-audio-enc.c

   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 libavdevice/alsa-audio-enc.c
  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 "libavformat/avformat.h"
  41 #include <alsa/asoundlib.h>
  42
  43 #include "alsa-audio.h"
  44
  45 av_cold static int audio_write_header(AVFormatContext *s1)
  46 {
  47     AlsaData *s = s1->priv_data;
  48     AVStream *st;
  49     unsigned int sample_rate;
  50     int codec_id;
  51     int res;
  52
  53     st = s1->streams[0];
  54     sample_rate = st->codec->sample_rate;
  55     codec_id    = st->codec->codec_id;
  56     res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
  57         st->codec->channels, &codec_id);
  58     if (sample_rate != st->codec->sample_rate) {
  59         av_log(s1, AV_LOG_ERROR,
  60                "sample rate %d not available, nearest is %d\n",
  61                st->codec->sample_rate, sample_rate);
  62         goto fail;
  63     }
  64
  65     return res;
  66
  67 fail:
  68     snd_pcm_close(s->h);
  69     return AVERROR(EIO);
  70 }
  71
  72 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
  73 {
  74     AlsaData *s = s1->priv_data;
  75     int res;
  76     int size     = pkt->size;
  77     uint8_t *buf = pkt->data;
  78
  79     while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) {
  80         if (res == -EAGAIN) {
  81
  82             return AVERROR(EAGAIN);
  83         }
  84
  85         if (ff_alsa_xrun_recover(s1, res) < 0) {
  86             av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
  87                    snd_strerror(res));
  88
  89             return AVERROR(EIO);
  90         }
  91     }
  92
  93     return 0;
  94 }
  95
  96 AVOutputFormat alsa_muxer = {
  97     "alsa",
  98     NULL_IF_CONFIG_SMALL("ALSA audio output"),
  99     "",
 100     "",
 101     sizeof(AlsaData),
 102     DEFAULT_CODEC_ID,
 103     CODEC_ID_NONE,
 104     audio_write_header,
 105     audio_write_packet,
 106     ff_alsa_close,
 107     .flags = AVFMT_NOFILE,
 108 };