From 30325d24d107dbf133da39f7c96d1510fd1c9449 Mon Sep 17 00:00:00 2001 From: erdgeist Date: Fri, 15 Aug 2025 12:42:40 +0200 Subject: Bump to codec2 version 1.2.0 --- phase.c | 242 ++++++++++++++++++++++++++++++---------------------------------- 1 file changed, 114 insertions(+), 128 deletions(-) (limited to 'phase.c') diff --git a/phase.c b/phase.c index e486613..dec8793 100644 --- a/phase.c +++ b/phase.c @@ -25,18 +25,19 @@ 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 +#include + +#include "comp.h" +#include "comp_prim.h" +#include "defines.h" +#include "kiss_fft.h" +#include "sine.h" /*---------------------------------------------------------------------------*\ @@ -47,25 +48,23 @@ \*---------------------------------------------------------------------------*/ -void sample_phase(MODEL *model, - COMP H[], - COMP A[] /* LPC analysis filter in freq domain */ - ) -{ - int m, b; - float r; +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); + r = TWO_PI / (FFT_ENC); - /* Sample phase at harmonics */ + /* 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 */ - } + 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() @@ -158,64 +157,56 @@ void sample_phase(MODEL *model, \*---------------------------------------------------------------------------*/ 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 */ + 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); + } - 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; + /* filter using LPC filter */ - /* modify sinusoidal phase */ + 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; - new_phi = atan2f(A_[m].imag, A_[m].real+1E-12); - model->phi[m] = new_phi; - } + /* modify sinusoidal phase */ + new_phi = atan2f(A_[m].imag, A_[m].real + 1E-12); + model->phi[m] = new_phi; + } } - /*---------------------------------------------------------------------------*\ FUNCTION....: mag_to_phase @@ -230,60 +221,55 @@ void phase_synth_zero_order( \*---------------------------------------------------------------------------*/ -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