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[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)
34#define FILTER_SHIFT 15
35
36typedef struct AVResampleContext{
37 short *filter_bank;
38 int filter_length;
39 int ideal_dst_incr;
40 int dst_incr;
41 int index;
42 int frac;
43 int src_incr;
44 int compensation_distance;
45}AVResampleContext;
46
47/**
48 * 0th order modified bessel function of the first kind.
49 */
50double bessel(double x){
51 double v=1;
52 double t=1;
53 int i;
54
55 for(i=1; i<50; i++){
56 t *= i;
57 v += pow(x*x/4, i)/(t*t);
58 }
59 return v;
60}
61
62/**
63 * builds a polyphase filterbank.
64 * @param factor resampling factor
65 * @param scale wanted sum of coefficients for each filter
66 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
67 */
68void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
69 int ph, i, v;
70 double x, y, w, tab[tap_count];
71 const int center= (tap_count-1)/2;
72
73 /* if upsampling, only need to interpolate, no filter */
74 if (factor > 1.0)
75 factor = 1.0;
76
77 for(ph=0;ph<phase_count;ph++) {
78 double norm = 0;
79 double e= 0;
80 for(i=0;i<tap_count;i++) {
81 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
82 if (x == 0) y = 1.0;
83 else y = sin(x) / x;
84 switch(type){
85 case 0:{
86 const float d= -0.5; //first order derivative = -0.5
87 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
88 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
89 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
90 break;}
91 case 1:
92 w = 2.0*x / (factor*tap_count) + M_PI;
93 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
94 break;
95 case 2:
96 w = 2.0*x / (factor*tap_count*M_PI);
08f7073a 97 y *= bessel(16*sqrt(FFMAX(1-w*w, 0)));
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98 break;
99 }
100
101 tab[i] = y;
102 norm += y;
103 }
104
105 /* normalize so that an uniform color remains the same */
106 for(i=0;i<tap_count;i++) {
bb22e8b1 107 v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767);
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108 filter[ph * tap_count + i] = v;
109 e += tab[i] * scale / norm - v;
110 }
111 }
112}
113
114/**
115 * initalizes a audio resampler.
116 * note, if either rate is not a integer then simply scale both rates up so they are
117 */
118AVResampleContext *av_resample_init(int out_rate, int in_rate){
119 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
120 double factor= FFMIN(out_rate / (double)in_rate, 1.0);
121
122 memset(c, 0, sizeof(AVResampleContext));
123
124 c->filter_length= ceil(16.0/factor);
125 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
126 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
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127 memcpy(&c->filter_bank[c->filter_length*PHASE_COUNT+1], c->filter_bank, (c->filter_length-1)*sizeof(short));
128 c->filter_bank[c->filter_length*PHASE_COUNT]= c->filter_bank[c->filter_length - 1];
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129
130 c->src_incr= out_rate;
131 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
132 c->index= -PHASE_COUNT*((c->filter_length-1)/2);
133
134 return c;
135}
136
137void av_resample_close(AVResampleContext *c){
138 av_freep(&c->filter_bank);
139 av_freep(&c);
140}
141
142void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
08f7073a 143// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
aaaf1635 144 c->compensation_distance= compensation_distance;
08f7073a 145 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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146}
147
148/**
149 * resamples.
150 * @param src an array of unconsumed samples
151 * @param consumed the number of samples of src which have been consumed are returned here
152 * @param src_size the number of unconsumed samples available
153 * @param dst_size the amount of space in samples available in dst
154 * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
155 * @return the number of samples written in dst or -1 if an error occured
156 */
157int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
158 int dst_index, i;
159 int index= c->index;
160 int frac= c->frac;
161 int dst_incr_frac= c->dst_incr % c->src_incr;
162 int dst_incr= c->dst_incr / c->src_incr;
80e85288 163 int compensation_distance= c->compensation_distance;
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164
165 for(dst_index=0; dst_index < dst_size; dst_index++){
166 short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
167 int sample_index= index >> PHASE_SHIFT;
168 int val=0;
80e85288 169
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170 if(sample_index < 0){
171 for(i=0; i<c->filter_length; i++)
b9d2085b 172 val += src[ABS(sample_index + i) % src_size] * filter[i];
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173 }else if(sample_index + c->filter_length > src_size){
174 break;
175 }else{
176#if 0
177 int64_t v=0;
178 int sub_phase= (frac<<12) / c->src_incr;
179 for(i=0; i<c->filter_length; i++){
180 int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
181 v += src[sample_index + i] * coeff;
182 }
183 val= v>>12;
184#else
185 for(i=0; i<c->filter_length; i++){
186 val += src[sample_index + i] * filter[i];
187 }
188#endif
189 }
190
191 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
192 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
193
194 frac += dst_incr_frac;
195 index += dst_incr;
196 if(frac >= c->src_incr){
197 frac -= c->src_incr;
198 index++;
199 }
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200
201 if(dst_index + 1 == compensation_distance){
202 compensation_distance= 0;
203 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
204 dst_incr= c->ideal_dst_incr / c->src_incr;
205 }
aaaf1635 206 }
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207 *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
208 index= FFMIN(index, 0);
209
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210 if(compensation_distance){
211 compensation_distance -= dst_index;
212 assert(compensation_distance > 0);
213 }
aaaf1635 214 if(update_ctx){
aaaf1635 215 c->frac= frac;
b9d2085b 216 c->index= index;
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217 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
218 c->compensation_distance= compensation_distance;
aaaf1635 219 }
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220#if 0
221 if(update_ctx && !c->compensation_distance){
222#undef rand
223 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
224av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
225 }
226#endif
227
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228 return dst_index;
229}