Imported Debian version 2.5.2~trusty
[deb_ffmpeg.git] / 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 };