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"
31 #include "libavutil/common.h"
33 #if FF_API_AVCODEC_RESAMPLE
35 #ifndef CONFIG_RESAMPLE_HP
36 #define FILTER_SHIFT 15
39 #define FELEM2 int32_t
40 #define FELEML int64_t
41 #define FELEM_MAX INT16_MAX
42 #define FELEM_MIN INT16_MIN
44 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
45 #define FILTER_SHIFT 30
48 #define FELEM2 int64_t
49 #define FELEML int64_t
50 #define FELEM_MAX INT32_MAX
51 #define FELEM_MIN INT32_MIN
52 #define WINDOW_TYPE 12
54 #define FILTER_SHIFT 0
59 #define WINDOW_TYPE 24
63 typedef struct AVResampleContext{
64 const AVClass *av_class;
72 int compensation_distance;
79 * 0th order modified bessel function of the first kind.
81 static double bessel(double x){
88 for(i=1; v != lastv; i++){
97 * Build a polyphase filterbank.
98 * @param factor resampling factor
99 * @param scale wanted sum of coefficients for each filter
100 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
101 * @return 0 on success, negative on error
103 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
106 double *tab = av_malloc(tap_count * sizeof(*tab));
107 const int center= (tap_count-1)/2;
110 return AVERROR(ENOMEM);
112 /* if upsampling, only need to interpolate, no filter */
116 for(ph=0;ph<phase_count;ph++) {
118 for(i=0;i<tap_count;i++) {
119 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
124 const float d= -0.5; //first order derivative = -0.5
125 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
126 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
127 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
130 w = 2.0*x / (factor*tap_count) + M_PI;
131 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
134 w = 2.0*x / (factor*tap_count*M_PI);
135 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
143 /* normalize so that an uniform color remains the same */
144 for(i=0;i<tap_count;i++) {
145 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
146 filter[ph * tap_count + i] = tab[i] / norm;
148 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
156 double sine[LEN + tap_count];
157 double filtered[LEN];
158 double maxff=-2, minff=2, maxsf=-2, minsf=2;
159 for(i=0; i<LEN; i++){
160 double ss=0, sf=0, ff=0;
161 for(j=0; j<LEN+tap_count; j++)
162 sine[j]= cos(i*j*M_PI/LEN);
163 for(j=0; j<LEN; j++){
166 for(k=0; k<tap_count; k++)
167 sum += filter[ph * tap_count + k] * sine[k+j];
168 filtered[j]= sum / (1<<FILTER_SHIFT);
169 ss+= sine[j + center] * sine[j + center];
170 ff+= filtered[j] * filtered[j];
171 sf+= sine[j + center] * filtered[j];
176 maxff= FFMAX(maxff, ff);
177 minff= FFMIN(minff, ff);
178 maxsf= FFMAX(maxsf, sf);
179 minsf= FFMIN(minsf, sf);
181 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);
193 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
194 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
195 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
196 int phase_count= 1<<phase_shift;
201 c->phase_shift= phase_shift;
202 c->phase_mask= phase_count-1;
205 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
206 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
209 if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
211 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
212 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
214 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
216 c->ideal_dst_incr= c->dst_incr;
218 c->index= -phase_count*((c->filter_length-1)/2);
222 av_free(c->filter_bank);
227 void av_resample_close(AVResampleContext *c){
228 av_freep(&c->filter_bank);
232 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
233 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
234 c->compensation_distance= compensation_distance;
235 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
238 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
242 int dst_incr_frac= c->dst_incr % c->src_incr;
243 int dst_incr= c->dst_incr / c->src_incr;
244 int compensation_distance= c->compensation_distance;
246 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
247 int64_t index2= ((int64_t)index)<<32;
248 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
249 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
251 for(dst_index=0; dst_index < dst_size; dst_index++){
252 dst[dst_index] = src[index2>>32];
255 index += dst_index * dst_incr;
256 index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
257 frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
259 for(dst_index=0; dst_index < dst_size; dst_index++){
260 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
261 int sample_index= index >> c->phase_shift;
264 if(sample_index < 0){
265 for(i=0; i<c->filter_length; i++)
266 val += src[FFABS(sample_index + i) % src_size] * filter[i];
267 }else if(sample_index + c->filter_length > src_size){
271 for(i=0; i<c->filter_length; i++){
272 val += src[sample_index + i] * (FELEM2)filter[i];
273 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
275 val+=(v2-val)*(FELEML)frac / c->src_incr;
277 for(i=0; i<c->filter_length; i++){
278 val += src[sample_index + i] * (FELEM2)filter[i];
282 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
283 dst[dst_index] = av_clip_int16(lrintf(val));
285 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
286 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
289 frac += dst_incr_frac;
291 if(frac >= c->src_incr){
296 if(dst_index + 1 == compensation_distance){
297 compensation_distance= 0;
298 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
299 dst_incr= c->ideal_dst_incr / c->src_incr;
303 *consumed= FFMAX(index, 0) >> c->phase_shift;
304 if(index>=0) index &= c->phase_mask;
306 if(compensation_distance){
307 compensation_distance -= dst_index;
308 av_assert2(compensation_distance > 0);
313 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
314 c->compensation_distance= compensation_distance;
317 if(update_ctx && !c->compensation_distance){
319 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
320 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);