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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/log.h"
29 #include "libavutil/avassert.h"
30 #include "swresample_internal.h"
31 
32 
33 typedef struct ResampleContext {
34  const AVClass *av_class;
36  int filter_length;
38  int ideal_dst_incr;
39  int dst_incr;
40  int index;
41  int frac;
42  int src_incr;
44  int phase_shift;
45  int phase_mask;
46  int linear;
48  int kaiser_beta;
49  double factor;
54 
55 /**
56  * 0th order modified bessel function of the first kind.
57  */
58 static double bessel(double x){
59  double v=1;
60  double lastv=0;
61  double t=1;
62  int i;
63  static const double inv[100]={
64  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),
65  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),
66  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),
67  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),
68  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),
69  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),
70  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),
71  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),
72  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),
73  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)
74  };
75 
76  x= x*x/4;
77  for(i=0; v != lastv; i++){
78  lastv=v;
79  t *= x*inv[i];
80  v += t;
81  av_assert2(i<99);
82  }
83  return v;
84 }
85 
86 /**
87  * builds a polyphase filterbank.
88  * @param factor resampling factor
89  * @param scale wanted sum of coefficients for each filter
90  * @param filter_type filter type
91  * @param kaiser_beta kaiser window beta
92  * @return 0 on success, negative on error
93  */
94 static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
95  int filter_type, int kaiser_beta){
96  int ph, i;
97  double x, y, w;
98  double *tab = av_malloc(tap_count * sizeof(*tab));
99  const int center= (tap_count-1)/2;
100 
101  if (!tab)
102  return AVERROR(ENOMEM);
103 
104  /* if upsampling, only need to interpolate, no filter */
105  if (factor > 1.0)
106  factor = 1.0;
107 
108  for(ph=0;ph<phase_count;ph++) {
109  double norm = 0;
110  for(i=0;i<tap_count;i++) {
111  x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
112  if (x == 0) y = 1.0;
113  else y = sin(x) / x;
114  switch(filter_type){
115  case SWR_FILTER_TYPE_CUBIC:{
116  const float d= -0.5; //first order derivative = -0.5
117  x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
118  if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
119  else y= d*(-4 + 8*x - 5*x*x + x*x*x);
120  break;}
122  w = 2.0*x / (factor*tap_count) + M_PI;
123  y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
124  break;
126  w = 2.0*x / (factor*tap_count*M_PI);
127  y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
128  break;
129  default:
130  av_assert0(0);
131  }
132 
133  tab[i] = y;
134  norm += y;
135  }
136 
137  /* normalize so that an uniform color remains the same */
138  switch(c->format){
139  case AV_SAMPLE_FMT_S16P:
140  for(i=0;i<tap_count;i++)
141  ((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
142  break;
143  case AV_SAMPLE_FMT_S32P:
144  for(i=0;i<tap_count;i++)
145  ((int32_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
146  break;
147  case AV_SAMPLE_FMT_FLTP:
148  for(i=0;i<tap_count;i++)
149  ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
150  break;
151  case AV_SAMPLE_FMT_DBLP:
152  for(i=0;i<tap_count;i++)
153  ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
154  break;
155  }
156  }
157 #if 0
158  {
159 #define LEN 1024
160  int j,k;
161  double sine[LEN + tap_count];
162  double filtered[LEN];
163  double maxff=-2, minff=2, maxsf=-2, minsf=2;
164  for(i=0; i<LEN; i++){
165  double ss=0, sf=0, ff=0;
166  for(j=0; j<LEN+tap_count; j++)
167  sine[j]= cos(i*j*M_PI/LEN);
168  for(j=0; j<LEN; j++){
169  double sum=0;
170  ph=0;
171  for(k=0; k<tap_count; k++)
172  sum += filter[ph * tap_count + k] * sine[k+j];
173  filtered[j]= sum / (1<<FILTER_SHIFT);
174  ss+= sine[j + center] * sine[j + center];
175  ff+= filtered[j] * filtered[j];
176  sf+= sine[j + center] * filtered[j];
177  }
178  ss= sqrt(2*ss/LEN);
179  ff= sqrt(2*ff/LEN);
180  sf= 2*sf/LEN;
181  maxff= FFMAX(maxff, ff);
182  minff= FFMIN(minff, ff);
183  maxsf= FFMAX(maxsf, sf);
184  minsf= FFMIN(minsf, sf);
185  if(i%11==0){
186  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);
187  minff=minsf= 2;
188  maxff=maxsf= -2;
189  }
190  }
191  }
192 #endif
193 
194  av_free(tab);
195  return 0;
196 }
197 
198 static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
199  double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,
200  double precision, int cheby){
201  double cutoff = cutoff0? cutoff0 : 0.97;
202  double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
203  int phase_count= 1<<phase_shift;
204 
205  if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
206  || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format
207  || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
208  c = av_mallocz(sizeof(*c));
209  if (!c)
210  return NULL;
211 
212  c->format= format;
213 
215 
216  switch(c->format){
217  case AV_SAMPLE_FMT_S16P:
218  c->filter_shift = 15;
219  break;
220  case AV_SAMPLE_FMT_S32P:
221  c->filter_shift = 30;
222  break;
223  case AV_SAMPLE_FMT_FLTP:
224  case AV_SAMPLE_FMT_DBLP:
225  c->filter_shift = 0;
226  break;
227  default:
228  av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
229  av_assert0(0);
230  }
231 
232  c->phase_shift = phase_shift;
233  c->phase_mask = phase_count - 1;
234  c->linear = linear;
235  c->factor = factor;
236  c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
237  c->filter_alloc = FFALIGN(c->filter_length, 8);
238  c->filter_bank = av_mallocz(c->filter_alloc*(phase_count+1)*c->felem_size);
239  c->filter_type = filter_type;
240  c->kaiser_beta = kaiser_beta;
241  if (!c->filter_bank)
242  goto error;
243  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))
244  goto error;
245  memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
246  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);
247  }
248 
249  c->compensation_distance= 0;
250  if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
251  goto error;
252  c->ideal_dst_incr= c->dst_incr;
253 
254  c->index= -phase_count*((c->filter_length-1)/2);
255  c->frac= 0;
256 
257  return c;
258 error:
259  av_free(c->filter_bank);
260  av_free(c);
261  return NULL;
262 }
263 
265  if(!*c)
266  return;
267  av_freep(&(*c)->filter_bank);
268  av_freep(c);
269 }
270 
271 static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
272  c->compensation_distance= compensation_distance;
273  if (compensation_distance)
274  c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
275  else
276  c->dst_incr = c->ideal_dst_incr;
277  return 0;
278 }
279 
280 #define TEMPLATE_RESAMPLE_S16
281 #include "resample_template.c"
282 #undef TEMPLATE_RESAMPLE_S16
283 
284 #define TEMPLATE_RESAMPLE_S32
285 #include "resample_template.c"
286 #undef TEMPLATE_RESAMPLE_S32
287 
288 #define TEMPLATE_RESAMPLE_FLT
289 #include "resample_template.c"
290 #undef TEMPLATE_RESAMPLE_FLT
291 
292 #define TEMPLATE_RESAMPLE_DBL
293 #include "resample_template.c"
294 #undef TEMPLATE_RESAMPLE_DBL
295 
296 // XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed
297 #if HAVE_MMXEXT_INLINE
298 
299 #include "x86/resample_mmx.h"
300 
301 #define TEMPLATE_RESAMPLE_S16_MMX2
302 #include "resample_template.c"
303 #undef TEMPLATE_RESAMPLE_S16_MMX2
304 
305 #if HAVE_SSSE3_INLINE
306 #define TEMPLATE_RESAMPLE_S16_SSSE3
307 #include "resample_template.c"
308 #undef TEMPLATE_RESAMPLE_S16_SSSE3
309 #endif
310 
311 #endif // HAVE_MMXEXT_INLINE
312 
313 static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
314  int i, ret= -1;
315  int av_unused mm_flags = av_get_cpu_flags();
316  int need_emms= 0;
317 
318  for(i=0; i<dst->ch_count; i++){
319 #if HAVE_MMXEXT_INLINE
320 #if HAVE_SSSE3_INLINE
321  if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_SSSE3)) ret= swri_resample_int16_ssse3(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
322  else
323 #endif
324  if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){
325  ret= swri_resample_int16_mmx2 (c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
326  need_emms= 1;
327  } else
328 #endif
329  if(c->format == AV_SAMPLE_FMT_S16P) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
330  else if(c->format == AV_SAMPLE_FMT_S32P) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
331  else if(c->format == AV_SAMPLE_FMT_FLTP) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
332  else if(c->format == AV_SAMPLE_FMT_DBLP) ret= swri_resample_double(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
333  }
334  if(need_emms)
335  emms_c();
336  return ret;
337 }
338 
339 static int64_t get_delay(struct SwrContext *s, int64_t base){
340  ResampleContext *c = s->resample;
341  int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
342  num <<= c->phase_shift;
343  num -= c->index;
344  num *= c->src_incr;
345  num -= c->frac;
346  return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);
347 }
348 
349 static int resample_flush(struct SwrContext *s) {
350  AudioData *a= &s->in_buffer;
351  int i, j, ret;
352  if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
353  return ret;
354  av_assert0(a->planar);
355  for(i=0; i<a->ch_count; i++){
356  for(j=0; j<s->in_buffer_count; j++){
357  memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
358  a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
359  }
360  }
361  s->in_buffer_count += (s->in_buffer_count+1)/2;
362  return 0;
363 }
364 
365 struct Resampler const swri_resampler={
371  get_delay,
372 };