oss_audio: Split muxer and demuxer
[libav.git] / libavdevice / oss_audio_dec.c
1 /*
2 * Linux audio play interface
3 * Copyright (c) 2000, 2001 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 #include "config.h"
23
24 #include <stdint.h>
25
26 #if HAVE_SOUNDCARD_H
27 #include <soundcard.h>
28 #else
29 #include <sys/soundcard.h>
30 #endif
31
32 #include <unistd.h>
33 #include <fcntl.h>
34 #include <sys/ioctl.h>
35
36 #include "libavutil/internal.h"
37 #include "libavutil/opt.h"
38 #include "libavutil/time.h"
39
40 #include "libavcodec/avcodec.h"
41
42 #include "libavformat/avformat.h"
43 #include "libavformat/internal.h"
44
45 #include "oss_audio.h"
46
47 static int audio_read_header(AVFormatContext *s1)
48 {
49 OSSAudioData *s = s1->priv_data;
50 AVStream *st;
51 int ret;
52
53 st = avformat_new_stream(s1, NULL);
54 if (!st) {
55 return AVERROR(ENOMEM);
56 }
57
58 ret = ff_oss_audio_open(s1, 0, s1->filename);
59 if (ret < 0) {
60 return AVERROR(EIO);
61 }
62
63 /* take real parameters */
64 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
65 st->codec->codec_id = s->codec_id;
66 st->codec->sample_rate = s->sample_rate;
67 st->codec->channels = s->channels;
68
69 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
70 return 0;
71 }
72
73 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
74 {
75 OSSAudioData *s = s1->priv_data;
76 int ret, bdelay;
77 int64_t cur_time;
78 struct audio_buf_info abufi;
79
80 if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
81 return ret;
82
83 ret = read(s->fd, pkt->data, pkt->size);
84 if (ret <= 0){
85 av_free_packet(pkt);
86 pkt->size = 0;
87 if (ret<0) return AVERROR(errno);
88 else return AVERROR_EOF;
89 }
90 pkt->size = ret;
91
92 /* compute pts of the start of the packet */
93 cur_time = av_gettime();
94 bdelay = ret;
95 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
96 bdelay += abufi.bytes;
97 }
98 /* subtract time represented by the number of bytes in the audio fifo */
99 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
100
101 /* convert to wanted units */
102 pkt->pts = cur_time;
103
104 if (s->flip_left && s->channels == 2) {
105 int i;
106 short *p = (short *) pkt->data;
107
108 for (i = 0; i < ret; i += 4) {
109 *p = ~*p;
110 p += 2;
111 }
112 }
113 return 0;
114 }
115
116 static int audio_read_close(AVFormatContext *s1)
117 {
118 OSSAudioData *s = s1->priv_data;
119
120 ff_oss_audio_close(s);
121 return 0;
122 }
123
124 static const AVOption options[] = {
125 { "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
126 { "channels", "", offsetof(OSSAudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
127 { NULL },
128 };
129
130 static const AVClass oss_demuxer_class = {
131 .class_name = "OSS demuxer",
132 .item_name = av_default_item_name,
133 .option = options,
134 .version = LIBAVUTIL_VERSION_INT,
135 };
136
137 AVInputFormat ff_oss_demuxer = {
138 .name = "oss",
139 .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
140 .priv_data_size = sizeof(OSSAudioData),
141 .read_header = audio_read_header,
142 .read_packet = audio_read_packet,
143 .read_close = audio_read_close,
144 .flags = AVFMT_NOFILE,
145 .priv_class = &oss_demuxer_class,
146 };