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: Oct 11 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 vorbis_gen_lpc(double *curve,double *lpc,lpc_lookup *l){
69 double aut[m+1],work[n+n],error;
73 /* input is a real curve. make it complex-real */
74 /* This mixes phase, but the LPC generation doesn't care. */
76 work[i*2]=curve[i]*fscale;
81 drft_backward(&l->fft,work);
83 /* The autocorrelation will not be circular. Shift, else we lose
84 most of the power in the edges. */
86 for(i=0,j=n/2;i<n/2;){
92 /* autocorrelation, p+1 lag coefficients */
97 for(i=j;i<n;i++)d+=work[i]*work[i-j];
101 /* Generate lpc coefficients from autocorr values */
105 memset(lpc,0,m*sizeof(double));
112 /* Sum up this iteration's reflection coefficient; note that in
113 Vorbis we don't save it. If anyone wants to recycle this code
114 and needs reflection coefficients, save the results of 'r' from
117 for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
120 /* Update LPC coefficients and total error */
125 lpc[j]+=r*lpc[i-1-j];
128 if(i%2)lpc[j]+=lpc[j]*r;
133 /* we need the error value to know how big an impulse to hit the
139 /* On top of this basic LPC infrastructure we introduce two modifications:
141 1) Filter generation is limited in the resolution of features it
142 can represent (this is more obvious when the filter is looked at as
143 a set of LSP coefficients). Human perception of the audio spectrum
144 is logarithmic not only in amplitude, but also frequency. Because
145 the high frequency features we'll need to encode will be broader
146 than the low frequency features, filter generation will be
147 dominated by higher frequencies (when most of the energy is in the
148 lowest frequencies, and greatest perceived resolution is in the
149 midrange). To avoid this effect, Vorbis encodes the frequency
150 spectrum with a biased log frequency scale. The intent is to
151 roughly equalize the sizes of the octaves (see xlogmap.h)
153 2) When we change the frequency scale, we also change the
154 (apparent) relative energies of the bands; that is, on a log scale
155 covering 5 octaves, the highest octave goes from being represented
156 in half the bins, to only 1/32 of the bins. If the amplitudes
157 remain the same, we have divided the energy represented by the
158 highest octave by 16 (as far as Levinson-Durbin is concerned).
159 This will seriously skew filter generation, which bases calculation
160 on the mean square error with respect to energy. Thus, Vorbis
161 normalizes the amplitudes of the log spectrum frequencies to keep
162 the relative octave energies correct. */
164 /* n == size of vector to be used for filter, m == order of filter,
165 oct == octaves in normalized scale, encode_p == encode (1) or
168 void lpc_init(lpc_lookup *l,int n, int mapped, int m, int oct, int encode_p){
169 double bias=LOG_BIAS(n,oct);
170 double scale=(float)mapped/(float)oct; /* where n==mapped */
173 memset(l,0,sizeof(lpc_lookup));
178 l->iscale=malloc(n*sizeof(int));
179 l->norm=malloc(n*sizeof(double));
182 /* how much 'real estate' in the log domain does the bin in the
183 linear domain represent? */
184 double logA=LOG_X(i-.5,bias);
185 double logB=LOG_X(i+.5,bias);
186 l->norm[i]=logB-logA; /* this much */
189 /* the scale is encode/decode specific for algebraic simplicity */
193 l->bscale=malloc(n*sizeof(int));
194 l->escale=malloc(n*sizeof(double));
197 l->escale[i]=LINEAR_X(i/scale,bias);
198 l->bscale[i]=rint(LOG_X(i,bias)*scale);
202 /* decode; encode may use this too */
204 drft_init(&l->fft,mapped*2);
206 double w=1./oct*M_PI;
208 l->iscale[i]=rint(LOG_X(i,bias)/oct*mapped);
209 if(l->iscale[i]>=l->ln)l->iscale[i]=l->ln-1;
214 void lpc_clear(lpc_lookup *l){
216 if(l->bscale)free(l->bscale);
217 if(l->escale)free(l->escale);
225 /* less efficient than the decode side (written for clarity). We're
226 not bottlenecked here anyway */
228 double vorbis_curve_to_lpc(double *curve,double *lpc,lpc_lookup *l){
229 /* map the input curve to a log curve for encoding */
231 /* for clarity, mapped and n are both represented although setting
232 'em equal is a decent rule of thumb. The below must be reworked
233 slightly if mapped != n */
239 /* fairly correct for low frequencies, naieve for high frequencies
240 (suffers from undersampling) */
241 for(i=0;i<mapped;i++){
242 double lin=l->escale[i];
245 double del=lin-floor(lin);
247 work[i]=(curve[a]/l->norm[a]*(1.-del)+
248 curve[b]/l->norm[b]*del);
252 return vorbis_gen_lpc(work,lpc,l);
256 /* One can do this the long way by generating the transfer function in
257 the time domain and taking the forward FFT of the result. The
258 results from direct calculation are cleaner and faster. If one
259 looks at the below in the context of the calling function, there's
260 lots of redundant trig, but at least it's clear */
262 /* This version does a linear curve generation and then later
263 interpolates the log curve from the linear curve. This could stand
264 optimization; it could both be more precise as well as not compute
265 quite a few unused values */
267 static void _vlpc_de_helper(double *curve,double *lpc,double amp,
270 memset(curve,0,sizeof(double)*l->ln*2);
273 curve[i*2+1]=lpc[i]/4/amp;
274 curve[i*2+2]=-lpc[i]/4/amp;
277 drft_backward(&l->fft,curve); /* reappropriated ;-) */
282 curve[0]=(1./(curve[0]+unit));
283 for(i=1;i<l->ln;i++){
284 double real=(curve[i]+curve[l2-i]);
285 double imag=(curve[i]-curve[l2-i]);
286 curve[i]=(1./hypot(real+unit,imag));
292 /* generate the whole freq response curve on an LPC IIR filter */
294 void vorbis_lpc_to_curve(double *curve,double *lpc,double amp,lpc_lookup *l){
295 double lcurve[l->ln*2];
298 _vlpc_de_helper(lcurve,lpc,amp,l);
301 curve[i]=lcurve[l->iscale[i]]*l->norm[i];
304 void vorbis_lpc_apply(double *residue,double *lpc,double amp,lpc_lookup *l){
305 double lcurve[l->ln*2];
308 _vlpc_de_helper(lcurve,lpc,amp,l);
311 residue[i]*=lcurve[l->iscale[i]]*l->norm[i];