make filter size, int32/int16 and a few other things selectable at compiletime
[libav.git] / libavcodec / resample2.c
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1/*
2 * audio resampling
3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
4 *
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
9 *
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
14 *
15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
18 *
19 */
20
21/**
22 * @file resample2.c
23 * audio resampling
24 * @author Michael Niedermayer <michaelni@gmx.at>
25 */
26
27#include "avcodec.h"
28#include "common.h"
1ac31760 29#include "dsputil.h"
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30
31#define PHASE_SHIFT 10
32#define PHASE_COUNT (1<<PHASE_SHIFT)
33#define PHASE_MASK (PHASE_COUNT-1)
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34#define FILTER_SIZE 16
35//#define LINEAR 1
36
37#if 1
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38#define FILTER_SHIFT 15
39
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40#define FELEM int16_t
41#define FELEM2 int32_t
42#define FELEM_MAX INT16_MAX
43#define FELEM_MIN INT16_MIN
44#else
45#define FILTER_SHIFT 24
46
47#define FELEM int32_t
48#define FELEM2 int64_t
49#define FELEM_MAX INT32_MAX
50#define FELEM_MIN INT32_MIN
51#endif
52
53
aaaf1635 54typedef struct AVResampleContext{
51a88020 55 FELEM *filter_bank;
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56 int filter_length;
57 int ideal_dst_incr;
58 int dst_incr;
59 int index;
60 int frac;
61 int src_incr;
62 int compensation_distance;
63}AVResampleContext;
64
65/**
66 * 0th order modified bessel function of the first kind.
67 */
68double bessel(double x){
69 double v=1;
70 double t=1;
71 int i;
72
73 for(i=1; i<50; i++){
74 t *= i;
75 v += pow(x*x/4, i)/(t*t);
76 }
77 return v;
78}
79
80/**
81 * builds a polyphase filterbank.
82 * @param factor resampling factor
83 * @param scale wanted sum of coefficients for each filter
84 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
85 */
51a88020 86void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
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87 int ph, i, v;
88 double x, y, w, tab[tap_count];
89 const int center= (tap_count-1)/2;
90
91 /* if upsampling, only need to interpolate, no filter */
92 if (factor > 1.0)
93 factor = 1.0;
94
95 for(ph=0;ph<phase_count;ph++) {
96 double norm = 0;
97 double e= 0;
98 for(i=0;i<tap_count;i++) {
99 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
100 if (x == 0) y = 1.0;
101 else y = sin(x) / x;
102 switch(type){
103 case 0:{
104 const float d= -0.5; //first order derivative = -0.5
105 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
106 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
107 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
108 break;}
109 case 1:
110 w = 2.0*x / (factor*tap_count) + M_PI;
111 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
112 break;
113 case 2:
114 w = 2.0*x / (factor*tap_count*M_PI);
08f7073a 115 y *= bessel(16*sqrt(FFMAX(1-w*w, 0)));
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116 break;
117 }
118
119 tab[i] = y;
120 norm += y;
121 }
122
123 /* normalize so that an uniform color remains the same */
124 for(i=0;i<tap_count;i++) {
51a88020 125 v = clip(lrintf(tab[i] * scale / norm + e), FELEM_MIN, FELEM_MAX);
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126 filter[ph * tap_count + i] = v;
127 e += tab[i] * scale / norm - v;
128 }
129 }
130}
131
132/**
133 * initalizes a audio resampler.
134 * note, if either rate is not a integer then simply scale both rates up so they are
135 */
136AVResampleContext *av_resample_init(int out_rate, int in_rate){
137 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
138 double factor= FFMIN(out_rate / (double)in_rate, 1.0);
139
140 memset(c, 0, sizeof(AVResampleContext));
141
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142 c->filter_length= ceil(FILTER_SIZE/factor);
143 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(FELEM));
aaaf1635 144 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
51a88020 145 memcpy(&c->filter_bank[c->filter_length*PHASE_COUNT+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
2ac615da 146 c->filter_bank[c->filter_length*PHASE_COUNT]= c->filter_bank[c->filter_length - 1];
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147
148 c->src_incr= out_rate;
149 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
150 c->index= -PHASE_COUNT*((c->filter_length-1)/2);
151
152 return c;
153}
154
155void av_resample_close(AVResampleContext *c){
156 av_freep(&c->filter_bank);
157 av_freep(&c);
158}
159
160void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
08f7073a 161// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
aaaf1635 162 c->compensation_distance= compensation_distance;
08f7073a 163 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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164}
165
166/**
167 * resamples.
168 * @param src an array of unconsumed samples
169 * @param consumed the number of samples of src which have been consumed are returned here
170 * @param src_size the number of unconsumed samples available
171 * @param dst_size the amount of space in samples available in dst
172 * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
173 * @return the number of samples written in dst or -1 if an error occured
174 */
175int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
176 int dst_index, i;
177 int index= c->index;
178 int frac= c->frac;
179 int dst_incr_frac= c->dst_incr % c->src_incr;
180 int dst_incr= c->dst_incr / c->src_incr;
80e85288 181 int compensation_distance= c->compensation_distance;
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182
183 for(dst_index=0; dst_index < dst_size; dst_index++){
51a88020 184 FELEM *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
aaaf1635 185 int sample_index= index >> PHASE_SHIFT;
51a88020 186 FELEM2 val=0;
80e85288 187
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188 if(sample_index < 0){
189 for(i=0; i<c->filter_length; i++)
b9d2085b 190 val += src[ABS(sample_index + i) % src_size] * filter[i];
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191 }else if(sample_index + c->filter_length > src_size){
192 break;
193 }else{
51a88020 194#ifdef LINEAR
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195 int64_t v=0;
196 int sub_phase= (frac<<12) / c->src_incr;
197 for(i=0; i<c->filter_length; i++){
51a88020 198 int64_t coeff= filter[i]*(FELEM2)(4096 - sub_phase) + filter[i + c->filter_length]*(FELEM2)sub_phase;
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199 v += src[sample_index + i] * coeff;
200 }
201 val= v>>12;
202#else
203 for(i=0; i<c->filter_length; i++){
51a88020 204 val += src[sample_index + i] * (FELEM2)filter[i];
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205 }
206#endif
207 }
208
209 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
210 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
211
212 frac += dst_incr_frac;
213 index += dst_incr;
214 if(frac >= c->src_incr){
215 frac -= c->src_incr;
216 index++;
217 }
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218
219 if(dst_index + 1 == compensation_distance){
220 compensation_distance= 0;
221 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
222 dst_incr= c->ideal_dst_incr / c->src_incr;
223 }
aaaf1635 224 }
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225 *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
226 index= FFMIN(index, 0);
227
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228 if(compensation_distance){
229 compensation_distance -= dst_index;
230 assert(compensation_distance > 0);
231 }
aaaf1635 232 if(update_ctx){
aaaf1635 233 c->frac= frac;
b9d2085b 234 c->index= index;
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235 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
236 c->compensation_distance= compensation_distance;
aaaf1635 237 }
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238#if 0
239 if(update_ctx && !c->compensation_distance){
240#undef rand
241 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
242av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
243 }
244#endif
245
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246 return dst_index;
247}