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
57static 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
92fail:
93    snd_pcm_close(s->h);
94    return AVERROR(EIO);
95}
96
97static 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
135static 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
141static 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    .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
147};
148
149AVInputFormat ff_alsa_demuxer = {
150    .name           = "alsa",
151    .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio input"),
152    .priv_data_size = sizeof(AlsaData),
153    .read_header    = audio_read_header,
154    .read_packet    = audio_read_packet,
155    .read_close     = ff_alsa_close,
156    .flags          = AVFMT_NOFILE,
157    .priv_class     = &alsa_demuxer_class,
158};
159