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Diffstat (limited to 'src/codec2_internal.h')
| -rw-r--r-- | src/codec2_internal.h | 109 |
1 files changed, 109 insertions, 0 deletions
diff --git a/src/codec2_internal.h b/src/codec2_internal.h new file mode 100644 index 0000000..b23ecaa --- /dev/null +++ b/src/codec2_internal.h @@ -0,0 +1,109 @@ +/*---------------------------------------------------------------------------*\ + + FILE........: codec2_internal.h + AUTHOR......: David Rowe + DATE CREATED: April 16 2012 + + Header file for Codec2 internal states, exposed via this header + file to assist in testing. + +\*---------------------------------------------------------------------------*/ + +/* + Copyright (C) 2012 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/>. +*/ + +#ifndef __CODEC2_INTERNAL__ +#define __CODEC2_INTERNAL__ +#include <stdbool.h> + +#include "codec2_fft.h" +#include "newamp1.h" +#include "newamp2.h" + +struct CODEC2 { + int mode; + C2CONST c2const; + int Fs; + int n_samp; + int m_pitch; + codec2_fft_cfg fft_fwd_cfg; /* forward FFT config */ + codec2_fftr_cfg fftr_fwd_cfg; /* forward real FFT config */ + float *w; /* [m_pitch] time domain hamming window */ + float W[FFT_ENC]; /* DFT of w[] */ + float *Pn; /* [2*n_samp] trapezoidal synthesis window */ + float *bpf_buf; /* buffer for band pass filter */ + float *Sn; /* [m_pitch] input speech */ + float hpf_states[2]; /* high pass filter states */ + void *nlp; /* pitch predictor states */ + int gray; /* non-zero for gray encoding */ + + codec2_fftr_cfg fftr_inv_cfg; /* inverse FFT config */ + float *Sn_; /* [2*n_samp] synthesised output speech */ + float ex_phase; /* excitation model phase track */ + float bg_est; /* background noise estimate for post filter */ + float prev_f0_enc; /* previous frame's f0 estimate */ + MODEL prev_model_dec; /* previous frame's model parameters */ + float prev_lsps_dec[LPC_ORD]; /* previous frame's LSPs */ + float prev_e_dec; /* previous frame's LPC energy */ + + int lpc_pf; /* LPC post filter on */ + int bass_boost; /* LPC post filter bass boost */ + float beta; /* LPC post filter parameters */ + float gamma; + + float xq_enc[2]; /* joint pitch and energy VQ states */ + float xq_dec[2]; + + int smoothing; /* enable smoothing for channels with errors */ + float *softdec; /* optional soft decn bits from demod */ + + /* newamp1 states */ + + float rate_K_sample_freqs_kHz[NEWAMP1_K]; + float prev_rate_K_vec_[NEWAMP1_K]; + float Wo_left; + int voicing_left; + codec2_fft_cfg phase_fft_fwd_cfg; + codec2_fft_cfg phase_fft_inv_cfg; + float se; /* running sum of squared error */ + unsigned int nse; /* number of terms in sum */ + float *user_rate_K_vec_no_mean_; /* optional, user supplied vector for quantisation experiments */ + bool post_filter_en; + float eq[NEWAMP1_K]; /* optional equaliser */ + bool eq_en; + + /*newamp2 states (also uses newamp1 states )*/ + float energy_prev; + float n2_rate_K_sample_freqs_kHz[NEWAMP2_K]; + float n2_prev_rate_K_vec_[NEWAMP2_K]; + float n2_pwb_rate_K_sample_freqs_kHz[NEWAMP2_16K_K]; + float n2_pwb_prev_rate_K_vec_[NEWAMP2_16K_K]; + + /* used to dump features for deep learning experiments */ + FILE *fmlfeat, *fmlmodel; + + /* encode/decode function pointers for the selected mode */ + void (*encode)(struct CODEC2 *c2, unsigned char * bits, short speech[]); + void (*decode)(struct CODEC2 *c2, short speech[], const unsigned char * bits); + void (*decode_ber)(struct CODEC2 *c2, short speech[], const unsigned char * bits, float ber_est); +}; + +// test and debug +void analyse_one_frame(struct CODEC2 *c2, MODEL *model, short speech[]); +void synthesise_one_frame(struct CODEC2 *c2, short speech[], MODEL *model, + COMP Aw[], float gain); +#endif |