1 /********************************************************************
3 * THIS FILE IS PART OF THE Ogg Vorbis SOFTWARE CODEC SOURCE CODE. *
4 * USE, DISTRIBUTION AND REPRODUCTION OF THIS SOURCE IS GOVERNED BY *
5 * THE GNU PUBLIC LICENSE 2, WHICH IS INCLUDED WITH THIS SOURCE. *
6 * PLEASE READ THESE TERMS DISTRIBUTING. *
8 * THE OggSQUISH SOURCE CODE IS (C) COPYRIGHT 1994-1999 *
9 * by 1999 Monty <monty@xiph.org> and The XIPHOPHORUS Company *
10 * http://www.xiph.org/ *
12 ********************************************************************
14 function: LPC low level routines
15 author: Monty <monty@xiph.org>
16 modifications by: Monty
17 last modification date: Aug 22 1999
19 ********************************************************************/
21 /* Some of these routines (autocorrelator, LPC coefficient estimator)
22 are derived from code written by Jutta Degener and Carsten Bormann;
23 thus we include their copyright below. The entirety of this file
24 is freely redistributable on the condition that both of these
25 copyright notices are preserved without modification. */
27 /* Preserved Copyright: *********************************************/
29 /* Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
30 Technische Universita"t Berlin
32 Any use of this software is permitted provided that this notice is not
33 removed and that neither the authors nor the Technische Universita"t
34 Berlin are deemed to have made any representations as to the
35 suitability of this software for any purpose nor are held responsible
36 for any defects of this software. THERE IS ABSOLUTELY NO WARRANTY FOR
39 As a matter of courtesy, the authors request to be informed about uses
40 this software has found, about bugs in this software, and about any
41 improvements that may be of general interest.
47 *********************************************************************/
57 /* This is pared down for Vorbis where we only use LPC to encode
58 spectral envelope curves. Thus we only are interested in
59 generating the coefficients and recovering the curve from the
60 coefficients. Autocorrelation LPC coeff generation algorithm
61 invented by N. Levinson in 1947, modified by J. Durbin in 1959. */
63 /* Input : n element envelope curve
64 Output: m lpc coefficients, excitation energy */
66 double memcof(double *data, int n, int m, double *d){
68 double p=0.,wk1[n],wk2[n],wkm[m],xms;
70 memset(wk1,0,sizeof(wk1));
71 memset(wk2,0,sizeof(wk2));
72 memset(wkm,0,sizeof(wkm));
74 for (j=0;j<n;j++) p += data[j]*data[j];
80 for (j=2;j<=n-1;j++) {
86 double num=0.,denom=0.;
87 for (j=1;j<=(n-k);j++) {
89 num += wk1[j-1]*wk2[j-1];
90 denom += wk1[j-1]*wk1[j-1] + wk2[j-1]*wk2[j-1];
95 xms *= (1.0-d[k-1]*d[k-1]);
97 for (i=1;i<=(k-1);i++)
98 d[i-1]=wkm[i-1]-d[k-1]*wkm[k-i-1];
100 if (k == m) return xms;
102 for (i=1;i<=k;i++) wkm[i-1]=d[i-1];
103 for (j=1;j<=(n-k-1);j++) {
105 wk1[j-1] -= wkm[k-1]*wk2[j-1];
106 wk2[j-1]=wk2[j]-wkm[k-1]*wk1[j];
112 static double vorbis_gen_lpc(double *curve,int n,double *lpc,int m){
113 double aut[m+1],work[n+n],error;
117 /* input is a real curve. make it complex-real */
125 drft_backward(&dl,work);
128 /* The autocorrelation will not be circular. Shift, else we lose
129 most of the power in the edges. */
131 for(i=0,j=n/2;i<n/2;){
137 /* autocorrelation, p+1 lag coefficients */
142 for(i=j;i<n;i++)d+=work[i]*work[i-j];
146 /* Generate lpc coefficients from autocorr values */
150 memset(lpc,0,m*sizeof(double));
157 /* Sum up this iteration's reflection coefficient; note that in
158 Vorbis we don't save it. If anyone wants to recycle this code
159 and needs reflection coefficients, save the results of 'r' from
162 for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
165 /* Update LPC coefficients and total error */
170 lpc[j]+=r*lpc[i-1-j];
173 if(i%2)lpc[j]+=lpc[j]*r;
178 /* we need the error value to know how big an impulse to hit the
184 /* One can do this the long way by generating the transfer function in
185 the time domain and taking the forward FFT of the result. The
186 results from direct calculation are cleaner and faster. If one
187 looks at the below in the context of the calling function, there's
188 lots of redundant trig, but at least it's clear */
190 static double vorbis_lpc_magnitude(double w,double *lpc, int m){
192 double real=1,imag=0;
195 real+=lpc[k]*cos(wn);
196 imag+=lpc[k]*sin(wn);
199 return(1./sqrt(real*real+imag*imag));
202 /* On top of this basic LPC infrastructure we introduce two modifications:
204 1) Filter generation is limited in the resolution of features it
205 can represent (this is more obvious when the filter is looked at as
206 a set of LSP coefficients). Human perception of the audio spectrum
207 is logarithmic not only in amplitude, but also frequency. Because
208 the high frequency features we'll need to encode will be broader
209 than the low frequency features, filter generation will be
210 dominated by higher frequencies (when most of the energy is in the
211 lowest frequencies, and greatest perceived resolution is in the
212 midrange). To avoid this effect, Vorbis encodes the frequency
213 spectrum with a biased log frequency scale. The intent is to
214 roughly equalize the sizes of the octaves (see xlogmap.h)
216 2) When we change the frequency scale, we also change the
217 (apparent) relative energies of the bands; that is, on a log scale
218 covering 5 octaves, the highest octave goes from being represented
219 in half the bins, to only 1/32 of the bins. If the amplitudes
220 remain the same, we have divided the energy represented by the
221 highest octave by 16 (as far as Levinson-Durbin is concerned).
222 This will seriously skew filter generation, which bases calculation
223 on the mean square error with respect to energy. Thus, Vorbis
224 normalizes the amplitudes of the log spectrum frequencies to keep
225 the relative octave energies correct. */
227 /* n == size of vector to be used for filter, m == order of filter,
228 oct == octaves in normalized scale, encode_p == encode (1) or
231 double lpc_init(lpc_lookup *l,int n, int m, int oct, int encode_p){
232 double bias=LOG_BIAS(n,oct);
233 double scale=(float)n/(float)oct; /* where n==mapped */
238 l->escale=malloc(n*sizeof(double));
239 l->dscale=malloc(n*sizeof(double));
240 l->norm=malloc(n*sizeof(double));
243 /* how much 'real estate' in the log domain does the bin in the
244 linear domain represent? */
245 double logA=LOG_X(i-.5,bias);
246 double logB=LOG_X(i+.5,bias);
247 l->norm[i]=logB-logA; /* this much */
250 /* the scale is encode/decode specific for algebraic simplicity */
256 l->escale[i]=LINEAR_X(i/scale,bias);
259 /* decode; encode may use this too */
262 double w=1./oct*M_PI;
264 l->dscale[i]=LOG_X(i,bias)*w;
268 void lpc_clear(lpc_lookup *l){
270 if(l->escale)free(l->escale);
277 /* less efficient than the decode side (written for clarity). We're
278 not bottlenecked here anyway */
280 double vorbis_curve_to_lpc(double *curve,double *lpc,lpc_lookup *l){
281 /* map the input curve to a log curve for encoding */
283 /* for clarity, mapped and n are both represented although setting
284 'em equal is a decent rule of thumb. */
292 /* fairly correct for low frequencies, naieve for high frequencies
293 (suffers from undersampling) */
295 for(i=0;i<mapped;i++){
296 double lin=l->escale[i];
299 double del=lin-floor(lin);
301 work[i]=(curve[a]/l->norm[a]*(1.-del)+
302 curve[b]/l->norm[b]*del);
306 memcpy(curve,work,sizeof(work));
308 return vorbis_gen_lpc(work,mapped,lpc,l->m);
311 /* generate the whole freq response curve on an LPC IIR filter */
313 void vorbis_lpc_to_curve(double *curve,double *lpc,double amp,lpc_lookup *l){
316 curve[i]=vorbis_lpc_magnitude(l->dscale[i],lpc,l->m)*amp*l->norm[i];
319 /* find frequency response of LPC filter only at nonsero residue
320 points and apply the envelope to the residue */
322 void vorbis_lpc_apply(double *residue,double *lpc,double amp,lpc_lookup *l){
326 residue[i]*=vorbis_lpc_magnitude(l->dscale[i],lpc,l->m)*amp*l->norm[i];