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
23 * @file libavcodec/resample2.c
25 * @author Michael Niedermayer <michaelni@gmx.at>
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 AVResampleContext{
60 const AVClass *av_class;
68 int compensation_distance;
75 * 0th order modified bessel function of the first kind.
77 static double bessel(double x){
84 for(i=1; v != lastv; i++){
93 * builds a polyphase filterbank.
94 * @param factor resampling factor
95 * @param scale wanted sum of coefficients for each filter
96 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
98 static void build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
100 double x, y, w, tab[tap_count];
101 const int center= (tap_count-1)/2;
103 /* if upsampling, only need to interpolate, no filter */
107 for(ph=0;ph<phase_count;ph++) {
109 for(i=0;i<tap_count;i++) {
110 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
115 const float d= -0.5; //first order derivative = -0.5
116 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
117 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
118 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
121 w = 2.0*x / (factor*tap_count) + M_PI;
122 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
125 w = 2.0*x / (factor*tap_count*M_PI);
126 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
134 /* normalize so that an uniform color remains the same */
135 for(i=0;i<tap_count;i++) {
136 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
137 filter[ph * tap_count + i] = tab[i] / norm;
139 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
147 double sine[LEN + tap_count];
148 double filtered[LEN];
149 double maxff=-2, minff=2, maxsf=-2, minsf=2;
150 for(i=0; i<LEN; i++){
151 double ss=0, sf=0, ff=0;
152 for(j=0; j<LEN+tap_count; j++)
153 sine[j]= cos(i*j*M_PI/LEN);
154 for(j=0; j<LEN; j++){
157 for(k=0; k<tap_count; k++)
158 sum += filter[ph * tap_count + k] * sine[k+j];
159 filtered[j]= sum / (1<<FILTER_SHIFT);
160 ss+= sine[j + center] * sine[j + center];
161 ff+= filtered[j] * filtered[j];
162 sf+= sine[j + center] * filtered[j];
167 maxff= FFMAX(maxff, ff);
168 minff= FFMIN(minff, ff);
169 maxsf= FFMAX(maxsf, sf);
170 minsf= FFMIN(minsf, sf);
172 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);
181 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
182 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
183 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
184 int phase_count= 1<<phase_shift;
186 c->phase_shift= phase_shift;
187 c->phase_mask= phase_count-1;
190 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
191 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
192 build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
193 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
194 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
196 c->src_incr= out_rate;
197 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
198 c->index= -phase_count*((c->filter_length-1)/2);
203 void av_resample_close(AVResampleContext *c){
204 av_freep(&c->filter_bank);
208 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
209 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
210 c->compensation_distance= compensation_distance;
211 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
214 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
218 int dst_incr_frac= c->dst_incr % c->src_incr;
219 int dst_incr= c->dst_incr / c->src_incr;
220 int compensation_distance= c->compensation_distance;
222 if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
223 int64_t index2= ((int64_t)index)<<32;
224 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
225 dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
227 for(dst_index=0; dst_index < dst_size; dst_index++){
228 dst[dst_index] = src[index2>>32];
231 frac += dst_index * dst_incr_frac;
232 index += dst_index * dst_incr;
233 index += frac / c->src_incr;
236 for(dst_index=0; dst_index < dst_size; dst_index++){
237 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
238 int sample_index= index >> c->phase_shift;
241 if(sample_index < 0){
242 for(i=0; i<c->filter_length; i++)
243 val += src[FFABS(sample_index + i) % src_size] * filter[i];
244 }else if(sample_index + c->filter_length > src_size){
248 for(i=0; i<c->filter_length; i++){
249 val += src[sample_index + i] * (FELEM2)filter[i];
250 v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
252 val+=(v2-val)*(FELEML)frac / c->src_incr;
254 for(i=0; i<c->filter_length; i++){
255 val += src[sample_index + i] * (FELEM2)filter[i];
259 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
260 dst[dst_index] = av_clip_int16(lrintf(val));
262 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
263 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
266 frac += dst_incr_frac;
268 if(frac >= c->src_incr){
273 if(dst_index + 1 == compensation_distance){
274 compensation_distance= 0;
275 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
276 dst_incr= c->ideal_dst_incr / c->src_incr;
280 *consumed= FFMAX(index, 0) >> c->phase_shift;
281 if(index>=0) index &= c->phase_mask;
283 if(compensation_distance){
284 compensation_distance -= dst_index;
285 assert(compensation_distance > 0);
290 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
291 c->compensation_distance= compensation_distance;
294 if(update_ctx && !c->compensation_distance){
296 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
297 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);