325 lines
11 KiB
C
325 lines
11 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 resample2.c
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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 "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
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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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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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}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 t=1;
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int i;
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x= x*x/4;
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for(i=1; i<50; i++){
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t *= x/(i*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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*/
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void av_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, tab[tap_count];
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const int center= (tap_count-1)/2;
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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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}
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/**
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* Initializes an audio resampler.
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* Note, if either rate is not an integer then simply scale both rates up so they are.
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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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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->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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av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
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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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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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return c;
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}
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void av_resample_close(AVResampleContext *c){
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av_freep(&c->filter_bank);
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av_freep(&c);
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}
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/**
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* Compensates samplerate/timestamp drift. The compensation is done by changing
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* the resampler parameters, so no audible clicks or similar distortions occur
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* @param compensation_distance distance in output samples over which the compensation should be performed
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* @param sample_delta number of output samples which should be output less
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*
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* example: av_resample_compensate(c, 10, 500)
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* here instead of 510 samples only 500 samples would be output
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*
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* note, due to rounding the actual compensation might be slightly different,
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* especially if the compensation_distance is large and the in_rate used during init is small
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*/
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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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}
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/**
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* resamples.
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* @param src an array of unconsumed samples
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* @param consumed the number of samples of src which have been consumed are returned here
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* @param src_size the number of unconsumed samples available
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* @param dst_size the amount of space in samples available in dst
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* @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
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* @return the number of samples written in dst or -1 if an error occurred
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*/
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int av_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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