mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-22 01:05:05 +02:00
321 lines
10 KiB
C
321 lines
10 KiB
C
/*
|
|
* audio resampling
|
|
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
|
|
*
|
|
* This file is part of Libav.
|
|
*
|
|
* Libav 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.
|
|
*
|
|
* Libav 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 Libav; 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 "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{
|
|
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;
|
|
}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;
|
|
|
|
x= x*x/4;
|
|
for(i=1; v != lastv; i++){
|
|
lastv=v;
|
|
t *= x/(i*i);
|
|
v += t;
|
|
}
|
|
return v;
|
|
}
|
|
|
|
/**
|
|
* Build 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 *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;
|
|
|
|
if (!c)
|
|
return NULL;
|
|
|
|
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));
|
|
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->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;
|
|
error:
|
|
av_free(c->filter_bank);
|
|
av_free(c);
|
|
return NULL;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|