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/*---------------------------------------------------------------------------*\
FILE........: newamp1.h
AUTHOR......: David Rowe
DATE CREATED: Jan 2017
Quantisation functions for the sinusoidal coder, using "newamp1"
algorithm that resamples variable rate L [Am} to a fixed rate K then
VQs.
\*---------------------------------------------------------------------------*/
/*
Copyright David Rowe 2017
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/>.
*/
#ifndef __NEWAMP1__
#define __NEWAMP1__
#define NEWAMP1_N_INDEXES 4 /* Number of indexes to pack: vq1, vq2, energy, Wo */
#define NEWAMP1_PHASE_NFFT 128 /* size of FFT used for phase synthesis */
#define NEWAMP1_K 20 /* rate K vector length */
#include "codec2_fft.h"
#include "comp.h"
void interp_para(float y[], float xp[], float yp[], int np, float x[], int n);
float ftomel(float fHz);
void mel_sample_freqs_kHz(float rate_K_sample_freqs_kHz[], int K, float mel_start, float mel_end);
void resample_const_rate_f(C2CONST *c2const, MODEL *model, float rate_K_vec[], float rate_K_sample_freqs_kHz[], int K);
float rate_K_mbest_encode(int *indexes, float *x, float *xq, int ndim, int mbest_entries);
void post_filter_newamp1(float vec[], float sample_freq_kHz[], int K, float pf_gain);
void interp_Wo_v(float Wo_[], int L_[], int voicing_[], float Wo1, float Wo2, int voicing1, int voicing2);
void resample_rate_L(C2CONST *c2const, MODEL *model, float rate_K_vec[], float rate_K_sample_freqs_kHz[], int K);
void determine_phase(C2CONST *c2const, COMP H[], MODEL *model, int Nfft, codec2_fft_cfg fwd_cfg, codec2_fft_cfg inv_cfg);
void newamp1_model_to_indexes(C2CONST *c2const,
int indexes[],
MODEL *model,
float rate_K_vec[],
float rate_K_sample_freqs_kHz[],
int K,
float *mean,
float rate_K_vec_no_mean[],
float rate_K_vec_no_mean_[],
float *se,
float *eq,
int eq_en);
void newamp1_indexes_to_rate_K_vec(float rate_K_vec_[],
float rate_K_vec_no_mean_[],
float rate_K_sample_freqs_kHz[],
int K,
float *mean_,
int indexes[],
float user_rate_K_vec_no_mean_[],
int post_filter_en);
void newamp1_interpolate(float interpolated_surface_[], float left_vec[], float right_vec[], int K);
void newamp1_indexes_to_model(C2CONST *c2const,
MODEL model_[],
COMP H[],
float interpolated_surface_[],
float prev_rate_K_vec_[],
float *Wo_left,
int *voicing_left,
float rate_K_sample_freqs_kHz[],
int K,
codec2_fft_cfg fwd_cfg,
codec2_fft_cfg inv_cfg,
int indexes[],
float user_rate_K_vec_no_mean_[],
int post_filter_en);
#endif
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