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
|
/*
* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This library 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 of the License, or (at your option) any later version.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/**
* @file resample2.c
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "avcodec.h"
#include "common.h"
#include "dsputil.h"
#define PHASE_SHIFT 10
#define PHASE_COUNT (1<<PHASE_SHIFT)
#define PHASE_MASK (PHASE_COUNT-1)
#define FILTER_SHIFT 15
typedef struct AVResampleContext{
short *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
double bessel(double x){
double v=1;
double t=1;
int i;
for(i=1; i<50; i++){
t *= i;
v += pow(x*x/4, i)/(t*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->kaiser windowed sinc beta=16
*/
void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i, v;
double x, y, w, tab[tap_count];
const int center= (tap_count-1)/2;
/* 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;
double e= 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;
case 2:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(16*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++) {
v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767);
filter[ph * tap_count + i] = v;
e += tab[i] * scale / norm - v;
}
}
}
/**
* initalizes a audio resampler.
* note, if either rate is not a integer then simply scale both rates up so they are
*/
AVResampleContext *av_resample_init(int out_rate, int in_rate){
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
double factor= FFMIN(out_rate / (double)in_rate, 1.0);
memset(c, 0, sizeof(AVResampleContext));
c->filter_length= ceil(16.0/factor);
c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 1;
c->src_incr= out_rate;
c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
c->index= -PHASE_COUNT*((c->filter_length-1)/2);
return c;
}
void av_resample_close(AVResampleContext *c){
av_freep(&c->filter_bank);
av_freep(&c);
}
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
// 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;
}
/**
* resamples.
* @param src an array of unconsumed samples
* @param consumed the number of samples of src which have been consumed are returned here
* @param src_size the number of unconsumed samples available
* @param dst_size the amount of space in samples available in dst
* @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
* @return the number of samples written in dst or -1 if an error occured
*/
int av_resample(AVResampleContext *c, short *dst, 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;
if(c->compensation_distance && c->compensation_distance < dst_size)
dst_size= c->compensation_distance;
for(dst_index=0; dst_index < dst_size; dst_index++){
short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
int sample_index= index >> PHASE_SHIFT;
int val=0;
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[ABS(sample_index + i) % src_size] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
}else{
#if 0
int64_t v=0;
int sub_phase= (frac<<12) / c->src_incr;
for(i=0; i<c->filter_length; i++){
int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
v += src[sample_index + i] * coeff;
}
val= v>>12;
#else
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * filter[i];
}
#endif
}
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
}
*consumed= FFMAX(index, 0) >> PHASE_SHIFT;
index= FFMIN(index, 0);
if(update_ctx){
if(c->compensation_distance){
c->compensation_distance -= dst_index;
if(!c->compensation_distance)
c->dst_incr= c->ideal_dst_incr;
}
c->frac= frac;
c->index= index;
}
#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;
}
|
