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https://github.com/FFmpeg/FFmpeg.git
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b5875b9111
Similar to libswscale this does resampling and format convertion, just for audio instead of video. changing sampling rate, sample formats, channel layouts and sample packing all in one with a very simple public interface. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
353 lines
12 KiB
C
353 lines
12 KiB
C
/*
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* audio resampling
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* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of FFmpeg.
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*
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* 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
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* 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
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* 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
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* @author Michael Niedermayer <michaelni@gmx.at>
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*/
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#include "libavutil/log.h"
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#include "swresample_internal.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
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#define FELEM2 int32_t
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#define FELEML int64_t
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#define FELEM_MAX INT16_MAX
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#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
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#define FELEM2 int64_t
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#define FELEML int64_t
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#define FELEM_MAX INT32_MAX
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#define FELEM_MIN INT32_MIN
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#define WINDOW_TYPE 12
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#else
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#define FILTER_SHIFT 0
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#define FELEM double
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#define FELEM2 double
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#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;
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int ideal_dst_incr;
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int dst_incr;
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int index;
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int frac;
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int src_incr;
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int compensation_distance;
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int phase_shift;
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int phase_mask;
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int linear;
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double factor;
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}AVResampleContext;
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/**
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* 0th order modified bessel function of the first kind.
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*/
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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;
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int i;
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static const double inv[100]={
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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),
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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),
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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),
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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),
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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),
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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),
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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),
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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),
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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),
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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)
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};
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x= x*x/4;
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for(i=0; v != lastv; i++){
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lastv=v;
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t *= x*inv[i];
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v += t;
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}
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return v;
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}
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/**
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* builds a polyphase filterbank.
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* @param factor resampling factor
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* @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;
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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)
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return AVERROR(ENOMEM);
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/* if upsampling, only need to interpolate, no filter */
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if (factor > 1.0)
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factor = 1.0;
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for(ph=0;ph<phase_count;ph++) {
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double norm = 0;
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for(i=0;i<tap_count;i++) {
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x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
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if (x == 0) y = 1.0;
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else y = sin(x) / x;
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switch(type){
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case 0:{
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const float d= -0.5; //first order derivative = -0.5
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x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
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if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
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else y= d*(-4 + 8*x - 5*x*x + x*x*x);
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break;}
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case 1:
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w = 2.0*x / (factor*tap_count) + M_PI;
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y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
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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;
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}
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tab[i] = y;
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norm += y;
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}
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/* normalize so that an uniform color remains the same */
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for(i=0;i<tap_count;i++) {
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#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
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filter[ph * tap_count + i] = tab[i] / norm;
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#else
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filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
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#endif
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}
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}
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#if 0
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{
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#define LEN 1024
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int j,k;
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double sine[LEN + tap_count];
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double filtered[LEN];
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double maxff=-2, minff=2, maxsf=-2, minsf=2;
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for(i=0; i<LEN; i++){
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double ss=0, sf=0, ff=0;
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for(j=0; j<LEN+tap_count; j++)
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sine[j]= cos(i*j*M_PI/LEN);
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for(j=0; j<LEN; j++){
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double sum=0;
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ph=0;
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for(k=0; k<tap_count; k++)
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sum += filter[ph * tap_count + k] * sine[k+j];
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filtered[j]= sum / (1<<FILTER_SHIFT);
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ss+= sine[j + center] * sine[j + center];
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ff+= filtered[j] * filtered[j];
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sf+= sine[j + center] * filtered[j];
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}
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ss= sqrt(2*ss/LEN);
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ff= sqrt(2*ff/LEN);
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sf= 2*sf/LEN;
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maxff= FFMAX(maxff, ff);
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minff= FFMIN(minff, ff);
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maxsf= FFMAX(maxsf, sf);
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minsf= FFMIN(minsf, sf);
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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;
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maxff=maxsf= -2;
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}
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}
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}
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#endif
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av_free(tab);
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return 0;
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}
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AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
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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 || c->phase_shift!=phase_shift || c->linear!=linear || c->factor != factor
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|| c->filter_length!=FFMAX((int)ceil(filter_size/factor), 1)){
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c= av_mallocz(sizeof(AVResampleContext));
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if (!c)
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return NULL;
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c->phase_shift= phase_shift;
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c->phase_mask= phase_count-1;
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c->linear= linear;
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c->factor= factor;
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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)
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goto error;
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if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
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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));
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c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
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}
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c->compensation_distance= 0;
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c->src_incr= out_rate;
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c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
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c->index= -phase_count*((c->filter_length-1)/2);
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c->frac= 0;
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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);
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return NULL;
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}
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void swr_resample_free(AVResampleContext **c){
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if(!*c)
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return;
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av_freep(&(*c)->filter_bank);
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av_freep(c);
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}
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void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
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AVResampleContext *c= s->resample;
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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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}
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int swr_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
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int dst_index, i;
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int index= c->index;
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int frac= c->frac;
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int dst_incr_frac= c->dst_incr % c->src_incr;
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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;
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int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
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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];
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index2 += incr;
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}
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frac += dst_index * dst_incr_frac;
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index += dst_index * dst_incr;
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index += frac / c->src_incr;
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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);
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int sample_index= index >> c->phase_shift;
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FELEM2 val=0;
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if(sample_index < 0){
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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){
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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];
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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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}
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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;
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dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
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#endif
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frac += dst_incr_frac;
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index += dst_incr;
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if(frac >= c->src_incr){
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frac -= c->src_incr;
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index++;
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}
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if(dst_index + 1 == compensation_distance){
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compensation_distance= 0;
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dst_incr_frac= c->ideal_dst_incr % c->src_incr;
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dst_incr= c->ideal_dst_incr / c->src_incr;
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}
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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){
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compensation_distance -= dst_index;
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assert(compensation_distance > 0);
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}
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if(update_ctx){
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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;
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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){
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#undef rand
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av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
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av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
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}
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#endif
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return dst_index;
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}
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int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
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int i, ret= -1;
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for(i=0; i<dst->ch_count; i++){
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ret= swr_resample(c, dst->ch[i], src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
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}
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return ret;
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}
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