polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample...
[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"
29
30#define PHASE_SHIFT 10
31#define PHASE_COUNT (1<<PHASE_SHIFT)
32#define PHASE_MASK (PHASE_COUNT-1)
33#define FILTER_SHIFT 15
34
35typedef struct AVResampleContext{
36 short *filter_bank;
37 int filter_length;
38 int ideal_dst_incr;
39 int dst_incr;
40 int index;
41 int frac;
42 int src_incr;
43 int compensation_distance;
44}AVResampleContext;
45
46/**
47 * 0th order modified bessel function of the first kind.
48 */
49double bessel(double x){
50 double v=1;
51 double t=1;
52 int i;
53
54 for(i=1; i<50; i++){
55 t *= i;
56 v += pow(x*x/4, i)/(t*t);
57 }
58 return v;
59}
60
61/**
62 * builds a polyphase filterbank.
63 * @param factor resampling factor
64 * @param scale wanted sum of coefficients for each filter
65 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
66 */
67void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
68 int ph, i, v;
69 double x, y, w, tab[tap_count];
70 const int center= (tap_count-1)/2;
71
72 /* if upsampling, only need to interpolate, no filter */
73 if (factor > 1.0)
74 factor = 1.0;
75
76 for(ph=0;ph<phase_count;ph++) {
77 double norm = 0;
78 double e= 0;
79 for(i=0;i<tap_count;i++) {
80 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
81 if (x == 0) y = 1.0;
82 else y = sin(x) / x;
83 switch(type){
84 case 0:{
85 const float d= -0.5; //first order derivative = -0.5
86 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
87 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
88 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
89 break;}
90 case 1:
91 w = 2.0*x / (factor*tap_count) + M_PI;
92 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
93 break;
94 case 2:
95 w = 2.0*x / (factor*tap_count*M_PI);
96 y *= bessel(16*sqrt(FFMAX(1-w*w, 0))) / bessel(16);
97 break;
98 }
99
100 tab[i] = y;
101 norm += y;
102 }
103
104 /* normalize so that an uniform color remains the same */
105 for(i=0;i<tap_count;i++) {
106 v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767);
107 filter[ph * tap_count + i] = v;
108 e += tab[i] * scale / norm - v;
109 }
110 }
111}
112
113/**
114 * initalizes a audio resampler.
115 * note, if either rate is not a integer then simply scale both rates up so they are
116 */
117AVResampleContext *av_resample_init(int out_rate, int in_rate){
118 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
119 double factor= FFMIN(out_rate / (double)in_rate, 1.0);
120
121 memset(c, 0, sizeof(AVResampleContext));
122
123 c->filter_length= ceil(16.0/factor);
124 c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
125 av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
126 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 1]= (1<<FILTER_SHIFT)-1;
127 c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 1;
128
129 c->src_incr= out_rate;
130 c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
131 c->index= -PHASE_COUNT*((c->filter_length-1)/2);
132
133 return c;
134}
135
136void av_resample_close(AVResampleContext *c){
137 av_freep(&c->filter_bank);
138 av_freep(&c);
139}
140
141void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
142 assert(!c->compensation_distance); //FIXME
143
144 c->compensation_distance= compensation_distance;
145 c->dst_incr-= c->ideal_dst_incr * sample_delta / compensation_distance;
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;
163
164 if(c->compensation_distance && c->compensation_distance < dst_size)
165 dst_size= c->compensation_distance;
166
167 for(dst_index=0; dst_index < dst_size; dst_index++){
168 short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
169 int sample_index= index >> PHASE_SHIFT;
170 int val=0;
171
172 if(sample_index < 0){
173 for(i=0; i<c->filter_length; i++)
174 val += src[ABS(sample_index + i)] * filter[i];
175 }else if(sample_index + c->filter_length > src_size){
176 break;
177 }else{
178#if 0
179 int64_t v=0;
180 int sub_phase= (frac<<12) / c->src_incr;
181 for(i=0; i<c->filter_length; i++){
182 int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
183 v += src[sample_index + i] * coeff;
184 }
185 val= v>>12;
186#else
187 for(i=0; i<c->filter_length; i++){
188 val += src[sample_index + i] * filter[i];
189 }
190#endif
191 }
192
193 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
194 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
195
196 frac += dst_incr_frac;
197 index += dst_incr;
198 if(frac >= c->src_incr){
199 frac -= c->src_incr;
200 index++;
201 }
202 }
203 if(update_ctx){
204 if(c->compensation_distance){
205 c->compensation_distance -= index;
206 if(!c->compensation_distance)
207 c->dst_incr= c->ideal_dst_incr;
208 }
209 c->frac= frac;
210 c->index=0;
211 }
212 *consumed= index >> PHASE_SHIFT;
213 return dst_index;
214}