view resample2.c @ 8006:c7c1e85d14bc libavcodec

Rename variables to clarify the channel coupling element and corresponding target channel element. Patch by Alex Converse (alex converse gmail com)
author superdump
date Mon, 06 Oct 2008 16:22:11 +0000
parents 3f819263176e
children e9d9d946f213
line wrap: on
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/*
 * audio resampling
 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file resample2.c
 * audio resampling
 * @author Michael Niedermayer <michaelni@gmx.at>
 */

#include "avcodec.h"
#include "dsputil.h"

#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15

#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
#define FILTER_SHIFT 30

#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
#define FILTER_SHIFT 0

#define FELEM double
#define FELEM2 double
#define FELEML double
#define WINDOW_TYPE 24
#endif


typedef struct AVResampleContext{
    FELEM *filter_bank;
    int filter_length;
    int ideal_dst_incr;
    int dst_incr;
    int index;
    int frac;
    int src_incr;
    int compensation_distance;
    int phase_shift;
    int phase_mask;
    int linear;
}AVResampleContext;

/**
 * 0th order modified bessel function of the first kind.
 */
static double bessel(double x){
    double v=1;
    double t=1;
    int i;

    x= x*x/4;
    for(i=1; i<50; i++){
        t *= x/(i*i);
        v += t;
    }
    return v;
}

/**
 * builds a polyphase filterbank.
 * @param factor resampling factor
 * @param scale wanted sum of coefficients for each filter
 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
 */
void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
    int ph, i;
    double x, y, w, tab[tap_count];
    const int center= (tap_count-1)/2;

    /* if upsampling, only need to interpolate, no filter */
    if (factor > 1.0)
        factor = 1.0;

    for(ph=0;ph<phase_count;ph++) {
        double norm = 0;
        for(i=0;i<tap_count;i++) {
            x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
            if (x == 0) y = 1.0;
            else        y = sin(x) / x;
            switch(type){
            case 0:{
                const float d= -0.5; //first order derivative = -0.5
                x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
                if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
                else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
                break;}
            case 1:
                w = 2.0*x / (factor*tap_count) + M_PI;
                y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
                break;
            default:
                w = 2.0*x / (factor*tap_count*M_PI);
                y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
                break;
            }

            tab[i] = y;
            norm += y;
        }

        /* normalize so that an uniform color remains the same */
        for(i=0;i<tap_count;i++) {
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
            filter[ph * tap_count + i] = tab[i] / norm;
#else
            filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
#endif
        }
    }
#if 0
    {
#define LEN 1024
        int j,k;
        double sine[LEN + tap_count];
        double filtered[LEN];
        double maxff=-2, minff=2, maxsf=-2, minsf=2;
        for(i=0; i<LEN; i++){
            double ss=0, sf=0, ff=0;
            for(j=0; j<LEN+tap_count; j++)
                sine[j]= cos(i*j*M_PI/LEN);
            for(j=0; j<LEN; j++){
                double sum=0;
                ph=0;
                for(k=0; k<tap_count; k++)
                    sum += filter[ph * tap_count + k] * sine[k+j];
                filtered[j]= sum / (1<<FILTER_SHIFT);
                ss+= sine[j + center] * sine[j + center];
                ff+= filtered[j] * filtered[j];
                sf+= sine[j + center] * filtered[j];
            }
            ss= sqrt(2*ss/LEN);
            ff= sqrt(2*ff/LEN);
            sf= 2*sf/LEN;
            maxff= FFMAX(maxff, ff);
            minff= FFMIN(minff, ff);
            maxsf= FFMAX(maxsf, sf);
            minsf= FFMIN(minsf, sf);
            if(i%11==0){
                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);
                minff=minsf= 2;
                maxff=maxsf= -2;
            }
        }
    }
#endif
}

/**
 * Initializes an audio resampler.
 * Note, if either rate is not an integer then simply scale both rates up so they are.
 */
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
    AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
    int phase_count= 1<<phase_shift;

    c->phase_shift= phase_shift;
    c->phase_mask= phase_count-1;
    c->linear= linear;

    c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
    c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
    av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
    memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
    c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];

    c->src_incr= out_rate;
    c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
    c->index= -phase_count*((c->filter_length-1)/2);

    return c;
}

void av_resample_close(AVResampleContext *c){
    av_freep(&c->filter_bank);
    av_freep(&c);
}

/**
 * Compensates samplerate/timestamp drift. The compensation is done by changing
 * the resampler parameters, so no audible clicks or similar distortions occur
 * @param compensation_distance distance in output samples over which the compensation should be performed
 * @param sample_delta number of output samples which should be output less
 *
 * example: av_resample_compensate(c, 10, 500)
 * here instead of 510 samples only 500 samples would be output
 *
 * note, due to rounding the actual compensation might be slightly different,
 * especially if the compensation_distance is large and the in_rate used during init is small
 */
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
//    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
    c->compensation_distance= compensation_distance;
    c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
}

/**
 * resamples.
 * @param src an array of unconsumed samples
 * @param consumed the number of samples of src which have been consumed are returned here
 * @param src_size the number of unconsumed samples available
 * @param dst_size the amount of space in samples available in dst
 * @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.
 * @return the number of samples written in dst or -1 if an error occurred
 */
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
    int dst_index, i;
    int index= c->index;
    int frac= c->frac;
    int dst_incr_frac= c->dst_incr % c->src_incr;
    int dst_incr=      c->dst_incr / c->src_incr;
    int compensation_distance= c->compensation_distance;

  if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
        int64_t index2= ((int64_t)index)<<32;
        int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
        dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);

        for(dst_index=0; dst_index < dst_size; dst_index++){
            dst[dst_index] = src[index2>>32];
            index2 += incr;
        }
        frac += dst_index * dst_incr_frac;
        index += dst_index * dst_incr;
        index += frac / c->src_incr;
        frac %= c->src_incr;
  }else{
    for(dst_index=0; dst_index < dst_size; dst_index++){
        FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
        int sample_index= index >> c->phase_shift;
        FELEM2 val=0;

        if(sample_index < 0){
            for(i=0; i<c->filter_length; i++)
                val += src[FFABS(sample_index + i) % src_size] * filter[i];
        }else if(sample_index + c->filter_length > src_size){
            break;
        }else if(c->linear){
            FELEM2 v2=0;
            for(i=0; i<c->filter_length; i++){
                val += src[sample_index + i] * (FELEM2)filter[i];
                v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
            }
            val+=(v2-val)*(FELEML)frac / c->src_incr;
        }else{
            for(i=0; i<c->filter_length; i++){
                val += src[sample_index + i] * (FELEM2)filter[i];
            }
        }

#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
        dst[dst_index] = av_clip_int16(lrintf(val));
#else
        val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
        dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
#endif

        frac += dst_incr_frac;
        index += dst_incr;
        if(frac >= c->src_incr){
            frac -= c->src_incr;
            index++;
        }

        if(dst_index + 1 == compensation_distance){
            compensation_distance= 0;
            dst_incr_frac= c->ideal_dst_incr % c->src_incr;
            dst_incr=      c->ideal_dst_incr / c->src_incr;
        }
    }
  }
    *consumed= FFMAX(index, 0) >> c->phase_shift;
    if(index>=0) index &= c->phase_mask;

    if(compensation_distance){
        compensation_distance -= dst_index;
        assert(compensation_distance > 0);
    }
    if(update_ctx){
        c->frac= frac;
        c->index= index;
        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
        c->compensation_distance= compensation_distance;
    }
#if 0
    if(update_ctx && !c->compensation_distance){
#undef rand
        av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
    }
#endif

    return dst_index;
}