2 * ALSA input and output
3 * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
4 * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
6 * This file is part of FFmpeg.
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.
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.
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
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 )
30 * This avdevice decoder allows to capture audio from an ALSA (Advanced
31 * Linux Sound Architecture) device.
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".
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.
40 * The PTS are an Unix time in microsecond.
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
48 #include <alsa/asoundlib.h>
49 #include "libavutil/opt.h"
50 #include "libavutil/mathematics.h"
53 #include "alsa-audio.h"
55 static av_cold int audio_read_header(AVFormatContext *s1,
56 AVFormatParameters *ap)
58 AlsaData *s = s1->priv_data;
61 enum CodecID codec_id;
64 st = av_new_stream(s1, 0);
66 av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
68 return AVERROR(ENOMEM);
70 codec_id = s1->audio_codec_id;
72 ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
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 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
84 o = 2 * M_PI * s->period_size / s->sample_rate * 1.5; // bandwidth: 1.5Hz
85 s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
97 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
99 AlsaData *s = s1->priv_data;
102 snd_pcm_sframes_t delay = 0;
104 if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
108 while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
109 if (res == -EAGAIN) {
112 return AVERROR(EAGAIN);
114 if (ff_alsa_xrun_recover(s1, res) < 0) {
115 av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
121 ff_timefilter_reset(s->timefilter);
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, res);
129 pkt->size = res * s->frame_size;
134 static const AVOption options[] = {
135 { "sample_rate", "", offsetof(AlsaData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
136 { "channels", "", offsetof(AlsaData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
140 static const AVClass alsa_demuxer_class = {
141 .class_name = "ALSA demuxer",
142 .item_name = av_default_item_name,
144 .version = LIBAVUTIL_VERSION_INT,
147 AVInputFormat ff_alsa_demuxer = {
149 NULL_IF_CONFIG_SMALL("ALSA audio input"),
155 .flags = AVFMT_NOFILE,
156 .priv_class = &alsa_demuxer_class,