code.delx.au - pulseaudio/blob - src/pulsecore/ffmpeg/resample2.c

   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 resample2.c
  24  * audio resampling
  25  * @author Michael Niedermayer <michaelni@gmx.at>
  26  */
  27
  28 #include "avcodec.h"
  29 #include "dsputil.h"
  30
  31 #ifndef CONFIG_RESAMPLE_HP
  32 #define FILTER_SHIFT 15
  33
  34 #define FELEM int16_t
  35 #define FELEM2 int32_t
  36 #define FELEML int64_t
  37 #define FELEM_MAX INT16_MAX
  38 #define FELEM_MIN INT16_MIN
  39 #define WINDOW_TYPE 9
  40 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
  41 #define FILTER_SHIFT 30
  42
  43 #define FELEM int32_t
  44 #define FELEM2 int64_t
  45 #define FELEML int64_t
  46 #define FELEM_MAX INT32_MAX
  47 #define FELEM_MIN INT32_MIN
  48 #define WINDOW_TYPE 12
  49 #else
  50 #define FILTER_SHIFT 0
  51
  52 #define FELEM double
  53 #define FELEM2 double
  54 #define FELEML double
  55 #define WINDOW_TYPE 24
  56 #endif
  57
  58
  59 typedef struct AVResampleContext{
  60     FELEM *filter_bank;
  61     int filter_length;
  62     int ideal_dst_incr;
  63     int dst_incr;
  64     int index;
  65     int frac;
  66     int src_incr;
  67     int compensation_distance;
  68     int phase_shift;
  69     int phase_mask;
  70     int linear;
  71 }AVResampleContext;
  72
  73 /**
  74  * 0th order modified bessel function of the first kind.
  75  */
  76 static double bessel(double x){
  77     double v=1;
  78     double t=1;
  79     int i;
  80
  81     x= x*x/4;
  82     for(i=1; i<50; i++){
  83         t *= x/(i*i);
  84         v += t;
  85     }
  86     return v;
  87 }
  88
  89 /**
  90  * builds a polyphase filterbank.
  91  * @param factor resampling factor
  92  * @param scale wanted sum of coefficients for each filter
  93  * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
  94  */
  95 void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
  96     int ph, i;
  97     double x, y, w, tab[tap_count];
  98     const int center= (tap_count-1)/2;
  99
 100     /* if upsampling, only need to interpolate, no filter */
 101     if (factor > 1.0)
 102         factor = 1.0;
 103
 104     for(ph=0;ph<phase_count;ph++) {
 105         double norm = 0;
 106         for(i=0;i<tap_count;i++) {
 107             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
 108             if (x == 0) y = 1.0;
 109             else        y = sin(x) / x;
 110             switch(type){
 111             case 0:{
 112                 const float d= -0.5; //first order derivative = -0.5
 113                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
 114                 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
 115                 else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
 116                 break;}
 117             case 1:
 118                 w = 2.0*x / (factor*tap_count) + M_PI;
 119                 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
 120                 break;
 121             default:
 122                 w = 2.0*x / (factor*tap_count*M_PI);
 123                 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
 124                 break;
 125             }
 126
 127             tab[i] = y;
 128             norm += y;
 129         }
 130
 131         /* normalize so that an uniform color remains the same */
 132         for(i=0;i<tap_count;i++) {
 133 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
 134             filter[ph * tap_count + i] = tab[i] / norm;
 135 #else
 136             filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
 137 #endif
 138         }
 139     }
 140 #if 0
 141     {
 142 #define LEN 1024
 143         int j,k;
 144         double sine[LEN + tap_count];
 145         double filtered[LEN];
 146         double maxff=-2, minff=2, maxsf=-2, minsf=2;
 147         for(i=0; i<LEN; i++){
 148             double ss=0, sf=0, ff=0;
 149             for(j=0; j<LEN+tap_count; j++)
 150                 sine[j]= cos(i*j*M_PI/LEN);
 151             for(j=0; j<LEN; j++){
 152                 double sum=0;
 153                 ph=0;
 154                 for(k=0; k<tap_count; k++)
 155                     sum += filter[ph * tap_count + k] * sine[k+j];
 156                 filtered[j]= sum / (1<<FILTER_SHIFT);
 157                 ss+= sine[j + center] * sine[j + center];
 158                 ff+= filtered[j] * filtered[j];
 159                 sf+= sine[j + center] * filtered[j];
 160             }
 161             ss= sqrt(2*ss/LEN);
 162             ff= sqrt(2*ff/LEN);
 163             sf= 2*sf/LEN;
 164             maxff= FFMAX(maxff, ff);
 165             minff= FFMIN(minff, ff);
 166             maxsf= FFMAX(maxsf, sf);
 167             minsf= FFMIN(minsf, sf);
 168             if(i%11==0){
 169                 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);
 170                 minff=minsf= 2;
 171                 maxff=maxsf= -2;
 172             }
 173         }
 174     }
 175 #endif
 176 }
 177
 178 /**
 179  * Initializes an audio resampler.
 180  * Note, if either rate is not an integer then simply scale both rates up so they are.
 181  */
 182 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
 183     AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
 184     double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
 185     int phase_count= 1<<phase_shift;
 186
 187     c->phase_shift= phase_shift;
 188     c->phase_mask= phase_count-1;
 189     c->linear= linear;
 190
 191     c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
 192     c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
 193     av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
 194     memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
 195     c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
 196
 197     c->src_incr= out_rate;
 198     c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
 199     c->index= -phase_count*((c->filter_length-1)/2);
 200
 201     return c;
 202 }
 203
 204 void av_resample_close(AVResampleContext *c){
 205     av_freep(&c->filter_bank);
 206     av_freep(&c);
 207 }
 208
 209 /**
 210  * Compensates samplerate/timestamp drift. The compensation is done by changing
 211  * the resampler parameters, so no audible clicks or similar distortions ocur
 212  * @param compensation_distance distance in output samples over which the compensation should be performed
 213  * @param sample_delta number of output samples which should be output less
 214  *
 215  * example: av_resample_compensate(c, 10, 500)
 216  * here instead of 510 samples only 500 samples would be output
 217  *
 218  * note, due to rounding the actual compensation might be slightly different,
 219  * especially if the compensation_distance is large and the in_rate used during init is small
 220  */
 221 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
 222 //    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
 223     c->compensation_distance= compensation_distance;
 224     c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
 225 }
 226
 227 /**
 228  * resamples.
 229  * @param src an array of unconsumed samples
 230  * @param consumed the number of samples of src which have been consumed are returned here
 231  * @param src_size the number of unconsumed samples available
 232  * @param dst_size the amount of space in samples available in dst
 233  * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
 234  * @return the number of samples written in dst or -1 if an error occured
 235  */
 236 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
 237     int dst_index, i;
 238     int index= c->index;
 239     int frac= c->frac;
 240     int dst_incr_frac= c->dst_incr % c->src_incr;
 241     int dst_incr=      c->dst_incr / c->src_incr;
 242     int compensation_distance= c->compensation_distance;
 243
 244   if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
 245         int64_t index2= ((int64_t)index)<<32;
 246         int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
 247         dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
 248
 249         for(dst_index=0; dst_index < dst_size; dst_index++){
 250             dst[dst_index] = src[index2>>32];
 251             index2 += incr;
 252         }
 253         frac += dst_index * dst_incr_frac;
 254         index += dst_index * dst_incr;
 255         index += frac / c->src_incr;
 256         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         assert(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 #if 0
 316     if(update_ctx && !c->compensation_distance){
 317 #undef rand
 318         av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
 319 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
 320     }
 321 #endif
 322
 323     return dst_index;
 324 }