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upload sdk
git-svn-id: svn://kolibrios.org@4349 a494cfbc-eb01-0410-851d-a64ba20cac60
This commit is contained in:
Sergey Semyonov (Serge)
372
contrib/sdk/sources/ffmpeg/libswresample/resample.c
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372
contrib/sdk/sources/ffmpeg/libswresample/resample.c
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@@ -0,0 +1,372 @@
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/*
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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_clipl_int32(llrint(tab[i] * scale / norm));
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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_calloc(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_freep(&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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