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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 can 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.  
  50. #include "libavutil/internal.h"
  51. #include "libavutil/mathematics.h"
  52. #include "libavutil/opt.h"
  53. #include "libavutil/time.h"
  54.  
  55. #include "libavformat/internal.h"
  56.  
  57. #include "avdevice.h"
  58. #include "alsa.h"
  59.  
  60. static av_cold int audio_read_header(AVFormatContext *s1)
  61. {
  62.     AlsaData *s = s1->priv_data;
  63.     AVStream *st;
  64.     int ret;
  65.     enum AVCodecID codec_id;
  66.  
  67.     st = avformat_new_stream(s1, NULL);
  68.     if (!st) {
  69.         av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
  70.  
  71.         return AVERROR(ENOMEM);
  72.     }
  73.     codec_id    = s1->audio_codec_id;
  74.  
  75.     ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
  76.         &codec_id);
  77.     if (ret < 0) {
  78.         return AVERROR(EIO);
  79.     }
  80.  
  81.     /* take real parameters */
  82.     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
  83.     st->codec->codec_id    = codec_id;
  84.     st->codec->sample_rate = s->sample_rate;
  85.     st->codec->channels    = s->channels;
  86.     st->codec->frame_size = s->frame_size;
  87.     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
  88.     /* microseconds instead of seconds, MHz instead of Hz */
  89.     s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
  90.                                       s->period_size, 1.5E-6);
  91.     if (!s->timefilter)
  92.         goto fail;
  93.  
  94.     return 0;
  95.  
  96. fail:
  97.     snd_pcm_close(s->h);
  98.     return AVERROR(EIO);
  99. }
  100.  
  101. static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
  102. {
  103.     AlsaData *s  = s1->priv_data;
  104.     int res;
  105.     int64_t dts;
  106.     snd_pcm_sframes_t delay = 0;
  107.  
  108.     if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
  109.         return AVERROR(EIO);
  110.     }
  111.  
  112.     while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
  113.         if (res == -EAGAIN) {
  114.             av_free_packet(pkt);
  115.  
  116.             return AVERROR(EAGAIN);
  117.         }
  118.         if (ff_alsa_xrun_recover(s1, res) < 0) {
  119.             av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
  120.                    snd_strerror(res));
  121.             av_free_packet(pkt);
  122.  
  123.             return AVERROR(EIO);
  124.         }
  125.         ff_timefilter_reset(s->timefilter);
  126.     }
  127.  
  128.     dts = av_gettime();
  129.     snd_pcm_delay(s->h, &delay);
  130.     dts -= av_rescale(delay + res, 1000000, s->sample_rate);
  131.     pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
  132.     s->last_period = res;
  133.  
  134.     pkt->size = res * s->frame_size;
  135.  
  136.     return 0;
  137. }
  138.  
  139. static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
  140. {
  141.     return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
  142. }
  143.  
  144. static const AVOption options[] = {
  145.     { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  146.     { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  147.     { NULL },
  148. };
  149.  
  150. static const AVClass alsa_demuxer_class = {
  151.     .class_name     = "ALSA demuxer",
  152.     .item_name      = av_default_item_name,
  153.     .option         = options,
  154.     .version        = LIBAVUTIL_VERSION_INT,
  155.     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
  156. };
  157.  
  158. AVInputFormat ff_alsa_demuxer = {
  159.     .name           = "alsa",
  160.     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
  161.     .priv_data_size = sizeof(AlsaData),
  162.     .read_header    = audio_read_header,
  163.     .read_packet    = audio_read_packet,
  164.     .read_close     = ff_alsa_close,
  165.     .get_device_list = audio_get_device_list,
  166.     .flags          = AVFMT_NOFILE,
  167.     .priv_class     = &alsa_demuxer_class,
  168. };
  169.