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ac25b31ede
After making some blind tests on a small collection of music samples for home usage. It turned out that the default cutoff was too low. The impact of filter_size was not clearly distinguishable (the results were on the edge) with the music samples but turned out to be clearly audible in some synthetic samples. Thanks to Daniel for helping out with the listening tests. Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
373 lines
14 KiB
C
373 lines
14 KiB
C
/*
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* audio resampling
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* Copyright (c) 2004-2012 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 "libavutil/avassert.h"
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#include "swresample_internal.h"
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typedef struct ResampleContext {
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const AVClass *av_class;
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uint8_t *filter_bank;
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int filter_length;
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int filter_alloc;
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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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enum SwrFilterType filter_type;
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int kaiser_beta;
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double factor;
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enum AVSampleFormat format;
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int felem_size;
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int filter_shift;
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} ResampleContext;
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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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av_assert2(i<99);
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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 filter_type filter type
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* @param kaiser_beta kaiser window beta
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* @return 0 on success, negative on error
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*/
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static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale,
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int filter_type, int kaiser_beta){
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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(filter_type){
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case SWR_FILTER_TYPE_CUBIC:{
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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 SWR_FILTER_TYPE_BLACKMAN_NUTTALL:
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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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case SWR_FILTER_TYPE_KAISER:
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w = 2.0*x / (factor*tap_count*M_PI);
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y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0)));
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break;
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default:
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av_assert0(0);
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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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switch(c->format){
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case AV_SAMPLE_FMT_S16P:
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for(i=0;i<tap_count;i++)
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((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX);
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break;
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case AV_SAMPLE_FMT_S32P:
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for(i=0;i<tap_count;i++)
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((int32_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
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break;
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case AV_SAMPLE_FMT_FLTP:
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for(i=0;i<tap_count;i++)
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((float*)filter)[ph * alloc + i] = tab[i] * scale / norm;
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break;
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case AV_SAMPLE_FMT_DBLP:
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for(i=0;i<tap_count;i++)
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((double*)filter)[ph * alloc + i] = tab[i] * scale / norm;
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break;
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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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static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta,
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double precision, int cheby){
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double cutoff = cutoff0? cutoff0 : 0.97;
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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) || c->format != format
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|| c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) {
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c = av_mallocz(sizeof(*c));
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if (!c)
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return NULL;
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c->format= format;
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c->felem_size= av_get_bytes_per_sample(c->format);
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switch(c->format){
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case AV_SAMPLE_FMT_S16P:
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c->filter_shift = 15;
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break;
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case AV_SAMPLE_FMT_S32P:
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c->filter_shift = 30;
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break;
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case AV_SAMPLE_FMT_FLTP:
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case AV_SAMPLE_FMT_DBLP:
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c->filter_shift = 0;
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break;
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default:
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av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n");
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av_assert0(0);
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}
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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_alloc = FFALIGN(c->filter_length, 8);
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c->filter_bank = av_mallocz(c->filter_alloc*(phase_count+1)*c->felem_size);
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c->filter_type = filter_type;
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c->kaiser_beta = kaiser_beta;
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if (!c->filter_bank)
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goto error;
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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))
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goto error;
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memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size);
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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);
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}
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c->compensation_distance= 0;
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if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
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goto error;
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c->ideal_dst_incr= c->dst_incr;
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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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static void resample_free(ResampleContext **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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static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){
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c->compensation_distance= compensation_distance;
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if (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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else
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c->dst_incr = c->ideal_dst_incr;
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return 0;
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}
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#define TEMPLATE_RESAMPLE_S16
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#include "resample_template.c"
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#undef TEMPLATE_RESAMPLE_S16
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#define TEMPLATE_RESAMPLE_S32
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#include "resample_template.c"
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#undef TEMPLATE_RESAMPLE_S32
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#define TEMPLATE_RESAMPLE_FLT
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#include "resample_template.c"
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#undef TEMPLATE_RESAMPLE_FLT
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#define TEMPLATE_RESAMPLE_DBL
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#include "resample_template.c"
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#undef TEMPLATE_RESAMPLE_DBL
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// XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed
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#if HAVE_MMXEXT_INLINE
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#include "x86/resample_mmx.h"
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#define TEMPLATE_RESAMPLE_S16_MMX2
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#include "resample_template.c"
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#undef TEMPLATE_RESAMPLE_S16_MMX2
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#if HAVE_SSSE3_INLINE
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#define TEMPLATE_RESAMPLE_S16_SSSE3
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#include "resample_template.c"
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#undef TEMPLATE_RESAMPLE_S16_SSSE3
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#endif
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#endif // HAVE_MMXEXT_INLINE
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static int multiple_resample(ResampleContext *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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int av_unused mm_flags = av_get_cpu_flags();
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int need_emms= 0;
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for(i=0; i<dst->ch_count; i++){
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#if HAVE_MMXEXT_INLINE
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#if HAVE_SSSE3_INLINE
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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);
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else
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#endif
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if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){
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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);
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need_emms= 1;
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} else
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#endif
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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);
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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);
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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);
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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);
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}
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if(need_emms)
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emms_c();
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return ret;
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}
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static int64_t get_delay(struct SwrContext *s, int64_t base){
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ResampleContext *c = s->resample;
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int64_t num = s->in_buffer_count - (c->filter_length-1)/2;
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num <<= c->phase_shift;
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num -= c->index;
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num *= c->src_incr;
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num -= c->frac;
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return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift);
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}
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static int resample_flush(struct SwrContext *s) {
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AudioData *a= &s->in_buffer;
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int i, j, ret;
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if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
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return ret;
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av_assert0(a->planar);
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for(i=0; i<a->ch_count; i++){
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for(j=0; j<s->in_buffer_count; j++){
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memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
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a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
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}
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}
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s->in_buffer_count += (s->in_buffer_count+1)/2;
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return 0;
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}
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struct Resampler const swri_resampler={
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resample_init,
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resample_free,
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multiple_resample,
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resample_flush,
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set_compensation,
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get_delay,
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};
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