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
|
/*
* Musepack decoder core
* Copyright (c) 2006 Konstantin Shishkov
*
* 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
*/
/**
* @file
* Musepack decoder core
* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
* divided into 32 subbands.
*/
#include "libavutil/attributes.h"
#include "avcodec.h"
#include "mpegaudiodsp.h"
#include "mpegaudio.h"
#include "mpc.h"
#include "mpcdata.h"
av_cold void ff_mpc_init(void)
{
ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
}
/**
* Process decoded Musepack data and produce PCM
*/
static void mpc_synth(MPCContext *c, int16_t **out, int channels)
{
int dither_state = 0;
int i, ch;
for(ch = 0; ch < channels; ch++){
for(i = 0; i < SAMPLES_PER_BAND; i++) {
ff_mpa_synth_filter_fixed(&c->mpadsp,
c->synth_buf[ch], &(c->synth_buf_offset[ch]),
ff_mpa_synth_window_fixed, &dither_state,
out[ch] + 32 * i, 1,
c->sb_samples[ch][i]);
}
}
}
void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, int16_t **out,
int channels)
{
int i, j, ch;
Band *bands = c->bands;
int off;
float mul;
/* dequantize */
memset(c->sb_samples, 0, sizeof(c->sb_samples));
off = 0;
for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
for(ch = 0; ch < 2; ch++){
if(bands[i].res[ch]){
j = 0;
mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0] & 0xFF];
for(; j < 12; j++)
c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1] & 0xFF];
for(; j < 24; j++)
c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2] & 0xFF];
for(; j < 36; j++)
c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
}
}
if(bands[i].msf){
int t1, t2;
for(j = 0; j < SAMPLES_PER_BAND; j++){
t1 = c->sb_samples[0][j][i];
t2 = c->sb_samples[1][j][i];
c->sb_samples[0][j][i] = t1 + t2;
c->sb_samples[1][j][i] = t1 - t2;
}
}
}
mpc_synth(c, out, channels);
}
|
