resample2.c 9.99 KB
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/*
 * audio resampling
 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
 *
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 * This file is part of FFmpeg.
 *
 * 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
 * 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
 * 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
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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
 * @author Michael Niedermayer <michaelni@gmx.at>
 */

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#include "libavutil/avassert.h"
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#include "avcodec.h"
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#include "dsputil.h"
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#include "libavutil/common.h"
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#if FF_API_AVCODEC_RESAMPLE

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#ifndef CONFIG_RESAMPLE_HP
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#define FILTER_SHIFT 15

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#define FELEM int16_t
#define FELEM2 int32_t
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#define FELEML int64_t
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#define FELEM_MAX INT16_MAX
#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
#define FELEM2 int64_t
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#define FELEML int64_t
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#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
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#define WINDOW_TYPE 12
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#else
#define FILTER_SHIFT 0

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#define FELEM double
#define FELEM2 double
#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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    const AVClass *av_class;
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    FELEM *filter_bank;
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    int filter_length;
    int ideal_dst_incr;
    int dst_incr;
    int index;
    int frac;
    int src_incr;
    int compensation_distance;
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    int phase_shift;
    int phase_mask;
    int linear;
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}AVResampleContext;

/**
 * 0th order modified bessel function of the first kind.
 */
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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;
    int i;
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    x= x*x/4;
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    for(i=1; v != lastv; i++){
        lastv=v;
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        t *= x/(i*i);
        v += t;
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    }
    return v;
}

/**
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 * Build a polyphase filterbank.
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 * @param factor resampling factor
 * @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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 * @return 0 on success, negative on error
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 */
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static int 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;
    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)
        return AVERROR(ENOMEM);

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    /* 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;
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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;
            }

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

        /* normalize so that an uniform color remains the same */
        for(i=0;i<tap_count;i++) {
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#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
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        }
    }
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#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){
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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;
                maxff=maxsf= -2;
            }
        }
    }
#endif
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    av_free(tab);
    return 0;
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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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    if (!c)
        return NULL;

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    c->phase_shift= phase_shift;
    c->phase_mask= phase_count-1;
    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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    if (!c->filter_bank)
        goto error;
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    if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
        goto error;
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    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];
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    if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
        goto error;
    c->ideal_dst_incr= c->dst_incr;

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    c->index= -phase_count*((c->filter_length-1)/2);
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    return c;
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error:
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    av_free(c->filter_bank);
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    av_free(c);
    return NULL;
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}

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

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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}

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;
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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;
        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);
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        for(dst_index=0; dst_index < dst_size; dst_index++){
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            dst[dst_index] = src[index2>>32];
            index2 += incr;
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        }
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        index += dst_index * dst_incr;
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        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
        frac   = (frac + dst_index * (int64_t)dst_incr_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);
        int sample_index= index >> c->phase_shift;
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        FELEM2 val=0;
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        if(sample_index < 0){
            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){
            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];
                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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#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;
        dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
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#endif
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        frac += dst_incr_frac;
        index += dst_incr;
        if(frac >= c->src_incr){
            frac -= c->src_incr;
            index++;
        }
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        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;
        }
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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){
        compensation_distance -= dst_index;
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        av_assert2(compensation_distance > 0);
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    }
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    if(update_ctx){
        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;
        c->compensation_distance= compensation_distance;
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    }
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    return dst_index;
}
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#endif