libavcodec/resample2.c

   1 /*
   2  * audio resampling
   3  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
   4  *
   5  * This library is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2 of the License, or (at your option) any later version.
   9  *
  10  * This library is distributed in the hope that it will be useful,
  11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with this library; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  18  *
  19  */
  20
  21 /**
  22  * @file resample2.c
  23  * audio resampling
  24  * @author Michael Niedermayer <michaelni@gmx.at>
  25  */
  26
  27 #include "avcodec.h"
  28 #include "common.h"
  29
  30 #define PHASE_SHIFT 10
  31 #define PHASE_COUNT (1<<PHASE_SHIFT)
  32 #define PHASE_MASK (PHASE_COUNT-1)
  33 #define FILTER_SHIFT 15
  34
  35 typedef struct AVResampleContext{
  36     short *filter_bank;
  37     int filter_length;
  38     int ideal_dst_incr;
  39     int dst_incr;
  40     int index;
  41     int frac;
  42     int src_incr;
  43     int compensation_distance;
  44 }AVResampleContext;
  45
  46 /**
  47  * 0th order modified bessel function of the first kind.
  48  */
  49 double bessel(double x){
  50     double v=1;
  51     double t=1;
  52     int i;
  53
  54     for(i=1; i<50; i++){
  55         t *= i;
  56         v += pow(x*x/4, i)/(t*t);
  57     }
  58     return v;
  59 }
  60
  61 /**
  62  * builds a polyphase filterbank.
  63  * @param factor resampling factor
  64  * @param scale wanted sum of coefficients for each filter
  65  * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
  66  */
  67 void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
  68     int ph, i, v;
  69     double x, y, w, tab[tap_count];
  70     const int center= (tap_count-1)/2;
  71
  72     /* if upsampling, only need to interpolate, no filter */
  73     if (factor > 1.0)
  74         factor = 1.0;
  75
  76     for(ph=0;ph<phase_count;ph++) {
  77         double norm = 0;
  78         double e= 0;
  79         for(i=0;i<tap_count;i++) {
  80             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
  81             if (x == 0) y = 1.0;
  82             else        y = sin(x) / x;
  83             switch(type){
  84             case 0:{
  85                 const float d= -0.5; //first order derivative = -0.5
  86                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
  87                 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
  88                 else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
  89                 break;}
  90             case 1:
  91                 w = 2.0*x / (factor*tap_count) + M_PI;
  92                 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
  93                 break;
  94             case 2:
  95                 w = 2.0*x / (factor*tap_count*M_PI);
  96                 y *= bessel(16*sqrt(FFMAX(1-w*w, 0))) / bessel(16);
  97                 break;
  98             }
  99
 100             tab[i] = y;
 101             norm += y;
 102         }
 103
 104         /* normalize so that an uniform color remains the same */
 105         for(i=0;i<tap_count;i++) {
 106             v = clip(lrintf(tab[i] * scale / norm) + e, -32768, 32767);
 107             filter[ph * tap_count + i] = v;
 108             e += tab[i] * scale / norm - v;
 109         }
 110     }
 111 }
 112
 113 /**
 114  * initalizes a audio resampler.
 115  * note, if either rate is not a integer then simply scale both rates up so they are
 116  */
 117 AVResampleContext *av_resample_init(int out_rate, int in_rate){
 118     AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
 119     double factor= FFMIN(out_rate / (double)in_rate, 1.0);
 120
 121     memset(c, 0, sizeof(AVResampleContext));
 122
 123     c->filter_length= ceil(16.0/factor);
 124     c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
 125     av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
 126     c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 1]= (1<<FILTER_SHIFT)-1;
 127     c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 1;
 128
 129     c->src_incr= out_rate;
 130     c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
 131     c->index= -PHASE_COUNT*((c->filter_length-1)/2);
 132
 133     return c;
 134 }
 135
 136 void av_resample_close(AVResampleContext *c){
 137     av_freep(&c->filter_bank);
 138     av_freep(&c);
 139 }
 140
 141 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
 142     assert(!c->compensation_distance); //FIXME
 143
 144     c->compensation_distance= compensation_distance;
 145     c->dst_incr-= c->ideal_dst_incr * sample_delta / compensation_distance;
 146 }
 147
 148 /**
 149  * resamples.
 150  * @param src an array of unconsumed samples
 151  * @param consumed the number of samples of src which have been consumed are returned here
 152  * @param src_size the number of unconsumed samples available
 153  * @param dst_size the amount of space in samples available in dst
 154  * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
 155  * @return the number of samples written in dst or -1 if an error occured
 156  */
 157 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
 158     int dst_index, i;
 159     int index= c->index;
 160     int frac= c->frac;
 161     int dst_incr_frac= c->dst_incr % c->src_incr;
 162     int dst_incr=      c->dst_incr / c->src_incr;
 163
 164     if(c->compensation_distance && c->compensation_distance < dst_size)
 165         dst_size= c->compensation_distance;
 166
 167     for(dst_index=0; dst_index < dst_size; dst_index++){
 168         short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
 169         int sample_index= index >> PHASE_SHIFT;
 170         int val=0;
 171
 172         if(sample_index < 0){
 173             for(i=0; i<c->filter_length; i++)
 174                 val += src[ABS(sample_index + i)] * filter[i];
 175         }else if(sample_index + c->filter_length > src_size){
 176             break;
 177         }else{
 178 #if 0
 179             int64_t v=0;
 180             int sub_phase= (frac<<12) / c->src_incr;
 181             for(i=0; i<c->filter_length; i++){
 182                 int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
 183                 v += src[sample_index + i] * coeff;
 184             }
 185             val= v>>12;
 186 #else
 187             for(i=0; i<c->filter_length; i++){
 188                 val += src[sample_index + i] * filter[i];
 189             }
 190 #endif
 191         }
 192
 193         val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
 194         dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
 195
 196         frac += dst_incr_frac;
 197         index += dst_incr;
 198         if(frac >= c->src_incr){
 199             frac -= c->src_incr;
 200             index++;
 201         }
 202     }
 203     if(update_ctx){
 204         if(c->compensation_distance){
 205             c->compensation_distance -= index;
 206             if(!c->compensation_distance)
 207                 c->dst_incr= c->ideal_dst_incr;
 208         }
 209         c->frac= frac;
 210         c->index=0;
 211     }
 212     *consumed= index >> PHASE_SHIFT;
 213     return dst_index;
 214 }