Bump to codec2 version 1.2.0erdgeist-bump-to-1.2.0

1 files changed, 62 insertions, 67 deletions

diff --git a/codec2_internal.h b/codec2_internal.h
index b46e358..32cd7eb 100644
--- a/codec2_internal.h
+++ b/codec2_internal.h
@@ -28,81 +28,76 @@
 
 #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 */
-    int            post_filter_en;
-    float          eq[NEWAMP1_K];            /* optional equaliser */
-    int            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;
-
-    /* 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);
+  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;
+
+  /* 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);
+                          COMP Aw[], float gain);
 #endif