1 files changed, 0 insertions, 472 deletions
diff --git a/gr-codec2-vocoder/src/lib/codec2/interp.c b/gr-codec2-vocoder/src/lib/codec2/interp.c
deleted file mode 100644
index 135d8c9e79..0000000000
--- a/gr-codec2-vocoder/src/lib/codec2/interp.c
+++ /dev/null
@@ -1,472 +0,0 @@
-/*---------------------------------------------------------------------------*\
-
-  FILE........: interp.c
-  AUTHOR......: David Rowe
-  DATE CREATED: 9/10/09
-
-  Interpolation of 20ms frames to 10ms frames.
-
-\*---------------------------------------------------------------------------*/
-
-/*
-  Copyright (C) 2009 David Rowe
-
-  All rights reserved.
-
-  This program is free software; you can redistribute it and/or modify
-  it under the terms of the GNU Lesser General Public License version 2.1, as
-  published by the Free Software Foundation.  This program 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 General Public
-  License for more details.
-
-  You should have received a copy of the GNU Lesser General Public License
-  along with this program; if not, see <http://www.gnu.org/licenses/>.
-*/
-
-#include <assert.h>
-#include <math.h>
-#include <string.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "defines.h"
-#include "interp.h"
-#include "lsp.h"
-#include "quantise.h"
-#include "dump.h"
-
-float sample_log_amp(MODEL *model, float w);
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: interp()	     
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 22/8/10 
-        
-  Given two frames decribed by model parameters 20ms apart, determines
-  the model parameters of the 10ms frame between them.  Assumes
-  voicing is available for middle (interpolated) frame.  Outputs are
-  amplitudes and Wo for the interpolated frame.
-
-  This version can interpolate the amplitudes between two frames of
-  different Wo and L.
-
-  This version works by log linear interpolation, but listening tests
-  showed it creates problems in background noise, e.g. hts2a and mmt1.
-  When this function is used (--dec mode) bg noise appears to be
-  amplitude modulated, and gets louder.  The interp_lsp() function
-  below seems to do a better job.
-  
-\*---------------------------------------------------------------------------*/
-
-void interpolate(
-  MODEL *interp,    /* interpolated model params                     */
-  MODEL *prev,      /* previous frames model params                  */
-  MODEL *next       /* next frames model params                      */
-)
-{
-    int   l;
-    float w,log_amp;
-
-    /* Wo depends on voicing of this and adjacent frames */
-
-    if (interp->voiced) {
-	if (prev->voiced && next->voiced)
-	    interp->Wo = (prev->Wo + next->Wo)/2.0;
-	if (!prev->voiced && next->voiced)
-	    interp->Wo = next->Wo;
-	if (prev->voiced && !next->voiced)
-	    interp->Wo = prev->Wo;
-    }
-    else {
-	interp->Wo = TWO_PI/P_MAX;
-    }
-    interp->L = PI/interp->Wo;
-
-    /* Interpolate amplitudes using linear interpolation in log domain */
-
-    for(l=1; l<=interp->L; l++) {
-	w = l*interp->Wo;
-	log_amp = (sample_log_amp(prev, w) + sample_log_amp(next, w))/2.0;
-	interp->A[l] = pow(10.0, log_amp);
-    }
-}
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: sample_log_amp()
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 22/8/10 
-        
-  Samples the amplitude envelope at an arbitrary frequency w.  Uses
-  linear interpolation in the log domain to sample between harmonic
-  amplitudes.
-  
-\*---------------------------------------------------------------------------*/
-
-float sample_log_amp(MODEL *model, float w)
-{
-    int   m;
-    float f, log_amp;
-
-    assert(w > 0.0); assert (w <= PI);
-
-    m = 0;
-    while ((m+1)*model->Wo < w) m++;
-    f = (w - m*model->Wo)/model->Wo;
-    assert(f <= 1.0);
-
-    if (m < 1) {
-	log_amp = f*log10(model->A[1] + 1E-6);
-    }
-    else if ((m+1) > model->L) {
-	log_amp = (1.0-f)*log10(model->A[model->L] + 1E-6);
-    }
-    else {
-	log_amp = (1.0-f)*log10(model->A[m] + 1E-6) + 
-                  f*log10(model->A[m+1] + 1E-6);
-	//printf("m=%d A[m] %f A[m+1] %f x %f %f %f\n", m, model->A[m], 
-	//       model->A[m+1], pow(10.0, log_amp),
-	//       (1-f), f);
-    }
-
-    return log_amp;
-}
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: sample_log_amp_quad()
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 9 March 2011 
-        
-  Samples the amplitude envelope at an arbitrary frequency w.  Uses
-  quadratic interpolation in the log domain to sample between harmonic
-  amplitudes.
-  
-  y(x) = ax*x + bx + c
-
-  We assume three points are x=-1, x=0, x=1, which we map to m-1,m,m+1
-
-  c = y(0)
-  b = (y(1) - y(-1))/2
-  a = y(-1) + b - y(0)
-
-\*---------------------------------------------------------------------------*/
-
-float sample_log_amp_quad(MODEL *model, float w)
-{
-    int   m;
-    float a,b,c,x, log_amp;
-
-    assert(w > 0.0); assert (w <= PI);
-
-    m = floor(w/model->Wo + 0.5);
-    if (m < 2) m = 2;
-    if (m > (model->L-1)) m = model->L-1;
-    c = log10(model->A[m]+1E-6);
-    b = (log10(model->A[m+1]+1E-6) - log10(model->A[m-1]+1E-6))/2.0;
-    a = log10(model->A[m-1]+1E-6) + b - c;
-    x = (w - m*model->Wo)/model->Wo;
-
-    log_amp = a*x*x + b*x + c;
-    //printf("m=%d A[m-1] %f A[m] %f A[m+1] %f w %f x %f log_amp %f\n", m,
-    //	   model->A[m-1], 
-    //	   model->A[m], model->A[m+1], w, x, pow(10.0, log_amp));
-    return log_amp;
-}
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: sample_log_amp_quad_nl()
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 10 March 2011
-        
-  Samples the amplitude envelope at an arbitrary frequency w.  Uses
-  quadratic interpolation in the log domain to sample between harmonic
-  amplitudes.  This version can handle non-linear steps along a freq
-  axis defined by arbitrary steps.
-  
-  y(x) = ax*x + bx + c
-
-  We assume three points are (x_1,y_1), (0,y0) and (x1,y1).
-
-\*---------------------------------------------------------------------------*/
-
-float sample_log_amp_quad_nl(
-			     float w[],     /* frequency points            */
-			     float A[],     /* for these amplitude samples */
-			     int np,        /* number of frequency points  */
-			     float w_sample /* frequency of new samples    */
-)
-{
-    int   m,i;
-    float a,b,c,x, log_amp, best_dist;
-    float x_1, x1;
-    float y_1, y0, y1;
-
-    //printf("w_sample  %f\n", w_sample);
-    assert(w_sample >= 0.0); assert (w_sample <= 1.1*PI);
-
-    /* find closest point to centre quadratic interpolator */
-
-    best_dist = 1E32;
-    for (i=0; i<np; i++)
-	if (fabs(w[i] - w_sample) < best_dist) {
-	    best_dist = fabs(w[i] - w_sample);
-	    m = i; 
-	}
-    
-    /* stay one point away from edge of array */
-
-    if (m < 1) m = 1;
-    if (m > (np-2)) m = np - 2;
-
-    /* find polynomial coeffs */
-
-    x_1 = w[m-1]- w[m]; x1 = w[m+1] - w[m];
-    y_1 = log10(A[m-1]+1E-6);
-    y0  = log10(A[m]+1E-6);
-    y1  = log10(A[m+1]+1E-6);
-
-    c = y0;
-    a = (y_1*x1 - y1*x_1 + c*x_1 - c*x1)/(x_1*x_1*x1 - x1*x1*x_1);
-    b = (y1 -a*x1*x1 - c)/x1;
-    x = w_sample - w[m];
-    
-    //printf("%f   %f  %f\n", w[0], w[1], w[2]);
-    //printf("%f %f  %f %f  %f %f\n", x_1, y_1, 0.0, y0, x1, y1);
-    log_amp = a*x*x + b*x + c;
-    //printf("a %f  b %f  c %f\n", a, b, c);
-    //printf("m=%d A[m-1] %f A[m] %f A[m+1] %f w_sample %f w[m] %f x %f log_amp %f\n", m,
-    //	   A[m-1], 
-    //	   A[m], A[m+1], w_sample, w[m], x, log_amp);
-    //exit(0);
-    return log_amp;
-}
-
-#define M_MAX 40
-
-float fres[] = {100,   200,  300,  400,  500,  600,  700,  800,  900, 1000,
-		1200, 1400, 1600, 1850, 2100, 2350, 2600, 2900, 3400, 3800};
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: resample_amp_nl()
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 7 March 2011
-        
-  Converts the current model with L {Am} samples spaced Wo apart to 
-  RES_POINTS samples spaced Wo/RES_POINTS apart.  Then subtracts
-  from the previous frames samples to get the delta.
-
-\*---------------------------------------------------------------------------*/
-
-void resample_amp_fixed(MODEL *model, 
-			float w[], float A[],
-			float wres[], float Ares[],
-			float AresdB_prev[], 
-			float AresdB[], 
-			float deltat[])
-{
-    int   i;
-
-    for(i=1; i<=model->L; i++) {
-	w[i-1] = i*model->Wo;
-	A[i-1] = model->A[i];
-    }
-
-    for(i=0; i<RES_POINTS; i++) {
-	wres[i] = fres[i]*PI/4000.0;
-    }
-    
-    for(i=0; i<RES_POINTS; i++) {
-	Ares[i] = pow(10.0,sample_log_amp_quad_nl(w, A, model->L, wres[i]));
-    }
-
-    /* work out delta T vector for this frame */
-
-    for(i=0; i<RES_POINTS; i++) {
-	AresdB[i] = 20.0*log10(Ares[i]);
-	deltat[i] = AresdB[i] - AresdB_prev[i];
-    }
-
-}
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: resample_amp_nl()
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 7 March 2011
-        
-  Converts the current model with L {Am} samples spaced Wo apart to M
-  samples spaced Wo/M apart.  Then converts back to L {Am} samples.
-  used to prototype constant rate Amplitude encoding ideas.
-  
-  Returns the SNR in dB.
-
-\*---------------------------------------------------------------------------*/
-
-float resample_amp_nl(MODEL *model, int m, float AresdB_prev[])
-{
-    int   i;
-    float w[MAX_AMP], A[MAX_AMP];
-    float wres[MAX_AMP], Ares[MAX_AMP], AresdB[MAX_AMP];
-    float signal, noise, snr;
-    float new_A;
-    float deltat[MAX_AMP], deltat_q[MAX_AMP], AresdB_q[MAX_AMP];
-
-    resample_amp_fixed(model, w, A, wres, Ares, AresdB_prev, AresdB, deltat);
-
-    /* quantise delta T vector */
-
-    for(i=0; i<RES_POINTS; i++) {
-	noise = 3.0*(1.0 - 2.0*rand()/RAND_MAX);
-	//noise = 0.0;
-	deltat_q[i] = deltat[i] + noise;
-    }
-
-    /* recover Ares vector */
-
-    for(i=0; i<RES_POINTS; i++) {
-	AresdB_q[i] = AresdB_prev[i] + deltat_q[i];
-	Ares[i] = pow(10.0, AresdB_q[i]/20.0);
-	//printf("%d %f %f\n", i, AresdB[i], AresdB_q[i]);
-    }
-
-    /* update memory based on version at decoder */
-
-    for(i=0; i<RES_POINTS; i++) {
-	AresdB_prev[i] = AresdB_q[i];
-    }
-
-#ifdef DUMP
-    dump_resample(wres,Ares,M_MAX);
-#endif
-
-    signal = noise = 0.0;
-    
-    for(i=1; i<model->L; i++) {
-	new_A = pow(10.0,sample_log_amp_quad_nl(wres, Ares, RES_POINTS, model->Wo*i));
-	signal += pow(model->A[i], 2.0);
-	noise  += pow(model->A[i] - new_A, 2.0);
-	//printf("%f %f\n", model->A[i], new_A);
-	model->A[i] = new_A;
-    }
-
-    snr = 10.0*log10(signal/noise);
-    printf("snr = %3.2f\n", snr);
-    //exit(0);
-    return snr;
-}
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: resample_amp()
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 10 March 2011
-        
-  Converts the current model with L {Am} samples spaced Wo apart to M
-  samples with a non-linear spacing.  Then converts back to L {Am}
-  samples.  used to prototype constant rate Amplitude encoding ideas.
-  
-  Returns the SNR in dB.
-
-\*---------------------------------------------------------------------------*/
-
-float resample_amp(MODEL *model, int m)
-{
-    int   i;
-    MODEL model_m;
-    float new_A, signal, noise, snr, log_amp_dB;
-    float n_db = 0.0;
-
-    model_m.Wo = PI/(float)m;
-    model_m.L = PI/model_m.Wo;
-
-    for(i=1; i<=model_m.L; i++) {
-	log_amp_dB    = 20.0*sample_log_amp_quad(model, i*model_m.Wo);
-	log_amp_dB   += n_db*(1.0 - 2.0*rand()/RAND_MAX);
-	model_m.A[i]  = pow(10,log_amp_dB/20.0);
-    }
-
-    //dump_resample(&model_m);
-
-    signal = noise = 0.0;
-    
-    for(i=1; i<model->L/4; i++) {
-	new_A = pow(10,sample_log_amp_quad(&model_m, i*model->Wo));
-	signal += pow(model->A[i], 2.0);
-	noise  += pow(model->A[i] - new_A, 2.0);
-	//printf("%f %f\n", model->A[i], new_A);
-	model->A[i] = new_A;
-    }
-
-    snr = 10.0*log10(signal/noise);
-    //printf("snr = %3.2f\n", snr);
-    //exit(0);
-    return snr;
-}
-
-/*---------------------------------------------------------------------------*\
-
-  FUNCTION....: interp_lsp()	     
-  AUTHOR......: David Rowe			      
-  DATE CREATED: 10 Nov 2010
-        
-  Given two frames decribed by model parameters 20ms apart, determines
-  the model parameters of the 10ms frame between them.  Assumes
-  voicing is available for middle (interpolated) frame.  Outputs are
-  amplitudes and Wo for the interpolated frame.
-
-  This version uses interpolation of LSPs, seems to do a better job
-  with bg noise.
-  
-\*---------------------------------------------------------------------------*/
-
-void interpolate_lsp(
-  MODEL *interp,    /* interpolated model params                     */
-  MODEL *prev,      /* previous frames model params                  */
-  MODEL *next,      /* next frames model params                      */
-  float *prev_lsps, /* previous frames LSPs                          */
-  float  prev_e,    /* previous frames LPC energy                    */
-  float *next_lsps, /* next frames LSPs                              */
-  float  next_e,    /* next frames LPC energy                        */
-  float *ak_interp  /* interpolated aks for this frame                */
-		     )
-{
-    int   l,i;
-    float lsps[LPC_ORD],e;
-    float snr;
-
-    /* Wo depends on voicing of this and adjacent frames */
-
-    if (interp->voiced) {
-	if (prev->voiced && next->voiced)
-	    interp->Wo = (prev->Wo + next->Wo)/2.0;
-	if (!prev->voiced && next->voiced)
-	    interp->Wo = next->Wo;
-	if (prev->voiced && !next->voiced)
-	    interp->Wo = prev->Wo;
-    }
-    else {
-	interp->Wo = TWO_PI/P_MAX;
-    }
-    interp->L = PI/interp->Wo;
-
-    /* interpolate LSPs */
-
-    for(i=0; i<LPC_ORD; i++) {
-	lsps[i] = (prev_lsps[i] + next_lsps[i])/2.0;
-    }
-
-    /* Interpolate LPC energy in log domain */
-
-    e = pow(10.0, (log10(prev_e) + log10(next_e))/2.0);
-
-    /* convert back to amplitudes */
-
-    lsp_to_lpc(lsps, ak_interp, LPC_ORD);
-    aks_to_M2(ak_interp, LPC_ORD, interp, e, &snr, 0); 
-}