From f02dfce6e6c34b3d8a7b8a0e784b506178e331fa Mon Sep 17 00:00:00 2001 From: "erdgeist@erdgeist.org" Date: Thu, 4 Jul 2019 23:26:09 +0200 Subject: stripdown of version 0.9 --- phase.c | 289 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 289 insertions(+) create mode 100644 phase.c (limited to 'phase.c') diff --git a/phase.c b/phase.c new file mode 100644 index 0000000..e486613 --- /dev/null +++ b/phase.c @@ -0,0 +1,289 @@ +/*---------------------------------------------------------------------------*\ + + 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 . +*/ + +#include "defines.h" +#include "phase.h" +#include "kiss_fft.h" +#include "comp.h" +#include "comp_prim.h" +#include "sine.h" + +#include +#include +#include +#include +#include + +/*---------------------------------------------------------------------------*\ + + sample_phase() + + Samples phase at centre of each harmonic from and array of FFT_ENC + DFT samples. + +\*---------------------------------------------------------------------------*/ + +void sample_phase(MODEL *model, + COMP H[], + COMP A[] /* LPC analysis filter in freq domain */ + ) +{ + int m, b; + float r; + + r = TWO_PI/(FFT_ENC); + + /* Sample phase at harmonics */ + + for(m=1; m<=model->L; m++) { + b = (int)(m*model->Wo/r + 0.5); + H[m] = cconj(A[b]); /* synth filter 1/A is opposite phase to analysis filter */ + } +} + + +/*---------------------------------------------------------------------------*\ + + 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 synthesis + filter excited by an impulse. In many Codec 2 modes the synthesis + filter is a LPC filter. 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_SAMP samples + over a synthesis frame of N_SAMP samples. For example if Wo is pi/20 + (200 Hz), then over a 10ms frame (N_SAMP=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_SAMP; + + 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 synthesis 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( + int n_samp, + MODEL *model, + float *ex_phase, /* excitation phase of fundamental */ + COMP H[] /* L synthesis filter freq domain samples */ + +) +{ + int m; + float new_phi; + COMP Ex[MAX_AMP+1]; /* excitation samples */ + COMP A_[MAX_AMP+1]; /* synthesised harmonic samples */ + + /* + 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+model->Wo)*N_SAMP/2; + */ + + ex_phase[0] += (model->Wo)*n_samp; + ex_phase[0] -= TWO_PI*floorf(ex_phase[0]/TWO_PI + 0.5); + + for(m=1; m<=model->L; m++) { + + /* generate excitation */ + + if (model->voiced) { + + Ex[m].real = cosf(ex_phase[0]*m); + Ex[m].imag = sinf(ex_phase[0]*m); + } + 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)codec2_rand()/CODEC2_RAND_MAX; + Ex[m].real = cosf(phi); + Ex[m].imag = sinf(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 = atan2f(A_[m].imag, A_[m].real+1E-12); + model->phi[m] = new_phi; + } + +} + + +/*---------------------------------------------------------------------------*\ + + FUNCTION....: mag_to_phase + AUTHOR......: David Rowe + DATE CREATED: Jan 2017 + + Algorithm for http://www.dsprelated.com/showcode/20.php ported to C. See + also Octave function mag_to_phase.m + + Given a magnitude spectrum in dB, returns a minimum-phase phase + spectra. + +\*---------------------------------------------------------------------------*/ + +void mag_to_phase(float phase[], /* Nfft/2+1 output phase samples in radians */ + float Gdbfk[], /* Nfft/2+1 postive freq amplitudes samples in dB */ + int Nfft, + codec2_fft_cfg fft_fwd_cfg, + codec2_fft_cfg fft_inv_cfg + ) +{ + COMP Sdb[Nfft], c[Nfft], cf[Nfft], Cf[Nfft]; + int Ns = Nfft/2+1; + int i; + + /* install negative frequency components, 1/Nfft takes into + account kiss fft lack of scaling on ifft */ + + Sdb[0].real = Gdbfk[0]; + Sdb[0].imag = 0.0; + for(i=1; i