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resample2.c
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1 /*
2  * audio resampling
3  * Copyright (c) 2004 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 "avcodec.h"
30 #include "libavutil/common.h"
31 
32 #if FF_API_AVCODEC_RESAMPLE
33 
34 #ifndef CONFIG_RESAMPLE_HP
35 #define FILTER_SHIFT 15
36 
37 #define FELEM int16_t
38 #define FELEM2 int32_t
39 #define FELEML int64_t
40 #define FELEM_MAX INT16_MAX
41 #define FELEM_MIN INT16_MIN
42 #define WINDOW_TYPE 9
43 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
44 #define FILTER_SHIFT 30
45 
46 #define FELEM int32_t
47 #define FELEM2 int64_t
48 #define FELEML int64_t
49 #define FELEM_MAX INT32_MAX
50 #define FELEM_MIN INT32_MIN
51 #define WINDOW_TYPE 12
52 #else
53 #define FILTER_SHIFT 0
54 
55 #define FELEM double
56 #define FELEM2 double
57 #define FELEML double
58 #define WINDOW_TYPE 24
59 #endif
60 
61 
62 typedef struct AVResampleContext{
63  const AVClass *av_class;
65  int filter_length;
66  int ideal_dst_incr;
67  int dst_incr;
68  int index;
69  int frac;
70  int src_incr;
71  int compensation_distance;
72  int phase_shift;
73  int phase_mask;
74  int linear;
75 }AVResampleContext;
76 
77 /**
78  * 0th order modified bessel function of the first kind.
79  */
80 static double bessel(double x){
81  double v=1;
82  double lastv=0;
83  double t=1;
84  int i;
85 
86  x= x*x/4;
87  for(i=1; v != lastv; i++){
88  lastv=v;
89  t *= x/(i*i);
90  v += t;
91  }
92  return v;
93 }
94 
95 /**
96  * Build a polyphase filterbank.
97  * @param factor resampling factor
98  * @param scale wanted sum of coefficients for each filter
99  * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
100  * @return 0 on success, negative on error
101  */
102 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
103  int ph, i;
104  double x, y, w;
105  double *tab = av_malloc(tap_count * sizeof(*tab));
106  const int center= (tap_count-1)/2;
107 
108  if (!tab)
109  return AVERROR(ENOMEM);
110 
111  /* if upsampling, only need to interpolate, no filter */
112  if (factor > 1.0)
113  factor = 1.0;
114 
115  for(ph=0;ph<phase_count;ph++) {
116  double norm = 0;
117  for(i=0;i<tap_count;i++) {
118  x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
119  if (x == 0) y = 1.0;
120  else y = sin(x) / x;
121  switch(type){
122  case 0:{
123  const float d= -0.5; //first order derivative = -0.5
124  x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
125  if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
126  else y= d*(-4 + 8*x - 5*x*x + x*x*x);
127  break;}
128  case 1:
129  w = 2.0*x / (factor*tap_count) + M_PI;
130  y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
131  break;
132  default:
133  w = 2.0*x / (factor*tap_count*M_PI);
134  y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
135  break;
136  }
137 
138  tab[i] = y;
139  norm += y;
140  }
141 
142  /* normalize so that an uniform color remains the same */
143  for(i=0;i<tap_count;i++) {
144 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
145  filter[ph * tap_count + i] = tab[i] / norm;
146 #else
147  filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
148 #endif
149  }
150  }
151 #if 0
152  {
153 #define LEN 1024
154  int j,k;
155  double sine[LEN + tap_count];
156  double filtered[LEN];
157  double maxff=-2, minff=2, maxsf=-2, minsf=2;
158  for(i=0; i<LEN; i++){
159  double ss=0, sf=0, ff=0;
160  for(j=0; j<LEN+tap_count; j++)
161  sine[j]= cos(i*j*M_PI/LEN);
162  for(j=0; j<LEN; j++){
163  double sum=0;
164  ph=0;
165  for(k=0; k<tap_count; k++)
166  sum += filter[ph * tap_count + k] * sine[k+j];
167  filtered[j]= sum / (1<<FILTER_SHIFT);
168  ss+= sine[j + center] * sine[j + center];
169  ff+= filtered[j] * filtered[j];
170  sf+= sine[j + center] * filtered[j];
171  }
172  ss= sqrt(2*ss/LEN);
173  ff= sqrt(2*ff/LEN);
174  sf= 2*sf/LEN;
175  maxff= FFMAX(maxff, ff);
176  minff= FFMIN(minff, ff);
177  maxsf= FFMAX(maxsf, sf);
178  minsf= FFMIN(minsf, sf);
179  if(i%11==0){
180  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  minff=minsf= 2;
182  maxff=maxsf= -2;
183  }
184  }
185  }
186 #endif
187 
188  av_free(tab);
189  return 0;
190 }
191 
192 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
193  AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
194  double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
195  int phase_count= 1<<phase_shift;
196 
197  if (!c)
198  return NULL;
199 
200  c->phase_shift= phase_shift;
201  c->phase_mask= phase_count-1;
202  c->linear= linear;
203 
204  c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
205  c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
206  if (!c->filter_bank)
207  goto error;
208  if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
209  goto error;
210  memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
211  c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
212 
213  if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
214  goto error;
215  c->ideal_dst_incr= c->dst_incr;
216 
217  c->index= -phase_count*((c->filter_length-1)/2);
218 
219  return c;
220 error:
221  av_free(c->filter_bank);
222  av_free(c);
223  return NULL;
224 }
225 
226 void av_resample_close(AVResampleContext *c){
227  av_freep(&c->filter_bank);
228  av_freep(&c);
229 }
230 
231 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
232 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
233  c->compensation_distance= compensation_distance;
234  c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
235 }
236 
237 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
238  int dst_index, i;
239  int index= c->index;
240  int frac= c->frac;
241  int dst_incr_frac= c->dst_incr % c->src_incr;
242  int dst_incr= c->dst_incr / c->src_incr;
243  int compensation_distance= c->compensation_distance;
244 
245  if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
246  int64_t index2= ((int64_t)index)<<32;
247  int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
248  dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
249 
250  for(dst_index=0; dst_index < dst_size; dst_index++){
251  dst[dst_index] = src[index2>>32];
252  index2 += incr;
253  }
254  index += dst_index * dst_incr;
255  index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
256  frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
257  }else{
258  for(dst_index=0; dst_index < dst_size; dst_index++){
259  FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
260  int sample_index= index >> c->phase_shift;
261  FELEM2 val=0;
262 
263  if(sample_index < 0){
264  for(i=0; i<c->filter_length; i++)
265  val += src[FFABS(sample_index + i) % src_size] * filter[i];
266  }else if(sample_index + c->filter_length > src_size){
267  break;
268  }else if(c->linear){
269  FELEM2 v2=0;
270  for(i=0; i<c->filter_length; i++){
271  val += src[sample_index + i] * (FELEM2)filter[i];
272  v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
273  }
274  val+=(v2-val)*(FELEML)frac / c->src_incr;
275  }else{
276  for(i=0; i<c->filter_length; i++){
277  val += src[sample_index + i] * (FELEM2)filter[i];
278  }
279  }
280 
281 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
282  dst[dst_index] = av_clip_int16(lrintf(val));
283 #else
284  val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
285  dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
286 #endif
287 
288  frac += dst_incr_frac;
289  index += dst_incr;
290  if(frac >= c->src_incr){
291  frac -= c->src_incr;
292  index++;
293  }
294 
295  if(dst_index + 1 == compensation_distance){
296  compensation_distance= 0;
297  dst_incr_frac= c->ideal_dst_incr % c->src_incr;
298  dst_incr= c->ideal_dst_incr / c->src_incr;
299  }
300  }
301  }
302  *consumed= FFMAX(index, 0) >> c->phase_shift;
303  if(index>=0) index &= c->phase_mask;
304 
305  if(compensation_distance){
306  compensation_distance -= dst_index;
307  av_assert2(compensation_distance > 0);
308  }
309  if(update_ctx){
310  c->frac= frac;
311  c->index= index;
312  c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
313  c->compensation_distance= compensation_distance;
314  }
315 
316  return dst_index;
317 }
318 
319 #endif