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+/*---------------------------------------------------------------------------*\
+
+ FILE........: newamp1.c
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Quantisation functions for the sinusoidal coder, using "newamp1"
+ algorithm that resamples variable rate L [Am} to a fixed rate K then
+ VQs.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ Copyright David Rowe 2017
+
+ 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 <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+
+#include "defines.h"
+#include "phase.h"
+#include "quantise.h"
+#include "mbest.h"
+#include "newamp1.h"
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: interp_para()
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ General 2nd order parabolic interpolator. Used splines originally,
+ but this is much simpler and we don't need much accuracy. Given two
+ vectors of points xp and yp, find interpolated values y at points x.
+
+\*---------------------------------------------------------------------------*/
+
+void interp_para(float y[], float xp[], float yp[], int np, float x[], int n)
+{
+ assert(np >= 3);
+
+ int k,i;
+ float xi, x1, y1, x2, y2, x3, y3, a, b;
+
+ k = 0;
+ for (i=0; i<n; i++) {
+ xi = x[i];
+
+ /* k is index into xp of where we start 3 points used to form parabola */
+
+ while ((xp[k+1] < xi) && (k < (np-3)))
+ k++;
+
+ x1 = xp[k]; y1 = yp[k]; x2 = xp[k+1]; y2 = yp[k+1]; x3 = xp[k+2]; y3 = yp[k+2];
+
+ //printf("k: %d np: %d i: %d xi: %f x1: %f y1: %f\n", k, np, i, xi, x1, y1);
+
+ a = ((y3-y2)/(x3-x2)-(y2-y1)/(x2-x1))/(x3-x1);
+ b = ((y3-y2)/(x3-x2)*(x2-x1)+(y2-y1)/(x2-x1)*(x3-x2))/(x3-x1);
+
+ y[i] = a*(xi-x2)*(xi-x2) + b*(xi-x2) + y2;
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: ftomel()
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Non linear sampling of frequency axis, reducing the "rate" is a
+ first step before VQ
+
+\*---------------------------------------------------------------------------*/
+
+float ftomel(float fHz) {
+ float mel = floorf(2595.0*log10f(1.0 + fHz/700.0)+0.5);
+ return mel;
+}
+
+void mel_sample_freqs_kHz(float rate_K_sample_freqs_kHz[], int K, float mel_start, float mel_end)
+{
+ float step = (mel_end-mel_start)/(K-1);
+ float mel;
+ int k;
+
+ mel = mel_start;
+ for (k=0; k<K; k++) {
+ rate_K_sample_freqs_kHz[k] = 0.7*(POW10F(mel/2595.0) - 1.0);
+ mel += step;
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: resample_const_rate_f()
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Resample Am from time-varying rate L=floor(pi/Wo) to fixed rate K.
+
+\*---------------------------------------------------------------------------*/
+
+void resample_const_rate_f(C2CONST *c2const, MODEL *model, float rate_K_vec[], float rate_K_sample_freqs_kHz[], int K)
+{
+ int m;
+ float AmdB[MAX_AMP+1], rate_L_sample_freqs_kHz[MAX_AMP+1], AmdB_peak;
+
+ /* convert rate L=pi/Wo amplitude samples to fixed rate K */
+
+ AmdB_peak = -100.0;
+ for(m=1; m<=model->L; m++) {
+ AmdB[m] = 20.0*log10f(model->A[m]+1E-16);
+ if (AmdB[m] > AmdB_peak) {
+ AmdB_peak = AmdB[m];
+ }
+ rate_L_sample_freqs_kHz[m] = m*model->Wo*(c2const->Fs/2000.0)/M_PI;
+ //printf("m: %d AmdB: %f AmdB_peak: %f sf: %f\n", m, AmdB[m], AmdB_peak, rate_L_sample_freqs_kHz[m]);
+ }
+
+ /* clip between peak and peak -50dB, to reduce dynamic range */
+
+ for(m=1; m<=model->L; m++) {
+ if (AmdB[m] < (AmdB_peak-50.0)) {
+ AmdB[m] = AmdB_peak-50.0;
+ }
+ }
+
+ interp_para(rate_K_vec, &rate_L_sample_freqs_kHz[1], &AmdB[1], model->L, rate_K_sample_freqs_kHz, K);
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: rate_K_mbest_encode
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Two stage rate K newamp1 VQ quantiser using mbest search.
+
+\*---------------------------------------------------------------------------*/
+
+float rate_K_mbest_encode(int *indexes, float *x, float *xq, int ndim, int mbest_entries)
+{
+ int i, j, n1, n2;
+ const float *codebook1 = newamp1vq_cb[0].cb;
+ const float *codebook2 = newamp1vq_cb[1].cb;
+ struct MBEST *mbest_stage1, *mbest_stage2;
+ float target[ndim];
+ int index[MBEST_STAGES];
+ float mse, tmp;
+
+ /* codebook is compiled for a fixed K */
+
+ assert(ndim == newamp1vq_cb[0].k);
+
+ mbest_stage1 = mbest_create(mbest_entries);
+ mbest_stage2 = mbest_create(mbest_entries);
+ for(i=0; i<MBEST_STAGES; i++)
+ index[i] = 0;
+
+ /* Stage 1 */
+
+ mbest_search(codebook1, x, ndim, newamp1vq_cb[0].m, mbest_stage1, index);
+
+ /* Stage 2 */
+
+ for (j=0; j<mbest_entries; j++) {
+ index[1] = n1 = mbest_stage1->list[j].index[0];
+ for(i=0; i<ndim; i++)
+ target[i] = x[i] - codebook1[ndim*n1+i];
+ mbest_search(codebook2, target, ndim, newamp1vq_cb[1].m, mbest_stage2, index);
+ }
+
+ n1 = mbest_stage2->list[0].index[1];
+ n2 = mbest_stage2->list[0].index[0];
+ mse = 0.0;
+ for (i=0;i<ndim;i++) {
+ tmp = codebook1[ndim*n1+i] + codebook2[ndim*n2+i];
+ mse += (x[i]-tmp)*(x[i]-tmp);
+ xq[i] = tmp;
+ }
+
+ mbest_destroy(mbest_stage1);
+ mbest_destroy(mbest_stage2);
+
+ indexes[0] = n1; indexes[1] = n2;
+
+ return mse;
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: post_filter
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Post Filter, has a big impact on speech quality after VQ. When used
+ on a mean removed rate K vector, it raises formants, and suppresses
+ anti-formants. As it manipulates amplitudes, we normalise energy to
+ prevent clipping or large level variations. pf_gain of 1.2 to 1.5
+ (dB) seems to work OK. Good area for further investigations and
+ improvements in speech quality.
+
+\*---------------------------------------------------------------------------*/
+
+void post_filter_newamp1(float vec[], float sample_freq_kHz[], int K, float pf_gain)
+{
+ int k;
+
+ /*
+ vec is rate K vector describing spectrum of current frame lets
+ pre-emp before applying PF. 20dB/dec over 300Hz. Postfilter
+ affects energy of frame so we measure energy before and after
+ and normalise. Plenty of room for experimentation here.
+ */
+
+ float pre[K];
+ float e_before = 0.0;
+ float e_after = 0.0;
+ for(k=0; k<K; k++) {
+ pre[k] = 20.0*log10f(sample_freq_kHz[k]/0.3);
+ vec[k] += pre[k];
+ e_before += POW10F(vec[k]/10.0);
+ vec[k] *= pf_gain;
+ e_after += POW10F(vec[k]/10.0);
+ }
+
+ float gain = e_after/e_before;
+ float gaindB = 10*log10f(gain);
+
+ for(k=0; k<K; k++) {
+ vec[k] -= gaindB;
+ vec[k] -= pre[k];
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: interp_Wo_v
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Decoder side interpolation of Wo and voicing, to go from 25 Hz
+ sample rate used over channel to 100Hz internal sample rate of Codec 2.
+
+\*---------------------------------------------------------------------------*/
+
+void interp_Wo_v(float Wo_[], int L_[], int voicing_[], float Wo1, float Wo2, int voicing1, int voicing2)
+{
+ int i;
+ int M = 4; /* interpolation rate */
+
+ for(i=0; i<M; i++)
+ voicing_[i] = 0;
+
+ if (!voicing1 && !voicing2) {
+ for(i=0; i<M; i++)
+ Wo_[i] = 2.0*M_PI/100.0;
+ }
+
+ if (voicing1 && !voicing2) {
+ Wo_[0] = Wo_[1] = Wo1;
+ Wo_[2] = Wo_[3] = 2.0*M_PI/100.0;
+ voicing_[0] = voicing_[1] = 1;
+ }
+
+ if (!voicing1 && voicing2) {
+ Wo_[0] = Wo_[1] = 2.0*M_PI/100.0;
+ Wo_[2] = Wo_[3] = Wo2;
+ voicing_[2] = voicing_[3] = 1;
+ }
+
+ if (voicing1 && voicing2) {
+ float c;
+ for(i=0,c=1.0; i<M; i++,c-=1.0/M) {
+ Wo_[i] = Wo1*c + Wo2*(1.0-c);
+ voicing_[i] = 1;
+ }
+ }
+
+ for(i=0; i<M; i++) {
+ L_[i] = floorf(M_PI/Wo_[i]);
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: resample_rate_L
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Decoder side conversion of rate K vector back to rate L.
+
+\*---------------------------------------------------------------------------*/
+
+void resample_rate_L(C2CONST *c2const, MODEL *model, float rate_K_vec[], float rate_K_sample_freqs_kHz[], int K)
+{
+ float rate_K_vec_term[K+2], rate_K_sample_freqs_kHz_term[K+2];
+ float AmdB[MAX_AMP+1], rate_L_sample_freqs_kHz[MAX_AMP+1];
+ int m,k;
+
+ /* terminate either end of the rate K vecs with 0dB points */
+
+ rate_K_vec_term[0] = rate_K_vec_term[K+1] = 0.0;
+ rate_K_sample_freqs_kHz_term[0] = 0.0;
+ rate_K_sample_freqs_kHz_term[K+1] = 4.0;
+
+ for(k=0; k<K; k++) {
+ rate_K_vec_term[k+1] = rate_K_vec[k];
+ rate_K_sample_freqs_kHz_term[k+1] = rate_K_sample_freqs_kHz[k];
+ //printf("k: %d f: %f rate_K: %f\n", k, rate_K_sample_freqs_kHz[k], rate_K_vec[k]);
+ }
+
+ for(m=1; m<=model->L; m++) {
+ rate_L_sample_freqs_kHz[m] = m*model->Wo*(c2const->Fs/2000.0)/M_PI;
+ }
+
+ interp_para(&AmdB[1], rate_K_sample_freqs_kHz_term, rate_K_vec_term, K+2, &rate_L_sample_freqs_kHz[1], model->L);
+ for(m=1; m<=model->L; m++) {
+ model->A[m] = POW10F(AmdB[m]/20.0);
+ // printf("m: %d f: %f AdB: %f A: %f\n", m, rate_L_sample_freqs_kHz[m], AmdB[m], model->A[m]);
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: determine_phase
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ Given a magnitude spectrum determine a phase spectrum, used for
+ phase synthesis with newamp1.
+
+\*---------------------------------------------------------------------------*/
+
+void determine_phase(C2CONST *c2const, COMP H[], MODEL *model, int Nfft, codec2_fft_cfg fwd_cfg, codec2_fft_cfg inv_cfg)
+{
+ int i,m,b;
+ int Ns = Nfft/2+1;
+ float Gdbfk[Ns], sample_freqs_kHz[Ns], phase[Ns];
+ float AmdB[MAX_AMP+1], rate_L_sample_freqs_kHz[MAX_AMP+1];
+
+ for(m=1; m<=model->L; m++) {
+ assert(model->A[m] != 0.0);
+ AmdB[m] = 20.0*log10f(model->A[m]);
+ rate_L_sample_freqs_kHz[m] = (float)m*model->Wo*(c2const->Fs/2000.0)/M_PI;
+ }
+
+ for(i=0; i<Ns; i++) {
+ sample_freqs_kHz[i] = (c2const->Fs/1000.0)*(float)i/Nfft;
+ }
+
+ interp_para(Gdbfk, &rate_L_sample_freqs_kHz[1], &AmdB[1], model->L, sample_freqs_kHz, Ns);
+ mag_to_phase(phase, Gdbfk, Nfft, fwd_cfg, inv_cfg);
+
+ for(m=1; m<=model->L; m++) {
+ b = floorf(0.5+m*model->Wo*Nfft/(2.0*M_PI));
+ H[m].real = cosf(phase[b]); H[m].imag = sinf(phase[b]);
+ }
+}
+
+/*---------------------------------------------------------------------------* \
+
+ FUNCTION....: determine_autoc
+ AUTHOR......: David Rowe
+ DATE CREATED: April 2020
+
+ Determine autocorrelation coefficients from model params, for machine
+ learning experiments.
+
+\*---------------------------------------------------------------------------*/
+
+void determine_autoc(C2CONST *c2const, float Rk[], int order, MODEL *model, int Nfft, codec2_fft_cfg fwd_cfg, codec2_fft_cfg inv_cfg)
+{
+ int i,m;
+ int Ns = Nfft/2+1;
+ float Gdbfk[Ns], sample_freqs_kHz[Ns];
+ float AmdB[MAX_AMP+1], rate_L_sample_freqs_kHz[MAX_AMP+1];
+
+ /* interpolate in the log domain */
+ for(m=1; m<=model->L; m++) {
+ assert(model->A[m] != 0.0);
+ AmdB[m] = 20.0*log10f(model->A[m]);
+ rate_L_sample_freqs_kHz[m] = (float)m*model->Wo*(c2const->Fs/2000.0)/M_PI;
+ }
+
+ for(i=0; i<Ns; i++) {
+ sample_freqs_kHz[i] = (c2const->Fs/1000.0)*(float)i/Nfft;
+ }
+
+ interp_para(Gdbfk, &rate_L_sample_freqs_kHz[1], &AmdB[1], model->L, sample_freqs_kHz, Ns);
+
+ COMP S[Nfft], R[Nfft];
+
+ /* install negative frequency components, convert to mag squared of spectrum */
+ S[0].real = pow(10.0, Gdbfk[0]/10.0);
+ S[0].imag = 0.0;
+ for(i=1; i<Ns; i++) {
+ S[i].real = S[Nfft-i].real = pow(10.0, Gdbfk[i]/10.0);
+ S[i].imag = S[Nfft-i].imag = 0.0;
+ }
+
+ /* IDFT of mag squared is autocorrelation function */
+ codec2_fft(inv_cfg, S, R);
+ for(int k=0; k<order+1; k++) Rk[k] = R[k].real;
+}
+
+
+/* update and optionally run "front eq" equaliser on before VQ */
+void newamp1_eq(float rate_K_vec_no_mean[], float eq[], int K, int eq_en) {
+ static float ideal[] = {8,10,12,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,-20};
+ float gain = 0.02;
+ float update;
+
+ for(int k=0; k<K; k++) {
+ update = rate_K_vec_no_mean[k] - ideal[k];
+ eq[k] = (1.0-gain)*eq[k] + gain*update;
+ if (eq[k] < 0.0) eq[k] = 0.0;
+ if (eq_en)
+ rate_K_vec_no_mean[k] -= eq[k];
+ }
+}
+
+/*---------------------------------------------------------------------------* \
+
+ FUNCTION....: newamp1_model_to_indexes
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ newamp1 encoder for amplitdues {Am}. Given the rate L model
+ parameters, outputs VQ and energy quantiser indexes.
+
+\*---------------------------------------------------------------------------*/
+
+void newamp1_model_to_indexes(C2CONST *c2const,
+ int indexes[],
+ MODEL *model,
+ float rate_K_vec[],
+ float rate_K_sample_freqs_kHz[],
+ int K,
+ float *mean,
+ float rate_K_vec_no_mean[],
+ float rate_K_vec_no_mean_[],
+ float *se,
+ float *eq,
+ int eq_en
+ )
+{
+ int k;
+
+ /* convert variable rate L to fixed rate K */
+ resample_const_rate_f(c2const, model, rate_K_vec, rate_K_sample_freqs_kHz, K);
+
+ /* remove mean */
+ float sum = 0.0;
+ for(k=0; k<K; k++)
+ sum += rate_K_vec[k];
+ *mean = sum/K;
+ for(k=0; k<K; k++)
+ rate_K_vec_no_mean[k] = rate_K_vec[k] - *mean;
+
+ /* update and optionally run "front eq" equaliser on before VQ */
+ newamp1_eq(rate_K_vec_no_mean, eq, K, eq_en);
+
+ /* two stage VQ */
+ rate_K_mbest_encode(indexes, rate_K_vec_no_mean, rate_K_vec_no_mean_, K, NEWAMP1_VQ_MBEST_DEPTH);
+
+ /* running sum of squared error for variance calculation */
+ for(k=0; k<K; k++)
+ *se += (float)pow(rate_K_vec_no_mean[k]-rate_K_vec_no_mean_[k],2.0);
+
+ /* scalar quantise mean (effectively the frame energy) */
+ float w[1] = {1.0};
+ float se_mean;
+ indexes[2] = quantise(newamp1_energy_cb[0].cb,
+ mean,
+ w,
+ newamp1_energy_cb[0].k,
+ newamp1_energy_cb[0].m,
+ &se_mean);
+
+ /* scalar quantise Wo. We steal the smallest Wo index to signal
+ an unvoiced frame */
+ if (model->voiced) {
+ int index = encode_log_Wo(c2const, model->Wo, 6);
+ if (index == 0) {
+ index = 1;
+ }
+ indexes[3] = index;
+ }
+ else {
+ indexes[3] = 0;
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: newamp1_interpolate
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+\*---------------------------------------------------------------------------*/
+
+void newamp1_interpolate(float interpolated_surface_[], float left_vec[], float right_vec[], int K)
+{
+ int i, k;
+ int M = 4;
+ float c;
+
+ /* (linearly) interpolate 25Hz amplitude vectors back to 100Hz */
+
+ for(i=0,c=1.0; i<M; i++,c-=1.0/M) {
+ for(k=0; k<K; k++) {
+ interpolated_surface_[i*K+k] = left_vec[k]*c + right_vec[k]*(1.0-c);
+ }
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: newamp1_indexes_to_rate_K_vec
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ newamp1 decoder for amplitudes {Am}. Given the rate K VQ and energy
+ indexes, outputs rate K vector.
+
+\*---------------------------------------------------------------------------*/
+
+void newamp1_indexes_to_rate_K_vec(float rate_K_vec_[],
+ float rate_K_vec_no_mean_[],
+ float rate_K_sample_freqs_kHz[],
+ int K,
+ float *mean_,
+ int indexes[],
+ float user_rate_K_vec_no_mean_[],
+ int post_filter_en)
+{
+ int k;
+ const float *codebook1 = newamp1vq_cb[0].cb;
+ const float *codebook2 = newamp1vq_cb[1].cb;
+ int n1 = indexes[0];
+ int n2 = indexes[1];
+
+ if (user_rate_K_vec_no_mean_ == NULL) {
+ /* normal operation */
+ for(k=0; k<K; k++) {
+ rate_K_vec_no_mean_[k] = codebook1[K*n1+k] + codebook2[K*n2+k];
+ }
+ } else {
+ /* for development we can optionally inject the quantised rate K vector here */
+ for(k=0; k<K; k++)
+ rate_K_vec_no_mean_[k] = user_rate_K_vec_no_mean_[k];
+ }
+
+ if (post_filter_en)
+ post_filter_newamp1(rate_K_vec_no_mean_, rate_K_sample_freqs_kHz, K, 1.5);
+
+ *mean_ = newamp1_energy_cb[0].cb[indexes[2]];
+
+ for(k=0; k<K; k++) {
+ rate_K_vec_[k] = rate_K_vec_no_mean_[k] + *mean_;
+ }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: newamp1_indexes_to_model
+ AUTHOR......: David Rowe
+ DATE CREATED: Jan 2017
+
+ newamp1 decoder.
+
+\*---------------------------------------------------------------------------*/
+
+void newamp1_indexes_to_model(C2CONST *c2const,
+ MODEL model_[],
+ COMP H[],
+ float *interpolated_surface_,
+ float prev_rate_K_vec_[],
+ float *Wo_left,
+ int *voicing_left,
+ float rate_K_sample_freqs_kHz[],
+ int K,
+ codec2_fft_cfg fwd_cfg,
+ codec2_fft_cfg inv_cfg,
+ int indexes[],
+ float user_rate_K_vec_no_mean_[],
+ int post_filter_en)
+{
+ float rate_K_vec_[K], rate_K_vec_no_mean_[K], mean_, Wo_right;
+ int voicing_right, k;
+ int M = 4;
+
+ /* extract latest rate K vector */
+
+ newamp1_indexes_to_rate_K_vec(rate_K_vec_,
+ rate_K_vec_no_mean_,
+ rate_K_sample_freqs_kHz,
+ K,
+ &mean_,
+ indexes,
+ user_rate_K_vec_no_mean_,
+ post_filter_en);
+
+ /* decode latest Wo and voicing */
+
+ if (indexes[3]) {
+ Wo_right = decode_log_Wo(c2const, indexes[3], 6);
+ voicing_right = 1;
+ }
+ else {
+ Wo_right = 2.0*M_PI/100.0;
+ voicing_right = 0;
+ }
+
+ /* interpolate 25Hz rate K vec back to 100Hz */
+
+ float *left_vec = prev_rate_K_vec_;
+ float *right_vec = rate_K_vec_;
+ newamp1_interpolate(interpolated_surface_, left_vec, right_vec, K);
+
+ /* interpolate 25Hz v and Wo back to 100Hz */
+
+ float aWo_[M];
+ int avoicing_[M], aL_[M], i;
+
+ interp_Wo_v(aWo_, aL_, avoicing_, *Wo_left, Wo_right, *voicing_left, voicing_right);
+
+ /* back to rate L amplitudes, synthesise phase for each frame */
+
+ for(i=0; i<M; i++) {
+ model_[i].Wo = aWo_[i];
+ model_[i].L = aL_[i];
+ model_[i].voiced = avoicing_[i];
+
+ resample_rate_L(c2const, &model_[i], &interpolated_surface_[K*i], rate_K_sample_freqs_kHz, K);
+ determine_phase(c2const, &H[(MAX_AMP+1)*i], &model_[i], NEWAMP1_PHASE_NFFT, fwd_cfg, inv_cfg);
+ }
+
+ /* update memories for next time */
+
+ for(k=0; k<K; k++) {
+ prev_rate_K_vec_[k] = rate_K_vec_[k];
+ }
+ *Wo_left = Wo_right;
+ *voicing_left = voicing_right;
+
+}
+
generated by cgit v1.2.3 (git 2.46.0) at 2026-01-27 00:44:50 +0000