/********************************************************************
* *
- * THIS FILE IS PART OF THE Ogg Vorbis SOFTWARE CODEC SOURCE CODE. *
- * USE, DISTRIBUTION AND REPRODUCTION OF THIS SOURCE IS GOVERNED BY *
- * THE GNU PUBLIC LICENSE 2, WHICH IS INCLUDED WITH THIS SOURCE. *
- * PLEASE READ THESE TERMS DISTRIBUTING. *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
- * THE OggSQUISH SOURCE CODE IS (C) COPYRIGHT 1994-2000 *
- * by Monty <monty@xiph.org> and The XIPHOPHORUS Company *
- * http://www.xiph.org/ *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
+ * by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: LPC low level routines
- last mod: $Id: lpc.c,v 1.10 1999/12/30 07:26:40 xiphmont Exp $
********************************************************************/
*********************************************************************/
#include <stdlib.h>
-#include <stdio.h>
#include <string.h>
#include <math.h>
#include "os.h"
#include "smallft.h"
#include "lpc.h"
-#include "xlogmap.h"
+#include "scales.h"
+#include "misc.h"
-/* This is pared down for Vorbis. Autocorrelation LPC coeff generation
- algorithm invented by N. Levinson in 1947, modified by J. Durbin in
- 1959. */
+/* Autocorrelation LPC coeff generation algorithm invented by
+ N. Levinson in 1947, modified by J. Durbin in 1959. */
/* Input : n elements of time doamin data
Output: m lpc coefficients, excitation energy */
-
-double vorbis_lpc_from_data(double *data,double *lpc,int n,int m){
- double *aut=alloca(sizeof(double)*(m+1));
+float vorbis_lpc_from_data(float *data,float *lpci,int n,int m){
+ double *aut=alloca(sizeof(*aut)*(m+1));
+ double *lpc=alloca(sizeof(*lpc)*(m));
double error;
+ double epsilon;
int i,j;
/* autocorrelation, p+1 lag coefficients */
-
j=m+1;
while(j--){
- double d=0;
- for(i=j;i<n;i++)d+=data[i]*data[i-j];
+ double d=0; /* double needed for accumulator depth */
+ for(i=j;i<n;i++)d+=(double)data[i]*data[i-j];
aut[j]=d;
}
-
+
/* Generate lpc coefficients from autocorr values */
- error=aut[0];
- if(error==0){
- memset(lpc,0,m*sizeof(double));
- return 0;
- }
-
+ /* set our noise floor to about -100dB */
+ error=aut[0] * (1. + 1e-10);
+ epsilon=1e-9*aut[0]+1e-10;
+
for(i=0;i<m;i++){
- double r=-aut[i+1];
+ double r= -aut[i+1];
+
+ if(error<epsilon){
+ memset(lpc+i,0,(m-i)*sizeof(*lpc));
+ goto done;
+ }
/* Sum up this iteration's reflection coefficient; note that in
Vorbis we don't save it. If anyone wants to recycle this code
each iteration. */
for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
- r/=error;
+ r/=error;
/* Update LPC coefficients and total error */
-
+
lpc[i]=r;
for(j=0;j<i/2;j++){
double tmp=lpc[j];
+
lpc[j]+=r*lpc[i-1-j];
lpc[i-1-j]+=r*tmp;
}
- if(i%2)lpc[j]+=lpc[j]*r;
-
- error*=1.0-r*r;
- }
-
- /* we need the error value to know how big an impulse to hit the
- filter with later */
-
- return error;
-}
-
-/* Input : n element envelope spectral curve
- Output: m lpc coefficients, excitation energy */
-
-double vorbis_lpc_from_spectrum(double *curve,double *lpc,lpc_lookup *l){
- int n=l->ln;
- int m=l->m;
- double *work=alloca(sizeof(double)*(n+n));
- double fscale=.5/n;
- int i,j;
-
- /* input is a real curve. make it complex-real */
- /* This mixes phase, but the LPC generation doesn't care. */
- for(i=0;i<n;i++){
- work[i*2]=curve[i]*fscale;
- work[i*2+1]=0;
- }
-
- n*=2;
- drft_backward(&l->fft,work);
-
- /* The autocorrelation will not be circular. Shift, else we lose
- most of the power in the edges. */
-
- for(i=0,j=n/2;i<n/2;){
- double temp=work[i];
- work[i++]=work[j];
- work[j++]=temp;
- }
-
- return(vorbis_lpc_from_data(work,lpc,n,m));
-}
+ if(i&1)lpc[j]+=lpc[j]*r;
-/* On top of this basic LPC infrastructure we introduce two modifications:
-
- 1) Filter generation is limited in the resolution of features it
- can represent (this is more obvious when the filter is looked at as
- a set of LSP coefficients). Human perception of the audio spectrum
- is logarithmic not only in amplitude, but also frequency. Because
- the high frequency features we'll need to encode will be broader
- than the low frequency features, filter generation will be
- dominated by higher frequencies (when most of the energy is in the
- lowest frequencies, and greatest perceived resolution is in the
- midrange). To avoid this effect, Vorbis encodes the frequency
- spectrum with a biased log frequency scale. The intent is to
- roughly equalize the sizes of the octaves (see xlogmap.h)
-
- 2) When we change the frequency scale, we also change the
- (apparent) relative energies of the bands; that is, on a log scale
- covering 5 octaves, the highest octave goes from being represented
- in half the bins, to only 1/32 of the bins. If the amplitudes
- remain the same, we have divided the energy represented by the
- highest octave by 16 (as far as Levinson-Durbin is concerned).
- This will seriously skew filter generation, which bases calculation
- on the mean square error with respect to energy. Thus, Vorbis
- normalizes the amplitudes of the log spectrum frequencies to keep
- the relative octave energies correct. */
-
-/* n == size of vector to be used for filter, m == order of filter,
- oct == octaves in normalized scale, encode_p == encode (1) or
- decode (0) */
-
-void lpc_init(lpc_lookup *l,int n, int mapped, int m, int oct, int encode_p){
- double bias=LOG_BIAS(n,oct);
- double scale=(float)mapped/(float)oct; /* where n==mapped */
- int i;
-
- memset(l,0,sizeof(lpc_lookup));
-
- l->n=n;
- l->ln=mapped;
- l->m=m;
- l->iscale=malloc(n*sizeof(int));
- l->ifrac=malloc(n*sizeof(double));
- l->norm=malloc(n*sizeof(double));
+ error*=1.-r*r;
- for(i=0;i<n;i++){
- /* how much 'real estate' in the log domain does the bin in the
- linear domain represent? */
- double logA=LOG_X(i,bias);
- double logB=LOG_X(i+1.,bias);
- l->norm[i]=logB-logA; /* this much */
}
- /* the scale is encode/decode specific for algebraic simplicity */
-
- if(encode_p){
- /* encode */
- l->escale=malloc(mapped*sizeof(double));
- l->uscale=malloc(n*sizeof(int));
-
- /* undersample guard */
- for(i=0;i<n;i++){
- l->uscale[i]=rint(LOG_X(i,bias)/oct*mapped);
- }
+ done:
- for(i=0;i<mapped;i++){
- l->escale[i]=LINEAR_X(i/scale,bias);
- l->uscale[(int)(floor(l->escale[i]))]=-1;
- l->uscale[(int)(ceil(l->escale[i]))]=-1;
- }
-
-
- }
- /* decode; encode may use this too */
-
- drft_init(&l->fft,mapped*2);
- for(i=0;i<n;i++){
- double is=LOG_X(i,bias)/oct*mapped;
- if(is<0.)is=0.;
-
- l->iscale[i]=floor(is);
- if(l->iscale[i]>=l->ln-1)l->iscale[i]=l->ln-2;
-
- l->ifrac[i]=is-floor(is);
- if(l->ifrac[i]>1.)l->ifrac[i]=1.;
-
- }
-}
-
-void lpc_clear(lpc_lookup *l){
- if(l){
- if(l->escale)free(l->escale);
- drft_clear(&l->fft);
- free(l->iscale);
- free(l->ifrac);
- free(l->norm);
- }
-}
-
-
-/* less efficient than the decode side (written for clarity). We're
- not bottlenecked here anyway */
-
-double vorbis_curve_to_lpc(double *curve,double *lpc,lpc_lookup *l){
- /* map the input curve to a log curve for encoding */
-
- /* for clarity, mapped and n are both represented although setting
- 'em equal is a decent rule of thumb. The below must be reworked
- slightly if mapped != n */
-
- int mapped=l->ln;
- double *work=alloca(sizeof(double)*mapped);
- int i;
-
- /* fairly correct for low frequencies, naieve for high frequencies
- (suffers from undersampling) */
- for(i=0;i<mapped;i++){
- double lin=l->escale[i];
- int a=floor(lin);
- int b=ceil(lin);
- double del=lin-floor(lin);
-
- work[i]=(curve[a]/l->norm[a]*(1.-del)+
- curve[b]/l->norm[b]*del);
-
- }
-
- /* for(i=0;i<l->n;i++)
- if(l->uscale[i]>0)
- if(work[l->uscale[i]]<curve[i])work[l->uscale[i]]=curve[i];*/
-
-#ifdef ANALYSIS
+ /* slightly damp the filter */
{
- int j;
- FILE *out;
- char buffer[80];
- static int frameno=0;
-
- sprintf(buffer,"preloglpc%d.m",frameno++);
- out=fopen(buffer,"w+");
- for(j=0;j<l->ln;j++)
- fprintf(out,"%g\n",work[j]);
- fclose(out);
- }
-#endif
-
- return vorbis_lpc_from_spectrum(work,lpc,l);
-}
-
-
-/* One can do this the long way by generating the transfer function in
- the time domain and taking the forward FFT of the result. The
- results from direct calculation are cleaner and faster.
-
- This version does a linear curve generation and then later
- interpolates the log curve from the linear curve. This could stand
- optimization; it could both be more precise as well as not compute
- quite a few unused values */
-
-void _vlpc_de_helper(double *curve,double *lpc,double amp,
- lpc_lookup *l){
- int i;
- memset(curve,0,sizeof(double)*l->ln*2);
- if(amp==0)return;
-
- for(i=0;i<l->m;i++){
- curve[i*2+1]=lpc[i]/4/amp;
- curve[i*2+2]=-lpc[i]/4/amp;
- }
-
- drft_backward(&l->fft,curve); /* reappropriated ;-) */
-
- {
- int l2=l->ln*2;
- double unit=1./amp;
- curve[0]=(1./(curve[0]*2+unit));
- for(i=1;i<l->ln;i++){
- double real=(curve[i]+curve[l2-i]);
- double imag=(curve[i]-curve[l2-i]);
- curve[i]=(1./hypot(real+unit,imag));
- }
- }
-}
-
-
-/* generate the whole freq response curve on an LPC IIR filter */
-
-void vorbis_lpc_to_curve(double *curve,double *lpc,double amp,lpc_lookup *l){
- double *lcurve=alloca(sizeof(double)*(l->ln*2));
- int i;
-
- _vlpc_de_helper(lcurve,lpc,amp,l);
-
-#ifdef ANALYSIS
- {
- int j;
- FILE *out;
- char buffer[80];
- static int frameno=0;
-
- sprintf(buffer,"loglpc%d.m",frameno++);
- out=fopen(buffer,"w+");
- for(j=0;j<l->ln;j++)
- fprintf(out,"%g\n",lcurve[j]);
- fclose(out);
- }
-#endif
-
- if(amp==0)return;
-
- for(i=0;i<l->n;i++){
- int ii=l->iscale[i];
- curve[i]=((1.-l->ifrac[i])*lcurve[ii]+
- l->ifrac[i]*lcurve[ii+1])*l->norm[i];
- }
-
-}
-
-void vorbis_lpc_apply(double *residue,double *lpc,double amp,lpc_lookup *l){
- double *lcurve=alloca(sizeof(double)*((l->ln+l->n)*2));
- int i;
-
- if(amp==0){
- memset(residue,0,l->n*sizeof(double));
- }else{
-
- _vlpc_de_helper(lcurve,lpc,amp,l);
-
- for(i=0;i<l->n;i++){
- if(residue[i]!=0){
- int ii=l->iscale[i];
- residue[i]*=((1.-l->ifrac[i])*lcurve[ii]+
- l->ifrac[i]*lcurve[ii+1])*l->norm[i];
- }
+ double g = .99;
+ double damp = g;
+ for(j=0;j<m;j++){
+ lpc[j]*=damp;
+ damp*=g;
}
}
-}
-
-/* subtract or add an lpc filter to data. Vorbis doesn't actually use this. */
-
-void vorbis_lpc_residue(double *coeff,double *prime,int m,
- double *data,long n){
-
- /* in: coeff[0...m-1] LPC coefficients
- prime[0...m-1] initial values
- data[0...n-1] data samples
- out: data[0...n-1] residuals from LPC prediction */
- long i,j;
- double *work=alloca(sizeof(double)*(m+n));
- double y;
+ for(j=0;j<m;j++)lpci[j]=(float)lpc[j];
- if(!prime)
- for(i=0;i<m;i++)
- work[i]=0;
- else
- for(i=0;i<m;i++)
- work[i]=prime[i];
+ /* we need the error value to know how big an impulse to hit the
+ filter with later */
- for(i=0;i<n;i++){
- y=0;
- for(j=0;j<m;j++)
- y-=work[i+j]*coeff[m-j-1];
-
- work[i+m]=data[i];
- data[i]-=y;
- }
+ return error;
}
+void vorbis_lpc_predict(float *coeff,float *prime,int m,
+ float *data,long n){
-void vorbis_lpc_predict(double *coeff,double *prime,int m,
- double *data,long n){
-
- /* in: coeff[0...m-1] LPC coefficients
+ /* in: coeff[0...m-1] LPC coefficients
prime[0...m-1] initial values (allocated size of n+m-1)
- data[0...n-1] residuals from LPC prediction
out: data[0...n-1] data samples */
long i,j,o,p;
- double y;
- double *work=alloca(sizeof(double)*(m+n));
+ float y;
+ float *work=alloca(sizeof(*work)*(m+n));
if(!prime)
for(i=0;i<m;i++)
- work[i]=0.;
+ work[i]=0.f;
else
for(i=0;i<m;i++)
work[i]=prime[i];
for(i=0;i<n;i++){
- y=data[i];
+ y=0;
o=i;
p=m;
for(j=0;j<m;j++)
y-=work[o++]*coeff[--p];
-
+
data[i]=work[o]=y;
}
}
-
-
projects / platform / upstream / libvorbis.git / blobdiff