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Diffstat (limited to 'gr-vocoder/lib/codec2/phase.c')
| -rw-r--r-- | gr-vocoder/lib/codec2/phase.c | 262 |
1 files changed, 0 insertions, 262 deletions
diff --git a/gr-vocoder/lib/codec2/phase.c b/gr-vocoder/lib/codec2/phase.c deleted file mode 100644 index 69cc6697f..000000000 --- a/gr-vocoder/lib/codec2/phase.c +++ /dev/null @@ -1,262 +0,0 @@ -/*---------------------------------------------------------------------------*\ - - FILE........: phase.c - AUTHOR......: David Rowe - DATE CREATED: 1/2/09 - - Functions for modelling and synthesising phase. - -\*---------------------------------------------------------------------------*/ - -/* - Copyright (C) 2009 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/>. -*/ - -#include "defines.h" -#include "phase.h" -#include "fft.h" -#include "comp.h" -#include "glottal.c" - -#include <assert.h> -#include <math.h> -#include <string.h> -#include <stdlib.h> - -#define GLOTTAL_FFT_SIZE 512 - -/*---------------------------------------------------------------------------*\ - - aks_to_H() - - Samples the complex LPC synthesis filter spectrum at the harmonic - frequencies. - -\*---------------------------------------------------------------------------*/ - -void aks_to_H( - MODEL *model, /* model parameters */ - float aks[], /* LPC's */ - float G, /* energy term */ - COMP H[], /* complex LPC spectral samples */ - int order -) -{ - COMP Pw[FFT_DEC]; /* power spectrum */ - int i,m; /* loop variables */ - int am,bm; /* limits of current band */ - float r; /* no. rads/bin */ - float Em; /* energy in band */ - float Am; /* spectral amplitude sample */ - int b; /* centre bin of harmonic */ - float phi_; /* phase of LPC spectra */ - - r = TWO_PI/(FFT_DEC); - - /* Determine DFT of A(exp(jw)) ------------------------------------------*/ - - for(i=0; i<FFT_DEC; i++) { - Pw[i].real = 0.0; - Pw[i].imag = 0.0; - } - - for(i=0; i<=order; i++) - Pw[i].real = aks[i]; - - fft(&Pw[0].real,FFT_DEC,-1); - - /* Sample magnitude and phase at harmonics */ - - for(m=1; m<=model->L; m++) { - am = floor((m - 0.5)*model->Wo/r + 0.5); - bm = floor((m + 0.5)*model->Wo/r + 0.5); - b = floor(m*model->Wo/r + 0.5); - - Em = 0.0; - for(i=am; i<bm; i++) - Em += G/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag); - Am = sqrt(fabs(Em/(bm-am))); - - phi_ = -atan2(Pw[b].imag,Pw[b].real); - H[m].real = Am*cos(phi_); - H[m].imag = Am*sin(phi_); - } -} - - -/*---------------------------------------------------------------------------*\ - - phase_synth_zero_order() - - Synthesises phases based on SNR and a rule based approach. No phase - parameters are required apart from the SNR (which can be reduced to a - 1 bit V/UV decision per frame). - - The phase of each harmonic is modelled as the phase of a LPC - synthesis filter excited by an impulse. Unlike the first order - model the position of the impulse is not transmitted, so we create - an excitation pulse train using a rule based approach. - - Consider a pulse train with a pulse starting time n=0, with pulses - repeated at a rate of Wo, the fundamental frequency. A pulse train - in the time domain is equivalent to harmonics in the frequency - domain. We can make an excitation pulse train using a sum of - sinsusoids: - - for(m=1; m<=L; m++) - ex[n] = cos(m*Wo*n) - - Note: the Octave script ../octave/phase.m is an example of this if - you would like to try making a pulse train. - - The phase of each excitation harmonic is: - - arg(E[m]) = mWo - - where E[m] are the complex excitation (freq domain) samples, - arg(x), just returns the phase of a complex sample x. - - As we don't transmit the pulse position for this model, we need to - synthesise it. Now the excitation pulses occur at a rate of Wo. - This means the phase of the first harmonic advances by N samples - over a synthesis frame of N samples. For example if Wo is pi/20 - (200 Hz), then over a 10ms frame (N=80 samples), the phase of the - first harmonic would advance (pi/20)*80 = 4*pi or two complete - cycles. - - We generate the excitation phase of the fundamental (first - harmonic): - - arg[E[1]] = Wo*N; - - We then relate the phase of the m-th excitation harmonic to the - phase of the fundamental as: - - arg(E[m]) = m*arg(E[1]) - - This E[m] then gets passed through the LPC synthesis filter to - determine the final harmonic phase. - - Comparing to speech synthesised using original phases: - - - Through headphones speech synthesised with this model is not as - good. Through a loudspeaker it is very close to original phases. - - - If there are voicing errors, the speech can sound clicky or - staticy. If V speech is mistakenly declared UV, this model tends to - synthesise impulses or clicks, as there is usually very little shift or - dispersion through the LPC filter. - - - When combined with LPC amplitude modelling there is an additional - drop in quality. I am not sure why, theory is interformant energy - is raised making any phase errors more obvious. - - NOTES: - - 1/ This synthesis model is effectively the same as a simple LPC-10 - vocoders, and yet sounds much better. Why? Conventional wisdom - (AMBE, MELP) says mixed voicing is required for high quality - speech. - - 2/ I am pretty sure the Lincoln Lab sinusoidal coding guys (like xMBE - also from MIT) first described this zero phase model, I need to look - up the paper. - - 3/ Note that this approach could cause some discontinuities in - the phase at the edge of synthesis frames, as no attempt is made - to make sure that the phase tracks are continuous (the excitation - phases are continuous, but not the final phases after filtering - by the LPC spectra). Technically this is a bad thing. However - this may actually be a good thing, disturbing the phase tracks a - bit. More research needed, e.g. test a synthesis model that adds - a small delta-W to make phase tracks line up for voiced - harmonics. - -\*---------------------------------------------------------------------------*/ - -void phase_synth_zero_order( - MODEL *model, - float aks[], - float *ex_phase, /* excitation phase of fundamental */ - int order -) -{ - int m; - float new_phi; - COMP Ex[MAX_AMP]; /* excitation samples */ - COMP A_[MAX_AMP]; /* synthesised harmonic samples */ - COMP H[MAX_AMP]; /* LPC freq domain samples */ - float G; - float jitter = 0.0; - float r; - int b; - - G = 1.0; - aks_to_H(model, aks, G, H, order); - - /* - Update excitation fundamental phase track, this sets the position - of each pitch pulse during voiced speech. After much experiment - I found that using just this frame's Wo improved quality for UV - sounds compared to interpolating two frames Wo like this: - - ex_phase[0] += (*prev_Wo+mode->Wo)*N/2; - */ - - ex_phase[0] += (model->Wo)*N; - ex_phase[0] -= TWO_PI*floor(ex_phase[0]/TWO_PI + 0.5); - r = TWO_PI/GLOTTAL_FFT_SIZE; - - for(m=1; m<=model->L; m++) { - - /* generate excitation */ - - if (model->voiced) { - /* I think adding a little jitter helps improve low pitch - males like hts1a. This moves the onset of each harmonic - over at +/- 0.25 of a sample. - */ - jitter = 0.25*(1.0 - 2.0*rand()/RAND_MAX); - b = floor(m*model->Wo/r + 0.5); - if (b > ((GLOTTAL_FFT_SIZE/2)-1)) { - b = (GLOTTAL_FFT_SIZE/2)-1; - } - Ex[m].real = cos(ex_phase[0]*m - jitter*model->Wo*m + glottal[b]); - Ex[m].imag = sin(ex_phase[0]*m - jitter*model->Wo*m + glottal[b]); - } - else { - - /* When a few samples were tested I found that LPC filter - phase is not needed in the unvoiced case, but no harm in - keeping it. - */ - float phi = TWO_PI*(float)rand()/RAND_MAX; - Ex[m].real = cos(phi); - Ex[m].imag = sin(phi); - } - - /* filter using LPC filter */ - - A_[m].real = H[m].real*Ex[m].real - H[m].imag*Ex[m].imag; - A_[m].imag = H[m].imag*Ex[m].real + H[m].real*Ex[m].imag; - - /* modify sinusoidal phase */ - - new_phi = atan2(A_[m].imag, A_[m].real+1E-12); - model->phi[m] = new_phi; - } - -} |