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/*
* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This library 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 of the License, or (at your option) any later version.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/**
* @file resample2.c
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "avcodec.h"
#include "common.h"
#include "dsputil.h"
#define PHASE_SHIFT 10
#define PHASE_COUNT (1<<PHASE_SHIFT)
#define PHASE_MASK (PHASE_COUNT-1)
#define FILTER_SHIFT 15
typedef struct AVResampleContext{
short *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
double bessel(double x){
double v=1;
double t=1;
int i;
for(i=1; i<50; i++){
t *= i;
v += pow(x*x/4, i)/(t*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->kaiser windowed sinc beta=16
*/
void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i, v;
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;
double e= 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;
case 2:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(16*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++) {
v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767);
filter[ph * tap_count + i] = v;
e += tab[i] * scale / norm - v;
}
}
}
/**
* initalizes a audio resampler.
* note, if either rate is not a integer then simply scale both rates up so they are
*/
AVResampleContext *av_resample_init(int out_rate, int in_rate){
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
double factor= FFMIN(out_rate / (double)in_rate, 1.0);
memset(c, 0, sizeof(AVResampleContext));
c->filter_length= ceil(16.0/factor);
c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 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);
}
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 wont 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 occured
*/
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;
if(c->compensation_distance && c->compensation_distance < dst_size)
dst_size= c->compensation_distance;
for(dst_index=0; dst_index < dst_size; dst_index++){
short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
int sample_index= index >> PHASE_SHIFT;
int val=0;
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[ABS(sample_index + i) % src_size] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
}else{
#if 0
int64_t v=0;
int sub_phase= (frac<<12) / c->src_incr;
for(i=0; i<c->filter_length; i++){
int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
v += src[sample_index + i] * coeff;
}
val= v>>12;
#else
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * filter[i];
}
#endif
}
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
}
*consumed= FFMAX(index, 0) >> PHASE_SHIFT;
index= FFMIN(index, 0);
if(update_ctx){
if(c->compensation_distance){
c->compensation_distance -= dst_index;
if(!c->compensation_distance)
c->dst_incr= c->ideal_dst_incr;
}
c->frac= frac;
c->index= index;
}
#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;
}
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