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i've tested, now you can too

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This commit is contained in:
Brian West
2012-12-20 20:08:42 -06:00
parent d3fcfa8245
commit d67b96af8a
94 changed files with 25305 additions and 2005 deletions
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libs/libcodec2/src/sine.c
+163 -45
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@@ -22,8 +22,7 @@
License for more details.
You should have received a copy of the GNU Lesser General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
along with this program; if not, see <http://www.gnu.org/licenses/>.
*/
/*---------------------------------------------------------------------------*\
@@ -38,7 +37,9 @@
#include "defines.h"
#include "sine.h"
#include "four1.h"
#include "kiss_fft.h"
#define HPF_BETA 0.125
/*---------------------------------------------------------------------------*\
@@ -65,9 +66,10 @@ void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
\*---------------------------------------------------------------------------*/
void make_analysis_window(float w[],COMP W[])
void make_analysis_window(kiss_fft_cfg fft_fwd_cfg, float w[], COMP W[])
{
float m;
COMP wshift[FFT_ENC];
COMP temp;
int i,j;
@@ -122,15 +124,15 @@ void make_analysis_window(float w[],COMP W[])
*/
for(i=0; i<FFT_ENC; i++) {
W[i].real = 0.0;
W[i].imag = 0.0;
wshift[i].real = 0.0;
wshift[i].imag = 0.0;
}
for(i=0; i<NW/2; i++)
W[i].real = w[i+M/2];
wshift[i].real = w[i+M/2];
for(i=FFT_ENC-NW/2,j=M/2-NW/2; i<FFT_ENC; i++,j++)
W[i].real = w[j];
wshift[i].real = w[j];
four1(&W[-1].imag,FFT_ENC,-1); /* "Numerical Recipes in C" FFT */
kiss_fft(fft_fwd_cfg, (kiss_fft_cpx *)wshift, (kiss_fft_cpx *)W);
/*
Re-arrange W[] to be symmetrical about FFT_ENC/2. Makes later
@@ -167,6 +169,26 @@ void make_analysis_window(float w[],COMP W[])
}
/*---------------------------------------------------------------------------*\
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
@@ -177,13 +199,14 @@ void make_analysis_window(float w[],COMP W[])
\*---------------------------------------------------------------------------*/
void dft_speech(COMP Sw[], float Sn[], float w[])
void dft_speech(kiss_fft_cfg fft_fwd_cfg, COMP Sw[], float Sn[], float w[])
{
int i;
int i;
COMP sw[FFT_ENC];
for(i=0; i<FFT_ENC; i++) {
Sw[i].real = 0.0;
Sw[i].imag = 0.0;
sw[i].real = 0.0;
sw[i].imag = 0.0;
}
/* Centre analysis window on time axis, we need to arrange input
@@ -192,14 +215,14 @@ void dft_speech(COMP Sw[], float Sn[], float w[])
/* move 2nd half to start of FFT input vector */
for(i=0; i<NW/2; i++)
Sw[i].real = Sn[i+M/2]*w[i+M/2];
sw[i].real = Sn[i+M/2]*w[i+M/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/2-NW/2]*w[i+M/2-NW/2];
sw[FFT_ENC-NW/2+i].real = Sn[i+M/2-NW/2]*w[i+M/2-NW/2];
four1(&Sw[-1].imag,FFT_ENC,-1);
kiss_fft(fft_fwd_cfg, (kiss_fft_cpx *)sw, (kiss_fft_cpx *)Sw);
}
/*---------------------------------------------------------------------------*\
@@ -362,38 +385,38 @@ float est_voicing_mbe(
MODEL *model,
COMP Sw[],
COMP W[],
float f0,
COMP Sw_[] /* DFT of all voiced synthesised signal for f0 */
/* useful for debugging/dump file */
)
COMP Sw_[], /* DFT of all voiced synthesised signal */
/* useful for debugging/dump file */
COMP Ew[], /* DFT of error */
float prev_Wo)
{
int i,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;
float error; /* accumulated error between original and synthesised */
float Wo;
float sig, snr;
float elow, ehigh, eratio;
float dF0, sixty;
sig = 0.0;
sig = 1E-4;
for(l=1; l<=model->L/4; l++) {
sig += model->A[l]*model->A[l];
}
for(i=0; i<FFT_ENC; i++) {
Sw_[i].real = 0.0;
Sw_[i].imag = 0.0;
Ew[i].real = 0.0;
Ew[i].imag = 0.0;
}
L = floor((FS/2.0)/f0);
Wo = f0*(TWO_PI/FS);
error = 0.0;
Wo = model->Wo;
error = 1E-4;
/* Just test across the harmonics in the first 1000 Hz (L/4) */
for(l=1; l<=L/4; l++) {
for(l=1; l<=model->L/4; l++) {
Am.real = 0.0;
Am.imag = 0.0;
den = 0.0;
@@ -418,16 +441,73 @@ float est_voicing_mbe(
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);
Ew[m].real = Sw[m].real - Sw_[m].real;
Ew[m].imag = Sw[m].imag - Sw_[m].imag;
error += Ew[m].real*Ew[m].real;
error += Ew[m].imag*Ew[m].imag;
}
}
snr = 10.0*log10(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.
*/
elow = ehigh = 1E-4;
for(l=1; l<=model->L/2; l++) {
elow += model->A[l]*model->A[l];
}
for(l=model->L/2; l<=model->L; l++) {
ehigh += model->A[l]*model->A[l];
}
eratio = 10.0*log10(elow/ehigh);
dF0 = 0.0;
/* 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;
/* If pitch is jumping about it's likely this is UV */
/* 13 Feb 2012 - this seems to add some V errors so comment out for now. Maybe
double check on bg noise files
dF0 = (model->Wo - prev_Wo)*FS/TWO_PI;
if (fabs(dF0) > 15.0)
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/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;
}
@@ -471,20 +551,22 @@ void make_synthesis_window(float Pn[])
DATE CREATED: 20/2/95
Synthesise a speech signal in the frequency domain from the
sinusodal model parameters. Uses overlap-add a triangular window to
smoothly interpolate betwen frames.
sinusodal model parameters. Uses overlap-add with a trapezoidal
window to smoothly interpolate betwen frames.
\*---------------------------------------------------------------------------*/
void synthesise(
float Sn_[], /* time domain synthesised signal */
MODEL *model, /* ptr to model parameters for this frame */
float Pn[], /* time domain Parzen window */
int shift /* used to handle transition frames */
kiss_fft_cfg fft_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]; /* DFT of synthesised signal */
COMP sw_[FFT_DEC]; /* synthesised signal */
if (shift) {
/* Update memories */
@@ -500,10 +582,30 @@ void synthesise(
Sw_[i].imag = 0.0;
}
/* Now set up frequency domain synthesised speech */
/*
Nov 2010 - found that synthesis using time domain cos() functions
gives better results for synthesis frames greater than 10ms. Inverse
FFT synthesis using a 512 pt FFT works well for 10ms window. I think
(but am not sure) that the problem is related to the quantisation of
the harmonic frequencies to the FFT bin size, e.g. there is a
8000/512 Hz step between FFT bins. For some reason this makes
the speech from longer frame > 10ms sound poor. The effect can also
be seen when synthesising test signals like single sine waves, some
sort of amplitude modulation at the frame rate.
Another possibility is using a larger FFT size (1024 or 2048).
*/
#define FFT_SYNTHESIS
#ifdef FFT_SYNTHESIS
/* Now set up frequency domain synthesised speech */
for(l=1; l<=model->L; l++) {
//for(l=model->L/2; l<=model->L; l++) {
//for(l=1; l<=model->L/4; l++) {
b = floor(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]*cos(model->phi[l]);
Sw_[b].imag = model->A[l]*sin(model->phi[l]);
Sw_[FFT_DEC-b].real = Sw_[b].real;
@@ -512,19 +614,35 @@ void synthesise(
/* Perform inverse DFT */
four1(&Sw_[-1].imag,FFT_DEC,1);
kiss_fft(fft_inv_cfg, (kiss_fft_cpx *)Sw_, (kiss_fft_cpx *)sw_);
#else
/*
Direct time domain synthesis using the cos() function. Works
well at 10ms and 20ms frames rates. Note synthesis window is
still used to handle overlap-add between adjacent frames. This
could be simplified as we don't need to synthesise where Pn[]
is zero.
*/
for(l=1; l<=model->L; l++) {
for(i=0,j=-N+1; i<N-1; i++,j++) {
Sw_[FFT_DEC-N+1+i].real += 2.0*model->A[l]*cos(j*model->Wo*l + model->phi[l]);
}
for(i=N-1,j=0; i<2*N; i++,j++)
Sw_[j].real += 2.0*model->A[l]*cos(j*model->Wo*l + model->phi[l]);
}
#endif
/* Overlap add to previous samples */
for(i=0; i<N-1; i++) {
Sn_[i] += Sw_[FFT_DEC-N+1+i].real*Pn[i];
Sn_[i] += sw_[FFT_DEC-N+1+i].real*Pn[i];
}
if (shift)
for(i=N-1,j=0; i<2*N; i++,j++)
Sn_[i] = Sw_[j].real*Pn[i];
Sn_[i] = sw_[j].real*Pn[i];
else
for(i=N-1,j=0; i<2*N; i++,j++)
Sn_[i] += Sw_[j].real*Pn[i];
Sn_[i] += sw_[j].real*Pn[i];
}