Initial commitHEADmaster

* codec2 cut-down version 1.2.0
* Remove codebook and generation of sources
* remove c2dec c2enc binaries
* prepare for emscripten

1 files changed, 656 insertions, 0 deletions

diff --git a/src/newamp1.c b/src/newamp1.c
new file mode 100644
index 0000000..3ba2de0
--- /dev/null
+++ b/src/newamp1.c
@@ -0,0 +1,656 @@
+/*---------------------------------------------------------------------------*\
+
+  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 "newamp1.h"
+
+#include <assert.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "defines.h"
+#include "mbest.h"
+#include "phase.h"
+#include "quantise.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;
+}