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/log.h"
29 #include "swresample_internal.h"
31 #ifndef CONFIG_RESAMPLE_HP
32 #define FILTER_SHIFT 15
35 #define FELEM2 int32_t
36 #define FELEML int64_t
37 #define FELEM_MAX INT16_MAX
38 #define FELEM_MIN INT16_MIN
40 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
41 #define FILTER_SHIFT 30
44 #define FELEM2 int64_t
45 #define FELEML int64_t
46 #define FELEM_MAX INT32_MAX
47 #define FELEM_MIN INT32_MIN
48 #define WINDOW_TYPE 12
50 #define FILTER_SHIFT 0
55 #define WINDOW_TYPE 24
59 typedef struct ResampleContext {
60 const AVClass *av_class;
68 int compensation_distance;
76 * 0th order modified bessel function of the first kind.
78 static double bessel(double x){
83 static const double inv[100]={
84 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),
85 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),
86 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),
87 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),
88 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),
89 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),
90 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),
91 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),
92 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),
93 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)
97 for(i=0; v != lastv; i++){
106 * builds a polyphase filterbank.
107 * @param factor resampling factor
108 * @param scale wanted sum of coefficients for each filter
109 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
110 * @return 0 on success, negative on error
112 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
115 double *tab = av_malloc(tap_count * sizeof(*tab));
116 const int center= (tap_count-1)/2;
119 return AVERROR(ENOMEM);
121 /* if upsampling, only need to interpolate, no filter */
125 for(ph=0;ph<phase_count;ph++) {
127 for(i=0;i<tap_count;i++) {
128 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
133 const float d= -0.5; //first order derivative = -0.5
134 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
135 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
136 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
139 w = 2.0*x / (factor*tap_count) + M_PI;
140 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
143 w = 2.0*x / (factor*tap_count*M_PI);
144 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
152 /* normalize so that an uniform color remains the same */
153 for(i=0;i<tap_count;i++) {
154 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
155 filter[ph * tap_count + i] = tab[i] / norm;
157 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
165 double sine[LEN + tap_count];
166 double filtered[LEN];
167 double maxff=-2, minff=2, maxsf=-2, minsf=2;
168 for(i=0; i<LEN; i++){
169 double ss=0, sf=0, ff=0;
170 for(j=0; j<LEN+tap_count; j++)
171 sine[j]= cos(i*j*M_PI/LEN);
172 for(j=0; j<LEN; j++){
175 for(k=0; k<tap_count; k++)
176 sum += filter[ph * tap_count + k] * sine[k+j];
177 filtered[j]= sum / (1<<FILTER_SHIFT);
178 ss+= sine[j + center] * sine[j + center];
179 ff+= filtered[j] * filtered[j];
180 sf+= sine[j + center] * filtered[j];
185 maxff= FFMAX(maxff, ff);
186 minff= FFMIN(minff, ff);
187 maxsf= FFMAX(maxsf, sf);
188 minsf= FFMIN(minsf, sf);
190 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);
202 ResampleContext *swri_resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
203 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
204 int phase_count= 1<<phase_shift;
206 if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
207 || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
208 c = av_mallocz(sizeof(*c));
212 c->phase_shift = phase_shift;
213 c->phase_mask = phase_count - 1;
216 c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
217 c->filter_bank = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
220 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
222 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
223 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
226 c->compensation_distance= 0;
227 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
229 c->ideal_dst_incr= c->dst_incr;
231 c->index= -phase_count*((c->filter_length-1)/2);
236 av_free(c->filter_bank);
241 void swri_resample_free(ResampleContext **c){
244 av_freep(&(*c)->filter_bank);
248 void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
249 ResampleContext *c= s->resample;
250 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
251 c->compensation_distance= compensation_distance;
252 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
255 int swri_resample(ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx){
259 int dst_incr_frac= c->dst_incr % c->src_incr;
260 int dst_incr= c->dst_incr / c->src_incr;
261 int compensation_distance= c->compensation_distance;
263 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
264 int64_t index2= ((int64_t)index)<<32;
265 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
266 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
268 for(dst_index=0; dst_index < dst_size; dst_index++){
269 dst[dst_index] = src[index2>>32];
272 index += dst_index * dst_incr;
273 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
274 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
276 for(dst_index=0; dst_index < dst_size; dst_index++){
277 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
278 int sample_index= index >> c->phase_shift;
281 if(sample_index < 0){
282 for(i=0; i<c->filter_length; i++)
283 val += src[FFABS(sample_index + i) % src_size] * filter[i];
284 }else if(sample_index + c->filter_length > src_size){
288 for(i=0; i<c->filter_length; i++){
289 val += src[sample_index + i] * (FELEM2)filter[i];
290 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
292 val+=(v2-val)*(FELEML)frac / c->src_incr;
294 for(i=0; i<c->filter_length; i++){
295 val += src[sample_index + i] * (FELEM2)filter[i];
299 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
300 dst[dst_index] = av_clip_int16(lrintf(val));
302 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
303 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
306 frac += dst_incr_frac;
308 if(frac >= c->src_incr){
313 if(dst_index + 1 == compensation_distance){
314 compensation_distance= 0;
315 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
316 dst_incr= c->ideal_dst_incr / c->src_incr;
320 *consumed= FFMAX(index, 0) >> c->phase_shift;
321 if(index>=0) index &= c->phase_mask;
323 if(compensation_distance){
324 compensation_distance -= dst_index;
325 assert(compensation_distance > 0);
330 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
331 c->compensation_distance= compensation_distance;
334 if(update_ctx && !c->compensation_distance){
336 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
337 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
344 int swri_multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
347 for(i=0; i<dst->ch_count; i++){
348 ret= swri_resample(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);