11 files changed, 85 insertions, 246 deletions
diff --git a/codec2.c b/codec2.c
index 840fe21..2697356 100644
--- a/codec2.c
+++ b/codec2.c
@@ -29,6 +29,7 @@
 #include <assert.h>
 #include <stdio.h>
 #include <stdlib.h>
+#include <stdbool.h>
 #include <string.h>
 #include <math.h>
@@ -114,12 +115,25 @@ struct CODEC2 * codec2_create(int mode)
    struct CODEC2 *c2;
    int            i,l;
-#ifndef CORTEX_M4
+    // ALL POSSIBLE MODES MUST BE CHECKED HERE!
-    if (( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, mode)) || 
+    // we test if the desired mode is enabled at compile time
-        ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, mode)) ) {
+    // and return NULL if not
+    if (false == ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, mode) 
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_700, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_450, mode)
+                   || CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, mode)
+                ) ) 
+    {
        return NULL;
    }  
-#endif
    c2 = (struct CODEC2*)MALLOC(sizeof(struct CODEC2));
    if (c2 == NULL)
@@ -221,14 +235,15 @@ struct CODEC2 * codec2_create(int mode)
        int k;
        for(k=0; k<NEWAMP1_K; k++) {
            c2->prev_rate_K_vec_[k] = 0.0;
+            c2->eq[k] = 0.0;
        }
+        c2->eq_en = 0;
        c2->Wo_left = 0.0;
        c2->voicing_left = 0;;
        c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP1_PHASE_NFFT, 0, NULL, NULL);
        c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP1_PHASE_NFFT, 1, NULL, NULL);
    }
-#ifndef CORTEX_M4
    /* newamp2 initialisation */
    if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) {
@@ -255,7 +270,6 @@ struct CODEC2 * codec2_create(int mode)
        c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 0, NULL, NULL);
        c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 1, NULL, NULL);
    }
-#endif
    c2->fmlfeat = NULL;
@@ -302,7 +316,6 @@ struct CODEC2 * codec2_create(int mode)
        c2->decode = codec2_decode_1200;
    }
-#ifndef CORTEX_M4
    if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700, c2->mode))
    {
        c2->encode = codec2_encode_700;
@@ -314,13 +327,13 @@ struct CODEC2 * codec2_create(int mode)
        c2->encode = codec2_encode_700b;
        c2->decode = codec2_decode_700b;
    }
-#endif
    if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode))
    {
        c2->encode = codec2_encode_700c;
        c2->decode = codec2_decode_700c;
    }
-#ifndef CORTEX_M4
    if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode))
    {
        c2->encode = codec2_encode_450;
@@ -334,7 +347,6 @@ struct CODEC2 * codec2_create(int mode)
        c2->decode = codec2_decode_450pwb;
    }
-#endif
    
    return c2;
 }
@@ -1518,7 +1530,6 @@ void codec2_decode_1200(struct CODEC2 *c2, short speech[], const unsigned char *
 }
-#ifndef CORTEX_M4
 /*---------------------------------------------------------------------------*\
  FUNCTION....: codec2_encode_700
@@ -1926,7 +1937,6 @@ void codec2_decode_700b(struct CODEC2 *c2, short speech[], const unsigned char *
    for(i=0; i<LPC_ORD_LOW; i++)
        c2->prev_lsps_dec[i] = lsps[3][i];
 }
-#endif
 /*---------------------------------------------------------------------------*\
@@ -1987,7 +1997,7 @@ void codec2_encode_700c(struct CODEC2 *c2, unsigned char * bits, short speech[])
                             K,
                             &mean,
                             rate_K_vec_no_mean,
-                             rate_K_vec_no_mean_, &c2->se);
+                             rate_K_vec_no_mean_, &c2->se, c2->eq, c2->eq_en);
    c2->nse += K;
 #ifndef CORTEX_M4
@@ -2092,7 +2102,6 @@ float codec2_energy_700c(struct CODEC2 *c2, const unsigned char * bits)
    return POW10F(mean/10.0);
 }
-#ifndef CORTEX_M4
 float codec2_energy_450(struct CODEC2 *c2, const unsigned char * bits)
 {
    int     indexes[4];
@@ -2418,7 +2427,6 @@ void codec2_decode_450pwb(struct CODEC2 *c2, short speech[], const unsigned char
   }
 }
-#endif
 /*---------------------------------------------------------------------------* \
@@ -2719,3 +2727,8 @@ float *codec2_enable_user_ratek(struct CODEC2 *codec2_state, int *K) {
 void codec2_700c_post_filter(struct CODEC2 *codec2_state, int en) {
    codec2_state->post_filter_en = en;
 }
+void codec2_700c_eq(struct CODEC2 *codec2_state, int en) {
+    codec2_state->eq_en = en;
+    codec2_state->se = 0.0; codec2_state->nse = 0;
+}
diff --git a/codec2.h b/codec2.h
index 60532ca..709b462 100644
--- a/codec2.h
+++ b/codec2.h
@@ -111,13 +111,16 @@ int  codec2_rebuild_spare_bit(struct CODEC2 *codec2_state, int unpacked_bits[]);
 void codec2_set_natural_or_gray(struct CODEC2 *codec2_state, int gray);
 void codec2_set_softdec(struct CODEC2 *c2, float *softdec);
 float codec2_get_energy(struct CODEC2 *codec2_state, const unsigned char *bits);
+      
 // support for ML and VQ experiments
 void codec2_open_mlfeat(struct CODEC2 *codec2_state, char *filename);
 void codec2_load_codebook(struct CODEC2 *codec2_state, int num, char *filename);
 float codec2_get_var(struct CODEC2 *codec2_state);
 float *codec2_enable_user_ratek(struct CODEC2 *codec2_state, int *K);
+// 700C post filter and equaliser
 void codec2_700c_post_filter(struct CODEC2 *codec2_state, int en);
+void codec2_700c_eq(struct CODEC2 *codec2_state, int en);
      
 #endif
diff --git a/codec2_internal.h b/codec2_internal.h
index 498a6c4..b46e358 100644
--- a/codec2_internal.h
+++ b/codec2_internal.h
@@ -42,7 +42,7 @@ struct CODEC2 {
    codec2_fft_cfg  fft_fwd_cfg;           /* forward FFT config                        */
    codec2_fftr_cfg fftr_fwd_cfg;          /* forward real FFT config                   */
    float        *w;                       /* [m_pitch] time domain hamming window      */
-    COMP          W[FFT_ENC];              /* DFT of w[]                                */
+    float         W[FFT_ENC];              /* DFT of w[]                                */
    float        *Pn;                      /* [2*n_samp] trapezoidal synthesis window   */
    float        *bpf_buf;                 /* buffer for band pass filter               */
    float        *Sn;                      /* [m_pitch] input speech                    */
@@ -82,6 +82,8 @@ struct CODEC2 {
    unsigned int   nse;                      /* number of terms in sum       */
    float         *user_rate_K_vec_no_mean_; /* optional, user supplied vector for quantisation experiments */
    int            post_filter_en;
+    float          eq[NEWAMP1_K];            /* optional equaliser */
+    int            eq_en;
    
    /*newamp2 states (also uses newamp1 states )*/
    float          energy_prev;
diff --git a/newamp1.c b/newamp1.c
index 575620a..8980ac6 100644
--- a/newamp1.c
+++ b/newamp1.c
@@ -415,23 +415,38 @@ void newamp1_model_to_indexes(C2CONST *c2const,
                              float *mean,
                              float  rate_K_vec_no_mean[], 
                              float  rate_K_vec_no_mean_[],
-                              float *se
+                              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 and two stage VQ */
+    /* remove mean */
    float sum = 0.0;
    for(k=0; k<K; k++)
        sum += rate_K_vec[k];   
-    *mean = sum/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 */
+    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(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];
+    }
+    /* 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 */
@@ -439,7 +454,6 @@ void newamp1_model_to_indexes(C2CONST *c2const,
        *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, 
@@ -451,7 +465,6 @@ void newamp1_model_to_indexes(C2CONST *c2const,
    /* 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) {
diff --git a/newamp1.h b/newamp1.h
index 7783abc..8c18813 100644
--- a/newamp1.h
+++ b/newamp1.h
@@ -55,7 +55,9 @@ void newamp1_model_to_indexes(C2CONST *c2const,
                              float *mean,
                              float  rate_K_vec_no_mean[], 
                              float  rate_K_vec_no_mean_[],
-                              float *se);
+                              float *se,
+                              float *eq,
+                              int    eq_en);
 void newamp1_indexes_to_rate_K_vec(float  rate_K_vec_[],  
                                   float  rate_K_vec_no_mean_[],
                                   float  rate_K_sample_freqs_kHz[], 
diff --git a/nlp.c b/nlp.c
index 8c8d5f1..036f6be 100644
--- a/nlp.c
+++ b/nlp.c
@@ -53,7 +53,6 @@
 #define F0_MAX      500
 #define CNLP        0.3         /* post processor constant              */
 #define NLP_NTAP 48             /* Decimation LPF order */
-#undef  POST_PROCESS_MBE        /* choose post processor                */
 /* 8 to 16 kHz sample rate conversion */
@@ -132,10 +131,6 @@ typedef struct {
    FILE         *f;
 } NLP;
-#ifdef POST_PROCESS_MBE
-float test_candidate_mbe(COMP Sw[], COMP W[], float f0);
-float post_process_mbe(COMP Fw[], int pmin, int pmax, float gmax, COMP Sw[], COMP W[], float *prev_Wo);
-#endif
 float post_process_sub_multiples(COMP Fw[],
                                 int pmin, int pmax, float gmax, int gmax_bin,
                                 float *prev_f0);
@@ -258,7 +253,7 @@ float nlp(
  int    n,                     /* frames shift (no. new samples in Sn[])             */
  float *pitch,                 /* estimated pitch period in samples at current Fs    */
  COMP   Sw[],                  /* Freq domain version of Sn[]                        */
-  COMP   W[],                   /* Freq domain window                                 */
+  float  W[],                   /* Freq domain window                                 */
  float *prev_f0                /* previous pitch f0 in Hz, memory for pitch tracking */
 )
 {
@@ -389,11 +384,7 @@ float nlp(
    PROFILE_SAMPLE_AND_LOG(peakpick, magsq, "      peak pick");
-    #ifdef POST_PROCESS_MBE
-    best_f0 = post_process_mbe(Fw, pmin, pmax, gmax, Sw, W, prev_f0);
-    #else
    best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin, prev_f0);
-    #endif
    PROFILE_SAMPLE_AND_LOG(shiftmem, peakpick,  "      post process");
@@ -491,178 +482,6 @@ float post_process_sub_multiples(COMP Fw[],
    return best_f0;
 }
-#ifdef POST_PROCESS_MBE
-/*---------------------------------------------------------------------------*\
-  post_process_mbe()
-  Use the MBE pitch estimation algorithm to evaluate pitch candidates.  This
-  works OK but the accuracy at low F0 is affected by NW, the analysis window
-  size used for the DFT of the input speech Sw[].  Also favours high F0 in
-  the presence of background noise which causes periodic artifacts in the
-  synthesised speech.
-\*---------------------------------------------------------------------------*/
-float post_process_mbe(COMP Fw[], int pmin, int pmax, float gmax, COMP Sw[], COMP W[], float *prev_Wo)
-{
-  float candidate_f0;
-  float f0,best_f0;             /* fundamental frequency */
-  float e,e_min;                /* MBE cost function */
-  int   i;
-  #ifdef DUMP
-  float e_hz[F0_MAX];
-  #endif
-  #if !defined(NDEBUG) || defined(DUMP)
-  int   bin;
-  #endif
-  float f0_min, f0_max;
-  float f0_start, f0_end;
-  f0_min = (float)SAMPLE_RATE/pmax;
-  f0_max = (float)SAMPLE_RATE/pmin;
-  /* Now look for local maxima.  Each local maxima is a candidate
-     that we test using the MBE pitch estimation algotithm */
-  #ifdef DUMP
-  for(i=0; i<F0_MAX; i++)
-      e_hz[i] = -1;
-  #endif
-  e_min = 1E32;
-  best_f0 = 50;
-  for(i=PE_FFT_SIZE*DEC/pmax; i<=PE_FFT_SIZE*DEC/pmin; i++) {
-    if ((Fw[i].real > Fw[i-1].real) && (Fw[i].real > Fw[i+1].real)) {
-        /* local maxima found, lets test if it's big enough */
-        if (Fw[i].real > T*gmax) {
-            /* OK, sample MBE cost function over +/- 10Hz range in 2.5Hz steps */
-            candidate_f0 = (float)i*SAMPLE_RATE/(PE_FFT_SIZE*DEC);
-            f0_start = candidate_f0-20;
-            f0_end = candidate_f0+20;
-            if (f0_start < f0_min) f0_start = f0_min;
-            if (f0_end > f0_max) f0_end = f0_max;
-            for(f0=f0_start; f0<=f0_end; f0+= 2.5) {
-                e = test_candidate_mbe(Sw, W, f0);
-                #if !defined(NDEBUG) || defined(DUMP)
-                bin = floorf(f0); assert((bin > 0) && (bin < F0_MAX));
-                #endif
-                #ifdef DUMP
-                e_hz[bin] = e;
-                #endif
-                if (e < e_min) {
-                    e_min = e;
-                    best_f0 = f0;
-                }
-            }
-        }
-    }
-  }
-  /* finally sample MBE cost function around previous pitch estimate
-     (form of pitch tracking) */
-  candidate_f0 = *prev_Wo * SAMPLE_RATE/TWO_PI;
-  f0_start = candidate_f0-20;
-  f0_end = candidate_f0+20;
-  if (f0_start < f0_min) f0_start = f0_min;
-  if (f0_end > f0_max) f0_end = f0_max;
-  for(f0=f0_start; f0<=f0_end; f0+= 2.5) {
-      e = test_candidate_mbe(Sw, W, f0);
-      #if !defined(NDEBUG) || defined(DUMP)
-      bin = floorf(f0); assert((bin > 0) && (bin < F0_MAX));
-      #endif
-      #ifdef DUMP
-      e_hz[bin] = e;
-      #endif
-      if (e < e_min) {
-          e_min = e;
-          best_f0 = f0;
-      }
-  }
-  #ifdef DUMP
-  dump_e(e_hz);
-  #endif
-  return best_f0;
-}
-/*---------------------------------------------------------------------------*\
-  test_candidate_mbe()
-  Returns the error of the MBE cost function for the input f0.
-  Note: I think a lot of the operations below can be simplified as
-  W[].imag = 0 and has been normalised such that den always equals 1.
-\*---------------------------------------------------------------------------*/
-float test_candidate_mbe(
-    COMP  Sw[],
-    COMP  W[],
-    float f0
-)
-{
-    COMP  Sw_[FFT_ENC];   /* DFT of all voiced synthesised signal */
-    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 originl and synthesised */
-    float Wo;             /* current "test" fundamental freq. */
-    int   L;
-    L = floorf((SAMPLE_RATE/2.0)/f0);
-    Wo = f0*(2*PI/SAMPLE_RATE);
-    error = 0.0;
-    /* Just test across the harmonics in the first 1000 Hz (L/4) */
-    for(l=1; l<L/4; 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 */
-        for(m=al; m<bl; m++) {
-            offset = FFT_ENC/2 + m - l*Wo*FFT_ENC/TWO_PI + 0.5;
-            Am.real += Sw[m].real*W[offset].real + Sw[m].imag*W[offset].imag;
-            Am.imag += Sw[m].imag*W[offset].real - Sw[m].real*W[offset].imag;
-            den += W[offset].real*W[offset].real + W[offset].imag*W[offset].imag;
-        }
-        Am.real = Am.real/den;
-        Am.imag = Am.imag/den;
-        /* Determine error between estimated harmonic and original */
-        for(m=al; m<bl; m++) {
-            offset = FFT_ENC/2 + m - l*Wo*FFT_ENC/TWO_PI + 0.5;
-            Sw_[m].real = Am.real*W[offset].real - Am.imag*W[offset].imag;
-            Sw_[m].imag = Am.real*W[offset].imag + Am.imag*W[offset].real;
-            error += (Sw[m].real - Sw_[m].real)*(Sw[m].real - Sw_[m].real);
-            error += (Sw[m].imag - Sw_[m].imag)*(Sw[m].imag - Sw_[m].imag);
-        }
-    }
-    return error;
-}
-#endif
 /*---------------------------------------------------------------------------*\
  FUNCTION....: fdmdv_16_to_8()
diff --git a/nlp.h b/nlp.h
index 28a59c3..9e427c2 100644
--- a/nlp.h
+++ b/nlp.h
@@ -33,6 +33,6 @@
 void *nlp_create(C2CONST *c2const);
 void nlp_destroy(void *nlp_state);
 float nlp(void *nlp_state, float Sn[], int n, 
-          float *pitch_samples, COMP Sw[], COMP W[], float *prev_f0);
+          float *pitch_samples, COMP Sw[], float W[], float *prev_f0);
 #endif
diff --git a/sine.c b/sine.c
index 4e31f37..d912cd7 100644
--- a/sine.c
+++ b/sine.c
@@ -97,11 +97,10 @@ C2CONST c2const_create(int Fs, float framelength_s) {
 \*---------------------------------------------------------------------------*/
-void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[], COMP W[])
+void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[], float W[])
 {
  float m;
  COMP  wshift[FFT_ENC];
-  COMP  temp;
  int   i,j;
  int   m_pitch = c2const->m_pitch;
  int   nw      = c2const->nw;
@@ -156,6 +155,8 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
       nw/2              nw/2
  */
+  COMP temp[FFT_ENC];
  for(i=0; i<FFT_ENC; i++) {
    wshift[i].real = 0.0;
    wshift[i].imag = 0.0;
@@ -165,7 +166,7 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
  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, W);
+  codec2_fft(fft_fwd_cfg, wshift, temp);
  /*
      Re-arrange W[] to be symmetrical about FFT_ENC/2.  Makes later
@@ -192,12 +193,8 @@ void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[
  for(i=0; i<FFT_ENC/2; i++) {
-    temp.real = W[i].real;
+      W[i] = temp[i + FFT_ENC / 2].real;
-    temp.imag = W[i].imag;
+      W[i + FFT_ENC / 2] = temp[i].real;
-    W[i].real = W[i+FFT_ENC/2].real;
-    W[i].imag = W[i+FFT_ENC/2].imag;
-    W[i+FFT_ENC/2].real = temp.real;
-    W[i+FFT_ENC/2].imag = temp.imag;
  }
 }
@@ -402,39 +399,30 @@ void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float
 \*---------------------------------------------------------------------------*/
-void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[], int est_phase)
+void estimate_amplitudes(MODEL *model, COMP Sw[], float W[], int est_phase)
 {
  int   i,m;            /* loop variables */
  int   am,bm;          /* bounds of current harmonic */
-  int   b;              /* DFT bin of centre of current harmonic */
  float den;            /* denominator of amplitude expression */
-  float r, one_on_r;    /* number of rads/bin */
-  int   offset;
-  COMP  Am;
-  r = TWO_PI/FFT_ENC;
+  float r = TWO_PI/FFT_ENC;
-  one_on_r = 1.0/r;
+  float one_on_r = 1.0/r;
  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);
-    b = (int)(m*model->Wo/r + 0.5);
-    /* Estimate ampltude of harmonic */
-    den = 0.0;
-    Am.real = Am.imag = 0.0;
-    offset = FFT_ENC/2 - (int)(m*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;
-      Am.real += Sw[i].real*W[i + offset].real;
-      Am.imag += Sw[i].imag*W[i + offset].real;
    }
    model->A[m] = sqrtf(den);
    if (est_phase) {
+        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 */
@@ -459,7 +447,7 @@ float est_voicing_mbe(
                      C2CONST *c2const,
                      MODEL *model,
                      COMP   Sw[],
-                      COMP   W[]
+                      float  W[]
                      )
 {
    int   l,al,bl,m;    /* loop variables */
@@ -497,9 +485,9 @@ float est_voicing_mbe(
        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].real;
+            Am.real += Sw[m].real*W[offset+m];
-            Am.imag += Sw[m].imag*W[offset+m].real;
+            Am.imag += Sw[m].imag*W[offset+m];
-            den += W[offset+m].real*W[offset+m].real;
+            den += W[offset+m]*W[offset+m];
        }
        Am.real = Am.real/den;
@@ -507,10 +495,9 @@ float est_voicing_mbe(
        /* Determine error between estimated harmonic and original */
-// Redundant!        offset = FFT_ENC/2 - l*Wo*FFT_ENC/TWO_PI + 0.5;
        for(m=al; m<bl; m++) {
-            Ew.real = Sw[m].real - Am.real*W[offset+m].real;
+            Ew.real = Sw[m].real - Am.real*W[offset+m];
-            Ew.imag = Sw[m].imag - Am.imag*W[offset+m].real;
+            Ew.imag = Sw[m].imag - Am.imag*W[offset+m];
            error += Ew.real*Ew.real;
            error += Ew.imag*Ew.imag;
        }
diff --git a/sine.h b/sine.h
index 5c7455c..21d21fb 100644
--- a/sine.h
+++ b/sine.h
@@ -34,12 +34,12 @@
 C2CONST c2const_create(int Fs, float framelength_ms);
-void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[], COMP W[]);
+void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[], float W[]);
 float hpf(float x, float states[]);
 void dft_speech(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, COMP Sw[], float Sn[], float w[]);
 void two_stage_pitch_refinement(C2CONST *c2const, MODEL *model, COMP Sw[]);
-void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[], int est_phase);
+void estimate_amplitudes(MODEL *model, COMP Sw[], float W[], int est_phase);
-float est_voicing_mbe(C2CONST *c2const, MODEL *model, COMP Sw[], COMP W[]);
+float est_voicing_mbe(C2CONST *c2const, MODEL *model, COMP Sw[], float W[]);
 void make_synthesis_window(C2CONST *c2const, float Pn[]);
 void synthesise(int n_samp, codec2_fftr_cfg fftr_inv_cfg, float Sn_[], MODEL *model, float Pn[], int shift);
diff --git a/stripdown.sh b/stripdown.sh
index 2e4466e..c8d8332 100755
--- a/stripdown.sh
+++ b/stripdown.sh
@@ -23,7 +23,7 @@ for file in ${LIBSRC} ${LIBINC}; do
 done
 # fixup one include
-sed s:codec2/version.h:version.h: src/codec2.h > "${DESTDIR}"/codec2.h
+sed s:\<codec2/version.h\>:\"version.h\": src/codec2.h > "${DESTDIR}"/codec2.h
 cat > "${DESTDIR}"/debug_alloc.h <<'EOF'
 #define FREE free
@@ -51,7 +51,7 @@ cd ..
 rm -r "${BUILDDIR}"
 cat > "${DESTDIR}"/Makefile <<'EOF'
-CFLAGS =  -Wall -Wno-strict-overflow -std=gnu11 -fPIC -g -O2 -I.
+CFLAGS =  -Wall -Wno-strict-overflow -std=gnu11 -fPIC -g -O2 -I. -lm
 CFLAGS += -DHORUS_L2_RX -DINTERLEAVER -DRUN_TIME_TABLES -DSCRAMBLER -Dcodec2_EXPORTS
 CFLAGS += -Wno-incompatible-pointer-types-discards-qualifiers
diff --git a/version.h b/version.h
index fc9891d..bae2151 100644
--- a/version.h
+++ b/version.h
@@ -31,7 +31,7 @@
 #define CODEC2_VERSION_MAJOR 0
 #define CODEC2_VERSION_MINOR 9
-/* #undef CODEC2_VERSION_PATCH */
+#define CODEC2_VERSION_PATCH 2
-#define CODEC2_VERSION "0.9"
+#define CODEC2_VERSION "0.9.2"
 #endif //CODEC2_HAVE_VERSION