FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
alsa-audio-dec.c
Go to the documentation of this file.
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 
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 */
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 
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  }
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 
149  .name = "alsa",
150  .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
151  .priv_data_size = sizeof(AlsaData),
155  .flags = AVFMT_NOFILE,
156  .priv_class = &alsa_demuxer_class,
157 };