alsa-audio-dec: explicitly cast the delay to a signed int64
[libav.git] / libavdevice / alsa-audio-dec.c
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 Libav.
7 *
8 * Libav 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 * Libav 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 Libav; 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 #include "libavformat/internal.h"
51 #include "libavutil/opt.h"
52
53 #include "alsa-audio.h"
54
55 static av_cold int audio_read_header(AVFormatContext *s1)
56 {
57 AlsaData *s = s1->priv_data;
58 AVStream *st;
59 int ret;
60 enum AVCodecID codec_id;
61 snd_pcm_sw_params_t *sw_params;
62
63 st = avformat_new_stream(s1, NULL);
64 if (!st) {
65 av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
66
67 return AVERROR(ENOMEM);
68 }
69 codec_id = s1->audio_codec_id;
70
71 ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
72 &codec_id);
73 if (ret < 0) {
74 return AVERROR(EIO);
75 }
76
77 if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
78 av_log(s1, AV_LOG_WARNING,
79 "capture with some ALSA plugins, especially dsnoop, "
80 "may hang.\n");
81
82 ret = snd_pcm_sw_params_malloc(&sw_params);
83 if (ret < 0) {
84 av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
85 snd_strerror(ret));
86 goto fail;
87 }
88
89 snd_pcm_sw_params_current(s->h, sw_params);
90 snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
91
92 ret = snd_pcm_sw_params(s->h, sw_params);
93 snd_pcm_sw_params_free(sw_params);
94 if (ret < 0) {
95 av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
96 snd_strerror(ret));
97 goto fail;
98 }
99
100 /* take real parameters */
101 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
102 st->codec->codec_id = codec_id;
103 st->codec->sample_rate = s->sample_rate;
104 st->codec->channels = s->channels;
105 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
106
107 return 0;
108
109 fail:
110 snd_pcm_close(s->h);
111 return AVERROR(EIO);
112 }
113
114 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
115 {
116 AlsaData *s = s1->priv_data;
117 AVStream *st = s1->streams[0];
118 int res;
119 snd_htimestamp_t timestamp;
120 snd_pcm_uframes_t ts_delay;
121
122 if (av_new_packet(pkt, s->period_size) < 0) {
123 return AVERROR(EIO);
124 }
125
126 while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
127 if (res == -EAGAIN) {
128 av_free_packet(pkt);
129
130 return AVERROR(EAGAIN);
131 }
132 if (ff_alsa_xrun_recover(s1, res) < 0) {
133 av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
134 snd_strerror(res));
135 av_free_packet(pkt);
136
137 return AVERROR(EIO);
138 }
139 }
140
141 snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
142 ts_delay += res;
143 pkt->pts = timestamp.tv_sec * 1000000LL
144 + (timestamp.tv_nsec * st->codec->sample_rate
145 - (int64_t)ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
146 / (st->codec->sample_rate * 1000LL);
147
148 pkt->size = res * s->frame_size;
149
150 return 0;
151 }
152
153 static const AVOption options[] = {
154 { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
155 { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
156 { NULL },
157 };
158
159 static const AVClass alsa_demuxer_class = {
160 .class_name = "ALSA demuxer",
161 .item_name = av_default_item_name,
162 .option = options,
163 .version = LIBAVUTIL_VERSION_INT,
164 };
165
166 AVInputFormat ff_alsa_demuxer = {
167 .name = "alsa",
168 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
169 .priv_data_size = sizeof(AlsaData),
170 .read_header = audio_read_header,
171 .read_packet = audio_read_packet,
172 .read_close = ff_alsa_close,
173 .flags = AVFMT_NOFILE,
174 .priv_class = &alsa_demuxer_class,
175 };