290 lines
9.1 KiB
C
Executable File
290 lines
9.1 KiB
C
Executable File
/*---------------------------------------------------------------------------*\
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FILE........: phase.c
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AUTHOR......: David Rowe
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DATE CREATED: 1/2/09
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Functions for modelling and synthesising phase.
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\*---------------------------------------------------------------------------*/
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/*
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Copyright (C) 2009 David Rowe
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All rights reserved.
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License version 2.1, as
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published by the Free Software Foundation. This program is
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distributed in the hope that it will be useful, but WITHOUT ANY
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WARRANTY; without even the implied warranty of MERCHANTABILITY or
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FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public
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License for more details.
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You should have received a copy of the GNU Lesser General Public License
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along with this program; if not,see <http://www.gnu.org/licenses/>.
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*/
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#include "defines.h"
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#include "phase.h"
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#include "kiss_fft.h"
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#include "comp.h"
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#include "comp_prim.h"
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#include "sine.h"
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#include <assert.h>
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#include <ctype.h>
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#include <math.h>
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#include <string.h>
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#include <stdlib.h>
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/*---------------------------------------------------------------------------*\
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sample_phase()
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Samples phase at centre of each harmonic from and array of FFT_ENC
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DFT samples.
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\*---------------------------------------------------------------------------*/
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void sample_phase(MODEL *model,
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COMP H[],
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COMP A[] /* LPC analysis filter in freq domain */
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)
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{
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int m, b;
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float r;
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r = TWO_PI/(FFT_ENC);
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/* Sample phase at harmonics */
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for(m=1; m<=model->L; m++) {
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b = (int)(m*model->Wo/r + 0.5);
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H[m] = cconj(A[b]); /* synth filter 1/A is opposite phase to analysis filter */
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}
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}
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/*---------------------------------------------------------------------------*\
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phase_synth_zero_order()
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Synthesises phases based on SNR and a rule based approach. No phase
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parameters are required apart from the SNR (which can be reduced to a
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1 bit V/UV decision per frame).
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The phase of each harmonic is modelled as the phase of a synthesis
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filter excited by an impulse. In many Codec 2 modes the synthesis
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filter is a LPC filter. Unlike the first order model the position
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of the impulse is not transmitted, so we create an excitation pulse
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train using a rule based approach.
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Consider a pulse train with a pulse starting time n=0, with pulses
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repeated at a rate of Wo, the fundamental frequency. A pulse train
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in the time domain is equivalent to harmonics in the frequency
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domain. We can make an excitation pulse train using a sum of
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sinsusoids:
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for(m=1; m<=L; m++)
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ex[n] = cos(m*Wo*n)
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Note: the Octave script ../octave/phase.m is an example of this if
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you would like to try making a pulse train.
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The phase of each excitation harmonic is:
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arg(E[m]) = mWo
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where E[m] are the complex excitation (freq domain) samples,
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arg(x), just returns the phase of a complex sample x.
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As we don't transmit the pulse position for this model, we need to
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synthesise it. Now the excitation pulses occur at a rate of Wo.
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This means the phase of the first harmonic advances by N_SAMP samples
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over a synthesis frame of N_SAMP samples. For example if Wo is pi/20
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(200 Hz), then over a 10ms frame (N_SAMP=80 samples), the phase of the
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first harmonic would advance (pi/20)*80 = 4*pi or two complete
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cycles.
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We generate the excitation phase of the fundamental (first
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harmonic):
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arg[E[1]] = Wo*N_SAMP;
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We then relate the phase of the m-th excitation harmonic to the
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phase of the fundamental as:
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arg(E[m]) = m*arg(E[1])
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This E[m] then gets passed through the LPC synthesis filter to
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determine the final harmonic phase.
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Comparing to speech synthesised using original phases:
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- Through headphones speech synthesised with this model is not as
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good. Through a loudspeaker it is very close to original phases.
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- If there are voicing errors, the speech can sound clicky or
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staticy. If V speech is mistakenly declared UV, this model tends to
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synthesise impulses or clicks, as there is usually very little shift or
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dispersion through the LPC synthesis filter.
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- When combined with LPC amplitude modelling there is an additional
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drop in quality. I am not sure why, theory is interformant energy
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is raised making any phase errors more obvious.
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NOTES:
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1/ This synthesis model is effectively the same as a simple LPC-10
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vocoders, and yet sounds much better. Why? Conventional wisdom
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(AMBE, MELP) says mixed voicing is required for high quality
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speech.
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2/ I am pretty sure the Lincoln Lab sinusoidal coding guys (like xMBE
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also from MIT) first described this zero phase model, I need to look
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up the paper.
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3/ Note that this approach could cause some discontinuities in
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the phase at the edge of synthesis frames, as no attempt is made
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to make sure that the phase tracks are continuous (the excitation
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phases are continuous, but not the final phases after filtering
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by the LPC spectra). Technically this is a bad thing. However
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this may actually be a good thing, disturbing the phase tracks a
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bit. More research needed, e.g. test a synthesis model that adds
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a small delta-W to make phase tracks line up for voiced
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harmonics.
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\*---------------------------------------------------------------------------*/
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void phase_synth_zero_order(
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int n_samp,
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MODEL *model,
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float *ex_phase, /* excitation phase of fundamental */
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COMP H[] /* L synthesis filter freq domain samples */
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)
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{
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int m;
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float new_phi;
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COMP Ex[MAX_AMP+1]; /* excitation samples */
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COMP A_[MAX_AMP+1]; /* synthesised harmonic samples */
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/*
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Update excitation fundamental phase track, this sets the position
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of each pitch pulse during voiced speech. After much experiment
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I found that using just this frame's Wo improved quality for UV
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sounds compared to interpolating two frames Wo like this:
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ex_phase[0] += (*prev_Wo+model->Wo)*N_SAMP/2;
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*/
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ex_phase[0] += (model->Wo)*n_samp;
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ex_phase[0] -= TWO_PI*floorf(ex_phase[0]/TWO_PI + 0.5);
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for(m=1; m<=model->L; m++) {
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/* generate excitation */
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if (model->voiced) {
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Ex[m].real = cosf(ex_phase[0]*m);
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Ex[m].imag = sinf(ex_phase[0]*m);
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}
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else {
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/* When a few samples were tested I found that LPC filter
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phase is not needed in the unvoiced case, but no harm in
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keeping it.
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*/
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float phi = TWO_PI*(float)codec2_rand()/CODEC2_RAND_MAX;
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Ex[m].real = cosf(phi);
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Ex[m].imag = sinf(phi);
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}
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/* filter using LPC filter */
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A_[m].real = H[m].real*Ex[m].real - H[m].imag*Ex[m].imag;
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A_[m].imag = H[m].imag*Ex[m].real + H[m].real*Ex[m].imag;
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/* modify sinusoidal phase */
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new_phi = atan2f(A_[m].imag, A_[m].real+1E-12);
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model->phi[m] = new_phi;
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}
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}
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/*---------------------------------------------------------------------------*\
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FUNCTION....: mag_to_phase
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AUTHOR......: David Rowe
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DATE CREATED: Jan 2017
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Algorithm for http://www.dsprelated.com/showcode/20.php ported to C. See
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also Octave function mag_to_phase.m
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Given a magnitude spectrum in dB, returns a minimum-phase phase
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spectra.
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\*---------------------------------------------------------------------------*/
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void mag_to_phase(float phase[], /* Nfft/2+1 output phase samples in radians */
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float Gdbfk[], /* Nfft/2+1 postive freq amplitudes samples in dB */
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int Nfft,
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codec2_fft_cfg fft_fwd_cfg,
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codec2_fft_cfg fft_inv_cfg
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)
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{
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COMP Sdb[Nfft], c[Nfft], cf[Nfft], Cf[Nfft];
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int Ns = Nfft/2+1;
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int i;
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/* install negative frequency components, 1/Nfft takes into
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account kiss fft lack of scaling on ifft */
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Sdb[0].real = Gdbfk[0];
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Sdb[0].imag = 0.0;
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for(i=1; i<Ns; i++) {
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Sdb[i].real = Sdb[Nfft-i].real = Gdbfk[i];
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Sdb[i].imag = Sdb[Nfft-i].imag = 0.0;
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}
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/* compute real cepstrum from log magnitude spectrum */
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codec2_fft(fft_inv_cfg, Sdb, c);
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for(i=0; i<Nfft; i++) {
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c[i].real /= (float)Nfft;
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c[i].imag /= (float)Nfft;
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}
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/* Fold cepstrum to reflect non-min-phase zeros inside unit circle */
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cf[0] = c[0];
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for(i=1; i<Ns-1; i++) {
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cf[i] = cadd(c[i],c[Nfft-i]);
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}
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cf[Ns-1] = c[Ns-1];
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for(i=Ns; i<Nfft; i++) {
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cf[i].real = 0.0;
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cf[i].imag = 0.0;
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}
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/* Cf = dB_magnitude + j * minimum_phase */
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codec2_fft(fft_fwd_cfg, cf, Cf);
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/* The maths says we are meant to be using log(x), not 20*log10(x),
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so we need to scale the phase to account for this:
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log(x) = 20*log10(x)/scale */
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float scale = (20.0/logf(10.0));
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for(i=0; i<Ns; i++) {
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phase[i] = Cf[i].imag/scale;
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}
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}
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