oss: add channels and sample_rate private options.
[libav.git] / libavdevice / alsa-audio-dec.c
CommitLineData
35fd8122
NG
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 *
2912e87a 6 * This file is part of Libav.
35fd8122 7 *
2912e87a 8 * Libav is free software; you can redistribute it and/or
35fd8122
NG
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 *
2912e87a 13 * Libav is distributed in the hope that it will be useful,
35fd8122
NG
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
2912e87a 19 * License along with Libav; if not, write to the Free Software
35fd8122
NG
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23/**
ba87f080 24 * @file
35fd8122
NG
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
35fd8122 48#include <alsa/asoundlib.h>
419bddd3 49#include "libavformat/avformat.h"
35fd8122
NG
50
51#include "alsa-audio.h"
52
244c8d10 53static av_cold int audio_read_header(AVFormatContext *s1,
35fd8122
NG
54 AVFormatParameters *ap)
55{
56 AlsaData *s = s1->priv_data;
57 AVStream *st;
58 int ret;
59 unsigned int sample_rate;
5c463aac 60 enum CodecID codec_id;
35fd8122
NG
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;
90d0379f 82 codec_id = s1->audio_codec_id;
35fd8122
NG
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 */
72415b2a 114 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
35fd8122
NG
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
c6610a21 166AVInputFormat ff_alsa_demuxer = {
35fd8122
NG
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};