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ffmpeg / libavcodec / resample2.c @ ca74c0a1

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/*
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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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}