i've tested, now you can too

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

View File

@@ -4,8 +4,8 @@
AUTHOR......: David Rowe
DATE CREATED: 23/3/93
Non Linear Pitch (NLP) estimation functions.
Non Linear Pitch (NLP) estimation functions.
\*---------------------------------------------------------------------------*/
/*
@@ -22,14 +22,13 @@
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/>.
*/
#include "defines.h"
#include "nlp.h"
#include "dump.h"
#include "four1.h"
#include "kiss_fft.h"
#include <assert.h>
#include <math.h>
@@ -60,7 +59,7 @@
/* 48 tap 600Hz low pass FIR filter coefficients */
float nlp_fir[] = {
const float nlp_fir[] = {
-1.0818124e-03,
-1.1008344e-03,
-9.2768838e-04,
@@ -112,12 +111,14 @@ float nlp_fir[] = {
};
typedef struct {
float sq[PMAX_M]; /* squared speech samples */
float mem_x,mem_y; /* memory for notch filter */
float mem_fir[NLP_NTAP]; /* decimation FIR filter memory */
float sq[PMAX_M]; /* squared speech samples */
float mem_x,mem_y; /* memory for notch filter */
float mem_fir[NLP_NTAP]; /* decimation FIR filter memory */
kiss_fft_cfg fft_cfg; /* kiss FFT config */
} NLP;
float post_process_mbe(COMP Fw[], int pmin, int pmax, float gmax);
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);
float post_process_sub_multiples(COMP Fw[],
int pmin, int pmax, float gmax, int gmax_bin,
float *prev_Wo);
@@ -146,20 +147,27 @@ void *nlp_create()
for(i=0; i<NLP_NTAP; i++)
nlp->mem_fir[i] = 0.0;
nlp->fft_cfg = kiss_fft_alloc (PE_FFT_SIZE, 0, NULL, NULL);
assert(nlp->fft_cfg != NULL);
return (void*)nlp;
}
/*---------------------------------------------------------------------------*\
nlp_destory()
nlp_destroy()
Initialisation function for NLP pitch estimator.
Shut down function for NLP pitch estimator.
\*---------------------------------------------------------------------------*/
void nlp_destroy(void *nlp_state)
{
NLP *nlp;
assert(nlp_state != NULL);
nlp = (NLP*)nlp_state;
KISS_FFT_FREE(nlp->fft_cfg);
free(nlp_state);
}
@@ -198,27 +206,30 @@ float nlp(
float Sn[], /* input speech vector */
int n, /* frames shift (no. new samples in Sn[]) */
int m, /* analysis window size */
int pmin, /* minimum pitch value */
int pmin, /* minimum pitch value */
int pmax, /* maximum pitch value */
float *pitch, /* estimated pitch period in samples */
COMP Sw[], /* Freq domain version of Sn[] */
COMP W[], /* Freq domain window */
float *prev_Wo
)
{
NLP *nlp;
float notch; /* current notch filter output */
COMP Fw[PE_FFT_SIZE]; /* DFT of squared signal */
float notch; /* current notch filter output */
COMP fw[PE_FFT_SIZE]; /* DFT of squared signal (input) */
COMP Fw[PE_FFT_SIZE]; /* DFT of squared signal (output) */
float gmax;
int gmax_bin;
int i,j;
float best_f0;
assert(nlp_state != NULL);
assert(m <= PMAX_M);
nlp = (NLP*)nlp_state;
/* Square, notch filter at DC, and LP filter vector */
for(i=m-n; i<M; i++) /* square latest speech samples */
for(i=m-n; i<m; i++) /* square latest speech samples */
nlp->sq[i] = Sn[i]*Sn[i];
for(i=m-n; i<m; i++) { /* notch filter at DC */
@@ -226,7 +237,14 @@ float nlp(
notch += COEFF*nlp->mem_y;
nlp->mem_x = nlp->sq[i];
nlp->mem_y = notch;
nlp->sq[i] = notch;
nlp->sq[i] = notch + 1.0; /* With 0 input vectors to codec,
kiss_fft() would take a long
time to execute when running in
real time. Problem was traced
to kiss_fft function call in
this function. Adding this small
constant fixed problem. Not
exactly sure why. */
}
for(i=m-n; i<m; i++) { /* FIR filter vector */
@@ -243,19 +261,24 @@ float nlp(
/* Decimate and DFT */
for(i=0; i<PE_FFT_SIZE; i++) {
Fw[i].real = 0.0;
Fw[i].imag = 0.0;
fw[i].real = 0.0;
fw[i].imag = 0.0;
}
for(i=0; i<m/DEC; i++) {
Fw[i].real = nlp->sq[i*DEC]*(0.5 - 0.5*cos(2*PI*i/(m/DEC-1)));
fw[i].real = nlp->sq[i*DEC]*(0.5 - 0.5*cos(2*PI*i/(m/DEC-1)));
}
#ifdef DUMP
dump_dec(Fw);
four1(&Fw[-1].imag,PE_FFT_SIZE,1);
#endif
kiss_fft(nlp->fft_cfg, (kiss_fft_cpx *)fw, (kiss_fft_cpx *)Fw);
for(i=0; i<PE_FFT_SIZE; i++)
Fw[i].real = Fw[i].real*Fw[i].real + Fw[i].imag*Fw[i].imag;
#ifdef DUMP
dump_sq(nlp->sq);
dump_Fw(Fw);
#endif
/* find global peak */
@@ -267,9 +290,13 @@ float nlp(
gmax_bin = i;
}
}
best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin,
prev_Wo);
//#define POST_PROCESS_MBE
#ifdef POST_PROCESS_MBE
best_f0 = post_process_mbe(Fw, pmin, pmax, gmax, Sw, W, prev_Wo);
#else
best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin, prev_Wo);
#endif
/* Shift samples in buffer to make room for new samples */
@@ -286,7 +313,7 @@ float nlp(
post_process_sub_multiples()
Given the global maximma of Fw[] we search interger submultiples for
Given the global maximma of Fw[] we search integer submultiples for
local maxima. If local maxima exist and they are above an
experimentally derived threshold (OK a magic number I pulled out of
the air) we choose the submultiple as the F0 estimate.
@@ -317,10 +344,10 @@ float post_process_sub_multiples(COMP Fw[],
/* post process estimate by searching submultiples */
mult = 2;
min_bin = PE_FFT_SIZE*DEC/pmax;
min_bin = PE_FFT_SIZE*DEC/pmax;
cmax_bin = gmax_bin;
prev_f0_bin = *prev_Wo*(4000.0/PI)*(PE_FFT_SIZE*DEC)/SAMPLE_RATE;
while(gmax_bin/mult >= min_bin) {
b = gmax_bin/mult; /* determine search interval */
@@ -339,7 +366,7 @@ float post_process_sub_multiples(COMP Fw[],
lmax = 0;
lmax_bin = bmin;
for (b=bmin; b<=bmax; b++) /* look for maximum in interval */
for (b=bmin; b<=bmax; b++) /* look for maximum in interval */
if (Fw[b].real > lmax) {
lmax = Fw[b].real;
lmax_bin = b;
@@ -359,3 +386,158 @@ float post_process_sub_multiples(COMP Fw[],
return best_f0;
}
/*---------------------------------------------------------------------------*\
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;
float e_hz[F0_MAX];
int bin;
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 */
for(i=0; i<F0_MAX; i++)
e_hz[i] = -1;
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);
bin = floor(f0); assert((bin > 0) && (bin < F0_MAX));
e_hz[bin] = e;
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);
bin = floor(f0); assert((bin > 0) && (bin < F0_MAX));
e_hz[bin] = e;
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 = floor((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 = ceil((l - 0.5)*Wo*FFT_ENC/TWO_PI);
bl = ceil((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;
}