ffmpeg/libavdevice/alsa-audio-dec.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
  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 int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
 136 {
 137     return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
 138 }
 139
 140 static const AVOption options[] = {
 141     { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
 142     { "channels",    "", offsetof(AlsaData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
 143     { NULL },
 144 };
 145
 146 static const AVClass alsa_demuxer_class = {
 147     .class_name     = "ALSA demuxer",
 148     .item_name      = av_default_item_name,
 149     .option         = options,
 150     .version        = LIBAVUTIL_VERSION_INT,
 151     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
 152 };
 153
 154 AVInputFormat ff_alsa_demuxer = {
 155     .name           = "alsa",
 156     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
 157     .priv_data_size = sizeof(AlsaData),
 158     .read_header    = audio_read_header,
 159     .read_packet    = audio_read_packet,
 160     .read_close     = ff_alsa_close,
 161     .get_device_list = audio_get_device_list,
 162     .flags          = AVFMT_NOFILE,
 163     .priv_class     = &alsa_demuxer_class,
 164 };