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

---
 sine.c | 689 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 689 insertions(+)
 create mode 100644 sine.c

(limited to 'sine.c')

diff --git a/sine.c b/sine.c
new file mode 100644
index 0000000..4e31f37
--- /dev/null
+++ b/sine.c
@@ -0,0 +1,689 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: sine.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 19/8/2010
+
+  Sinusoidal analysis and synthesis functions.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 1990-2010 David Rowe
+
+  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/>.
+*/
+
+/*---------------------------------------------------------------------------*\
+
+				INCLUDES
+
+\*---------------------------------------------------------------------------*/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+
+#include "defines.h"
+#include "sine.h"
+#include "kiss_fft.h"
+
+#define HPF_BETA 0.125
+
+/*---------------------------------------------------------------------------*\
+
+				HEADERS
+
+\*---------------------------------------------------------------------------*/
+
+void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
+			 float pstep);
+
+/*---------------------------------------------------------------------------*\
+
+				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.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);
+    */
+
+    return c2const;
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: make_analysis_window
+  AUTHOR......: David Rowe
+  DATE CREATED: 11/5/94
+
+  Init function that generates the time domain analysis window and it's DFT.
+
+\*---------------------------------------------------------------------------*/
+
+void make_analysis_window(C2CONST *c2const, codec2_fft_cfg fft_fwd_cfg, float w[], COMP W[])
+{
+  float m;
+  COMP  wshift[FFT_ENC];
+  COMP  temp;
+  int   i,j;
+  int   m_pitch = c2const->m_pitch;
+  int   nw      = c2const->nw;
+
+  /*
+     Generate Hamming window centered on M-sample pitch analysis window
+
+  0            M/2           M-1
+  |-------------|-------------|
+        |-------|-------|
+            nw samples
+
+     All our analysis/synthsis is centred on the M/2 sample.
+  */
+
+  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=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++) {
+    w[i] *= m;
+  }
+
+  /*
+     Generate DFT of analysis window, used for later processing.  Note
+     we modulo FFT_ENC shift the time domain window w[], this makes the
+     imaginary part of the DFT W[] equal to zero as the shifted w[] is
+     even about the n=0 time axis if nw is odd.  Having the imag part
+     of the DFT W[] makes computation easier.
+
+     0                      FFT_ENC-1
+     |-------------------------|
+
+      ----\               /----
+           \             /
+            \           /          <- shifted version of window w[n]
+             \         /
+              \       /
+               -------
+
+     |---------|     |---------|
+       nw/2              nw/2
+  */
+
+  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];
+
+  codec2_fft(fft_fwd_cfg, wshift, W);
+
+  /*
+      Re-arrange W[] to be symmetrical about FFT_ENC/2.  Makes later
+      analysis convenient.
+
+   Before:
+
+
+     0                 FFT_ENC-1
+     |----------|---------|
+     __                   _
+       \                 /
+        \_______________/
+
+   After:
+
+     0                 FFT_ENC-1
+     |----------|---------|
+               ___
+              /   \
+     ________/     \_______
+
+  */
+
+
+  for(i=0; i<FFT_ENC/2; i++) {
+    temp.real = W[i].real;
+    temp.imag = W[i].imag;
+    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;
+  }
+
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: hpf
+  AUTHOR......: David Rowe
+  DATE CREATED: 16 Nov 2010
+
+  High pass filter with a -3dB point of about 160Hz.
+
+    y(n) = -HPF_BETA*y(n-1) + x(n) - x(n-1)
+
+\*---------------------------------------------------------------------------*/
+
+float hpf(float x, float states[])
+{
+    states[0] = -HPF_BETA*states[0] + x - states[1];
+    states[1] = x;
+
+    return states[0];
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: dft_speech
+  AUTHOR......: David Rowe
+  DATE CREATED: 27/5/94
+
+  Finds the DFT of the current speech input speech frame.
+
+\*---------------------------------------------------------------------------*/
+
+// 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
+#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;
+    }
+
+    /* 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 */
+
+    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 */
+
+    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);
+}
+#else
+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++) {
+    sw[i] = 0.0;
+  }
+
+  /* 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 */
+
+  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];
+
+  codec2_fftr(fftr_fwd_cfg, sw, Sw);
+}
+#endif
+
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: two_stage_pitch_refinement
+  AUTHOR......: David Rowe
+  DATE CREATED: 27/5/94
+
+  Refines the current pitch estimate using the harmonic sum pitch
+  estimation technique.
+
+\*---------------------------------------------------------------------------*/
+
+void two_stage_pitch_refinement(C2CONST *c2const, MODEL *model, COMP Sw[])
+{
+  float pmin,pmax,pstep;	/* pitch refinment minimum, maximum and step */
+
+  /* Coarse refinement */
+
+  pmax = TWO_PI/model->Wo + 5;
+  pmin = TWO_PI/model->Wo - 5;
+  pstep = 1.0;
+  hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
+
+  /* Fine refinement */
+
+  pmax = TWO_PI/model->Wo + 1;
+  pmin = TWO_PI/model->Wo - 1;
+  pstep = 0.25;
+  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;
+
+  model->L = floorf(PI/model->Wo);
+
+  /* trap occasional round off issues with floorf() */
+  if (model->Wo*model->L >= 0.95*PI) {
+      model->L--;
+  }
+  assert(model->Wo*model->L < PI);
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: hs_pitch_refinement
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Harmonic sum pitch refinement function.
+
+ pmin   pitch search range minimum
+ pmax	pitch search range maximum
+ step   pitch search step size
+ model	current pitch estimate in model.Wo
+
+ model 	refined pitch estimate in model.Wo
+
+\*---------------------------------------------------------------------------*/
+
+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 */
+  Wom = model->Wo;
+  Em = 0.0;
+  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) {
+    E = 0.0;
+    Wo = TWO_PI/p;
+
+    /* 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;
+    }
+    /* Compare to see if this is a maximum */
+
+    if (E > Em) {
+      Em = E;
+      Wom = Wo;
+    }
+  }
+
+  model->Wo = Wom;
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: estimate_amplitudes
+  AUTHOR......: David Rowe
+  DATE CREATED: 27/5/94
+
+  Estimates the complex amplitudes of the harmonics.
+
+\*---------------------------------------------------------------------------*/
+
+void estimate_amplitudes(MODEL *model, COMP Sw[], COMP 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;
+  one_on_r = 1.0/r;
+
+  for(m=1; m<=model->L; m++) {
+    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) {
+
+        /* 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);
+    }
+  }
+}
+
+/*---------------------------------------------------------------------------*\
+
+  est_voicing_mbe()
+
+  Returns the error of the MBE cost function for a fiven 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 est_voicing_mbe(
+                      C2CONST *c2const,
+                      MODEL *model,
+                      COMP   Sw[],
+                      COMP   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;
+
+    /* 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);
+
+	/* 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].real;
+	    Am.imag += Sw[m].imag*W[offset+m].real;
+	    den += W[offset+m].real*W[offset+m].real;
+        }
+
+        Am.real = Am.real/den;
+        Am.imag = Am.imag/den;
+
+        /* 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.imag = Sw[m].imag - Am.imag*W[offset+m].real;
+	    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);
+
+    /* Look for Type 1 errors, strongly V speech that has been
+       accidentally declared UV */
+
+    if (model->voiced == 0)
+	if (eratio > 10.0)
+	    model->voiced = 1;
+
+    /* Look for Type 2 errors, strongly UV speech that has been
+       accidentally declared V */
+
+    if (model->voiced == 1) {
+	if (eratio < -10.0)
+	    model->voiced = 0;
+
+	/* 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. */
+
+	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;
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: make_synthesis_window
+  AUTHOR......: David Rowe
+  DATE CREATED: 11/5/94
+
+  Init function that generates the trapezoidal (Parzen) sythesis window.
+
+\*---------------------------------------------------------------------------*/
+
+void make_synthesis_window(C2CONST *c2const, float Pn[])
+{
+  int   i;
+  float win;
+  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;
+  win = 0.0;
+  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;
+  win = 1.0;
+  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;
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: synthesise
+  AUTHOR......: David Rowe
+  DATE CREATED: 20/2/95
+
+  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.
+
+\*---------------------------------------------------------------------------*/
+
+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;
+    }
+
+    /* 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]);
+    }
+
+    /* Perform inverse DFT */
+
+    codec2_fftri(fftr_inv_cfg, Sw_,sw_);
+
+    /* Overlap add to previous samples */
+
+    #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;
+    }
+
+    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);
+}
+
-- 
cgit v1.2.3