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