avdevice: Add missing header for NULL_IF_CONFIG_SMALL
[libav.git] / libavdevice / alsa-audio-dec.c
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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
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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,
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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
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23/**
ba87f080 24 * @file
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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>
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49
50#include "libavutil/internal.h"
51#include "libavutil/opt.h"
52
419bddd3 53#include "libavformat/avformat.h"
c3f9ebf7 54#include "libavformat/internal.h"
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55
56#include "alsa-audio.h"
57
6e9651d1 58static av_cold int audio_read_header(AVFormatContext *s1)
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59{
60 AlsaData *s = s1->priv_data;
61 AVStream *st;
62 int ret;
36ef5369 63 enum AVCodecID codec_id;
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64 snd_pcm_sw_params_t *sw_params;
65
3b3bbdd3 66 st = avformat_new_stream(s1, NULL);
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67 if (!st) {
68 av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
69
70 return AVERROR(ENOMEM);
71 }
90d0379f 72 codec_id = s1->audio_codec_id;
35fd8122 73
2ea8faf3 74 ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
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75 &codec_id);
76 if (ret < 0) {
77 return AVERROR(EIO);
78 }
79
80 if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
81 av_log(s1, AV_LOG_WARNING,
82 "capture with some ALSA plugins, especially dsnoop, "
83 "may hang.\n");
84
85 ret = snd_pcm_sw_params_malloc(&sw_params);
86 if (ret < 0) {
87 av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
88 snd_strerror(ret));
89 goto fail;
90 }
91
92 snd_pcm_sw_params_current(s->h, sw_params);
93 snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
94
95 ret = snd_pcm_sw_params(s->h, sw_params);
96 snd_pcm_sw_params_free(sw_params);
97 if (ret < 0) {
98 av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
99 snd_strerror(ret));
100 goto fail;
101 }
102
103 /* take real parameters */
72415b2a 104 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
35fd8122 105 st->codec->codec_id = codec_id;
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106 st->codec->sample_rate = s->sample_rate;
107 st->codec->channels = s->channels;
c3f9ebf7 108 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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109
110 return 0;
111
112fail:
113 snd_pcm_close(s->h);
114 return AVERROR(EIO);
115}
116
117static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
118{
119 AlsaData *s = s1->priv_data;
120 AVStream *st = s1->streams[0];
121 int res;
122 snd_htimestamp_t timestamp;
123 snd_pcm_uframes_t ts_delay;
124
125 if (av_new_packet(pkt, s->period_size) < 0) {
126 return AVERROR(EIO);
127 }
128
129 while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
130 if (res == -EAGAIN) {
131 av_free_packet(pkt);
132
133 return AVERROR(EAGAIN);
134 }
135 if (ff_alsa_xrun_recover(s1, res) < 0) {
136 av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
137 snd_strerror(res));
138 av_free_packet(pkt);
139
140 return AVERROR(EIO);
141 }
142 }
143
144 snd_pcm_htimestamp(s->h, &ts_delay, &timestamp);
145 ts_delay += res;
146 pkt->pts = timestamp.tv_sec * 1000000LL
147 + (timestamp.tv_nsec * st->codec->sample_rate
089fac77 148 - (int64_t)ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
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149 / (st->codec->sample_rate * 1000LL);
150
151 pkt->size = res * s->frame_size;
152
153 return 0;
154}
155
2ea8faf3 156static const AVOption options[] = {
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157 { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
158 { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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159 { NULL },
160};
161
162static const AVClass alsa_demuxer_class = {
163 .class_name = "ALSA demuxer",
164 .item_name = av_default_item_name,
165 .option = options,
166 .version = LIBAVUTIL_VERSION_INT,
167};
168
c6610a21 169AVInputFormat ff_alsa_demuxer = {
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170 .name = "alsa",
171 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
172 .priv_data_size = sizeof(AlsaData),
173 .read_header = audio_read_header,
174 .read_packet = audio_read_packet,
175 .read_close = ff_alsa_close,
176 .flags = AVFMT_NOFILE,
177 .priv_class = &alsa_demuxer_class,
35fd8122 178};