libavcodec/resample2.c
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 /*
  * audio resampling
  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
  *
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  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
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  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
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  * version 2.1 of the License, or (at your option) any later version.
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  *
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  * FFmpeg is distributed in the hope that it will be useful,
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  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
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  * License along with FFmpeg; if not, write to the Free Software
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  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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  */
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 /**
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  * @file
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  * audio resampling
  * @author Michael Niedermayer <michaelni@gmx.at>
  */
 
 #include "avcodec.h"
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 #include "dsputil.h"
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 #ifndef CONFIG_RESAMPLE_HP
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 #define FILTER_SHIFT 15
 
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 #define FELEM int16_t
 #define FELEM2 int32_t
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 #define FELEML int64_t
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 #define FELEM_MAX INT16_MAX
 #define FELEM_MIN INT16_MIN
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 #define WINDOW_TYPE 9
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 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
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 #define FILTER_SHIFT 30
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 #define FELEM int32_t
 #define FELEM2 int64_t
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 #define FELEML int64_t
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 #define FELEM_MAX INT32_MAX
 #define FELEM_MIN INT32_MIN
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 #define WINDOW_TYPE 12
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 #else
 #define FILTER_SHIFT 0
 
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 #define FELEM double
 #define FELEM2 double
 #define FELEML double
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 #define WINDOW_TYPE 24
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 #endif
 
 
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 typedef struct AVResampleContext{
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     const AVClass *av_class;
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     FELEM *filter_bank;
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     int filter_length;
     int ideal_dst_incr;
     int dst_incr;
     int index;
     int frac;
     int src_incr;
     int compensation_distance;
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     int phase_shift;
     int phase_mask;
     int linear;
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 }AVResampleContext;
 
 /**
  * 0th order modified bessel function of the first kind.
  */
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 static double bessel(double x){
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     double v=1;
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     double lastv=0;
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     double t=1;
     int i;
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     x= x*x/4;
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     for(i=1; v != lastv; i++){
         lastv=v;
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         t *= x/(i*i);
         v += t;
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     }
     return v;
 }
 
 /**
  * builds a polyphase filterbank.
  * @param factor resampling factor
  * @param scale wanted sum of coefficients for each filter
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  * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
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  * @return 0 on success, negative on error
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  */
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 static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
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     int ph, i;
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     double x, y, w;
     double *tab = av_malloc(tap_count * sizeof(*tab));
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     const int center= (tap_count-1)/2;
 
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     if (!tab)
         return AVERROR(ENOMEM);
 
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     /* if upsampling, only need to interpolate, no filter */
     if (factor > 1.0)
         factor = 1.0;
 
     for(ph=0;ph<phase_count;ph++) {
         double norm = 0;
         for(i=0;i<tap_count;i++) {
             x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
             if (x == 0) y = 1.0;
             else        y = sin(x) / x;
             switch(type){
             case 0:{
                 const float d= -0.5; //first order derivative = -0.5
                 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
                 if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
                 else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
                 break;}
             case 1:
                 w = 2.0*x / (factor*tap_count) + M_PI;
                 y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
                 break;
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             default:
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                 w = 2.0*x / (factor*tap_count*M_PI);
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                 y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
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                 break;
             }
 
             tab[i] = y;
             norm += y;
         }
 
         /* normalize so that an uniform color remains the same */
         for(i=0;i<tap_count;i++) {
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 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
             filter[ph * tap_count + i] = tab[i] / norm;
 #else
             filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
 #endif
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         }
     }
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 #if 0
     {
 #define LEN 1024
         int j,k;
         double sine[LEN + tap_count];
         double filtered[LEN];
         double maxff=-2, minff=2, maxsf=-2, minsf=2;
         for(i=0; i<LEN; i++){
             double ss=0, sf=0, ff=0;
             for(j=0; j<LEN+tap_count; j++)
                 sine[j]= cos(i*j*M_PI/LEN);
             for(j=0; j<LEN; j++){
                 double sum=0;
                 ph=0;
                 for(k=0; k<tap_count; k++)
                     sum += filter[ph * tap_count + k] * sine[k+j];
                 filtered[j]= sum / (1<<FILTER_SHIFT);
                 ss+= sine[j + center] * sine[j + center];
                 ff+= filtered[j] * filtered[j];
                 sf+= sine[j + center] * filtered[j];
             }
             ss= sqrt(2*ss/LEN);
             ff= sqrt(2*ff/LEN);
             sf= 2*sf/LEN;
             maxff= FFMAX(maxff, ff);
             minff= FFMIN(minff, ff);
             maxsf= FFMAX(maxsf, sf);
             minsf= FFMIN(minsf, sf);
             if(i%11==0){
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                 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);
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                 minff=minsf= 2;
                 maxff=maxsf= -2;
             }
         }
     }
 #endif
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     av_free(tab);
     return 0;
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 }
 
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 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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     AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
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     double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
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     int phase_count= 1<<phase_shift;
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     if (!c)
         return NULL;
 
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     c->phase_shift= phase_shift;
     c->phase_mask= phase_count-1;
     c->linear= linear;
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     c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
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     c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
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     if (!c->filter_bank)
         goto error;
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     if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
         goto error;
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     memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
     c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
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     if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
         goto error;
     c->ideal_dst_incr= c->dst_incr;
 
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     c->index= -phase_count*((c->filter_length-1)/2);
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     return c;
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 error:
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     av_free(c->filter_bank);
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     av_free(c);
     return NULL;
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 }
 
 void av_resample_close(AVResampleContext *c){
     av_freep(&c->filter_bank);
     av_freep(&c);
 }
 
 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
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 //    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
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     c->compensation_distance= compensation_distance;
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     c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
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 }
 
 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
     int dst_index, i;
     int index= c->index;
     int frac= c->frac;
     int dst_incr_frac= c->dst_incr % c->src_incr;
     int dst_incr=      c->dst_incr / c->src_incr;
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     int compensation_distance= c->compensation_distance;
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   if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
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         int64_t index2= ((int64_t)index)<<32;
         int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
         dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
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         for(dst_index=0; dst_index < dst_size; dst_index++){
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             dst[dst_index] = src[index2>>32];
             index2 += incr;
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         }
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         index += dst_index * dst_incr;
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         index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
         frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
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   }else{
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     for(dst_index=0; dst_index < dst_size; dst_index++){
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         FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
         int sample_index= index >> c->phase_shift;
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         FELEM2 val=0;
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         if(sample_index < 0){
             for(i=0; i<c->filter_length; i++)
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                 val += src[FFABS(sample_index + i) % src_size] * filter[i];
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         }else if(sample_index + c->filter_length > src_size){
             break;
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         }else if(c->linear){
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             FELEM2 v2=0;
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             for(i=0; i<c->filter_length; i++){
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                 val += src[sample_index + i] * (FELEM2)filter[i];
                 v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
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             }
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             val+=(v2-val)*(FELEML)frac / c->src_incr;
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         }else{
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             for(i=0; i<c->filter_length; i++){
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                 val += src[sample_index + i] * (FELEM2)filter[i];
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             }
         }
 
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 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
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         dst[dst_index] = av_clip_int16(lrintf(val));
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 #else
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         val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
         dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
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 #endif
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         frac += dst_incr_frac;
         index += dst_incr;
         if(frac >= c->src_incr){
             frac -= c->src_incr;
             index++;
         }
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         if(dst_index + 1 == compensation_distance){
             compensation_distance= 0;
             dst_incr_frac= c->ideal_dst_incr % c->src_incr;
             dst_incr=      c->ideal_dst_incr / c->src_incr;
         }
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     }
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   }
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     *consumed= FFMAX(index, 0) >> c->phase_shift;
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     if(index>=0) index &= c->phase_mask;
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     if(compensation_distance){
         compensation_distance -= dst_index;
         assert(compensation_distance > 0);
     }
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     if(update_ctx){
         c->frac= frac;
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         c->index= index;
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         c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
         c->compensation_distance= compensation_distance;
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     }
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 #if 0
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     if(update_ctx && !c->compensation_distance){
 #undef rand
         av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
 av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
     }
 #endif
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     return dst_index;
 }