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
24 * @file libavdevice/alsa-audio-dec.c
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 "libavformat/avformat.h"
49 #include <alsa/asoundlib.h>
51 #include "alsa-audio.h"
53 av_cold
static int audio_read_header(AVFormatContext
*s1
,
54 AVFormatParameters
*ap
)
56 AlsaData
*s
= s1
->priv_data
;
59 unsigned int sample_rate
;
60 enum CodecID codec_id
;
61 snd_pcm_sw_params_t
*sw_params
;
63 if (ap
->sample_rate
<= 0) {
64 av_log(s1
, AV_LOG_ERROR
, "Bad sample rate %d\n", ap
->sample_rate
);
69 if (ap
->channels
<= 0) {
70 av_log(s1
, AV_LOG_ERROR
, "Bad channels number %d\n", ap
->channels
);
75 st
= av_new_stream(s1
, 0);
77 av_log(s1
, AV_LOG_ERROR
, "Cannot add stream\n");
79 return AVERROR(ENOMEM
);
81 sample_rate
= ap
->sample_rate
;
82 codec_id
= ap
->audio_codec_id
;
84 ret
= ff_alsa_open(s1
, SND_PCM_STREAM_CAPTURE
, &sample_rate
, ap
->channels
,
90 if (snd_pcm_type(s
->h
) != SND_PCM_TYPE_HW
)
91 av_log(s1
, AV_LOG_WARNING
,
92 "capture with some ALSA plugins, especially dsnoop, "
95 ret
= snd_pcm_sw_params_malloc(&sw_params
);
97 av_log(s1
, AV_LOG_ERROR
, "cannot allocate software parameters structure (%s)\n",
102 snd_pcm_sw_params_current(s
->h
, sw_params
);
103 snd_pcm_sw_params_set_tstamp_mode(s
->h
, sw_params
, SND_PCM_TSTAMP_ENABLE
);
105 ret
= snd_pcm_sw_params(s
->h
, sw_params
);
106 snd_pcm_sw_params_free(sw_params
);
108 av_log(s1
, AV_LOG_ERROR
, "cannot install ALSA software parameters (%s)\n",
113 /* take real parameters */
114 st
->codec
->codec_type
= CODEC_TYPE_AUDIO
;
115 st
->codec
->codec_id
= codec_id
;
116 st
->codec
->sample_rate
= sample_rate
;
117 st
->codec
->channels
= ap
->channels
;
118 av_set_pts_info(st
, 64, 1, 1000000); /* 64 bits pts in us */
127 static int audio_read_packet(AVFormatContext
*s1
, AVPacket
*pkt
)
129 AlsaData
*s
= s1
->priv_data
;
130 AVStream
*st
= s1
->streams
[0];
132 snd_htimestamp_t timestamp
;
133 snd_pcm_uframes_t ts_delay
;
135 if (av_new_packet(pkt
, s
->period_size
) < 0) {
139 while ((res
= snd_pcm_readi(s
->h
, pkt
->data
, pkt
->size
/ s
->frame_size
)) < 0) {
140 if (res
== -EAGAIN
) {
143 return AVERROR(EAGAIN
);
145 if (ff_alsa_xrun_recover(s1
, res
) < 0) {
146 av_log(s1
, AV_LOG_ERROR
, "ALSA read error: %s\n",
154 snd_pcm_htimestamp(s
->h
, &ts_delay
, ×tamp
);
156 pkt
->pts
= timestamp
.tv_sec
* 1000000LL
157 + (timestamp
.tv_nsec
* st
->codec
->sample_rate
158 - ts_delay
* 1000000000LL + st
->codec
->sample_rate
* 500LL)
159 / (st
->codec
->sample_rate
* 1000LL);
161 pkt
->size
= res
* s
->frame_size
;
166 AVInputFormat alsa_demuxer
= {
168 NULL_IF_CONFIG_SMALL("ALSA audio input"),
174 .flags
= AVFMT_NOFILE
,