3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 * @author Michael Niedermayer <michaelni@gmx.at>
28 #include "libavutil/avassert.h"
32 #ifndef CONFIG_RESAMPLE_HP
33 #define FILTER_SHIFT 15
36 #define FELEM2 int32_t
37 #define FELEML int64_t
38 #define FELEM_MAX INT16_MAX
39 #define FELEM_MIN INT16_MIN
41 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
42 #define FILTER_SHIFT 30
45 #define FELEM2 int64_t
46 #define FELEML int64_t
47 #define FELEM_MAX INT32_MAX
48 #define FELEM_MIN INT32_MIN
49 #define WINDOW_TYPE 12
51 #define FILTER_SHIFT 0
56 #define WINDOW_TYPE 24
60 typedef struct AVResampleContext{
61 const AVClass *av_class;
69 int compensation_distance;
76 * 0th order modified bessel function of the first kind.
78 static double bessel(double x){
85 for(i=1; v != lastv; i++){
94 * Build a polyphase filterbank.
95 * @param factor resampling factor
96 * @param scale wanted sum of coefficients for each filter
97 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
98 * @return 0 on success, negative on error
100 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
103 double *tab = av_malloc(tap_count * sizeof(*tab));
104 const int center= (tap_count-1)/2;
107 return AVERROR(ENOMEM);
109 /* if upsampling, only need to interpolate, no filter */
113 for(ph=0;ph<phase_count;ph++) {
115 for(i=0;i<tap_count;i++) {
116 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
121 const float d= -0.5; //first order derivative = -0.5
122 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
123 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
124 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
127 w = 2.0*x / (factor*tap_count) + M_PI;
128 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
131 w = 2.0*x / (factor*tap_count*M_PI);
132 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
140 /* normalize so that an uniform color remains the same */
141 for(i=0;i<tap_count;i++) {
142 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
143 filter[ph * tap_count + i] = tab[i] / norm;
145 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
153 double sine[LEN + tap_count];
154 double filtered[LEN];
155 double maxff=-2, minff=2, maxsf=-2, minsf=2;
156 for(i=0; i<LEN; i++){
157 double ss=0, sf=0, ff=0;
158 for(j=0; j<LEN+tap_count; j++)
159 sine[j]= cos(i*j*M_PI/LEN);
160 for(j=0; j<LEN; j++){
163 for(k=0; k<tap_count; k++)
164 sum += filter[ph * tap_count + k] * sine[k+j];
165 filtered[j]= sum / (1<<FILTER_SHIFT);
166 ss+= sine[j + center] * sine[j + center];
167 ff+= filtered[j] * filtered[j];
168 sf+= sine[j + center] * filtered[j];
173 maxff= FFMAX(maxff, ff);
174 minff= FFMIN(minff, ff);
175 maxsf= FFMAX(maxsf, sf);
176 minsf= FFMIN(minsf, sf);
178 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);
190 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
191 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
192 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
193 int phase_count= 1<<phase_shift;
198 c->phase_shift= phase_shift;
199 c->phase_mask= phase_count-1;
202 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
203 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
206 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
208 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
209 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
211 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
213 c->ideal_dst_incr= c->dst_incr;
215 c->index= -phase_count*((c->filter_length-1)/2);
219 av_free(c->filter_bank);
224 void av_resample_close(AVResampleContext *c){
225 av_freep(&c->filter_bank);
229 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
230 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
231 c->compensation_distance= compensation_distance;
232 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
235 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
239 int dst_incr_frac= c->dst_incr % c->src_incr;
240 int dst_incr= c->dst_incr / c->src_incr;
241 int compensation_distance= c->compensation_distance;
243 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
244 int64_t index2= ((int64_t)index)<<32;
245 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
246 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
248 for(dst_index=0; dst_index < dst_size; dst_index++){
249 dst[dst_index] = src[index2>>32];
252 index += dst_index * dst_incr;
253 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
254 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
256 for(dst_index=0; dst_index < dst_size; dst_index++){
257 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
258 int sample_index= index >> c->phase_shift;
261 if(sample_index < 0){
262 for(i=0; i<c->filter_length; i++)
263 val += src[FFABS(sample_index + i) % src_size] * filter[i];
264 }else if(sample_index + c->filter_length > src_size){
268 for(i=0; i<c->filter_length; i++){
269 val += src[sample_index + i] * (FELEM2)filter[i];
270 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
272 val+=(v2-val)*(FELEML)frac / c->src_incr;
274 for(i=0; i<c->filter_length; i++){
275 val += src[sample_index + i] * (FELEM2)filter[i];
279 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
280 dst[dst_index] = av_clip_int16(lrintf(val));
282 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
283 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
286 frac += dst_incr_frac;
288 if(frac >= c->src_incr){
293 if(dst_index + 1 == compensation_distance){
294 compensation_distance= 0;
295 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
296 dst_incr= c->ideal_dst_incr / c->src_incr;
300 *consumed= FFMAX(index, 0) >> c->phase_shift;
301 if(index>=0) index &= c->phase_mask;
303 if(compensation_distance){
304 compensation_distance -= dst_index;
305 av_assert2(compensation_distance > 0);
310 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
311 c->compensation_distance= compensation_distance;
314 if(update_ctx && !c->compensation_distance){
316 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
317 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);