1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
|
/*
* Fixed-point MPEG audio decoder
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "config_components.h"
#include "libavutil/samplefmt.h"
#define USE_FLOATS 0
#include "mpegaudio.h"
#define SHR(a,b) (((int)(a))>>(b))
/* WARNING: only correct for positive numbers */
#define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
#define MULH3(x, y, s) MULH((s)*(x), y)
#define MULLx(x, y, s) MULL((int)(x),(y),s)
#define RENAME(a) a ## _fixed
#define OUT_FMT AV_SAMPLE_FMT_S16
#define OUT_FMT_P AV_SAMPLE_FMT_S16P
/* Intensity stereo table. See commit b91d46614df189e7905538e7f5c4ed9c7ed0d274
* (float based mp1/mp2/mp3 decoders.) for how they were created. */
static const int32_t is_table[2][16] = {
{ 0x000000, 0x1B0CB1, 0x2ED9EC, 0x400000, 0x512614, 0x64F34F, 0x800000 },
{ 0x800000, 0x64F34F, 0x512614, 0x400000, 0x2ED9EC, 0x1B0CB1, 0x000000 }
};
/* Antialiasing table. See commit ce4a29c066cddfc180979ed86396812f24337985
* (optimize antialias) for how they were created. */
static const int32_t csa_table[8][4] = {
{ 0x36E129F8, 0xDF128056, 0x15F3AA4E, 0xA831565E },
{ 0x386E75F2, 0xE1CF24A5, 0x1A3D9A97, 0xA960AEB3 },
{ 0x3CC6B73A, 0xEBF19FA6, 0x28B856E0, 0xAF2AE86C },
{ 0x3EEEA054, 0xF45B88BC, 0x334A2910, 0xB56CE868 },
{ 0x3FB6905C, 0xF9F27F18, 0x39A90F74, 0xBA3BEEBC },
{ 0x3FF23F20, 0xFD60D1E4, 0x3D531104, 0xBD6E92C4 },
{ 0x3FFE5932, 0xFF175EE4, 0x3F15B816, 0xBF1905B2 },
{ 0x3FFFE34A, 0xFFC3612F, 0x3FC34479, 0xBFC37DE5 }
};
#include "mpegaudiodec_template.c"
#if CONFIG_MP1_DECODER
const AVCodec ff_mp1_decoder = {
.name = "mp1",
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP1,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP2_DECODER
const AVCodec ff_mp2_decoder = {
.name = "mp2",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP3_DECODER
const AVCodec ff_mp3_decoder = {
.name = "mp3",
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP3ADU_DECODER
const AVCodec ff_mp3adu_decoder = {
.name = "mp3adu",
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ADU,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame_adu,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP3ON4_DECODER
const AVCodec ff_mp3on4_decoder = {
.name = "mp3on4",
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ON4,
.priv_data_size = sizeof(MP3On4DecodeContext),
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush_mp3on4,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};
#endif
|
