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
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
|
/*
* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* 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
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/log.h"
#include "swresample_internal.h"
#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
#define FILTER_SHIFT 0
#define FELEM double
#define FELEM2 double
#define FELEML double
#define WINDOW_TYPE 24
#endif
typedef struct AVResampleContext{
const AVClass *av_class;
FELEM *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
double factor;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
static double bessel(double x){
double v=1;
double lastv=0;
double t=1;
int i;
static const double inv[100]={
1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),
1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),
1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),
1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),
1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),
1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),
1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),
1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),
1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),
1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)
};
x= x*x/4;
for(i=0; v != lastv; i++){
lastv=v;
t *= x*inv[i];
v += t;
}
return v;
}
/**
* builds a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
* @return 0 on success, negative on error
*/
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i;
double x, y, w;
double *tab = av_malloc(tap_count * sizeof(*tab));
const int center= (tap_count-1)/2;
if (!tab)
return AVERROR(ENOMEM);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
for(ph=0;ph<phase_count;ph++) {
double norm = 0;
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch(type){
case 0:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case 1:
w = 2.0*x / (factor*tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
break;
default:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
break;
}
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
#endif
}
}
#if 0
{
#define LEN 1024
int j,k;
double sine[LEN + tap_count];
double filtered[LEN];
double maxff=-2, minff=2, maxsf=-2, minsf=2;
for(i=0; i<LEN; i++){
double ss=0, sf=0, ff=0;
for(j=0; j<LEN+tap_count; j++)
sine[j]= cos(i*j*M_PI/LEN);
for(j=0; j<LEN; j++){
double sum=0;
ph=0;
for(k=0; k<tap_count; k++)
sum += filter[ph * tap_count + k] * sine[k+j];
filtered[j]= sum / (1<<FILTER_SHIFT);
ss+= sine[j + center] * sine[j + center];
ff+= filtered[j] * filtered[j];
sf+= sine[j + center] * filtered[j];
}
ss= sqrt(2*ss/LEN);
ff= sqrt(2*ff/LEN);
sf= 2*sf/LEN;
maxff= FFMAX(maxff, ff);
minff= FFMIN(minff, ff);
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
minff=minsf= 2;
maxff=maxsf= -2;
}
}
}
#endif
av_free(tab);
return 0;
}
AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
|| c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
c = av_mallocz(sizeof(AVResampleContext));
if (!c)
return NULL;
c->phase_shift = phase_shift;
c->phase_mask = phase_count - 1;
c->linear = linear;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
if (!c->filter_bank)
goto error;
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
goto error;
memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
}
c->compensation_distance= 0;
if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
goto error;
c->ideal_dst_incr= c->dst_incr;
c->index= -phase_count*((c->filter_length-1)/2);
c->frac= 0;
return c;
error:
av_free(c->filter_bank);
av_free(c);
return NULL;
}
void swr_resample_free(AVResampleContext **c){
if(!*c)
return;
av_freep(&(*c)->filter_bank);
av_freep(c);
}
void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
AVResampleContext *c= s->resample;
// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
c->compensation_distance= compensation_distance;
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
}
int swr_resample(AVResampleContext *c, short *dst, const short *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int compensation_distance= c->compensation_distance;
if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
int64_t index2= ((int64_t)index)<<32;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
for(dst_index=0; dst_index < dst_size; dst_index++){
dst[dst_index] = src[index2>>32];
index2 += incr;
}
frac += dst_index * dst_incr_frac;
index += dst_index * dst_incr;
index += frac / c->src_incr;
frac %= c->src_incr;
}else{
for(dst_index=0; dst_index < dst_size; dst_index++){
FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
int sample_index= index >> c->phase_shift;
FELEM2 val=0;
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
}else if(c->linear){
FELEM2 v2=0;
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
}
val+=(v2-val)*(FELEML)frac / c->src_incr;
}else{
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
}
}
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
#endif
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
if(dst_index + 1 == compensation_distance){
compensation_distance= 0;
dst_incr_frac= c->ideal_dst_incr % c->src_incr;
dst_incr= c->ideal_dst_incr / c->src_incr;
}
}
}
*consumed= FFMAX(index, 0) >> c->phase_shift;
if(index>=0) index &= c->phase_mask;
if(compensation_distance){
compensation_distance -= dst_index;
assert(compensation_distance > 0);
}
if(update_ctx){
c->frac= frac;
c->index= index;
c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
c->compensation_distance= compensation_distance;
}
#if 0
if(update_ctx && !c->compensation_distance){
#undef rand
av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
}
#endif
return dst_index;
}
int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i, ret= -1;
for(i=0; i<dst->ch_count; i++){
ret= swr_resample(c, (short*)dst->ch[i], (const short*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
}
return ret;
}
|