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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 library 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 of the License, or (at your option) any later version.
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 *
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 * This library 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 this library; if not, write to the Free Software
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 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
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 *
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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 "common.h"
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#include "dsputil.h"
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#define PHASE_SHIFT 10
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#define PHASE_COUNT (1<<PHASE_SHIFT)
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#define PHASE_MASK (PHASE_COUNT-1)
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#define FILTER_SHIFT 15
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typedef struct AVResampleContext{
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    short *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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}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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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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    for(i=1; i<50; i++){
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        t *= i;
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        v += pow(x*x/4, i)/(t*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->kaiser windowed sinc beta=16
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 */
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void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
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    int ph, i, v;
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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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        double e= 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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            case 2:
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                w = 2.0*x / (factor*tap_count*M_PI);
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                y *= bessel(16*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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            v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767);
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            filter[ph * tap_count + i] = v;
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            e += tab[i] * scale / norm - v;
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        }
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    }
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}
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/**
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 * initalizes a audio resampler.
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 * note, if either rate is not a 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){
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    AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
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    double factor= FFMIN(out_rate / (double)in_rate, 1.0);
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    memset(c, 0, sizeof(AVResampleContext));
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    c->filter_length= ceil(16.0/factor);
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    c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
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    av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
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    c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
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    c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 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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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 wont 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 occured
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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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    for(dst_index=0; dst_index < dst_size; dst_index++){
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        short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
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        int sample_index= index >> PHASE_SHIFT;
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        int 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[ABS(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{
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#if 0
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            int64_t v=0;
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            int sub_phase= (frac<<12) / c->src_incr;
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            for(i=0; i<c->filter_length; i++){
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                int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
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                v += src[sample_index + i] * coeff;
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            }
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            val= v>>12;
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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] * filter[i];
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            }
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#endif
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        }
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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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        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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    *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
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    index= FFMIN(index, 0);
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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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}