FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavdevice/alsa_dec.c
Date: 2024-02-16 17:37:06
Exec Total Coverage
Lines: 0 51 0.0%
Functions: 0 3 0.0%
Branches: 0 18 0.0%

Line Branch Exec Source
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 can 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
50 #include "libavutil/internal.h"
51 #include "libavutil/mathematics.h"
52 #include "libavutil/opt.h"
53 #include "libavutil/time.h"
54
55 #include "libavformat/internal.h"
56
57 #include "avdevice.h"
58 #include "alsa.h"
59
60 static av_cold int audio_read_header(AVFormatContext *s1)
61 {
62 AlsaData *s = s1->priv_data;
63 AVStream *st;
64 int ret;
65 enum AVCodecID codec_id;
66
67 st = avformat_new_stream(s1, NULL);
68 if (!st) {
69 av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
70
71 return AVERROR(ENOMEM);
72 }
73 codec_id = s1->audio_codec_id;
74
75 ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
76 &codec_id);
77 if (ret < 0) {
78 return AVERROR(EIO);
79 }
80
81 /* take real parameters */
82 st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
83 st->codecpar->codec_id = codec_id;
84 st->codecpar->sample_rate = s->sample_rate;
85 st->codecpar->ch_layout.nb_channels = s->channels;
86 st->codecpar->frame_size = s->frame_size;
87 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
88 /* microseconds instead of seconds, MHz instead of Hz */
89 s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
90 s->period_size, 1.5E-6);
91 if (!s->timefilter)
92 goto fail;
93
94 return 0;
95
96 fail:
97 snd_pcm_close(s->h);
98 return AVERROR(EIO);
99 }
100
101 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
102 {
103 AlsaData *s = s1->priv_data;
104 int res;
105 int64_t dts;
106 snd_pcm_sframes_t delay = 0;
107
108 if (!s->pkt->data) {
109 int ret = av_new_packet(s->pkt, s->period_size * s->frame_size);
110 if (ret < 0)
111 return ret;
112 s->pkt->size = 0;
113 }
114
115 do {
116 while ((res = snd_pcm_readi(s->h, s->pkt->data + s->pkt->size, s->period_size - s->pkt->size / s->frame_size)) < 0) {
117 if (res == -EAGAIN) {
118 return AVERROR(EAGAIN);
119 }
120 s->pkt->size = 0;
121 if (ff_alsa_xrun_recover(s1, res) < 0) {
122 av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
123 snd_strerror(res));
124 return AVERROR(EIO);
125 }
126 ff_timefilter_reset(s->timefilter);
127 }
128 s->pkt->size += res * s->frame_size;
129 } while (s->pkt->size < s->period_size * s->frame_size);
130
131 av_packet_move_ref(pkt, s->pkt);
132 dts = av_gettime();
133 snd_pcm_delay(s->h, &delay);
134 dts -= av_rescale(delay + res, 1000000, s->sample_rate);
135 pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
136 s->last_period = res;
137
138 return 0;
139 }
140
141 static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
142 {
143 return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
144 }
145
146 static const AVOption options[] = {
147 { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
148 { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
149 { NULL },
150 };
151
152 static const AVClass alsa_demuxer_class = {
153 .class_name = "ALSA indev",
154 .item_name = av_default_item_name,
155 .option = options,
156 .version = LIBAVUTIL_VERSION_INT,
157 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
158 };
159
160 const AVInputFormat ff_alsa_demuxer = {
161 .name = "alsa",
162 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
163 .priv_data_size = sizeof(AlsaData),
164 .read_header = audio_read_header,
165 .read_packet = audio_read_packet,
166 .read_close = ff_alsa_close,
167 .get_device_list = audio_get_device_list,
168 .flags = AVFMT_NOFILE,
169 .priv_class = &alsa_demuxer_class,
170 };
171