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
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
|
/*
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "libavutil/mem.h"
#include "dct32.h"
#include "mathops.h"
#include "mpegaudiodsp.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"
#if CONFIG_FLOAT
#define RENAME(n) n##_float
static inline float round_sample(float *sum)
{
float sum1=*sum;
*sum = 0;
return sum1;
}
#define MACS(rt, ra, rb) rt+=(ra)*(rb)
#define MULS(ra, rb) ((ra)*(rb))
#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
#else
#define RENAME(n) n##_fixed
#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
static inline int round_sample(int64_t *sum)
{
int sum1;
sum1 = (int)((*sum) >> OUT_SHIFT);
*sum &= (1<<OUT_SHIFT)-1;
return av_clip_int16(sum1);
}
# define MULS(ra, rb) MUL64(ra, rb)
# define MACS(rt, ra, rb) MAC64(rt, ra, rb)
# define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
#endif
DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
#define SUM8(op, sum, w, p) \
{ \
op(sum, (w)[0 * 64], (p)[0 * 64]); \
op(sum, (w)[1 * 64], (p)[1 * 64]); \
op(sum, (w)[2 * 64], (p)[2 * 64]); \
op(sum, (w)[3 * 64], (p)[3 * 64]); \
op(sum, (w)[4 * 64], (p)[4 * 64]); \
op(sum, (w)[5 * 64], (p)[5 * 64]); \
op(sum, (w)[6 * 64], (p)[6 * 64]); \
op(sum, (w)[7 * 64], (p)[7 * 64]); \
}
#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
{ \
INTFLOAT tmp;\
tmp = p[0 * 64];\
op1(sum1, (w1)[0 * 64], tmp);\
op2(sum2, (w2)[0 * 64], tmp);\
tmp = p[1 * 64];\
op1(sum1, (w1)[1 * 64], tmp);\
op2(sum2, (w2)[1 * 64], tmp);\
tmp = p[2 * 64];\
op1(sum1, (w1)[2 * 64], tmp);\
op2(sum2, (w2)[2 * 64], tmp);\
tmp = p[3 * 64];\
op1(sum1, (w1)[3 * 64], tmp);\
op2(sum2, (w2)[3 * 64], tmp);\
tmp = p[4 * 64];\
op1(sum1, (w1)[4 * 64], tmp);\
op2(sum2, (w2)[4 * 64], tmp);\
tmp = p[5 * 64];\
op1(sum1, (w1)[5 * 64], tmp);\
op2(sum2, (w2)[5 * 64], tmp);\
tmp = p[6 * 64];\
op1(sum1, (w1)[6 * 64], tmp);\
op2(sum2, (w2)[6 * 64], tmp);\
tmp = p[7 * 64];\
op1(sum1, (w1)[7 * 64], tmp);\
op2(sum2, (w2)[7 * 64], tmp);\
}
void RENAME(ff_mpadsp_apply_window)(MPA_INT *synth_buf, MPA_INT *window,
int *dither_state, OUT_INT *samples,
int incr)
{
register const MPA_INT *w, *w2, *p;
int j;
OUT_INT *samples2;
#if CONFIG_FLOAT
float sum, sum2;
#else
int64_t sum, sum2;
#endif
/* copy to avoid wrap */
memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
samples2 = samples + 31 * incr;
w = window;
w2 = window + 31;
sum = *dither_state;
p = synth_buf + 16;
SUM8(MACS, sum, w, p);
p = synth_buf + 48;
SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
samples += incr;
w++;
/* we calculate two samples at the same time to avoid one memory
access per two sample */
for(j=1;j<16;j++) {
sum2 = 0;
p = synth_buf + 16 + j;
SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
p = synth_buf + 48 - j;
SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
*samples = round_sample(&sum);
samples += incr;
sum += sum2;
*samples2 = round_sample(&sum);
samples2 -= incr;
w++;
w2--;
}
p = synth_buf + 32;
SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
*dither_state= sum;
}
/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
32 samples. */
void RENAME(ff_mpa_synth_filter)(MPADSPContext *s, MPA_INT *synth_buf_ptr,
int *synth_buf_offset,
MPA_INT *window, int *dither_state,
OUT_INT *samples, int incr,
MPA_INT *sb_samples)
{
MPA_INT *synth_buf;
int offset;
offset = *synth_buf_offset;
synth_buf = synth_buf_ptr + offset;
s->RENAME(dct32)(synth_buf, sb_samples);
s->RENAME(apply_window)(synth_buf, window, dither_state, samples, incr);
offset = (offset - 32) & 511;
*synth_buf_offset = offset;
}
void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
{
int i, j;
/* max = 18760, max sum over all 16 coefs : 44736 */
for(i=0;i<257;i++) {
INTFLOAT v;
v = ff_mpa_enwindow[i];
#if CONFIG_FLOAT
v *= 1.0 / (1LL<<(16 + FRAC_BITS));
#endif
window[i] = v;
if ((i & 63) != 0)
v = -v;
if (i != 0)
window[512 - i] = v;
}
// Needed for avoiding shuffles in ASM implementations
for(i=0; i < 8; i++)
for(j=0; j < 16; j++)
window[512+16*i+j] = window[64*i+32-j];
for(i=0; i < 8; i++)
for(j=0; j < 16; j++)
window[512+128+16*i+j] = window[64*i+48-j];
}
|