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1 | /* |
2 | * audio resampling | |
3 | * Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at> | |
4 | * | |
5 | * This file is part of FFmpeg. | |
6 | * | |
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. | |
11 | * | |
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. | |
16 | * | |
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 | |
20 | */ | |
21 | ||
22 | /** | |
23 | * @file | |
24 | * audio resampling | |
25 | * @author Michael Niedermayer <michaelni@gmx.at> | |
26 | */ | |
27 | ||
28 | #include "libavutil/avassert.h" | |
29 | #include "resample.h" | |
30 | ||
31 | /** | |
32 | * 0th order modified bessel function of the first kind. | |
33 | */ | |
34 | static double bessel(double x){ | |
35 | double v=1; | |
36 | double lastv=0; | |
37 | double t=1; | |
38 | int i; | |
39 | static const double inv[100]={ | |
40 | 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), | |
41 | 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), | |
42 | 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), | |
43 | 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), | |
44 | 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), | |
45 | 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), | |
46 | 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), | |
47 | 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), | |
48 | 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), | |
49 | 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) | |
50 | }; | |
51 | ||
52 | x= x*x/4; | |
53 | for(i=0; v != lastv; i++){ | |
54 | lastv=v; | |
55 | t *= x*inv[i]; | |
56 | v += t; | |
57 | av_assert2(i<99); | |
58 | } | |
59 | return v; | |
60 | } | |
61 | ||
62 | /** | |
63 | * builds a polyphase filterbank. | |
64 | * @param factor resampling factor | |
65 | * @param scale wanted sum of coefficients for each filter | |
66 | * @param filter_type filter type | |
67 | * @param kaiser_beta kaiser window beta | |
68 | * @return 0 on success, negative on error | |
69 | */ | |
70 | static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, | |
71 | int filter_type, int kaiser_beta){ | |
72 | int ph, i; | |
73 | double x, y, w; | |
74 | double *tab = av_malloc_array(tap_count, sizeof(*tab)); | |
75 | const int center= (tap_count-1)/2; | |
76 | ||
77 | if (!tab) | |
78 | return AVERROR(ENOMEM); | |
79 | ||
80 | /* if upsampling, only need to interpolate, no filter */ | |
81 | if (factor > 1.0) | |
82 | factor = 1.0; | |
83 | ||
84 | for(ph=0;ph<phase_count;ph++) { | |
85 | double norm = 0; | |
86 | for(i=0;i<tap_count;i++) { | |
87 | x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; | |
88 | if (x == 0) y = 1.0; | |
89 | else y = sin(x) / x; | |
90 | switch(filter_type){ | |
91 | case SWR_FILTER_TYPE_CUBIC:{ | |
92 | const float d= -0.5; //first order derivative = -0.5 | |
93 | x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); | |
94 | if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); | |
95 | else y= d*(-4 + 8*x - 5*x*x + x*x*x); | |
96 | break;} | |
97 | case SWR_FILTER_TYPE_BLACKMAN_NUTTALL: | |
98 | w = 2.0*x / (factor*tap_count) + M_PI; | |
99 | y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w); | |
100 | break; | |
101 | case SWR_FILTER_TYPE_KAISER: | |
102 | w = 2.0*x / (factor*tap_count*M_PI); | |
103 | y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); | |
104 | break; | |
105 | default: | |
106 | av_assert0(0); | |
107 | } | |
108 | ||
109 | tab[i] = y; | |
110 | norm += y; | |
111 | } | |
112 | ||
113 | /* normalize so that an uniform color remains the same */ | |
114 | switch(c->format){ | |
115 | case AV_SAMPLE_FMT_S16P: | |
116 | for(i=0;i<tap_count;i++) | |
117 | ((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX); | |
118 | break; | |
119 | case AV_SAMPLE_FMT_S32P: | |
120 | for(i=0;i<tap_count;i++) | |
121 | ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm)); | |
122 | break; | |
123 | case AV_SAMPLE_FMT_FLTP: | |
124 | for(i=0;i<tap_count;i++) | |
125 | ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm; | |
126 | break; | |
127 | case AV_SAMPLE_FMT_DBLP: | |
128 | for(i=0;i<tap_count;i++) | |
129 | ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm; | |
130 | break; | |
131 | } | |
132 | } | |
133 | #if 0 | |
134 | { | |
135 | #define LEN 1024 | |
136 | int j,k; | |
137 | double sine[LEN + tap_count]; | |
138 | double filtered[LEN]; | |
139 | double maxff=-2, minff=2, maxsf=-2, minsf=2; | |
140 | for(i=0; i<LEN; i++){ | |
141 | double ss=0, sf=0, ff=0; | |
142 | for(j=0; j<LEN+tap_count; j++) | |
143 | sine[j]= cos(i*j*M_PI/LEN); | |
144 | for(j=0; j<LEN; j++){ | |
145 | double sum=0; | |
146 | ph=0; | |
147 | for(k=0; k<tap_count; k++) | |
148 | sum += filter[ph * tap_count + k] * sine[k+j]; | |
149 | filtered[j]= sum / (1<<FILTER_SHIFT); | |
150 | ss+= sine[j + center] * sine[j + center]; | |
151 | ff+= filtered[j] * filtered[j]; | |
152 | sf+= sine[j + center] * filtered[j]; | |
153 | } | |
154 | ss= sqrt(2*ss/LEN); | |
155 | ff= sqrt(2*ff/LEN); | |
156 | sf= 2*sf/LEN; | |
157 | maxff= FFMAX(maxff, ff); | |
158 | minff= FFMIN(minff, ff); | |
159 | maxsf= FFMAX(maxsf, sf); | |
160 | minsf= FFMIN(minsf, sf); | |
161 | if(i%11==0){ | |
162 | 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); | |
163 | minff=minsf= 2; | |
164 | maxff=maxsf= -2; | |
165 | } | |
166 | } | |
167 | } | |
168 | #endif | |
169 | ||
170 | av_free(tab); | |
171 | return 0; | |
172 | } | |
173 | ||
174 | static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, | |
175 | double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, | |
176 | double precision, int cheby) | |
177 | { | |
178 | double cutoff = cutoff0? cutoff0 : 0.97; | |
179 | double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); | |
180 | int phase_count= 1<<phase_shift; | |
181 | ||
182 | if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor | |
183 | || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format | |
184 | || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { | |
185 | c = av_mallocz(sizeof(*c)); | |
186 | if (!c) | |
187 | return NULL; | |
188 | ||
189 | c->format= format; | |
190 | ||
191 | c->felem_size= av_get_bytes_per_sample(c->format); | |
192 | ||
193 | switch(c->format){ | |
194 | case AV_SAMPLE_FMT_S16P: | |
195 | c->filter_shift = 15; | |
196 | break; | |
197 | case AV_SAMPLE_FMT_S32P: | |
198 | c->filter_shift = 30; | |
199 | break; | |
200 | case AV_SAMPLE_FMT_FLTP: | |
201 | case AV_SAMPLE_FMT_DBLP: | |
202 | c->filter_shift = 0; | |
203 | break; | |
204 | default: | |
205 | av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n"); | |
206 | av_assert0(0); | |
207 | } | |
208 | ||
209 | if (filter_size/factor > INT32_MAX/256) { | |
210 | av_log(NULL, AV_LOG_ERROR, "Filter length too large\n"); | |
211 | goto error; | |
212 | } | |
213 | ||
214 | c->phase_shift = phase_shift; | |
215 | c->phase_mask = phase_count - 1; | |
216 | c->linear = linear; | |
217 | c->factor = factor; | |
218 | c->filter_length = FFMAX((int)ceil(filter_size/factor), 1); | |
219 | c->filter_alloc = FFALIGN(c->filter_length, 8); | |
220 | c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size); | |
221 | c->filter_type = filter_type; | |
222 | c->kaiser_beta = kaiser_beta; | |
223 | if (!c->filter_bank) | |
224 | goto error; | |
225 | 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)) | |
226 | goto error; | |
227 | memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size); | |
228 | 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); | |
229 | } | |
230 | ||
231 | c->compensation_distance= 0; | |
232 | if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2)) | |
233 | goto error; | |
234 | c->ideal_dst_incr = c->dst_incr; | |
235 | c->dst_incr_div = c->dst_incr / c->src_incr; | |
236 | c->dst_incr_mod = c->dst_incr % c->src_incr; | |
237 | ||
238 | c->index= -phase_count*((c->filter_length-1)/2); | |
239 | c->frac= 0; | |
240 | ||
241 | swri_resample_dsp_init(c); | |
242 | ||
243 | return c; | |
244 | error: | |
245 | av_freep(&c->filter_bank); | |
246 | av_free(c); | |
247 | return NULL; | |
248 | } | |
249 | ||
250 | static void resample_free(ResampleContext **c){ | |
251 | if(!*c) | |
252 | return; | |
253 | av_freep(&(*c)->filter_bank); | |
254 | av_freep(c); | |
255 | } | |
256 | ||
257 | static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ | |
258 | c->compensation_distance= compensation_distance; | |
259 | if (compensation_distance) | |
260 | c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance; | |
261 | else | |
262 | c->dst_incr = c->ideal_dst_incr; | |
263 | ||
264 | c->dst_incr_div = c->dst_incr / c->src_incr; | |
265 | c->dst_incr_mod = c->dst_incr % c->src_incr; | |
266 | ||
267 | return 0; | |
268 | } | |
269 | ||
270 | static int swri_resample(ResampleContext *c, | |
271 | uint8_t *dst, const uint8_t *src, int *consumed, | |
272 | int src_size, int dst_size, int update_ctx) | |
273 | { | |
274 | if (c->filter_length == 1 && c->phase_shift == 0) { | |
275 | int index= c->index; | |
276 | int frac= c->frac; | |
277 | int64_t index2= (1LL<<32)*c->frac/c->src_incr + (1LL<<32)*index; | |
278 | int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr; | |
279 | int new_size = (src_size * (int64_t)c->src_incr - frac + c->dst_incr - 1) / c->dst_incr; | |
280 | ||
281 | dst_size= FFMIN(dst_size, new_size); | |
282 | c->dsp.resample_one(dst, src, dst_size, index2, incr); | |
283 | ||
284 | index += dst_size * c->dst_incr_div; | |
285 | index += (frac + dst_size * (int64_t)c->dst_incr_mod) / c->src_incr; | |
286 | av_assert2(index >= 0); | |
287 | *consumed= index; | |
288 | if (update_ctx) { | |
289 | c->frac = (frac + dst_size * (int64_t)c->dst_incr_mod) % c->src_incr; | |
290 | c->index = 0; | |
291 | } | |
292 | } else { | |
293 | int64_t end_index = (1LL + src_size - c->filter_length) << c->phase_shift; | |
294 | int64_t delta_frac = (end_index - c->index) * c->src_incr - c->frac; | |
295 | int delta_n = (delta_frac + c->dst_incr - 1) / c->dst_incr; | |
296 | ||
297 | dst_size = FFMIN(dst_size, delta_n); | |
298 | if (dst_size > 0) { | |
299 | *consumed = c->dsp.resample(c, dst, src, dst_size, update_ctx); | |
300 | } else { | |
301 | *consumed = 0; | |
302 | } | |
303 | } | |
304 | ||
305 | return dst_size; | |
306 | } | |
307 | ||
308 | static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ | |
309 | int i, ret= -1; | |
310 | int av_unused mm_flags = av_get_cpu_flags(); | |
311 | int need_emms = c->format == AV_SAMPLE_FMT_S16P && ARCH_X86_32 && | |
312 | (mm_flags & (AV_CPU_FLAG_MMX2 | AV_CPU_FLAG_SSE2)) == AV_CPU_FLAG_MMX2; | |
313 | int64_t max_src_size = (INT64_MAX >> (c->phase_shift+1)) / c->src_incr; | |
314 | ||
315 | if (c->compensation_distance) | |
316 | dst_size = FFMIN(dst_size, c->compensation_distance); | |
317 | src_size = FFMIN(src_size, max_src_size); | |
318 | ||
319 | for(i=0; i<dst->ch_count; i++){ | |
320 | ret= swri_resample(c, dst->ch[i], src->ch[i], | |
321 | consumed, src_size, dst_size, i+1==dst->ch_count); | |
322 | } | |
323 | if(need_emms) | |
324 | emms_c(); | |
325 | ||
326 | if (c->compensation_distance) { | |
327 | c->compensation_distance -= ret; | |
328 | if (!c->compensation_distance) { | |
329 | c->dst_incr = c->ideal_dst_incr; | |
330 | c->dst_incr_div = c->dst_incr / c->src_incr; | |
331 | c->dst_incr_mod = c->dst_incr % c->src_incr; | |
332 | } | |
333 | } | |
334 | ||
335 | return ret; | |
336 | } | |
337 | ||
338 | static int64_t get_delay(struct SwrContext *s, int64_t base){ | |
339 | ResampleContext *c = s->resample; | |
340 | int64_t num = s->in_buffer_count - (c->filter_length-1)/2; | |
341 | num <<= c->phase_shift; | |
342 | num -= c->index; | |
343 | num *= c->src_incr; | |
344 | num -= c->frac; | |
345 | return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); | |
346 | } | |
347 | ||
348 | static int resample_flush(struct SwrContext *s) { | |
349 | AudioData *a= &s->in_buffer; | |
350 | int i, j, ret; | |
351 | if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0) | |
352 | return ret; | |
353 | av_assert0(a->planar); | |
354 | for(i=0; i<a->ch_count; i++){ | |
355 | for(j=0; j<s->in_buffer_count; j++){ | |
356 | memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps, | |
357 | a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps); | |
358 | } | |
359 | } | |
360 | s->in_buffer_count += (s->in_buffer_count+1)/2; | |
361 | return 0; | |
362 | } | |
363 | ||
364 | // in fact the whole handle multiple ridiculously small buffers might need more thinking... | |
365 | static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src, | |
366 | int in_count, int *out_idx, int *out_sz) | |
367 | { | |
368 | int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res; | |
369 | ||
370 | if (c->index >= 0) | |
371 | return 0; | |
372 | ||
373 | if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0) | |
374 | return res; | |
375 | ||
376 | // copy | |
377 | for (n = *out_sz; n < num; n++) { | |
378 | for (ch = 0; ch < src->ch_count; ch++) { | |
379 | memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size), | |
380 | src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size); | |
381 | } | |
382 | } | |
383 | ||
384 | // if not enough data is in, return and wait for more | |
385 | if (num < c->filter_length + 1) { | |
386 | *out_sz = num; | |
387 | *out_idx = c->filter_length; | |
388 | return INT_MAX; | |
389 | } | |
390 | ||
391 | // else invert | |
392 | for (n = 1; n <= c->filter_length; n++) { | |
393 | for (ch = 0; ch < src->ch_count; ch++) { | |
394 | memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size), | |
395 | dst->ch[ch] + ((c->filter_length + n) * c->felem_size), | |
396 | c->felem_size); | |
397 | } | |
398 | } | |
399 | ||
400 | res = num - *out_sz; | |
401 | *out_idx = c->filter_length + (c->index >> c->phase_shift); | |
f6fa7814 DM |
402 | *out_sz = FFMAX(*out_sz + c->filter_length, |
403 | 1 + c->filter_length * 2) - *out_idx; | |
2ba45a60 | 404 | c->index &= c->phase_mask; |
2ba45a60 | 405 | |
f6fa7814 | 406 | return FFMAX(res, 0); |
2ba45a60 DM |
407 | } |
408 | ||
409 | struct Resampler const swri_resampler={ | |
410 | resample_init, | |
411 | resample_free, | |
412 | multiple_resample, | |
413 | resample_flush, | |
414 | set_compensation, | |
415 | get_delay, | |
416 | invert_initial_buffer, | |
417 | }; |