FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libswresample/resample.c
Date: 2025-11-28 06:05:02
Exec Total Coverage
Lines: 243 294 82.7%
Functions: 9 10 90.0%
Branches: 146 189 77.2%
Line Branch Exec Source
1 /*
2 * audio resampling
3 * Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
4 * bessel function: Copyright (c) 2006 Xiaogang Zhang
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23 /**
24 * @file
25 * audio resampling
26 * @author Michael Niedermayer <michaelni@gmx.at>
27 */
28
29 #include "libavutil/avassert.h"
30 #include "libavutil/mem.h"
31 #include "resample.h"
32
33 /**
34 * builds a polyphase filterbank.
35 * @param factor resampling factor
36 * @param scale wanted sum of coefficients for each filter
37 * @param filter_type filter type
38 * @param kaiser_beta kaiser window beta
39 * @return 0 on success, negative on error
40 */
41 724 static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
42 int filter_type, double kaiser_beta){
43 int ph, i;
44
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 724 int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1;
45 double x, y, w, t, s;
46 724 double *tab = av_malloc_array(tap_count+1, sizeof(*tab));
47 724 double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut));
48 724 const int center= (tap_count-1)/2;
49 724 double norm = 0;
50 724 int ret = AVERROR(ENOMEM);
51
52
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 724 if (!tab || !sin_lut)
53 ✗ goto fail;
54
55
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 724 av_assert0(tap_count == 1 || tap_count % 2 == 0);
56
57 /* if upsampling, only need to interpolate, no filter */
58
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 724 if (factor > 1.0)
59 ✗ factor = 1.0;
60
61
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 724 if (factor == 1.0) {
62
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 118328 for (ph = 0; ph < ph_nb; ph++)
63
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 117953 sin_lut[ph] = sin(M_PI * ph / phase_count) * (center & 1 ? 1 : -1);
64 }
65
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 225452 for(ph = 0; ph < ph_nb; ph++) {
66 224728 s = sin_lut[ph];
67
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 42402490 for(i=0;i<tap_count;i++) {
68 42177762 x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
69
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 42177762 if (x == 0) y = 1.0;
70
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 42177038 else if (factor == 1.0)
71 3772571 y = s / x;
72 else
73 38404467 y = sin(x) / x;
74
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 42177762 switch(filter_type){
75 ✗ case SWR_FILTER_TYPE_CUBIC:{
76 ✗ const float d= -0.5; //first order derivative = -0.5
77 ✗ x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
78 ✗ if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
79 ✗ else y= d*(-4 + 8*x - 5*x*x + x*x*x);
80 ✗ break;}
81 ✗ case SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
82 ✗ w = 2.0*x / (factor*tap_count);
83 ✗ t = -cos(w);
84 ✗ y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t);
85 ✗ break;
86 42177762 case SWR_FILTER_TYPE_KAISER:
87 42177762 w = 2.0*x / (factor*tap_count*M_PI);
88
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 42177762 y *= av_bessel_i0(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
89 42177762 break;
90 ✗ default:
91 ✗ av_assert0(0);
92 }
93
94 42177762 tab[i] = y;
95 42177762 s = -s;
96
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 42177762 if (!ph)
97 107786 norm += y;
98 }
99
100 /* normalize so that an uniform color remains the same */
101
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 224728 switch(c->format){
102 60220 case AV_SAMPLE_FMT_S16P:
103
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 10932903 for(i=0;i<tap_count;i++)
104 10872683 ((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm));
105
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 60220 if (phase_count % 2) break;
106
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 10721045 for (i = 0; i < tap_count; i++)
107 10666284 ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i];
108 54761 break;
109 49672 case AV_SAMPLE_FMT_S32P:
110
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 10024056 for(i=0;i<tap_count;i++)
111 9974384 ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm));
112
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 49672 if (phase_count % 2) break;
113
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 9817748 for (i = 0; i < tap_count; i++)
114 9773384 ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i];
115 44364 break;
116 59008 case AV_SAMPLE_FMT_FLTP:
117
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 10890007 for(i=0;i<tap_count;i++)
118 10830999 ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
119
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 59008 if (phase_count % 2) break;
120
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 10682609 for (i = 0; i < tap_count; i++)
121 10628954 ((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i];
122 53655 break;
123 55828 case AV_SAMPLE_FMT_DBLP:
124
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 10555524 for(i=0;i<tap_count;i++)
125 10499696 ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
126
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 55828 if (phase_count % 2) break;
127
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 10349216 for (i = 0; i < tap_count; i++)
128 10298696 ((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i];
129 50520 break;
130 }
131 }
132 #if 0
133 {
134 #define LEN 1024
135 int j,k;
136 double sine[LEN + tap_count];
137 double filtered[LEN];
138 double maxff=-2, minff=2, maxsf=-2, minsf=2;
139 for(i=0; i<LEN; i++){
140 double ss=0, sf=0, ff=0;
141 for(j=0; j<LEN+tap_count; j++)
142 sine[j]= cos(i*j*M_PI/LEN);
143 for(j=0; j<LEN; j++){
144 double sum=0;
145 ph=0;
146 for(k=0; k<tap_count; k++)
147 sum += filter[ph * tap_count + k] * sine[k+j];
148 filtered[j]= sum / (1<<FILTER_SHIFT);
149 ss+= sine[j + center] * sine[j + center];
150 ff+= filtered[j] * filtered[j];
151 sf+= sine[j + center] * filtered[j];
152 }
153 ss= sqrt(2*ss/LEN);
154 ff= sqrt(2*ff/LEN);
155 sf= 2*sf/LEN;
156 maxff= FFMAX(maxff, ff);
157 minff= FFMIN(minff, ff);
158 maxsf= FFMAX(maxsf, sf);
159 minsf= FFMIN(minsf, sf);
160 if(i%11==0){
161 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);
162 minff=minsf= 2;
163 maxff=maxsf= -2;
164 }
165 }
166 }
167 #endif
168
169 724 ret = 0;
170 724 fail:
171 724 av_free(tab);
172 724 av_free(sin_lut);
173 724 return ret;
174 }
175
176 3079 static void resample_free(ResampleContext **cc){
177 3079 ResampleContext *c = *cc;
178
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 3079 if(!c)
179 2404 return;
180 675 av_freep(&c->filter_bank);
181 675 av_freep(cc);
182 }
183
184 682 static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
185 double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta,
186 double precision, int cheby, int exact_rational)
187 {
188
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 682 double cutoff = cutoff0? cutoff0 : 0.97;
189
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 682 double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
190 682 int phase_count= 1<<phase_shift;
191 682 int phase_count_compensation = phase_count;
192 682 int filter_length = FFMAX((int)ceil(filter_size/factor), 1);
193
194
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 682 if (filter_length > 1)
195 680 filter_length = FFALIGN(filter_length, 2);
196
197
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 682 if (exact_rational) {
198 int phase_count_exact, phase_count_exact_den;
199
200 474 av_reduce(&phase_count_exact, &phase_count_exact_den, out_rate, in_rate, INT_MAX);
201
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 474 if (phase_count_exact <= phase_count) {
202 342 phase_count_compensation = phase_count_exact * (phase_count / phase_count_exact);
203 342 phase_count = phase_count_exact;
204 }
205 }
206
207
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 682 if (!c || c->phase_count != phase_count || c->linear!=linear || c->factor != factor
208
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 7 || c->filter_length != filter_length || c->format != format
209
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 7 || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
210 675 resample_free(&c);
211 675 c = av_mallocz(sizeof(*c));
212
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 675 if (!c)
213 ✗ return NULL;
214
215 675 c->format= format;
216
217 675 c->felem_size= av_get_bytes_per_sample(c->format);
218
219
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 675 switch(c->format){
220 176 case AV_SAMPLE_FMT_S16P:
221 176 c->filter_shift = 15;
222 176 break;
223 144 case AV_SAMPLE_FMT_S32P:
224 144 c->filter_shift = 30;
225 144 break;
226 355 case AV_SAMPLE_FMT_FLTP:
227 case AV_SAMPLE_FMT_DBLP:
228 355 c->filter_shift = 0;
229 355 break;
230 ✗ default:
231 ✗ av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
232 ✗ av_assert0(0);
233 }
234
235
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 675 if (filter_size/factor > INT32_MAX/256) {
236 ✗ av_log(NULL, AV_LOG_ERROR, "Filter length too large\n");
237 ✗ goto error;
238 }
239
240 675 c->phase_count = phase_count;
241 675 c->linear = linear;
242 675 c->factor = factor;
243 675 c->filter_length = filter_length;
244 675 c->filter_alloc = FFALIGN(c->filter_length, 8);
245 675 c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size);
246 675 c->filter_type = filter_type;
247 675 c->kaiser_beta = kaiser_beta;
248 675 c->phase_count_compensation = phase_count_compensation;
249
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 675 if (!c->filter_bank)
250 ✗ goto error;
251
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 675 if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta))
252 ✗ goto error;
253 675 memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
254 675 memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
255 }
256
257 682 c->compensation_distance= 0;
258
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 682 if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
259 ✗ goto error;
260
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 7711 while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
261 7029 c->dst_incr *= 2;
262 7029 c->src_incr *= 2;
263 }
264 682 c->ideal_dst_incr = c->dst_incr;
265 682 c->dst_incr_div = c->dst_incr / c->src_incr;
266 682 c->dst_incr_mod = c->dst_incr % c->src_incr;
267
268 682 c->index= -phase_count*((c->filter_length-1)/2);
269 682 c->frac= 0;
270
271 682 swri_resample_dsp_init(c);
272
273 682 return c;
274 ✗ error:
275 ✗ av_freep(&c->filter_bank);
276 ✗ av_free(c);
277 ✗ return NULL;
278 }
279
280 127 static int rebuild_filter_bank_with_compensation(ResampleContext *c)
281 {
282 uint8_t *new_filter_bank;
283 int new_src_incr, new_dst_incr;
284 127 int phase_count = c->phase_count_compensation;
285 int ret;
286
287
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 127 if (phase_count == c->phase_count)
288 78 return 0;
289
290
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 49 av_assert0(!c->frac && !c->dst_incr_mod);
291
292 49 new_filter_bank = av_calloc(c->filter_alloc, (phase_count + 1) * c->felem_size);
293
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 49 if (!new_filter_bank)
294 ✗ return AVERROR(ENOMEM);
295
296 49 ret = build_filter(c, new_filter_bank, c->factor, c->filter_length, c->filter_alloc,
297 49 phase_count, 1 << c->filter_shift, c->filter_type, c->kaiser_beta);
298
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 49 if (ret < 0) {
299 ✗ av_freep(&new_filter_bank);
300 ✗ return ret;
301 }
302 49 memcpy(new_filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, new_filter_bank, (c->filter_alloc-1)*c->felem_size);
303 49 memcpy(new_filter_bank + (c->filter_alloc*phase_count )*c->felem_size, new_filter_bank + (c->filter_alloc - 1)*c->felem_size, c->felem_size);
304
305
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 49 if (!av_reduce(&new_src_incr, &new_dst_incr, c->src_incr,
306 49 c->dst_incr * (int64_t)(phase_count/c->phase_count), INT32_MAX/2))
307 {
308 ✗ av_freep(&new_filter_bank);
309 ✗ return AVERROR(EINVAL);
310 }
311
312 49 c->src_incr = new_src_incr;
313 49 c->dst_incr = new_dst_incr;
314
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 571 while (c->dst_incr < (1<<20) && c->src_incr < (1<<20)) {
315 522 c->dst_incr *= 2;
316 522 c->src_incr *= 2;
317 }
318 49 c->ideal_dst_incr = c->dst_incr;
319 49 c->dst_incr_div = c->dst_incr / c->src_incr;
320 49 c->dst_incr_mod = c->dst_incr % c->src_incr;
321 49 c->index *= phase_count / c->phase_count;
322 49 c->phase_count = phase_count;
323 49 av_freep(&c->filter_bank);
324 49 c->filter_bank = new_filter_bank;
325 49 return 0;
326 }
327
328 127 static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
329 int ret;
330
331
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 127 if (compensation_distance && sample_delta) {
332 127 ret = rebuild_filter_bank_with_compensation(c);
333
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 127 if (ret < 0)
334 ✗ return ret;
335 }
336
337 127 c->compensation_distance= compensation_distance;
338
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 127 if (compensation_distance)
339 127 c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
340 else
341 ✗ c->dst_incr = c->ideal_dst_incr;
342
343 127 c->dst_incr_div = c->dst_incr / c->src_incr;
344 127 c->dst_incr_mod = c->dst_incr % c->src_incr;
345
346 127 return 0;
347 }
348
349 64135 static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
350 int i;
351 64135 int64_t max_src_size = (INT64_MAX/2 / c->phase_count) / c->src_incr;
352
353
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 64135 if (c->compensation_distance)
354 775 dst_size = FFMIN(dst_size, c->compensation_distance);
355 64135 src_size = FFMIN(src_size, max_src_size);
356
357 64135 *consumed = 0;
358
359
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 64217 if (c->filter_length == 1 && c->phase_count == 1) {
360 82 int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*c->index + 1;
361 82 int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr + 1;
362 82 int new_size = (src_size * (int64_t)c->src_incr - c->frac + c->dst_incr - 1) / c->dst_incr;
363
364 82 dst_size = FFMAX(FFMIN(dst_size, new_size), 0);
365
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 82 if (dst_size > 0) {
366
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 160 for (i = 0; i < dst->ch_count; i++) {
367 80 c->dsp.resample_one(dst->ch[i], src->ch[i], dst_size, index2, incr);
368
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 80 if (i+1 == dst->ch_count) {
369 80 c->index += dst_size * c->dst_incr_div;
370 80 c->index += (c->frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr;
371 av_assert2(c->index >= 0);
372 80 *consumed = c->index;
373 80 c->frac = (c->frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr;
374 80 c->index = 0;
375 }
376 }
377 }
378 } else {
379 64053 int64_t end_index = (1LL + src_size - c->filter_length) * c->phase_count;
380 64053 int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac;
381 64053 int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr;
382 int (*resample_func)(struct ResampleContext *c, void *dst,
383 const void *src, int n, int update_ctx);
384
385 64053 dst_size = FFMAX(FFMIN(dst_size, delta_n), 0);
386
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 64053 if (dst_size > 0) {
387 /* resample_linear and resample_common should have same behavior
388 * when frac and dst_incr_mod are zero */
389
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 45454 resample_func = (c->linear && (c->frac || c->dst_incr_mod)) ?
390
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 104538 c->dsp.resample_linear : c->dsp.resample_common;
391
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 155289 for (i = 0; i < dst->ch_count; i++)
392 96205 *consumed = resample_func(c, dst->ch[i], src->ch[i], dst_size, i+1 == dst->ch_count);
393 }
394 }
395
396
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 64135 if (c->compensation_distance) {
397 775 c->compensation_distance -= dst_size;
398
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 775 if (!c->compensation_distance) {
399 ✗ c->dst_incr = c->ideal_dst_incr;
400 ✗ c->dst_incr_div = c->dst_incr / c->src_incr;
401 ✗ c->dst_incr_mod = c->dst_incr % c->src_incr;
402 }
403 }
404
405 64135 return dst_size;
406 }
407
408 16890 static int64_t get_delay(struct SwrContext *s, int64_t base){
409 16890 ResampleContext *c = s->resample;
410 16890 int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
411 16890 num *= c->phase_count;
412 16890 num -= c->index;
413 16890 num *= c->src_incr;
414 16890 num -= c->frac;
415 16890 return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr * c->phase_count);
416 }
417
418 ✗ static int64_t get_out_samples(struct SwrContext *s, int in_samples) {
419 ✗ ResampleContext *c = s->resample;
420 // The + 2 are added to allow implementations to be slightly inaccurate, they should not be needed currently.
421 // They also make it easier to proof that changes and optimizations do not
422 // break the upper bound.
423 ✗ int64_t num = s->in_buffer_count + 2LL + in_samples;
424 ✗ num *= c->phase_count;
425 ✗ num -= c->index;
426 ✗ num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) * c->phase_count, AV_ROUND_UP) + 2;
427
428 ✗ if (c->compensation_distance) {
429 ✗ if (num > INT_MAX)
430 ✗ return AVERROR(EINVAL);
431
432 ✗ num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);
433 }
434 ✗ return num;
435 }
436
437 657 static int resample_flush(struct SwrContext *s) {
438 657 ResampleContext *c = s->resample;
439 657 AudioData *a= &s->in_buffer;
440 int i, j, ret;
441 657 int reflection = (FFMIN(s->in_buffer_count, c->filter_length) + 1) / 2;
442
443
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 657 if((ret = swri_realloc_audio(a, s->in_buffer_index + s->in_buffer_count + reflection)) < 0)
444 ✗ return ret;
445
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 657 av_assert0(a->planar);
446
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 1341 for(i=0; i<a->ch_count; i++){
447
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 51970 for(j=0; j<reflection; j++){
448 51286 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
449 51286 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
450 }
451 }
452 657 s->in_buffer_count += reflection;
453 657 return 0;
454 }
455
456 // in fact the whole handle multiple ridiculously small buffers might need more thinking...
457 19404 static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src,
458 int in_count, int *out_idx, int *out_sz)
459 {
460 19404 int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;
461
462
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 19404 if (c->index >= 0)
463 18637 return 0;
464
465
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 767 if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)
466 ✗ return res;
467
468 // copy
469
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 105213 for (n = *out_sz; n < num; n++) {
470
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 210091 for (ch = 0; ch < src->ch_count; ch++) {
471 105645 memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
472 105645 src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);
473 }
474 }
475
476 // if not enough data is in, return and wait for more
477
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 767 if (num < c->filter_length + 1) {
478 87 *out_sz = num;
479 87 *out_idx = c->filter_length;
480 87 return INT_MAX;
481 }
482
483 // else invert
484
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 104446 for (n = 1; n <= c->filter_length; n++) {
485
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 208686 for (ch = 0; ch < src->ch_count; ch++) {
486 104920 memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),
487 104920 dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
488 104920 c->felem_size);
489 }
490 }
491
492 680 res = num - *out_sz;
493 680 *out_idx = c->filter_length;
494
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 51883 while (c->index < 0) {
495 51203 --*out_idx;
496 51203 c->index += c->phase_count;
497 }
498 680 *out_sz = FFMAX(*out_sz + c->filter_length,
499 680 1 + c->filter_length * 2) - *out_idx;
500
501 680 return FFMAX(res, 0);
502 }
503
504 const struct Resampler swri_resampler = {
505 .init = resample_init,
506 .free = resample_free,
507 .multiple_resample = multiple_resample,
508 .flush = resample_flush,
509 .set_compensation = set_compensation,
510 .get_delay = get_delay,
511 .invert_initial_buffer = invert_initial_buffer,
512 .get_out_samples = get_out_samples,
513 };
514