Bump to codec2 version 1.2.0erdgeist-bump-to-1.2.0

1 files changed, 286 insertions, 305 deletions

diff --git a/sine.c b/sine.c
index d912cd7..814c269 100644
--- a/sine.c
+++ b/sine.c
@@ -27,64 +27,66 @@
 
 /*---------------------------------------------------------------------------*\
 
-                                INCLUDES
+                                INCLUDES
 
 \*---------------------------------------------------------------------------*/
 
-#include <stdlib.h>
-#include <stdio.h>
+#include "sine.h"
+
 #include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
 
 #include "defines.h"
-#include "sine.h"
 #include "kiss_fft.h"
 
 #define HPF_BETA 0.125
 
 /*---------------------------------------------------------------------------*\
 
-                                HEADERS
+                                HEADERS
 
 \*---------------------------------------------------------------------------*/
 
 void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
-                         float pstep);
+                         float pstep);
 
 /*---------------------------------------------------------------------------*\
 
-                                FUNCTIONS
+                                FUNCTIONS
 
 \*---------------------------------------------------------------------------*/
 
 C2CONST c2const_create(int Fs, float framelength_s) {
-    C2CONST c2const;
-
-    assert((Fs == 8000) || (Fs = 16000));
-    c2const.Fs = Fs;
-    c2const.n_samp = round(Fs*framelength_s);
-    c2const.max_amp = floor(Fs*P_MIN_S/2);
-    c2const.p_min = floor(Fs*P_MIN_S);
-    c2const.p_max = floor(Fs*P_MAX_S);
-    c2const.m_pitch = floor(Fs*M_PITCH_S);
-    c2const.Wo_min = TWO_PI/c2const.p_max;
-    c2const.Wo_max = TWO_PI/c2const.p_min;
-
-    if (Fs == 8000) {
-        c2const.nw = 279;
-    } else {
-        c2const.nw = 511;  /* actually a bit shorter in time but lets us maintain constant FFT size */
-    }
+  C2CONST c2const;
+
+  assert((Fs == 8000) || (Fs == 16000));
+  c2const.Fs = Fs;
+  c2const.n_samp = round(Fs * framelength_s);
+  c2const.max_amp = floor(Fs * P_MAX_S / 2);
+  c2const.p_min = floor(Fs * P_MIN_S);
+  c2const.p_max = floor(Fs * P_MAX_S);
+  c2const.m_pitch = floor(Fs * M_PITCH_S);
+  c2const.Wo_min = TWO_PI / c2const.p_max;
+  c2const.Wo_max = TWO_PI / c2const.p_min;
+
+  if (Fs == 8000) {
+    c2const.nw = 279;
+  } else {
+    c2const.nw = 511; /* actually a bit shorter in time but lets us maintain
+                         constant FFT size */
+  }
 
-    c2const.tw = Fs*TW_S;
+  c2const.tw = Fs * TW_S;
 
-    /*
-    fprintf(stderr, "max_amp: %d m_pitch: %d\n", c2const.n_samp, c2const.m_pitch);
-    fprintf(stderr, "p_min: %d p_max: %d\n", c2const.p_min, c2const.p_max);
-    fprintf(stderr, "Wo_min: %f Wo_max: %f\n", c2const.Wo_min, c2const.Wo_max);
-    fprintf(stderr, "nw: %d tw: %d\n", c2const.nw, c2const.tw);
-    */
+  /*
+  fprintf(stderr, "max_amp: %d m_pitch: %d\n", c2const.n_samp, c2const.m_pitch);
+  fprintf(stderr, "p_min: %d p_max: %d\n", c2const.p_min, c2const.p_max);
+  fprintf(stderr, "Wo_min: %f Wo_max: %f\n", c2const.Wo_min, c2const.Wo_max);
+  fprintf(stderr, "nw: %d tw: %d\n", c2const.nw, c2const.tw);
+  */
 
-    return c2const;
+  return c2const;
 }
 
 /*---------------------------------------------------------------------------*\
@@ -97,13 +99,13 @@ C2CONST c2const_create(int Fs, float framelength_s) {
 
 \*---------------------------------------------------------------------------*/
 
-void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[], float W[])
-{
+void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg,
+                          float w[], float W[]) {
   float m;
-  COMP  wshift[FFT_ENC];
-  int   i,j;
-  int   m_pitch = c2const->m_pitch;
-  int   nw      = c2const->nw;
+  COMP wshift[FFT_ENC];
+  int i, j;
+  int m_pitch = c2const->m_pitch;
+  int nw = c2const->nw;
 
   /*
      Generate Hamming window centered on M-sample pitch analysis window
@@ -117,20 +119,18 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
   */
 
   m = 0.0;
-  for(i=0; i<m_pitch/2-nw/2; i++)
-    w[i] = 0.0;
-  for(i=m_pitch/2-nw/2,j=0; i<m_pitch/2+nw/2; i++,j++) {
-    w[i] = 0.5 - 0.5*cosf(TWO_PI*j/(nw-1));
-    m += w[i]*w[i];
+  for (i = 0; i < m_pitch / 2 - nw / 2; i++) w[i] = 0.0;
+  for (i = m_pitch / 2 - nw / 2, j = 0; i < m_pitch / 2 + nw / 2; i++, j++) {
+    w[i] = 0.5 - 0.5 * cosf(TWO_PI * j / (nw - 1));
+    m += w[i] * w[i];
   }
-  for(i=m_pitch/2+nw/2; i<m_pitch; i++)
-    w[i] = 0.0;
+  for (i = m_pitch / 2 + nw / 2; i < m_pitch; i++) w[i] = 0.0;
 
   /* Normalise - makes freq domain amplitude estimation straight
      forward */
 
-  m = 1.0/sqrtf(m*FFT_ENC);
-  for(i=0; i<m_pitch; i++) {
+  m = 1.0 / sqrtf(m * FFT_ENC);
+  for (i = 0; i < m_pitch; i++) {
     w[i] *= m;
   }
 
@@ -157,14 +157,13 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
 
   COMP temp[FFT_ENC];
 
-  for(i=0; i<FFT_ENC; i++) {
+  for (i = 0; i < FFT_ENC; i++) {
     wshift[i].real = 0.0;
     wshift[i].imag = 0.0;
   }
-  for(i=0; i<nw/2; i++)
-    wshift[i].real = w[i+m_pitch/2];
-  for(i=FFT_ENC-nw/2,j=m_pitch/2-nw/2; i<FFT_ENC; i++,j++)
-   wshift[i].real = w[j];
+  for (i = 0; i < nw / 2; i++) wshift[i].real = w[i + m_pitch / 2];
+  for (i = FFT_ENC - nw / 2, j = m_pitch / 2 - nw / 2; i < FFT_ENC; i++, j++)
+    wshift[i].real = w[j];
 
   codec2_fft(fft_fwd_cfg, wshift, temp);
 
@@ -191,12 +190,10 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
 
   */
 
-
-  for(i=0; i<FFT_ENC/2; i++) {
-      W[i] = temp[i + FFT_ENC / 2].real;
-      W[i + FFT_ENC / 2] = temp[i].real;
+  for (i = 0; i < FFT_ENC / 2; i++) {
+    W[i] = temp[i + FFT_ENC / 2].real;
+    W[i + FFT_ENC / 2] = temp[i].real;
   }
-
 }
 
 /*---------------------------------------------------------------------------*\
@@ -211,12 +208,11 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
 
 \*---------------------------------------------------------------------------*/
 
-float hpf(float x, float states[])
-{
-    states[0] = -HPF_BETA*states[0] + x - states[1];
-    states[1] = x;
+float hpf(float x, float states[]) {
+  states[0] = -HPF_BETA * states[0] + x - states[1];
+  states[1] = x;
 
-    return states[0];
+  return states[0];
 }
 
 /*---------------------------------------------------------------------------*\
@@ -230,41 +226,43 @@ float hpf(float x, float states[])
 \*---------------------------------------------------------------------------*/
 
 // TODO: we can either go for a faster FFT using fftr and some stack usage
-// or we can reduce stack usage to almost zero on STM32 by switching to fft_inplace
+// or we can reduce stack usage to almost zero on STM32 by switching to
+// fft_inplace
 #if 1
-void dft_speech(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, COMP Sw[], float Sn[], float w[])
-{
-    int  i;
-    int  m_pitch = c2const->m_pitch;
-    int   nw      = c2const->nw;
-
-    for(i=0; i<FFT_ENC; i++) {
-        Sw[i].real = 0.0;
-        Sw[i].imag = 0.0;
-    }
+void dft_speech(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, COMP Sw[],
+                float Sn[], float w[]) {
+  int i;
+  int m_pitch = c2const->m_pitch;
+  int nw = c2const->nw;
+
+  for (i = 0; i < FFT_ENC; i++) {
+    Sw[i].real = 0.0;
+    Sw[i].imag = 0.0;
+  }
 
-    /* Centre analysis window on time axis, we need to arrange input
-       to FFT this way to make FFT phases correct */
+  /* Centre analysis window on time axis, we need to arrange input
+     to FFT this way to make FFT phases correct */
 
-    /* move 2nd half to start of FFT input vector */
+  /* move 2nd half to start of FFT input vector */
 
-    for(i=0; i<nw/2; i++)
-        Sw[i].real = Sn[i+m_pitch/2]*w[i+m_pitch/2];
+  for (i = 0; i < nw / 2; i++)
+    Sw[i].real = Sn[i + m_pitch / 2] * w[i + m_pitch / 2];
 
-    /* move 1st half to end of FFT input vector */
+  /* move 1st half to end of FFT input vector */
 
-    for(i=0; i<nw/2; i++)
-        Sw[FFT_ENC-nw/2+i].real = Sn[i+m_pitch/2-nw/2]*w[i+m_pitch/2-nw/2];
+  for (i = 0; i < nw / 2; i++)
+    Sw[FFT_ENC - nw / 2 + i].real =
+        Sn[i + m_pitch / 2 - nw / 2] * w[i + m_pitch / 2 - nw / 2];
 
-    codec2_fft_inplace(fft_fwd_cfg, Sw);
+  codec2_fft_inplace(fft_fwd_cfg, Sw);
 }
 #else
-void dft_speech(codec2_fftr_cfg fftr_fwd_cfg, COMP Sw[], float Sn[], float w[])
-{
-    int  i;
+void dft_speech(codec2_fftr_cfg fftr_fwd_cfg, COMP Sw[], float Sn[],
+                float w[]) {
+  int i;
   float sw[FFT_ENC];
 
-  for(i=0; i<FFT_ENC; i++) {
+  for (i = 0; i < FFT_ENC; i++) {
     sw[i] = 0.0;
   }
 
@@ -273,19 +271,18 @@ void dft_speech(codec2_fftr_cfg fftr_fwd_cfg, COMP Sw[], float Sn[], float w[])
 
   /* move 2nd half to start of FFT input vector */
 
-  for(i=0; i<nw/2; i++)
-    sw[i] = Sn[i+m_pitch/2]*w[i+m_pitch/2];
+  for (i = 0; i < nw / 2; i++) sw[i] = Sn[i + m_pitch / 2] * w[i + m_pitch / 2];
 
   /* move 1st half to end of FFT input vector */
 
-  for(i=0; i<nw/2; i++)
-    sw[FFT_ENC-nw/2+i] = Sn[i+m_pitch/2-nw/2]*w[i+m_pitch/2-nw/2];
+  for (i = 0; i < nw / 2; i++)
+    sw[FFT_ENC - nw / 2 + i] =
+        Sn[i + m_pitch / 2 - nw / 2] * w[i + m_pitch / 2 - nw / 2];
 
   codec2_fftr(fftr_fwd_cfg, sw, Sw);
 }
 #endif
 
-
 /*---------------------------------------------------------------------------*\
 
   FUNCTION....: two_stage_pitch_refinement
@@ -297,38 +294,35 @@ void dft_speech(codec2_fftr_cfg fftr_fwd_cfg, COMP Sw[], float Sn[], float w[])
 
 \*---------------------------------------------------------------------------*/
 
-void two_stage_pitch_refinement(C2CONST *c2const, MODEL *model, COMP Sw[])
-{
-  float pmin,pmax,pstep;        /* pitch refinment minimum, maximum and step */
+void two_stage_pitch_refinement(C2CONST *c2const, MODEL *model, COMP Sw[]) {
+  float pmin, pmax, pstep; /* pitch refinement minimum, maximum and step */
 
   /* Coarse refinement */
 
-  pmax = TWO_PI/model->Wo + 5;
-  pmin = TWO_PI/model->Wo - 5;
+  pmax = TWO_PI / model->Wo + 5;
+  pmin = TWO_PI / model->Wo - 5;
   pstep = 1.0;
-  hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
+  hs_pitch_refinement(model, Sw, pmin, pmax, pstep);
 
   /* Fine refinement */
 
-  pmax = TWO_PI/model->Wo + 1;
-  pmin = TWO_PI/model->Wo - 1;
+  pmax = TWO_PI / model->Wo + 1;
+  pmin = TWO_PI / model->Wo - 1;
   pstep = 0.25;
-  hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
+  hs_pitch_refinement(model, Sw, pmin, pmax, pstep);
 
   /* Limit range */
 
-  if (model->Wo < TWO_PI/c2const->p_max)
-    model->Wo = TWO_PI/c2const->p_max;
-  if (model->Wo > TWO_PI/c2const->p_min)
-    model->Wo = TWO_PI/c2const->p_min;
+  if (model->Wo < TWO_PI / c2const->p_max) model->Wo = TWO_PI / c2const->p_max;
+  if (model->Wo > TWO_PI / c2const->p_min) model->Wo = TWO_PI / c2const->p_min;
 
-  model->L = floorf(PI/model->Wo);
+  model->L = floorf(PI / model->Wo);
 
   /* trap occasional round off issues with floorf() */
-  if (model->Wo*model->L >= 0.95*PI) {
-      model->L--;
+  if (model->Wo * model->L >= 0.95 * PI) {
+    model->L--;
   }
-  assert(model->Wo*model->L < PI);
+  assert(model->Wo * model->L < PI);
 }
 
 /*---------------------------------------------------------------------------*\
@@ -348,35 +342,39 @@ void two_stage_pitch_refinement(C2CONST *c2const, MODEL *model, COMP Sw[])
 
 \*---------------------------------------------------------------------------*/
 
-void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float pstep)
-{
-  int m;                /* loop variable */
-  int b;                /* bin for current harmonic centre */
-  float E;              /* energy for current pitch*/
-  float Wo;             /* current "test" fundamental freq. */
-  float Wom;            /* Wo that maximises E */
-  float Em;             /* mamimum energy */
-  float r, one_on_r;    /* number of rads/bin */
-  float p;              /* current pitch */
+void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
+                         float pstep) {
+  int m;             /* loop variable */
+  int b;             /* bin for current harmonic centre */
+  float E;           /* energy for current pitch*/
+  float Wo;          /* current "test" fundamental freq. */
+  float Wom;         /* Wo that maximises E */
+  float Em;          /* mamimum energy */
+  float r, one_on_r; /* number of rads/bin */
+  float p;           /* current pitch */
 
   /* Initialisation */
 
-  model->L = PI/model->Wo;      /* use initial pitch est. for L */
+  model->L = PI / model->Wo; /* use initial pitch est. for L */
   Wom = model->Wo;
   Em = 0.0;
-  r = TWO_PI/FFT_ENC;
-  one_on_r = 1.0/r;
+  r = TWO_PI / FFT_ENC;
+  one_on_r = 1.0 / r;
 
   /* Determine harmonic sum for a range of Wo values */
 
-  for(p=pmin; p<=pmax; p+=pstep) {
+  for (p = pmin; p <= pmax; p += pstep) {
     E = 0.0;
-    Wo = TWO_PI/p;
+    Wo = TWO_PI / p;
+
+    float bFloat = Wo * one_on_r;
+    float currentBFloat = bFloat;
 
     /* Sum harmonic magnitudes */
-    for(m=1; m<=model->L; m++) {
-        b = (int)(m*Wo*one_on_r + 0.5);
-        E += Sw[b].real*Sw[b].real + Sw[b].imag*Sw[b].imag;
+    for (m = 1; m <= model->L; m++) {
+      b = (int)(currentBFloat + 0.5);
+      E += Sw[b].real * Sw[b].real + Sw[b].imag * Sw[b].imag;
+      currentBFloat += bFloat;
     }
     /* Compare to see if this is a maximum */
 
@@ -399,35 +397,35 @@ void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float
 
 \*---------------------------------------------------------------------------*/
 
-void estimate_amplitudes(MODEL *model, COMP Sw[], float W[], int est_phase)
-{
-  int   i,m;            /* loop variables */
-  int   am,bm;          /* bounds of current harmonic */
-  float den;            /* denominator of amplitude expression */
+void estimate_amplitudes(MODEL *model, COMP Sw[], float W[], int est_phase) {
+  int i, m;   /* loop variables */
+  int am, bm; /* bounds of current harmonic */
+  float den;  /* denominator of amplitude expression */
 
-  float r = TWO_PI/FFT_ENC;
-  float one_on_r = 1.0/r;
+  float r = TWO_PI / FFT_ENC;
+  float one_on_r = 1.0 / r;
 
-  for(m=1; m<=model->L; m++) {
+  for (m = 1; m <= model->L; m++) {
     /* Estimate ampltude of harmonic */
 
     den = 0.0;
-    am = (int)((m - 0.5)*model->Wo*one_on_r + 0.5);
-    bm = (int)((m + 0.5)*model->Wo*one_on_r + 0.5);
+    am = (int)((m - 0.5) * model->Wo * one_on_r + 0.5);
+    bm = (int)((m + 0.5) * model->Wo * one_on_r + 0.5);
 
-    for(i=am; i<bm; i++) {
-      den += Sw[i].real*Sw[i].real + Sw[i].imag*Sw[i].imag;
+    for (i = am; i < bm; i++) {
+      den += Sw[i].real * Sw[i].real + Sw[i].imag * Sw[i].imag;
     }
 
     model->A[m] = sqrtf(den);
 
     if (est_phase) {
-        int b = (int)(m*model->Wo/r + 0.5); /* DFT bin of centre of current harmonic */
+      int b = (int)(m * model->Wo / r +
+                    0.5); /* DFT bin of centre of current harmonic */
 
-        /* Estimate phase of harmonic, this is expensive in CPU for
-           embedded devicesso we make it an option */
+      /* Estimate phase of harmonic, this is expensive in CPU for
+         embedded devicesso we make it an option */
 
-        model->phi[m] = atan2f(Sw[b].imag,Sw[b].real);
+      model->phi[m] = atan2f(Sw[b].imag, Sw[b].real);
     }
   }
 }
@@ -443,119 +441,110 @@ void estimate_amplitudes(MODEL *model, COMP Sw[], float W[], int est_phase)
 
 \*---------------------------------------------------------------------------*/
 
-float est_voicing_mbe(
-                      C2CONST *c2const,
-                      MODEL *model,
-                      COMP   Sw[],
-                      float  W[]
-                      )
-{
-    int   l,al,bl,m;    /* loop variables */
-    COMP  Am;             /* amplitude sample for this band */
-    int   offset;         /* centers Hw[] about current harmonic */
-    float den;            /* denominator of Am expression */
-    float error;          /* accumulated error between original and synthesised */
-    float Wo;
-    float sig, snr;
-    float elow, ehigh, eratio;
-    float sixty;
-    COMP   Ew;
-    Ew.real = 0;
-    Ew.imag = 0;
-
-    int l_1000hz = model->L*1000.0/(c2const->Fs/2);
-    sig = 1E-4;
-    for(l=1; l<=l_1000hz; l++) {
-        sig += model->A[l]*model->A[l];
-    }
+float est_voicing_mbe(C2CONST *c2const, MODEL *model, COMP Sw[], float W[]) {
+  int l, al, bl, m; /* loop variables */
+  COMP Am;          /* amplitude sample for this band */
+  int offset;       /* centers Hw[] about current harmonic */
+  float den;        /* denominator of Am expression */
+  float error;      /* accumulated error between original and synthesised */
+  float Wo;
+  float sig, snr;
+  float elow, ehigh, eratio;
+  float sixty;
+  COMP Ew;
+  Ew.real = 0;
+  Ew.imag = 0;
+
+  int l_1000hz = model->L * 1000.0 / (c2const->Fs / 2);
+  sig = 1E-4;
+  for (l = 1; l <= l_1000hz; l++) {
+    sig += model->A[l] * model->A[l];
+  }
 
-    Wo = model->Wo;
-    error = 1E-4;
+  Wo = model->Wo;
+  error = 1E-4;
 
-    /* Just test across the harmonics in the first 1000 Hz */
+  /* Just test across the harmonics in the first 1000 Hz */
 
-    for(l=1; l<=l_1000hz; l++) {
-        Am.real = 0.0;
-        Am.imag = 0.0;
-        den = 0.0;
-        al = ceilf((l - 0.5)*Wo*FFT_ENC/TWO_PI);
-        bl = ceilf((l + 0.5)*Wo*FFT_ENC/TWO_PI);
+  for (l = 1; l <= l_1000hz; l++) {
+    Am.real = 0.0;
+    Am.imag = 0.0;
+    den = 0.0;
+    al = ceilf((l - 0.5) * Wo * FFT_ENC / TWO_PI);
+    bl = ceilf((l + 0.5) * Wo * FFT_ENC / TWO_PI);
 
-        /* Estimate amplitude of harmonic assuming harmonic is totally voiced */
+    /* Estimate amplitude of harmonic assuming harmonic is totally voiced */
 
-        offset = FFT_ENC/2 - l*Wo*FFT_ENC/TWO_PI + 0.5;
-        for(m=al; m<bl; m++) {
-            Am.real += Sw[m].real*W[offset+m];
-            Am.imag += Sw[m].imag*W[offset+m];
-            den += W[offset+m]*W[offset+m];
-        }
+    offset = FFT_ENC / 2 - l * Wo * FFT_ENC / TWO_PI + 0.5;
+    for (m = al; m < bl; m++) {
+      Am.real += Sw[m].real * W[offset + m];
+      Am.imag += Sw[m].imag * W[offset + m];
+      den += W[offset + m] * W[offset + m];
+    }
 
-        Am.real = Am.real/den;
-        Am.imag = Am.imag/den;
+    Am.real = Am.real / den;
+    Am.imag = Am.imag / den;
 
-        /* Determine error between estimated harmonic and original */
+    /* Determine error between estimated harmonic and original */
 
-        for(m=al; m<bl; m++) {
-            Ew.real = Sw[m].real - Am.real*W[offset+m];
-            Ew.imag = Sw[m].imag - Am.imag*W[offset+m];
-            error += Ew.real*Ew.real;
-            error += Ew.imag*Ew.imag;
-        }
+    for (m = al; m < bl; m++) {
+      Ew.real = Sw[m].real - Am.real * W[offset + m];
+      Ew.imag = Sw[m].imag - Am.imag * W[offset + m];
+      error += Ew.real * Ew.real;
+      error += Ew.imag * Ew.imag;
     }
+  }
 
-    snr = 10.0*log10f(sig/error);
-    if (snr > V_THRESH)
-        model->voiced = 1;
-    else
-        model->voiced = 0;
-
-    /* post processing, helps clean up some voicing errors ------------------*/
-
-    /*
-       Determine the ratio of low freqency to high frequency energy,
-       voiced speech tends to be dominated by low frequency energy,
-       unvoiced by high frequency. This measure can be used to
-       determine if we have made any gross errors.
-    */
-
-    int l_2000hz = model->L*2000.0/(c2const->Fs/2);
-    int l_4000hz = model->L*4000.0/(c2const->Fs/2);
-    elow = ehigh = 1E-4;
-    for(l=1; l<=l_2000hz; l++) {
-        elow += model->A[l]*model->A[l];
-    }
-    for(l=l_2000hz; l<=l_4000hz; l++) {
-        ehigh += model->A[l]*model->A[l];
-    }
-    eratio = 10.0*log10f(elow/ehigh);
+  snr = 10.0 * log10f(sig / error);
+  if (snr > V_THRESH)
+    model->voiced = 1;
+  else
+    model->voiced = 0;
+
+  /* post processing, helps clean up some voicing errors ------------------*/
 
-    /* Look for Type 1 errors, strongly V speech that has been
-       accidentally declared UV */
+  /*
+     Determine the ratio of low frequency to high frequency energy,
+     voiced speech tends to be dominated by low frequency energy,
+     unvoiced by high frequency. This measure can be used to
+     determine if we have made any gross errors.
+  */
+
+  int l_2000hz = model->L * 2000.0 / (c2const->Fs / 2);
+  int l_4000hz = model->L * 4000.0 / (c2const->Fs / 2);
+  elow = ehigh = 1E-4;
+  for (l = 1; l <= l_2000hz; l++) {
+    elow += model->A[l] * model->A[l];
+  }
+  for (l = l_2000hz; l <= l_4000hz; l++) {
+    ehigh += model->A[l] * model->A[l];
+  }
+  eratio = 10.0 * log10f(elow / ehigh);
 
-    if (model->voiced == 0)
-        if (eratio > 10.0)
-            model->voiced = 1;
+  /* Look for Type 1 errors, strongly V speech that has been
+     accidentally declared UV */
 
-    /* Look for Type 2 errors, strongly UV speech that has been
-       accidentally declared V */
+  if (model->voiced == 0)
+    if (eratio > 10.0) model->voiced = 1;
 
-    if (model->voiced == 1) {
-        if (eratio < -10.0)
-            model->voiced = 0;
+  /* Look for Type 2 errors, strongly UV speech that has been
+     accidentally declared V */
 
-        /* A common source of Type 2 errors is the pitch estimator
-           gives a low (50Hz) estimate for UV speech, which gives a
-           good match with noise due to the close harmoonic spacing.
-           These errors are much more common than people with 50Hz3
-           pitch, so we have just a small eratio threshold. */
+  if (model->voiced == 1) {
+    if (eratio < -10.0) model->voiced = 0;
 
-        sixty = 60.0*TWO_PI/c2const->Fs;
-        if ((eratio < -4.0) && (model->Wo <= sixty))
-            model->voiced = 0;
-    }
-    //printf(" v: %d snr: %f eratio: %3.2f %f\n",model->voiced,snr,eratio,dF0);
+    /* A common source of Type 2 errors is the pitch estimator
+       gives a low (50Hz) estimate for UV speech, which gives a
+       good match with noise due to the close harmoonic spacing.
+       These errors are much more common than people with 50Hz3
+       pitch, so we have just a small eratio threshold. */
 
-    return snr;
+    sixty = 60.0 * TWO_PI / c2const->Fs;
+    if ((eratio < -4.0) && (model->Wo <= sixty)) model->voiced = 0;
+  }
+  // printf(" v: %d snr: %f eratio: %3.2f %f\n",model->voiced,snr,eratio,dF0);
+
+  return snr;
 }
 
 /*---------------------------------------------------------------------------*\
@@ -564,32 +553,29 @@ float est_voicing_mbe(
   AUTHOR......: David Rowe
   DATE CREATED: 11/5/94
 
-  Init function that generates the trapezoidal (Parzen) sythesis window.
+  Init function that generates the trapezoidal (Parzen) synthesis window.
 
 \*---------------------------------------------------------------------------*/
 
-void make_synthesis_window(C2CONST *c2const, float Pn[])
-{
-  int   i;
+void make_synthesis_window(C2CONST *c2const, float Pn[]) {
+  int i;
   float win;
-  int   n_samp = c2const->n_samp;
-  int   tw     = c2const->tw;
+  int n_samp = c2const->n_samp;
+  int tw = c2const->tw;
 
   /* Generate Parzen window in time domain */
 
   win = 0.0;
-  for(i=0; i<n_samp/2-tw; i++)
-    Pn[i] = 0.0;
+  for (i = 0; i < n_samp / 2 - tw; i++) Pn[i] = 0.0;
   win = 0.0;
-  for(i=n_samp/2-tw; i<n_samp/2+tw; win+=1.0/(2*tw), i++ )
+  for (i = n_samp / 2 - tw; i < n_samp / 2 + tw; win += 1.0 / (2 * tw), i++)
     Pn[i] = win;
-  for(i=n_samp/2+tw; i<3*n_samp/2-tw; i++)
-    Pn[i] = 1.0;
+  for (i = n_samp / 2 + tw; i < 3 * n_samp / 2 - tw; i++) Pn[i] = 1.0;
   win = 1.0;
-  for(i=3*n_samp/2-tw; i<3*n_samp/2+tw; win-=1.0/(2*tw), i++)
+  for (i = 3 * n_samp / 2 - tw; i < 3 * n_samp / 2 + tw;
+       win -= 1.0 / (2 * tw), i++)
     Pn[i] = win;
-  for(i=3*n_samp/2+tw; i<2*n_samp; i++)
-    Pn[i] = 0.0;
+  for (i = 3 * n_samp / 2 + tw; i < 2 * n_samp; i++) Pn[i] = 0.0;
 }
 
 /*---------------------------------------------------------------------------*\
@@ -600,77 +586,72 @@ void make_synthesis_window(C2CONST *c2const, float Pn[])
 
   Synthesise a speech signal in the frequency domain from the
   sinusodal model parameters.  Uses overlap-add with a trapezoidal
-  window to smoothly interpolate betwen frames.
+  window to smoothly interpolate between frames.
 
 \*---------------------------------------------------------------------------*/
 
-void synthesise(
-  int    n_samp,
-  codec2_fftr_cfg fftr_inv_cfg,
-  float  Sn_[],         /* time domain synthesised signal              */
-  MODEL *model,         /* ptr to model parameters for this frame      */
-  float  Pn[],          /* time domain Parzen window                   */
-  int    shift          /* flag used to handle transition frames       */
-)
-{
-    int   i,l,j,b;              /* loop variables */
-    COMP  Sw_[FFT_DEC/2+1];     /* DFT of synthesised signal */
-    float sw_[FFT_DEC];         /* synthesised signal */
-
-    if (shift) {
-        /* Update memories */
-        for(i=0; i<n_samp-1; i++) {
-            Sn_[i] = Sn_[i+n_samp];
-        }
-        Sn_[n_samp-1] = 0.0;
+void synthesise(int n_samp, codec2_fftr_cfg fftr_inv_cfg,
+                float Sn_[],  /* time domain synthesised signal              */
+                MODEL *model, /* ptr to model parameters for this frame      */
+                float Pn[],   /* time domain Parzen window                   */
+                int shift     /* flag used to handle transition frames       */
+) {
+  int i, l, j, b;            /* loop variables */
+  COMP Sw_[FFT_DEC / 2 + 1]; /* DFT of synthesised signal */
+  float sw_[FFT_DEC];        /* synthesised signal */
+
+  if (shift) {
+    /* Update memories */
+    for (i = 0; i < n_samp - 1; i++) {
+      Sn_[i] = Sn_[i + n_samp];
     }
+    Sn_[n_samp - 1] = 0.0;
+  }
 
-    for(i=0; i<FFT_DEC/2+1; i++) {
-        Sw_[i].real = 0.0;
-        Sw_[i].imag = 0.0;
-    }
+  for (i = 0; i < FFT_DEC / 2 + 1; i++) {
+    Sw_[i].real = 0.0;
+    Sw_[i].imag = 0.0;
+  }
 
-    /* Now set up frequency domain synthesised speech */
+  /* Now set up frequency domain synthesised speech */
 
-    for(l=1; l<=model->L; l++) {
-        b = (int)(l*model->Wo*FFT_DEC/TWO_PI + 0.5);
-        if (b > ((FFT_DEC/2)-1)) {
-            b = (FFT_DEC/2)-1;
-        }
-        Sw_[b].real = model->A[l]*cosf(model->phi[l]);
-        Sw_[b].imag = model->A[l]*sinf(model->phi[l]);
+  for (l = 1; l <= model->L; l++) {
+    b = (int)(l * model->Wo * FFT_DEC / TWO_PI + 0.5);
+    if (b > ((FFT_DEC / 2) - 1)) {
+      b = (FFT_DEC / 2) - 1;
     }
+    Sw_[b].real = model->A[l] * cosf(model->phi[l]);
+    Sw_[b].imag = model->A[l] * sinf(model->phi[l]);
+  }
 
-    /* Perform inverse DFT */
+  /* Perform inverse DFT */
 
-    codec2_fftri(fftr_inv_cfg, Sw_,sw_);
+  codec2_fftri(fftr_inv_cfg, Sw_, sw_);
 
-    /* Overlap add to previous samples */
+  /* Overlap add to previous samples */
 
-    #ifdef USE_KISS_FFT
-    #define    FFTI_FACTOR ((float)1.0)
-    #else
-    #define    FFTI_FACTOR ((float32_t)FFT_DEC)
-    #endif
+#ifdef USE_KISS_FFT
+#define FFTI_FACTOR ((float)1.0)
+#else
+#define FFTI_FACTOR ((float32_t)FFT_DEC)
+#endif
 
-    for(i=0; i<n_samp-1; i++) {
-        Sn_[i] += sw_[FFT_DEC-n_samp+1+i]*Pn[i] * FFTI_FACTOR;
-    }
+  for (i = 0; i < n_samp - 1; i++) {
+    Sn_[i] += sw_[FFT_DEC - n_samp + 1 + i] * Pn[i] * FFTI_FACTOR;
+  }
 
-    if (shift)
-        for(i=n_samp-1,j=0; i<2*n_samp; i++,j++)
-            Sn_[i] = sw_[j]*Pn[i] * FFTI_FACTOR;
-    else
-        for(i=n_samp-1,j=0; i<2*n_samp; i++,j++)
-            Sn_[i] += sw_[j]*Pn[i] * FFTI_FACTOR;
+  if (shift)
+    for (i = n_samp - 1, j = 0; i < 2 * n_samp; i++, j++)
+      Sn_[i] = sw_[j] * Pn[i] * FFTI_FACTOR;
+  else
+    for (i = n_samp - 1, j = 0; i < 2 * n_samp; i++, j++)
+      Sn_[i] += sw_[j] * Pn[i] * FFTI_FACTOR;
 }
 
-
 /* todo: this should probably be in some states rather than a static */
 static unsigned long next = 1;
 
 int codec2_rand(void) {
-    next = next * 1103515245 + 12345;
-    return((unsigned)(next/65536) % 32768);
+  next = next * 1103515245 + 12345;
+  return ((unsigned)(next / 65536) % 32768);
 }
-