path: root/libswresample/resample2.c

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

#include "libavutil/log.h"
#include "swresample_internal.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{
    const AVClass *av_class;
    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;
    double factor;
}AVResampleContext;

/**
 * 0th order modified bessel function of the first kind.
 */
static double bessel(double x){
    double v=1;
    double lastv=0;
    double t=1;
    int i;
    static const double inv[100]={
 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),
 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),
 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),
 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),
 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),
 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),
 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),
 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),
 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),
 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)
    };

    x= x*x/4;
    for(i=0; v != lastv; i++){
        lastv=v;
        t *= x*inv[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
 * @return 0 on success, negative on error
 */
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
    int ph, i;
    double x, y, w;
    double *tab = av_malloc(tap_count * sizeof(*tab));
    const int center= (tap_count-1)/2;

    if (!tab)
        return AVERROR(ENOMEM);

    /* 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

    av_free(tab);
    return 0;
}

AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
    int phase_count= 1<<phase_shift;

    if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
           || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
        c = av_mallocz(sizeof(AVResampleContext));
        if (!c)
            return NULL;

        c->phase_shift   = phase_shift;
        c->phase_mask    = phase_count - 1;
        c->linear        = linear;
        c->factor        = factor;
        c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
        c->filter_bank   = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
        if (!c->filter_bank)
            goto error;
        if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
            goto error;
        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->compensation_distance= 0;
    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);
    c->frac= 0;

    return c;
error:
    av_free(c->filter_bank);
    av_free(c);
    return NULL;
}

void swr_resample_free(AVResampleContext **c){
    if(!*c)
        return;
    av_freep(&(*c)->filter_bank);
    av_freep(c);
}

void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
    AVResampleContext *c= s->resample;
//    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;
}

int swr_resample(AVResampleContext *c, short *dst, const 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;
}

int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
    int i, ret= -1;

    for(i=0; i<dst->ch_count; i++){
        ret= swr_resample(c, (short*)dst->ch[i], (const short*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
    }

    return ret;
}