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/avformat.h"
50
51#include "alsa-audio.h"
52
53static av_cold int audio_read_header(AVFormatContext *s1,
54                                     AVFormatParameters *ap)
55{
56    AlsaData *s = s1->priv_data;
57    AVStream *st;
58    int ret;
59    unsigned int sample_rate;
60    enum CodecID codec_id;
61    snd_pcm_sw_params_t *sw_params;
62
63    if (ap->sample_rate <= 0) {
64        av_log(s1, AV_LOG_ERROR, "Bad sample rate %d\n", ap->sample_rate);
65
66        return AVERROR(EIO);
67    }
68
69    if (ap->channels <= 0) {
70        av_log(s1, AV_LOG_ERROR, "Bad channels number %d\n", ap->channels);
71
72        return AVERROR(EIO);
73    }
74
75    st = av_new_stream(s1, 0);
76    if (!st) {
77        av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
78
79        return AVERROR(ENOMEM);
80    }
81    sample_rate = ap->sample_rate;
82    codec_id    = s1->audio_codec_id;
83
84    ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &sample_rate, ap->channels,
85        &codec_id);
86    if (ret < 0) {
87        return AVERROR(EIO);
88    }
89
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, "
93               "may hang.\n");
94
95    ret = snd_pcm_sw_params_malloc(&sw_params);
96    if (ret < 0) {
97        av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
98               snd_strerror(ret));
99        goto fail;
100    }
101
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);
104
105    ret = snd_pcm_sw_params(s->h, sw_params);
106    snd_pcm_sw_params_free(sw_params);
107    if (ret < 0) {
108        av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
109               snd_strerror(ret));
110        goto fail;
111    }
112
113    /* take real parameters */
114    st->codec->codec_type  = AVMEDIA_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 */
119
120    return 0;
121
122fail:
123    snd_pcm_close(s->h);
124    return AVERROR(EIO);
125}
126
127static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
128{
129    AlsaData *s  = s1->priv_data;
130    AVStream *st = s1->streams[0];
131    int res;
132    snd_htimestamp_t timestamp;
133    snd_pcm_uframes_t ts_delay;
134
135    if (av_new_packet(pkt, s->period_size) < 0) {
136        return AVERROR(EIO);
137    }
138
139    while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
140        if (res == -EAGAIN) {
141            av_free_packet(pkt);
142
143            return AVERROR(EAGAIN);
144        }
145        if (ff_alsa_xrun_recover(s1, res) < 0) {
146            av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
147                   snd_strerror(res));
148            av_free_packet(pkt);
149
150            return AVERROR(EIO);
151        }
152    }
153
154    snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
155    ts_delay += res;
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);
160
161    pkt->size = res * s->frame_size;
162
163    return 0;
164}
165
166AVInputFormat alsa_demuxer = {
167    "alsa",
168    NULL_IF_CONFIG_SMALL("ALSA audio input"),
169    sizeof(AlsaData),
170    NULL,
171    audio_read_header,
172    audio_read_packet,
173    ff_alsa_close,
174    .flags = AVFMT_NOFILE,
175};
176