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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: input
  26.  * @author Luca Abeni ( lucabe72 email it )
  27.  * @author Benoit Fouet ( benoit fouet free fr )
  28.  * @author Nicolas George ( nicolas george normalesup org )
  29.  *
  30.  * This avdevice decoder allows to capture audio from an ALSA (Advanced
  31.  * Linux Sound Architecture) device.
  32.  *
  33.  * The filename parameter is the name of an ALSA PCM device capable of
  34.  * capture, for example "default" or "plughw:1"; see the ALSA documentation
  35.  * for naming conventions. The empty string is equivalent to "default".
  36.  *
  37.  * The capture period is set to the lower value available for the device,
  38.  * which gives a low latency suitable for real-time capture.
  39.  *
  40.  * The PTS are an Unix time in microsecond.
  41.  *
  42.  * Due to a bug in the ALSA library
  43.  * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
  44.  * decoder does not work with certain ALSA plugins, especially the dsnoop
  45.  * plugin.
  46.  */
  47.  
  48. #include <alsa/asoundlib.h>
  49. #include "libavformat/internal.h"
  50. #include "libavutil/opt.h"
  51. #include "libavutil/mathematics.h"
  52. #include "libavutil/time.h"
  53.  
  54. #include "avdevice.h"
  55. #include "alsa-audio.h"
  56.  
  57. static av_cold int audio_read_header(AVFormatContext *s1)
  58. {
  59.     AlsaData *s = s1->priv_data;
  60.     AVStream *st;
  61.     int ret;
  62.     enum AVCodecID codec_id;
  63.  
  64.     st = avformat_new_stream(s1, NULL);
  65.     if (!st) {
  66.         av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
  67.  
  68.         return AVERROR(ENOMEM);
  69.     }
  70.     codec_id    = s1->audio_codec_id;
  71.  
  72.     ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
  73.         &codec_id);
  74.     if (ret < 0) {
  75.         return AVERROR(EIO);
  76.     }
  77.  
  78.     /* take real parameters */
  79.     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
  80.     st->codec->codec_id    = codec_id;
  81.     st->codec->sample_rate = s->sample_rate;
  82.     st->codec->channels    = s->channels;
  83.     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
  84.     /* microseconds instead of seconds, MHz instead of Hz */
  85.     s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
  86.                                       s->period_size, 1.5E-6);
  87.     if (!s->timefilter)
  88.         goto fail;
  89.  
  90.     return 0;
  91.  
  92. fail:
  93.     snd_pcm_close(s->h);
  94.     return AVERROR(EIO);
  95. }
  96.  
  97. static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
  98. {
  99.     AlsaData *s  = s1->priv_data;
  100.     int res;
  101.     int64_t dts;
  102.     snd_pcm_sframes_t delay = 0;
  103.  
  104.     if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
  105.         return AVERROR(EIO);
  106.     }
  107.  
  108.     while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
  109.         if (res == -EAGAIN) {
  110.             av_free_packet(pkt);
  111.  
  112.             return AVERROR(EAGAIN);
  113.         }
  114.         if (ff_alsa_xrun_recover(s1, res) < 0) {
  115.             av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
  116.                    snd_strerror(res));
  117.             av_free_packet(pkt);
  118.  
  119.             return AVERROR(EIO);
  120.         }
  121.         ff_timefilter_reset(s->timefilter);
  122.     }
  123.  
  124.     dts = av_gettime();
  125.     snd_pcm_delay(s->h, &delay);
  126.     dts -= av_rescale(delay + res, 1000000, s->sample_rate);
  127.     pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
  128.     s->last_period = res;
  129.  
  130.     pkt->size = res * s->frame_size;
  131.  
  132.     return 0;
  133. }
  134.  
  135. static const AVOption options[] = {
  136.     { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  137.     { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  138.     { NULL },
  139. };
  140.  
  141. static const AVClass alsa_demuxer_class = {
  142.     .class_name     = "ALSA demuxer",
  143.     .item_name      = av_default_item_name,
  144.     .option         = options,
  145.     .version        = LIBAVUTIL_VERSION_INT,
  146. };
  147.  
  148. AVInputFormat ff_alsa_demuxer = {
  149.     .name           = "alsa",
  150.     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
  151.     .priv_data_size = sizeof(AlsaData),
  152.     .read_header    = audio_read_header,
  153.     .read_packet    = audio_read_packet,
  154.     .read_close     = ff_alsa_close,
  155.     .flags          = AVFMT_NOFILE,
  156.     .priv_class     = &alsa_demuxer_class,
  157. };
  158.