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
26 * @author Michael Niedermayer <michaelni@gmx.at>
33 #ifndef CONFIG_RESAMPLE_HP
34 #define FILTER_SHIFT 15
37 #define FELEM2 int32_t
38 #define FELEML int64_t
39 #define FELEM_MAX INT16_MAX
40 #define FELEM_MIN INT16_MIN
42 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
43 #define FILTER_SHIFT 30
46 #define FELEM2 int64_t
47 #define FELEML int64_t
48 #define FELEM_MAX INT32_MAX
49 #define FELEM_MIN INT32_MIN
50 #define WINDOW_TYPE 12
52 #define FILTER_SHIFT 0
54 #define FELEM long double
55 #define FELEM2 long double
56 #define FELEML long double
57 #define WINDOW_TYPE 24
61 typedef struct AVResampleContext{
69 int compensation_distance;
76 * 0th order modified bessel function of the first kind.
78 static double bessel(double x){
92 * builds a polyphase filterbank.
93 * @param factor resampling factor
94 * @param scale wanted sum of coefficients for each filter
95 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
97 void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
99 double x, y, w, tab[tap_count];
100 const int center= (tap_count-1)/2;
102 /* if upsampling, only need to interpolate, no filter */
106 for(ph=0;ph<phase_count;ph++) {
108 for(i=0;i<tap_count;i++) {
109 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
114 const float d= -0.5; //first order derivative = -0.5
115 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
116 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
117 else y= d*(-4 + 8*x - 5*x*x + x*x*x);
120 w = 2.0*x / (factor*tap_count) + M_PI;
121 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
124 w = 2.0*x / (factor*tap_count*M_PI);
125 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
133 /* normalize so that an uniform color remains the same */
134 for(i=0;i<tap_count;i++) {
135 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
136 filter[ph * tap_count + i] = tab[i] / norm;
138 filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
146 double sine[LEN + tap_count];
147 double filtered[LEN];
148 double maxff=-2, minff=2, maxsf=-2, minsf=2;
149 for(i=0; i<LEN; i++){
150 double ss=0, sf=0, ff=0;
151 for(j=0; j<LEN+tap_count; j++)
152 sine[j]= cos(i*j*M_PI/LEN);
153 for(j=0; j<LEN; j++){
156 for(k=0; k<tap_count; k++)
157 sum += filter[ph * tap_count + k] * sine[k+j];
158 filtered[j]= sum / (1<<FILTER_SHIFT);
159 ss+= sine[j + center] * sine[j + center];
160 ff+= filtered[j] * filtered[j];
161 sf+= sine[j + center] * filtered[j];
166 maxff= FFMAX(maxff, ff);
167 minff= FFMIN(minff, ff);
168 maxsf= FFMAX(maxsf, sf);
169 minsf= FFMIN(minsf, sf);
171 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 * initalizes a audio resampler.
182 * note, if either rate is not a integer then simply scale both rates up so they are
184 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
185 AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
186 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
187 int phase_count= 1<<phase_shift;
189 c->phase_shift= phase_shift;
190 c->phase_mask= phase_count-1;
193 c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
194 c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
195 av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
196 memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
197 c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
199 c->src_incr= out_rate;
200 c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
201 c->index= -phase_count*((c->filter_length-1)/2);
206 void av_resample_close(AVResampleContext *c){
207 av_freep(&c->filter_bank);
212 * Compensates samplerate/timestamp drift. The compensation is done by changing
213 * the resampler parameters, so no audible clicks or similar distortions ocur
214 * @param compensation_distance distance in output samples over which the compensation should be performed
215 * @param sample_delta number of output samples which should be output less
217 * example: av_resample_compensate(c, 10, 500)
218 * here instead of 510 samples only 500 samples would be output
220 * note, due to rounding the actual compensation might be slightly different,
221 * especially if the compensation_distance is large and the in_rate used during init is small
223 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
224 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
225 c->compensation_distance= compensation_distance;
226 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
231 * @param src an array of unconsumed samples
232 * @param consumed the number of samples of src which have been consumed are returned here
233 * @param src_size the number of unconsumed samples available
234 * @param dst_size the amount of space in samples available in dst
235 * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
236 * @return the number of samples written in dst or -1 if an error occured
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 frac += dst_index * dst_incr_frac;
256 index += dst_index * dst_incr;
257 index += frac / c->src_incr;
260 for(dst_index=0; dst_index < dst_size; dst_index++){
261 FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
262 int sample_index= index >> c->phase_shift;
265 if(sample_index < 0){
266 for(i=0; i<c->filter_length; i++)
267 val += src[FFABS(sample_index + i) % src_size] * filter[i];
268 }else if(sample_index + c->filter_length > src_size){
272 int sub_phase= (frac<<8) / c->src_incr;
273 for(i=0; i<c->filter_length; i++){
274 FELEML coeff= filter[i]*(256 - sub_phase) + filter[i + c->filter_length]*sub_phase;
275 v += src[sample_index + i] * coeff;
279 for(i=0; i<c->filter_length; i++){
280 val += src[sample_index + i] * (FELEM2)filter[i];
284 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
285 dst[dst_index] = av_clip(lrintf(val), -32768, 32767);
287 val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
288 dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
291 frac += dst_incr_frac;
293 if(frac >= c->src_incr){
298 if(dst_index + 1 == compensation_distance){
299 compensation_distance= 0;
300 dst_incr_frac= c->ideal_dst_incr % c->src_incr;
301 dst_incr= c->ideal_dst_incr / c->src_incr;
305 *consumed= FFMAX(index, 0) >> c->phase_shift;
306 if(index>=0) index &= c->phase_mask;
308 if(compensation_distance){
309 compensation_distance -= dst_index;
310 assert(compensation_distance > 0);
315 c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
316 c->compensation_distance= compensation_distance;
319 if(update_ctx && !c->compensation_distance){
321 av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
322 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);