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[ffmpeg] / libavformat / pcmdec.c
1 /*
2  * RAW PCM demuxers
3  * Copyright (c) 2002 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg 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  * FFmpeg 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 FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21
22 #include "avformat.h"
23 #include "internal.h"
24 #include "pcm.h"
25 #include "libavutil/log.h"
26 #include "libavutil/opt.h"
27 #include "libavutil/avassert.h"
28
29 typedef struct PCMAudioDemuxerContext {
30     AVClass *class;
31     int sample_rate;
32     int channels;
33 } PCMAudioDemuxerContext;
34
35 static int pcm_read_header(AVFormatContext *s)
36 {
37     PCMAudioDemuxerContext *s1 = s->priv_data;
38     AVStream *st;
39
40     st = avformat_new_stream(s, NULL);
41     if (!st)
42         return AVERROR(ENOMEM);
43
44
45     st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
46     st->codec->codec_id    = s->iformat->raw_codec_id;
47     st->codec->sample_rate = s1->sample_rate;
48     st->codec->channels    = s1->channels;
49
50     st->codec->bits_per_coded_sample =
51         av_get_bits_per_sample(st->codec->codec_id);
52
53     av_assert0(st->codec->bits_per_coded_sample > 0);
54
55     st->codec->block_align =
56         st->codec->bits_per_coded_sample * st->codec->channels / 8;
57
58     avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
59     return 0;
60 }
61
62 static const AVOption pcm_options[] = {
63     { "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 44100}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
64     { "channels",    "", offsetof(PCMAudioDemuxerContext, channels),    AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
65     { NULL },
66 };
67
68 #define PCMDEF(name_, long_name_, ext, codec)               \
69 static const AVClass name_ ## _demuxer_class = {            \
70     .class_name = #name_ " demuxer",                        \
71     .item_name  = av_default_item_name,                     \
72     .option     = pcm_options,                              \
73     .version    = LIBAVUTIL_VERSION_INT,                    \
74 };                                                          \
75 AVInputFormat ff_pcm_ ## name_ ## _demuxer = {              \
76     .name           = #name_,                               \
77     .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
78     .priv_data_size = sizeof(PCMAudioDemuxerContext),       \
79     .read_header    = pcm_read_header,                      \
80     .read_packet    = ff_pcm_read_packet,                   \
81     .read_seek      = ff_pcm_read_seek,                     \
82     .flags          = AVFMT_GENERIC_INDEX,                  \
83     .extensions     = ext,                                  \
84     .raw_codec_id   = codec,                                \
85     .priv_class     = &name_ ## _demuxer_class,             \
86 };
87
88 PCMDEF(f64be, "PCM 64-bit floating-point big-endian",
89        NULL, AV_CODEC_ID_PCM_F64BE)
90
91 PCMDEF(f64le, "PCM 64-bit floating-point little-endian",
92        NULL, AV_CODEC_ID_PCM_F64LE)
93
94 PCMDEF(f32be, "PCM 32-bit floating-point big-endian",
95        NULL, AV_CODEC_ID_PCM_F32BE)
96
97 PCMDEF(f32le, "PCM 32-bit floating-point little-endian",
98        NULL, AV_CODEC_ID_PCM_F32LE)
99
100 PCMDEF(s32be, "PCM signed 32-bit big-endian",
101        NULL, AV_CODEC_ID_PCM_S32BE)
102
103 PCMDEF(s32le, "PCM signed 32-bit little-endian",
104        NULL, AV_CODEC_ID_PCM_S32LE)
105
106 PCMDEF(s24be, "PCM signed 24-bit big-endian",
107        NULL, AV_CODEC_ID_PCM_S24BE)
108
109 PCMDEF(s24le, "PCM signed 24-bit little-endian",
110        NULL, AV_CODEC_ID_PCM_S24LE)
111
112 PCMDEF(s16be, "PCM signed 16-bit big-endian",
113        AV_NE("sw", NULL), AV_CODEC_ID_PCM_S16BE)
114
115 PCMDEF(s16le, "PCM signed 16-bit little-endian",
116        AV_NE(NULL, "sw"), AV_CODEC_ID_PCM_S16LE)
117
118 PCMDEF(s8, "PCM signed 8-bit",
119        "sb", AV_CODEC_ID_PCM_S8)
120
121 PCMDEF(u32be, "PCM unsigned 32-bit big-endian",
122        NULL, AV_CODEC_ID_PCM_U32BE)
123
124 PCMDEF(u32le, "PCM unsigned 32-bit little-endian",
125        NULL, AV_CODEC_ID_PCM_U32LE)
126
127 PCMDEF(u24be, "PCM unsigned 24-bit big-endian",
128        NULL, AV_CODEC_ID_PCM_U24BE)
129
130 PCMDEF(u24le, "PCM unsigned 24-bit little-endian",
131        NULL, AV_CODEC_ID_PCM_U24LE)
132
133 PCMDEF(u16be, "PCM unsigned 16-bit big-endian",
134        AV_NE("uw", NULL), AV_CODEC_ID_PCM_U16BE)
135
136 PCMDEF(u16le, "PCM unsigned 16-bit little-endian",
137        AV_NE(NULL, "uw"), AV_CODEC_ID_PCM_U16LE)
138
139 PCMDEF(u8, "PCM unsigned 8-bit",
140        "ub", AV_CODEC_ID_PCM_U8)
141
142 PCMDEF(alaw, "PCM A-law",
143        "al", AV_CODEC_ID_PCM_ALAW)
144
145 PCMDEF(mulaw, "PCM mu-law",
146        "ul", AV_CODEC_ID_PCM_MULAW)
147
148 static const AVOption sln_options[] = {
149     { "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 8000}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
150     { "channels",    "", offsetof(PCMAudioDemuxerContext, channels),    AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
151     { NULL },
152 };
153
154 static const AVClass sln_demuxer_class = {
155     .class_name = "sln demuxer",
156     .item_name  = av_default_item_name,
157     .option     = sln_options,
158     .version    = LIBAVUTIL_VERSION_INT,
159 };
160
161 AVInputFormat ff_sln_demuxer = {
162     .name           = "sln",
163     .long_name      = NULL_IF_CONFIG_SMALL("Asterisk raw pcm"),
164     .priv_data_size = sizeof(PCMAudioDemuxerContext),
165     .read_header    = pcm_read_header,
166     .read_packet    = ff_pcm_read_packet,
167     .read_seek      = ff_pcm_read_seek,
168     .flags          = AVFMT_GENERIC_INDEX,
169     .extensions     = "sln",
170     .raw_codec_id   = AV_CODEC_ID_PCM_S16LE,
171     .priv_class     = &sln_demuxer_class,
172 };