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resample.c
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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){
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;}
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;
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 
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 
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 
242 
243  return c;
244 error:
245  av_freep(&c->filter_bank);
246  av_free(c);
247  return NULL;
248 }
249 
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){
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 
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 &&
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 *= 1 << 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 int64_t get_out_samples(struct SwrContext *s, int in_samples) {
349  ResampleContext *c = s->resample;
350  // The + 2 are added to allow implementations to be slightly inaccurate, they should not be needed currently.
351  // They also make it easier to proof that changes and optimizations do not
352  // break the upper bound.
353  int64_t num = s->in_buffer_count + 2LL + in_samples;
354  num *= 1 << c->phase_shift;
355  num -= c->index;
356  num = av_rescale_rnd(num, s->out_sample_rate, ((int64_t)s->in_sample_rate) << c->phase_shift, AV_ROUND_UP) + 2;
357 
358  if (c->compensation_distance) {
359  if (num > INT_MAX)
360  return AVERROR(EINVAL);
361 
362  num = FFMAX(num, (num * c->ideal_dst_incr - 1) / c->dst_incr + 1);
363  }
364  return num;
365 }
366 
367 static int resample_flush(struct SwrContext *s) {
368  AudioData *a= &s->in_buffer;
369  int i, j, ret;
370  if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
371  return ret;
372  av_assert0(a->planar);
373  for(i=0; i<a->ch_count; i++){
374  for(j=0; j<s->in_buffer_count; j++){
375  memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
376  a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
377  }
378  }
379  s->in_buffer_count += (s->in_buffer_count+1)/2;
380  return 0;
381 }
382 
383 // in fact the whole handle multiple ridiculously small buffers might need more thinking...
385  int in_count, int *out_idx, int *out_sz)
386 {
387  int n, ch, num = FFMIN(in_count + *out_sz, c->filter_length + 1), res;
388 
389  if (c->index >= 0)
390  return 0;
391 
392  if ((res = swri_realloc_audio(dst, c->filter_length * 2 + 1)) < 0)
393  return res;
394 
395  // copy
396  for (n = *out_sz; n < num; n++) {
397  for (ch = 0; ch < src->ch_count; ch++) {
398  memcpy(dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
399  src->ch[ch] + ((n - *out_sz) * c->felem_size), c->felem_size);
400  }
401  }
402 
403  // if not enough data is in, return and wait for more
404  if (num < c->filter_length + 1) {
405  *out_sz = num;
406  *out_idx = c->filter_length;
407  return INT_MAX;
408  }
409 
410  // else invert
411  for (n = 1; n <= c->filter_length; n++) {
412  for (ch = 0; ch < src->ch_count; ch++) {
413  memcpy(dst->ch[ch] + ((c->filter_length - n) * c->felem_size),
414  dst->ch[ch] + ((c->filter_length + n) * c->felem_size),
415  c->felem_size);
416  }
417  }
418 
419  res = num - *out_sz;
420  *out_idx = c->filter_length + (c->index >> c->phase_shift);
421  *out_sz = FFMAX(*out_sz + c->filter_length,
422  1 + c->filter_length * 2) - *out_idx;
423  c->index &= c->phase_mask;
424 
425  return FFMAX(res, 0);
426 }
427 
428 struct Resampler const swri_resampler={
434  get_delay,
437 };
float, planar
Definition: samplefmt.h:70
#define NULL
Definition: coverity.c:32
float v
const char * s
Definition: avisynth_c.h:631
static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed)
Definition: resample.c:308
double factor
Definition: resample.h:48
int out_sample_rate
output sample rate
Audio buffer used for intermediate storage between conversion phases.
Definition: audio_data.h:37
int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding rnd)
Rescale a 64-bit integer with specified rounding.
Definition: mathematics.c:63
int ch_count
number of channels
SwrFilterType
Resampling Filter Types.
Definition: swresample.h:170
#define AV_CPU_FLAG_MMX2
SSE integer functions or AMD MMX ext.
Definition: cpu.h:31
double, planar
Definition: samplefmt.h:71
int in_buffer_index
cached buffer position
AudioData in_buffer
cached audio data (convert and resample purpose)
struct ResampleContext * resample
resampling context
#define FFALIGN(x, a)
Definition: common.h:71
int filter_alloc
Definition: resample.h:34
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
enum AVSampleFormat format
Definition: resample.h:49
uint8_t
Round toward +infinity.
Definition: mathematics.h:74
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:63
static int swri_resample(ResampleContext *c, uint8_t *dst, const uint8_t *src, int *consumed, int src_size, int dst_size, int update_ctx)
Definition: resample.c:270
static void resample_free(ResampleContext **c)
Definition: resample.c:250
static double bessel(double x)
0th order modified bessel function of the first kind.
Definition: resample.c:34
int swri_realloc_audio(AudioData *a, int count)
Definition: swresample.c:381
int compensation_distance
Definition: resample.h:38
static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance)
Definition: resample.c:257
enum AVResampleFilterType filter_type
Definition: resample.h:42
int filter_shift
Definition: resample.h:51
struct Resampler const swri_resampler
Definition: resample.c:428
static int invert_initial_buffer(ResampleContext *c, AudioData *dst, const AudioData *src, int in_count, int *out_idx, int *out_sz)
Definition: resample.c:384
#define lrintf(x)
Definition: libm_mips.h:70
int dst_incr_mod
Definition: resample.h:38
Kaiser Windowed Sinc.
Definition: swresample.h:173
#define av_log(a,...)
int in_buffer_count
cached buffer length
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
Blackman Nuttall Windowed Sinc.
Definition: swresample.h:172
static int resample_flush(struct SwrContext *s)
Definition: resample.c:367
#define AVERROR(e)
Definition: error.h:43
The libswresample context.
unsigned int index
Definition: resample.h:35
int dst_incr_div
Definition: resample.h:37
simple assert() macros that are a bit more flexible than ISO C assert().
#define FFMAX(a, b)
Definition: common.h:64
int compensation_distance
Definition: resample2.c:71
int av_reduce(int *dst_num, int *dst_den, int64_t num, int64_t den, int64_t max)
Reduce a fraction.
Definition: rational.c:35
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:127
#define FFMIN(a, b)
Definition: common.h:66
float y
signed 32 bits, planar
Definition: samplefmt.h:69
static int64_t get_delay(struct SwrContext *s, int64_t base)
Definition: resample.c:338
int32_t
int in_sample_rate
input sample rate
static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, int filter_type, int kaiser_beta)
builds a polyphase filterbank.
Definition: resample.c:70
int bps
bytes per sample
int n
Definition: avisynth_c.h:547
static ResampleContext * resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby)
Definition: resample.c:174
struct ResampleContext::@194 dsp
#define FILTER_SHIFT
Definition: resample2.c:35
AVS_Value src
Definition: avisynth_c.h:482
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
#define llrint(x)
Definition: libm.h:112
int filter_length
Definition: resample.h:32
int index
Definition: gxfenc.c:89
void swri_resample_dsp_init(ResampleContext *c)
Definition: resample_dsp.c:46
#define LEN
static const int factor[16]
Definition: vf_pp7.c:75
int av_get_cpu_flags(void)
Return the flags which specify extensions supported by the CPU.
Definition: cpu.c:76
int(* resample)(struct ResampleContext *c, void *dst, const void *src, int n, int update_ctx)
Definition: resample.h:56
static int64_t get_out_samples(struct SwrContext *s, int in_samples)
Definition: resample.c:348
void * av_calloc(size_t nmemb, size_t size)
Allocate a block of nmemb * size bytes with alignment suitable for all memory accesses (including vec...
Definition: mem.c:260
void(* resample_one)(struct ResampleContext *c, void *dst0, int dst_index, const void *src0, unsigned int index, int frac)
Definition: resample.h:45
int ideal_dst_incr
Definition: resample.h:33
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:104
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
static double c[64]
#define av_free(p)
uint8_t * filter_bank
Definition: resample.h:31
static const struct twinvq_data tab
#define AV_CPU_FLAG_SSE2
PIV SSE2 functions.
Definition: cpu.h:34
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:68
int planar
1 if planar audio, 0 otherwise
#define M_PI
Definition: mathematics.h:46
#define av_malloc_array(a, b)
uint8_t * ch[SWR_CH_MAX]
samples buffer per channel
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
#define av_unused
Definition: attributes.h:118