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UHSDR/UHSDR-active-devel/mchf-eclipse/drivers/freedv/sine.c
Xavier f838fa66c4 first commit
2022-11-08 16:13:55 +01:00

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/*---------------------------------------------------------------------------*\
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;
sig = 1E-4;
for(l=1; l<=model->L/4; l++) {
sig += model->A[l]*model->A[l];
}
Wo = model->Wo;
error = 1E-4;
/* Just test across the harmonics in the first 1000 Hz */
int l_1000hz = model->L*1000.0/(c2const->Fs/2);
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);
}
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