From f02dfce6e6c34b3d8a7b8a0e784b506178e331fa Mon Sep 17 00:00:00 2001
From: "erdgeist@erdgeist.org" <erdgeist@bauklotz.fritz.box>
Date: Thu, 4 Jul 2019 23:26:09 +0200
Subject: stripdown of version 0.9

---
 newamp1.c | 625 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 625 insertions(+)
 create mode 100644 newamp1.c

(limited to 'newamp1.c')

diff --git a/newamp1.c b/newamp1.c
new file mode 100644
index 0000000..575620a
--- /dev/null
+++ b/newamp1.c
@@ -0,0 +1,625 @@
+/*---------------------------------------------------------------------------*\
+
+  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"
+
+#define NEWAMP1_VQ_MBEST_DEPTH 5  /* how many candidates we keep for each stage of mbest search */
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: interp_para()
+  AUTHOR......: David Rowe
+  DATE CREATED: Jan 2017
+
+  General 2nd order parabolic interpolator.  Used splines orginally,
+  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];
+  float w[ndim];
+  int   index[MBEST_STAGES];
+  float mse, tmp;
+
+  /* codebook is compiled for a fixed K */
+
+  assert(ndim == newamp1vq_cb[0].k);
+
+  /* equal weights, could be argued mel freq axis gives freq dep weighting */
+
+  for(i=0; i<ndim; i++)
+      w[i] = 1.0;
+
+  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, w, ndim, newamp1vq_cb[0].m, mbest_stage1, index);
+  MBEST_PRINT("Stage 1:", mbest_stage1);
+
+  /* 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, w, ndim, newamp1vq_cb[1].m, mbest_stage2, index);
+  }
+  MBEST_PRINT("Stage 2:", mbest_stage2);
+
+  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 supresses
+  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 experiment here as well.
+    */
+    
+    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 channle 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....: 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
+                              )
+{
+    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 and two stage VQ */
+
+    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;
+    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 += 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, synthesis 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;
+
+}
+
-- 
cgit v1.2.3