/*---------------------------------------------------------------------------*\ FILE........: codec2.c AUTHOR......: David Rowe DATE CREATED: 21/8/2010 Codec2 fully quantised encoder and decoder functions. If you want use codec2, the codec2_xxx functions are for you. \*---------------------------------------------------------------------------*/ /* Copyright (C) 2010 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 #include #include #include #include #include #include "defines.h" #include "codec2_fft.h" #include "sine.h" #include "nlp.h" #include "dump.h" #include "lpc.h" #include "quantise.h" #include "phase.h" #include "interp.h" #include "postfilter.h" #include "codec2.h" #include "lsp.h" #include "newamp2.h" #include "codec2_internal.h" #include "machdep.h" #include "bpf.h" #include "bpfb.h" #include "c2wideband.h" #include "debug_alloc.h" /*---------------------------------------------------------------------------* \ FUNCTION HEADERS \*---------------------------------------------------------------------------*/ 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); void codec2_encode_3200(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_3200(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_2400(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_2400(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_1600(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_1600(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_1400(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_1400(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_1300(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_1300(struct CODEC2 *c2, short speech[], const unsigned char * bits, float ber_est); void codec2_encode_1200(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_1200(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_700(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_700(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_700b(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_700b(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_700c(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_700c(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_encode_450(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_450(struct CODEC2 *c2, short speech[], const unsigned char * bits); void codec2_decode_450pwb(struct CODEC2 *c2, short speech[], const unsigned char * bits); static void ear_protection(float in_out[], int n); /*---------------------------------------------------------------------------*\ FUNCTIONS \*---------------------------------------------------------------------------*/ /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_create AUTHOR......: David Rowe DATE CREATED: 21/8/2010 Create and initialise an instance of the codec. Returns a pointer to the codec states or NULL on failure. One set of states is sufficient for a full duuplex codec (i.e. an encoder and decoder). You don't need separate states for encoders and decoders. See c2enc.c and c2dec.c for examples. \*---------------------------------------------------------------------------*/ //Don't create CODEC2_MODE_450PWB for Encoding as it has undefined behavior ! struct CODEC2 * codec2_create(int mode) { struct CODEC2 *c2; int i,l; // ALL POSSIBLE MODES MUST BE CHECKED HERE! // we test if the desired mode is enabled at compile time // and return NULL if not if (false == ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_700, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_450, mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, mode) ) ) { return NULL; } c2 = (struct CODEC2*)MALLOC(sizeof(struct CODEC2)); if (c2 == NULL) return NULL; c2->mode = mode; /* store constants in a few places for convenience */ if( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, mode) == 0){ c2->c2const = c2const_create(8000, N_S); }else{ c2->c2const = c2const_create(16000, N_S); } c2->Fs = c2->c2const.Fs; int n_samp = c2->n_samp = c2->c2const.n_samp; int m_pitch = c2->m_pitch = c2->c2const.m_pitch; c2->Pn = (float*)MALLOC(2*n_samp*sizeof(float)); if (c2->Pn == NULL) { return NULL; } c2->Sn_ = (float*)MALLOC(2*n_samp*sizeof(float)); if (c2->Sn_ == NULL) { FREE(c2->Pn); return NULL; } c2->w = (float*)MALLOC(m_pitch*sizeof(float)); if (c2->w == NULL) { FREE(c2->Pn); FREE(c2->Sn_); return NULL; } c2->Sn = (float*)MALLOC(m_pitch*sizeof(float)); if (c2->Sn == NULL) { FREE(c2->Pn); FREE(c2->Sn_); FREE(c2->w); return NULL; } for(i=0; iSn[i] = 1.0; c2->hpf_states[0] = c2->hpf_states[1] = 0.0; for(i=0; i<2*n_samp; i++) c2->Sn_[i] = 0; c2->fft_fwd_cfg = codec2_fft_alloc(FFT_ENC, 0, NULL, NULL); c2->fftr_fwd_cfg = codec2_fftr_alloc(FFT_ENC, 0, NULL, NULL); make_analysis_window(&c2->c2const, c2->fft_fwd_cfg, c2->w,c2->W); make_synthesis_window(&c2->c2const, c2->Pn); c2->fftr_inv_cfg = codec2_fftr_alloc(FFT_DEC, 1, NULL, NULL); quantise_init(); c2->prev_f0_enc = 1/P_MAX_S; c2->bg_est = 0.0; c2->ex_phase = 0.0; for(l=1; l<=MAX_AMP; l++) c2->prev_model_dec.A[l] = 0.0; c2->prev_model_dec.Wo = TWO_PI/c2->c2const.p_max; c2->prev_model_dec.L = PI/c2->prev_model_dec.Wo; c2->prev_model_dec.voiced = 0; for(i=0; iprev_lsps_dec[i] = i*PI/(LPC_ORD+1); } c2->prev_e_dec = 1; c2->nlp = nlp_create(&c2->c2const); if (c2->nlp == NULL) { return NULL; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, mode)) c2->gray = 0; // natural binary better for trellis decoding (hopefully added later) else c2->gray = 1; c2->lpc_pf = 1; c2->bass_boost = 1; c2->beta = LPCPF_BETA; c2->gamma = LPCPF_GAMMA; c2->xq_enc[0] = c2->xq_enc[1] = 0.0; c2->xq_dec[0] = c2->xq_dec[1] = 0.0; c2->smoothing = 0; c2->se = 0.0; c2->nse = 0; c2->user_rate_K_vec_no_mean_ = NULL; c2->post_filter_en = 1; c2->bpf_buf = (float*)MALLOC(sizeof(float)*(BPF_N+4*c2->n_samp)); assert(c2->bpf_buf != NULL); for(i=0; in_samp; i++) c2->bpf_buf[i] = 0.0; c2->softdec = NULL; /* newamp1 initialisation */ if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) { mel_sample_freqs_kHz(c2->rate_K_sample_freqs_kHz, NEWAMP1_K, ftomel(200.0), ftomel(3700.0) ); int k; for(k=0; kprev_rate_K_vec_[k] = 0.0; c2->eq[k] = 0.0; } c2->eq_en = 0; c2->Wo_left = 0.0; c2->voicing_left = 0;; c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP1_PHASE_NFFT, 0, NULL, NULL); c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP1_PHASE_NFFT, 1, NULL, NULL); } /* newamp2 initialisation */ if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) { n2_mel_sample_freqs_kHz(c2->n2_rate_K_sample_freqs_kHz, NEWAMP2_K); int k; for(k=0; kn2_prev_rate_K_vec_[k] = 0.0; } c2->Wo_left = 0.0; c2->voicing_left = 0;; c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 0, NULL, NULL); c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 1, NULL, NULL); } /* newamp2 PWB initialisation */ if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) { n2_mel_sample_freqs_kHz(c2->n2_pwb_rate_K_sample_freqs_kHz, NEWAMP2_16K_K); int k; for(k=0; kn2_pwb_prev_rate_K_vec_[k] = 0.0; } c2->Wo_left = 0.0; c2->voicing_left = 0;; c2->phase_fft_fwd_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 0, NULL, NULL); c2->phase_fft_inv_cfg = codec2_fft_alloc(NEWAMP2_PHASE_NFFT, 1, NULL, NULL); } c2->fmlfeat = NULL; // make sure that one of the two decode function pointers is empty // for the encode function pointer this is not required since we always set it // to a meaningful value c2->decode = NULL; c2->decode_ber = NULL; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, c2->mode)) { c2->encode = codec2_encode_3200; c2->decode = codec2_decode_3200; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, c2->mode)) { c2->encode = codec2_encode_2400; c2->decode = codec2_decode_2400; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, c2->mode)) { c2->encode = codec2_encode_1600; c2->decode = codec2_decode_1600; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, c2->mode)) { c2->encode = codec2_encode_1400; c2->decode = codec2_decode_1400; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) { c2->encode = codec2_encode_1300; c2->decode_ber = codec2_decode_1300; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) { c2->encode = codec2_encode_1200; c2->decode = codec2_decode_1200; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700, c2->mode)) { c2->encode = codec2_encode_700; c2->decode = codec2_decode_700; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, c2->mode)) { c2->encode = codec2_encode_700b; c2->decode = codec2_decode_700b; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) { c2->encode = codec2_encode_700c; c2->decode = codec2_decode_700c; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) { c2->encode = codec2_encode_450; c2->decode = codec2_decode_450; } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) { //Encode PWB doesnt make sense c2->encode = codec2_encode_450; c2->decode = codec2_decode_450pwb; } return c2; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_destroy AUTHOR......: David Rowe DATE CREATED: 21/8/2010 Destroy an instance of the codec. \*---------------------------------------------------------------------------*/ void codec2_destroy(struct CODEC2 *c2) { assert(c2 != NULL); FREE(c2->bpf_buf); nlp_destroy(c2->nlp); codec2_fft_free(c2->fft_fwd_cfg); codec2_fftr_free(c2->fftr_fwd_cfg); codec2_fftr_free(c2->fftr_inv_cfg); if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) { codec2_fft_free(c2->phase_fft_fwd_cfg); codec2_fft_free(c2->phase_fft_inv_cfg); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) { codec2_fft_free(c2->phase_fft_fwd_cfg); codec2_fft_free(c2->phase_fft_inv_cfg); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) { codec2_fft_free(c2->phase_fft_fwd_cfg); codec2_fft_free(c2->phase_fft_inv_cfg); } FREE(c2->Pn); FREE(c2->Sn); FREE(c2->w); FREE(c2->Sn_); FREE(c2); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_bits_per_frame AUTHOR......: David Rowe DATE CREATED: Nov 14 2011 Returns the number of bits per frame. \*---------------------------------------------------------------------------*/ int codec2_bits_per_frame(struct CODEC2 *c2) { if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, c2->mode)) return 64; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, c2->mode)) return 48; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, c2->mode)) return 64; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, c2->mode)) return 56; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) return 52; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) return 48; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700, c2->mode)) return 28; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, c2->mode)) return 28; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) return 28; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) return 18; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) return 18; return 0; /* shouldn't get here */ } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_samples_per_frame AUTHOR......: David Rowe DATE CREATED: Nov 14 2011 Returns the number of speech samples per frame. \*---------------------------------------------------------------------------*/ int codec2_samples_per_frame(struct CODEC2 *c2) { if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, c2->mode)) return 160; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, c2->mode)) return 160; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) return 320; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) return 640; return 0; /* shouldnt get here */ } void codec2_encode(struct CODEC2 *c2, unsigned char *bits, short speech[]) { assert(c2 != NULL); assert(c2->encode != NULL); c2->encode(c2, bits, speech); } void codec2_decode(struct CODEC2 *c2, short speech[], const unsigned char *bits) { codec2_decode_ber(c2, speech, bits, 0.0); } void codec2_decode_ber(struct CODEC2 *c2, short speech[], const unsigned char *bits, float ber_est) { assert(c2 != NULL); assert(c2->decode != NULL || c2->decode_ber != NULL); if (c2->decode != NULL) { c2->decode(c2, speech, bits); } else { c2->decode_ber(c2, speech, bits, ber_est); } } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_3200 AUTHOR......: David Rowe DATE CREATED: 13 Sep 2012 Encodes 160 speech samples (20ms of speech) into 64 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm twice. On the first frame we just send the voicing bits. On the second frame we send all model parameters. Compared to 2400 we use a larger number of bits for the LSPs and non-VQ pitch and energy. The bit allocation is: Parameter bits/frame -------------------------------------- Harmonic magnitudes (LSPs) 50 Pitch (Wo) 7 Energy 5 Voicing (10ms update) 2 TOTAL 64 \*---------------------------------------------------------------------------*/ void codec2_encode_3200(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float ak[LPC_ORD+1]; float lsps[LPC_ORD]; float e; int Wo_index, e_index; int lspd_indexes[LPC_ORD]; int i; unsigned int nbit = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* first 10ms analysis frame - we just want voicing */ analyse_one_frame(c2, &model, speech); pack(bits, &nbit, model.voiced, 1); /* second 10ms analysis frame */ analyse_one_frame(c2, &model, &speech[c2->n_samp]); pack(bits, &nbit, model.voiced, 1); Wo_index = encode_Wo(&c2->c2const, model.Wo, WO_BITS); pack(bits, &nbit, Wo_index, WO_BITS); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); e_index = encode_energy(e, E_BITS); pack(bits, &nbit, e_index, E_BITS); encode_lspds_scalar(lspd_indexes, lsps, LPC_ORD); for(i=0; ic2const, Wo_index, WO_BITS); model[1].L = PI/model[1].Wo; e_index = unpack(bits, &nbit, E_BITS); e[1] = decode_energy(e_index, E_BITS); for(i=0; iprev_model_dec, &model[1], c2->c2const.Wo_min); e[0] = interp_energy(c2->prev_e_dec, e[1]); /* LSPs are sampled every 20ms so we interpolate the frame in between, then recover spectral amplitudes */ interpolate_lsp_ver2(&lsps[0][0], c2->prev_lsps_dec, &lsps[1][0], 0.5, LPC_ORD); for(i=0; i<2; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); } /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[1]; c2->prev_e_dec = e[1]; for(i=0; iprev_lsps_dec[i] = lsps[1][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_2400 AUTHOR......: David Rowe DATE CREATED: 21/8/2010 Encodes 160 speech samples (20ms of speech) into 48 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm twice. On the first frame we just send the voicing bit. On the second frame we send all model parameters. The bit allocation is: Parameter bits/frame -------------------------------------- Harmonic magnitudes (LSPs) 36 Joint VQ of Energy and Wo 8 Voicing (10ms update) 2 Spare 2 TOTAL 48 \*---------------------------------------------------------------------------*/ void codec2_encode_2400(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float ak[LPC_ORD+1]; float lsps[LPC_ORD]; float e; int WoE_index; int lsp_indexes[LPC_ORD]; int i; int spare = 0; unsigned int nbit = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* first 10ms analysis frame - we just want voicing */ analyse_one_frame(c2, &model, speech); pack(bits, &nbit, model.voiced, 1); /* second 10ms analysis frame */ analyse_one_frame(c2, &model, &speech[c2->n_samp]); pack(bits, &nbit, model.voiced, 1); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); WoE_index = encode_WoE(&model, e, c2->xq_enc); pack(bits, &nbit, WoE_index, WO_E_BITS); encode_lsps_scalar(lsp_indexes, lsps, LPC_ORD); for(i=0; ic2const, &model[1], &e[1], c2->xq_dec, WoE_index); for(i=0; iprev_model_dec, &model[1], c2->c2const.Wo_min); e[0] = interp_energy(c2->prev_e_dec, e[1]); /* LSPs are sampled every 20ms so we interpolate the frame in between, then recover spectral amplitudes */ interpolate_lsp_ver2(&lsps[0][0], c2->prev_lsps_dec, &lsps[1][0], 0.5, LPC_ORD); for(i=0; i<2; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); /* dump parameters for deep learning experiments */ if (c2->fmlfeat != NULL) { /* 10 LSPs - energy - Wo - voicing flag - 10 LPCs */ fwrite(&lsps[i][0], LPC_ORD, sizeof(float), c2->fmlfeat); fwrite(&e[i], 1, sizeof(float), c2->fmlfeat); fwrite(&model[i].Wo, 1, sizeof(float), c2->fmlfeat); float voiced_float = model[i].voiced; fwrite(&voiced_float, 1, sizeof(float), c2->fmlfeat); fwrite(&ak[i][1], LPC_ORD, sizeof(float), c2->fmlfeat); } } /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[1]; c2->prev_e_dec = e[1]; for(i=0; iprev_lsps_dec[i] = lsps[1][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_1600 AUTHOR......: David Rowe DATE CREATED: Feb 28 2013 Encodes 320 speech samples (40ms of speech) into 64 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm 4 times: frame 0: voicing bit frame 1: voicing bit, Wo and E frame 2: voicing bit frame 3: voicing bit, Wo and E, scalar LSPs The bit allocation is: Parameter frame 2 frame 4 Total ------------------------------------------------------- Harmonic magnitudes (LSPs) 0 36 36 Pitch (Wo) 7 7 14 Energy 5 5 10 Voicing (10ms update) 2 2 4 TOTAL 14 50 64 \*---------------------------------------------------------------------------*/ void codec2_encode_1600(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float lsps[LPC_ORD]; float ak[LPC_ORD+1]; float e; int lsp_indexes[LPC_ORD]; int Wo_index, e_index; int i; unsigned int nbit = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* frame 1: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, speech); pack(bits, &nbit, model.voiced, 1); /* frame 2: - voicing, scalar Wo & E -------------------------------*/ analyse_one_frame(c2, &model, &speech[c2->n_samp]); pack(bits, &nbit, model.voiced, 1); Wo_index = encode_Wo(&c2->c2const, model.Wo, WO_BITS); pack(bits, &nbit, Wo_index, WO_BITS); /* need to run this just to get LPC energy */ e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); e_index = encode_energy(e, E_BITS); pack(bits, &nbit, e_index, E_BITS); /* frame 3: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, &speech[2*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); /* frame 4: - voicing, scalar Wo & E, scalar LSPs ------------------*/ analyse_one_frame(c2, &model, &speech[3*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); Wo_index = encode_Wo(&c2->c2const, model.Wo, WO_BITS); pack(bits, &nbit, Wo_index, WO_BITS); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); e_index = encode_energy(e, E_BITS); pack(bits, &nbit, e_index, E_BITS); encode_lsps_scalar(lsp_indexes, lsps, LPC_ORD); for(i=0; ic2const, Wo_index, WO_BITS); model[1].L = PI/model[1].Wo; e_index = unpack(bits, &nbit, E_BITS); e[1] = decode_energy(e_index, E_BITS); model[2].voiced = unpack(bits, &nbit, 1); model[3].voiced = unpack(bits, &nbit, 1); Wo_index = unpack(bits, &nbit, WO_BITS); model[3].Wo = decode_Wo(&c2->c2const, Wo_index, WO_BITS); model[3].L = PI/model[3].Wo; e_index = unpack(bits, &nbit, E_BITS); e[3] = decode_energy(e_index, E_BITS); for(i=0; iprev_model_dec, &model[1], c2->c2const.Wo_min); e[0] = interp_energy(c2->prev_e_dec, e[1]); interp_Wo(&model[2], &model[1], &model[3], c2->c2const.Wo_min); e[2] = interp_energy(e[1], e[3]); /* LSPs are sampled every 40ms so we interpolate the 3 frames in between, then recover spectral amplitudes */ for(i=0, weight=0.25; i<3; i++, weight += 0.25) { interpolate_lsp_ver2(&lsps[i][0], c2->prev_lsps_dec, &lsps[3][0], weight, LPC_ORD); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); } /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[3]; c2->prev_e_dec = e[3]; for(i=0; iprev_lsps_dec[i] = lsps[3][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_1400 AUTHOR......: David Rowe DATE CREATED: May 11 2012 Encodes 320 speech samples (40ms of speech) into 56 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm 4 times: frame 0: voicing bit frame 1: voicing bit, joint VQ of Wo and E frame 2: voicing bit frame 3: voicing bit, joint VQ of Wo and E, scalar LSPs The bit allocation is: Parameter frame 2 frame 4 Total ------------------------------------------------------- Harmonic magnitudes (LSPs) 0 36 36 Energy+Wo 8 8 16 Voicing (10ms update) 2 2 4 TOTAL 10 46 56 \*---------------------------------------------------------------------------*/ void codec2_encode_1400(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float lsps[LPC_ORD]; float ak[LPC_ORD+1]; float e; int lsp_indexes[LPC_ORD]; int WoE_index; int i; unsigned int nbit = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* frame 1: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, speech); pack(bits, &nbit, model.voiced, 1); /* frame 2: - voicing, joint Wo & E -------------------------------*/ analyse_one_frame(c2, &model, &speech[c2->n_samp]); pack(bits, &nbit, model.voiced, 1); /* need to run this just to get LPC energy */ e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); WoE_index = encode_WoE(&model, e, c2->xq_enc); pack(bits, &nbit, WoE_index, WO_E_BITS); /* frame 3: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, &speech[2*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); /* frame 4: - voicing, joint Wo & E, scalar LSPs ------------------*/ analyse_one_frame(c2, &model, &speech[3*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); WoE_index = encode_WoE(&model, e, c2->xq_enc); pack(bits, &nbit, WoE_index, WO_E_BITS); encode_lsps_scalar(lsp_indexes, lsps, LPC_ORD); for(i=0; ic2const, &model[1], &e[1], c2->xq_dec, WoE_index); model[2].voiced = unpack(bits, &nbit, 1); model[3].voiced = unpack(bits, &nbit, 1); WoE_index = unpack(bits, &nbit, WO_E_BITS); decode_WoE(&c2->c2const, &model[3], &e[3], c2->xq_dec, WoE_index); for(i=0; iprev_model_dec, &model[1], c2->c2const.Wo_min); e[0] = interp_energy(c2->prev_e_dec, e[1]); interp_Wo(&model[2], &model[1], &model[3], c2->c2const.Wo_min); e[2] = interp_energy(e[1], e[3]); /* LSPs are sampled every 40ms so we interpolate the 3 frames in between, then recover spectral amplitudes */ for(i=0, weight=0.25; i<3; i++, weight += 0.25) { interpolate_lsp_ver2(&lsps[i][0], c2->prev_lsps_dec, &lsps[3][0], weight, LPC_ORD); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); } /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[3]; c2->prev_e_dec = e[3]; for(i=0; iprev_lsps_dec[i] = lsps[3][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_1300 AUTHOR......: David Rowe DATE CREATED: March 14 2013 Encodes 320 speech samples (40ms of speech) into 52 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm 4 times: frame 0: voicing bit frame 1: voicing bit, frame 2: voicing bit frame 3: voicing bit, Wo and E, scalar LSPs The bit allocation is: Parameter frame 2 frame 4 Total ------------------------------------------------------- Harmonic magnitudes (LSPs) 0 36 36 Pitch (Wo) 0 7 7 Energy 0 5 5 Voicing (10ms update) 2 2 4 TOTAL 2 50 52 \*---------------------------------------------------------------------------*/ void codec2_encode_1300(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float lsps[LPC_ORD]; float ak[LPC_ORD+1]; float e; int lsp_indexes[LPC_ORD]; int Wo_index, e_index; int i; unsigned int nbit = 0; //#ifdef PROFILE //unsigned int quant_start; //#endif assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* frame 1: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, speech); pack_natural_or_gray(bits, &nbit, model.voiced, 1, c2->gray); /* frame 2: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, &speech[c2->n_samp]); pack_natural_or_gray(bits, &nbit, model.voiced, 1, c2->gray); /* frame 3: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, &speech[2*c2->n_samp]); pack_natural_or_gray(bits, &nbit, model.voiced, 1, c2->gray); /* frame 4: - voicing, scalar Wo & E, scalar LSPs ------------------*/ analyse_one_frame(c2, &model, &speech[3*c2->n_samp]); pack_natural_or_gray(bits, &nbit, model.voiced, 1, c2->gray); Wo_index = encode_Wo(&c2->c2const, model.Wo, WO_BITS); pack_natural_or_gray(bits, &nbit, Wo_index, WO_BITS, c2->gray); //#ifdef PROFILE //quant_start = machdep_profile_sample(); //#endif e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); e_index = encode_energy(e, E_BITS); pack_natural_or_gray(bits, &nbit, e_index, E_BITS, c2->gray); encode_lsps_scalar(lsp_indexes, lsps, LPC_ORD); for(i=0; igray); } //#ifdef PROFILE //machdep_profile_sample_and_log(quant_start, " quant/packing"); //#endif assert(nbit == (unsigned)codec2_bits_per_frame(c2)); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_decode_1300 AUTHOR......: David Rowe DATE CREATED: 11 May 2012 Decodes frames of 52 bits into 320 samples (40ms) of speech. \*---------------------------------------------------------------------------*/ static int frames; void codec2_decode_1300(struct CODEC2 *c2, short speech[], const unsigned char * bits, float ber_est) { MODEL model[4]; int lsp_indexes[LPC_ORD]; float lsps[4][LPC_ORD]; int Wo_index, e_index; float e[4]; float snr; float ak[4][LPC_ORD+1]; int i,j; unsigned int nbit = 0; float weight; COMP Aw[FFT_ENC]; //PROFILE_VAR(recover_start); assert(c2 != NULL); frames+= 4; /* only need to zero these out due to (unused) snr calculation */ for(i=0; i<4; i++) for(j=1; j<=MAX_AMP; j++) model[i].A[j] = 0.0; /* unpack bits from channel ------------------------------------*/ /* this will partially fill the model params for the 4 x 10ms frames */ model[0].voiced = unpack_natural_or_gray(bits, &nbit, 1, c2->gray); model[1].voiced = unpack_natural_or_gray(bits, &nbit, 1, c2->gray); model[2].voiced = unpack_natural_or_gray(bits, &nbit, 1, c2->gray); model[3].voiced = unpack_natural_or_gray(bits, &nbit, 1, c2->gray); Wo_index = unpack_natural_or_gray(bits, &nbit, WO_BITS, c2->gray); model[3].Wo = decode_Wo(&c2->c2const, Wo_index, WO_BITS); model[3].L = PI/model[3].Wo; e_index = unpack_natural_or_gray(bits, &nbit, E_BITS, c2->gray); e[3] = decode_energy(e_index, E_BITS); //fprintf(stderr, "%d %f\n", e_index, e[3]); for(i=0; igray); } decode_lsps_scalar(&lsps[3][0], lsp_indexes, LPC_ORD); check_lsp_order(&lsps[3][0], LPC_ORD); bw_expand_lsps(&lsps[3][0], LPC_ORD, 50.0, 100.0); if (ber_est > 0.15) { model[0].voiced = model[1].voiced = model[2].voiced = model[3].voiced = 0; e[3] = decode_energy(10, E_BITS); bw_expand_lsps(&lsps[3][0], LPC_ORD, 200.0, 200.0); //fprintf(stderr, "soft mute\n"); } /* interpolate ------------------------------------------------*/ /* Wo, energy, and LSPs are sampled every 40ms so we interpolate the 3 frames in between */ //PROFILE_SAMPLE(recover_start); for(i=0, weight=0.25; i<3; i++, weight += 0.25) { interpolate_lsp_ver2(&lsps[i][0], c2->prev_lsps_dec, &lsps[3][0], weight, LPC_ORD); interp_Wo2(&model[i], &c2->prev_model_dec, &model[3], weight, c2->c2const.Wo_min); e[i] = interp_energy2(c2->prev_e_dec, e[3],weight); } /* then recover spectral amplitudes */ for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); /* dump parameters for deep learning experiments */ if (c2->fmlfeat != NULL) { /* 10 LSPs - energy - Wo - voicing flag - 10 LPCs */ fwrite(&lsps[i][0], LPC_ORD, sizeof(float), c2->fmlfeat); fwrite(&e[i], 1, sizeof(float), c2->fmlfeat); fwrite(&model[i].Wo, 1, sizeof(float), c2->fmlfeat); float voiced_float = model[i].voiced; fwrite(&voiced_float, 1, sizeof(float), c2->fmlfeat); fwrite(&ak[i][1], LPC_ORD, sizeof(float), c2->fmlfeat); } } /* for(i=0; i<4; i++) { printf("%d Wo: %f L: %d v: %d\n", frames, model[i].Wo, model[i].L, model[i].voiced); } if (frames == 4*50) exit(0); */ //PROFILE_SAMPLE_AND_LOG2(recover_start, " recover"); #ifdef DUMP dump_lsp_(&lsps[3][0]); dump_ak_(&ak[3][0], LPC_ORD); #endif /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[3]; c2->prev_e_dec = e[3]; for(i=0; iprev_lsps_dec[i] = lsps[3][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_1200 AUTHOR......: David Rowe DATE CREATED: Nov 14 2011 Encodes 320 speech samples (40ms of speech) into 48 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm four times: frame 0: voicing bit frame 1: voicing bit, joint VQ of Wo and E frame 2: voicing bit frame 3: voicing bit, joint VQ of Wo and E, VQ LSPs The bit allocation is: Parameter frame 2 frame 4 Total ------------------------------------------------------- Harmonic magnitudes (LSPs) 0 27 27 Energy+Wo 8 8 16 Voicing (10ms update) 2 2 4 Spare 0 1 1 TOTAL 10 38 48 \*---------------------------------------------------------------------------*/ void codec2_encode_1200(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float lsps[LPC_ORD]; float lsps_[LPC_ORD]; float ak[LPC_ORD+1]; float e; int lsp_indexes[LPC_ORD]; int WoE_index; int i; int spare = 0; unsigned int nbit = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* frame 1: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, speech); pack(bits, &nbit, model.voiced, 1); /* frame 2: - voicing, joint Wo & E -------------------------------*/ analyse_one_frame(c2, &model, &speech[c2->n_samp]); pack(bits, &nbit, model.voiced, 1); /* need to run this just to get LPC energy */ e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); WoE_index = encode_WoE(&model, e, c2->xq_enc); pack(bits, &nbit, WoE_index, WO_E_BITS); /* frame 3: - voicing ---------------------------------------------*/ analyse_one_frame(c2, &model, &speech[2*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); /* frame 4: - voicing, joint Wo & E, scalar LSPs ------------------*/ analyse_one_frame(c2, &model, &speech[3*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD); WoE_index = encode_WoE(&model, e, c2->xq_enc); pack(bits, &nbit, WoE_index, WO_E_BITS); encode_lsps_vq(lsp_indexes, lsps, lsps_, LPC_ORD); for(i=0; ic2const, &model[1], &e[1], c2->xq_dec, WoE_index); model[2].voiced = unpack(bits, &nbit, 1); model[3].voiced = unpack(bits, &nbit, 1); WoE_index = unpack(bits, &nbit, WO_E_BITS); decode_WoE(&c2->c2const, &model[3], &e[3], c2->xq_dec, WoE_index); for(i=0; iprev_model_dec, &model[1], c2->c2const.Wo_min); e[0] = interp_energy(c2->prev_e_dec, e[1]); interp_Wo(&model[2], &model[1], &model[3], c2->c2const.Wo_min); e[2] = interp_energy(e[1], e[3]); /* LSPs are sampled every 40ms so we interpolate the 3 frames in between, then recover spectral amplitudes */ for(i=0, weight=0.25; i<3; i++, weight += 0.25) { interpolate_lsp_ver2(&lsps[i][0], c2->prev_lsps_dec, &lsps[3][0], weight, LPC_ORD); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); } /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[3]; c2->prev_e_dec = e[3]; for(i=0; iprev_lsps_dec[i] = lsps[3][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_700 AUTHOR......: David Rowe DATE CREATED: April 2015 Encodes 320 speech samples (40ms of speech) into 28 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm four times: frame 0: nothing frame 1: nothing frame 2: nothing frame 3: voicing bit, scalar Wo and E, 17 bit LSP MEL scalar, 2 spare The bit allocation is: Parameter frames 1-3 frame 4 Total ----------------------------------------------------------- Harmonic magnitudes (LSPs) 0 17 17 Energy 0 3 3 log Wo 0 5 5 Voicing 0 1 1 spare 0 2 2 TOTAL 0 28 28 \*---------------------------------------------------------------------------*/ void codec2_encode_700(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float lsps[LPC_ORD_LOW]; float mel[LPC_ORD_LOW]; float ak[LPC_ORD_LOW+1]; float e, f; int indexes[LPC_ORD_LOW]; int Wo_index, e_index, i; unsigned int nbit = 0; float bpf_out[4*c2->n_samp]; short bpf_speech[4*c2->n_samp]; int spare = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* band pass filter */ for(i=0; ibpf_buf[i] = c2->bpf_buf[4*c2->n_samp+i]; for(i=0; i<4*c2->n_samp; i++) c2->bpf_buf[BPF_N+i] = speech[i]; inverse_filter(&c2->bpf_buf[BPF_N], bpf, 4*c2->n_samp, bpf_out, BPF_N-1); for(i=0; i<4*c2->n_samp; i++) bpf_speech[i] = bpf_out[i]; /* frame 1 --------------------------------------------------------*/ analyse_one_frame(c2, &model, bpf_speech); /* frame 2 --------------------------------------------------------*/ analyse_one_frame(c2, &model, &bpf_speech[c2->n_samp]); /* frame 3 --------------------------------------------------------*/ analyse_one_frame(c2, &model, &bpf_speech[2*c2->n_samp]); /* frame 4: - voicing, scalar Wo & E, scalar LSPs -----------------*/ analyse_one_frame(c2, &model, &bpf_speech[3*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); Wo_index = encode_log_Wo(&c2->c2const, model.Wo, 5); pack_natural_or_gray(bits, &nbit, Wo_index, 5, c2->gray); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD_LOW); e_index = encode_energy(e, 3); pack_natural_or_gray(bits, &nbit, e_index, 3, c2->gray); for(i=0; igray); } pack_natural_or_gray(bits, &nbit, spare, 2, c2->gray); assert(nbit == (unsigned)codec2_bits_per_frame(c2)); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_decode_700 AUTHOR......: David Rowe DATE CREATED: April 2015 Decodes frames of 28 bits into 320 samples (40ms) of speech. \*---------------------------------------------------------------------------*/ void codec2_decode_700(struct CODEC2 *c2, short speech[], const unsigned char * bits) { MODEL model[4]; int indexes[LPC_ORD_LOW]; float mel[LPC_ORD_LOW]; float lsps[4][LPC_ORD_LOW]; int Wo_index, e_index; float e[4]; float snr, f_; float ak[4][LPC_ORD_LOW+1]; int i,j; unsigned int nbit = 0; float weight; COMP Aw[FFT_ENC]; assert(c2 != NULL); /* only need to zero these out due to (unused) snr calculation */ for(i=0; i<4; i++) for(j=1; j<=MAX_AMP; j++) model[i].A[j] = 0.0; /* unpack bits from channel ------------------------------------*/ model[3].voiced = unpack(bits, &nbit, 1); model[0].voiced = model[1].voiced = model[2].voiced = model[3].voiced; Wo_index = unpack_natural_or_gray(bits, &nbit, 5, c2->gray); model[3].Wo = decode_log_Wo(&c2->c2const, Wo_index, 5); model[3].L = PI/model[3].Wo; e_index = unpack_natural_or_gray(bits, &nbit, 3, c2->gray); e[3] = decode_energy(e_index, 3); for(i=0; igray); } decode_mels_scalar(mel, indexes, LPC_ORD_LOW); for(i=0; isoftdec) { float e = 0.0; for(i=9; i<9+17; i++) e += c2->softdec[i]*c2->softdec[i]; e /= 6.0; //fprintf(stderr, "e: %f\n", e); //if (e < 0.3) // bw_expand_lsps(&lsps[3][0], LPC_ORD_LOW, 150.0, 300.0); } #endif /* interpolate ------------------------------------------------*/ /* LSPs, Wo, and energy are sampled every 40ms so we interpolate the 3 frames in between, then recover spectral amplitudes */ for(i=0, weight=0.25; i<3; i++, weight += 0.25) { interpolate_lsp_ver2(&lsps[i][0], c2->prev_lsps_dec, &lsps[3][0], weight, LPC_ORD_LOW); interp_Wo2(&model[i], &c2->prev_model_dec, &model[3], weight, c2->c2const.Wo_min); e[i] = interp_energy2(c2->prev_e_dec, e[3],weight); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD_LOW); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD_LOW, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); } #ifdef DUMP dump_lsp_(&lsps[3][0]); dump_ak_(&ak[3][0], LPC_ORD_LOW); dump_model(&model[3]); if (c2->softdec) dump_softdec(c2->softdec, nbit); #endif /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[3]; c2->prev_e_dec = e[3]; for(i=0; iprev_lsps_dec[i] = lsps[3][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_700b AUTHOR......: David Rowe DATE CREATED: August 2015 Version b of 700 bit/s codec. After some experiments over the air I wanted was unhappy with the rate 700 codec so spent a few weeks trying to improve the speech quality. This version uses a wider BPF and vector quantised mel-lsps. Encodes 320 speech samples (40ms of speech) into 28 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm four times: frame 0: nothing frame 1: nothing frame 2: nothing frame 3: voicing bit, 5 bit scalar Wo and 3 bit E, 18 bit LSP MEL VQ, 1 spare The bit allocation is: Parameter frames 1-3 frame 4 Total ----------------------------------------------------------- Harmonic magnitudes (LSPs) 0 18 18 Energy 0 3 3 log Wo 0 5 5 Voicing 0 1 1 spare 0 1 1 TOTAL 0 28 28 \*---------------------------------------------------------------------------*/ void codec2_encode_700b(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; float lsps[LPC_ORD_LOW]; float mel[LPC_ORD_LOW]; float mel_[LPC_ORD_LOW]; float ak[LPC_ORD_LOW+1]; float e, f; int indexes[3]; int Wo_index, e_index, i; unsigned int nbit = 0; float bpf_out[4*c2->n_samp]; short bpf_speech[4*c2->n_samp]; int spare = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); /* band pass filter */ for(i=0; ibpf_buf[i] = c2->bpf_buf[4*c2->n_samp+i]; for(i=0; i<4*c2->n_samp; i++) c2->bpf_buf[BPF_N+i] = speech[i]; inverse_filter(&c2->bpf_buf[BPF_N], bpfb, 4*c2->n_samp, bpf_out, BPF_N-1); for(i=0; i<4*c2->n_samp; i++) bpf_speech[i] = bpf_out[i]; /* frame 1 --------------------------------------------------------*/ analyse_one_frame(c2, &model, bpf_speech); /* frame 2 --------------------------------------------------------*/ analyse_one_frame(c2, &model, &bpf_speech[c2->n_samp]); /* frame 3 --------------------------------------------------------*/ analyse_one_frame(c2, &model, &bpf_speech[2*c2->n_samp]); /* frame 4: - voicing, scalar Wo & E, VQ mel LSPs -----------------*/ analyse_one_frame(c2, &model, &bpf_speech[3*c2->n_samp]); pack(bits, &nbit, model.voiced, 1); Wo_index = encode_log_Wo(&c2->c2const, model.Wo, 5); pack_natural_or_gray(bits, &nbit, Wo_index, 5, c2->gray); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, c2->m_pitch, LPC_ORD_LOW); e_index = encode_energy(e, 3); pack_natural_or_gray(bits, &nbit, e_index, 3, c2->gray); for(i=0; igray); } pack_natural_or_gray(bits, &nbit, spare, 1, c2->gray); assert(nbit == (unsigned)codec2_bits_per_frame(c2)); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_decode_700b AUTHOR......: David Rowe DATE CREATED: August 2015 Decodes frames of 28 bits into 320 samples (40ms) of speech. \*---------------------------------------------------------------------------*/ void codec2_decode_700b(struct CODEC2 *c2, short speech[], const unsigned char * bits) { MODEL model[4]; int indexes[3]; float mel[LPC_ORD_LOW]; float lsps[4][LPC_ORD_LOW]; int Wo_index, e_index; float e[4]; float snr, f_; float ak[4][LPC_ORD_LOW+1]; int i,j; unsigned int nbit = 0; float weight; COMP Aw[FFT_ENC]; assert(c2 != NULL); /* only need to zero these out due to (unused) snr calculation */ for(i=0; i<4; i++) for(j=1; j<=MAX_AMP; j++) model[i].A[j] = 0.0; /* unpack bits from channel ------------------------------------*/ model[3].voiced = unpack(bits, &nbit, 1); model[0].voiced = model[1].voiced = model[2].voiced = model[3].voiced; Wo_index = unpack_natural_or_gray(bits, &nbit, 5, c2->gray); model[3].Wo = decode_log_Wo(&c2->c2const, Wo_index, 5); model[3].L = PI/model[3].Wo; e_index = unpack_natural_or_gray(bits, &nbit, 3, c2->gray); e[3] = decode_energy(e_index, 3); for(i=0; i<3; i++) { indexes[i] = unpack_natural_or_gray(bits, &nbit, lspmelvq_cb_bits(i), c2->gray); } lspmelvq_decode(indexes, mel, LPC_ORD_LOW); #define MEL_ROUND 10 for(i=1; iprev_lsps_dec, &lsps[3][0], weight, LPC_ORD_LOW); interp_Wo2(&model[i], &c2->prev_model_dec, &model[3], weight, c2->c2const.Wo_min); e[i] = interp_energy2(c2->prev_e_dec, e[3],weight); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD_LOW); aks_to_M2(c2->fftr_fwd_cfg, &ak[i][0], LPC_ORD_LOW, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma, Aw); apply_lpc_correction(&model[i]); synthesise_one_frame(c2, &speech[c2->n_samp*i], &model[i], Aw, 1.0); } #ifdef DUMP dump_lsp_(&lsps[3][0]); dump_ak_(&ak[3][0], LPC_ORD_LOW); dump_model(&model[3]); if (c2->softdec) dump_softdec(c2->softdec, nbit); #endif /* update memories for next frame ----------------------------*/ c2->prev_model_dec = model[3]; c2->prev_e_dec = e[3]; for(i=0; iprev_lsps_dec[i] = lsps[3][i]; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_700c AUTHOR......: David Rowe DATE CREATED: Jan 2017 Version c of 700 bit/s codec that uses newamp1 fixed rate VQ of amplitudes. Encodes 320 speech samples (40ms of speech) into 28 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm four times: frame 0: nothing frame 1: nothing frame 2: nothing frame 3: 18 bit 2 stage VQ (9 bits/stage), 4 bits energy, 6 bit scalar Wo/voicing. No spare bits. Voicing is encoded using the 0 index of the Wo quantiser. The bit allocation is: Parameter frames 1-3 frame 4 Total ----------------------------------------------------------- Harmonic magnitudes (rate k VQ) 0 18 18 Energy 0 4 4 log Wo/voicing 0 6 6 TOTAL 0 28 28 \*---------------------------------------------------------------------------*/ void codec2_encode_700c(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; int indexes[4], i, M=4; unsigned int nbit = 0; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); for(i=0; in_samp]); } int K = 20; float rate_K_vec[K], mean; float rate_K_vec_no_mean[K], rate_K_vec_no_mean_[K]; newamp1_model_to_indexes(&c2->c2const, indexes, &model, rate_K_vec, c2->rate_K_sample_freqs_kHz, K, &mean, rate_K_vec_no_mean, rate_K_vec_no_mean_, &c2->se, c2->eq, c2->eq_en); c2->nse += K; #ifndef CORTEX_M4 /* dump features for deep learning experiments */ if (c2->fmlfeat != NULL) { fwrite(&mean, 1, sizeof(float), c2->fmlfeat); fwrite(rate_K_vec_no_mean, K, sizeof(float), c2->fmlfeat); fwrite(rate_K_vec_no_mean_, K, sizeof(float), c2->fmlfeat); } #endif pack_natural_or_gray(bits, &nbit, indexes[0], 9, 0); pack_natural_or_gray(bits, &nbit, indexes[1], 9, 0); pack_natural_or_gray(bits, &nbit, indexes[2], 4, 0); pack_natural_or_gray(bits, &nbit, indexes[3], 6, 0); assert(nbit == (unsigned)codec2_bits_per_frame(c2)); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_decode_700c AUTHOR......: David Rowe DATE CREATED: August 2015 Decodes frames of 28 bits into 320 samples (40ms) of speech. \*---------------------------------------------------------------------------*/ void codec2_decode_700c(struct CODEC2 *c2, short speech[], const unsigned char * bits) { MODEL model[4]; int indexes[4]; int i; unsigned int nbit = 0; assert(c2 != NULL); /* unpack bits from channel ------------------------------------*/ indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[2] = unpack_natural_or_gray(bits, &nbit, 4, 0); indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0); int M = 4; COMP HH[M][MAX_AMP+1]; float interpolated_surface_[M][NEWAMP1_K]; newamp1_indexes_to_model(&c2->c2const, model, (COMP*)HH, (float*)interpolated_surface_, c2->prev_rate_K_vec_, &c2->Wo_left, &c2->voicing_left, c2->rate_K_sample_freqs_kHz, NEWAMP1_K, c2->phase_fft_fwd_cfg, c2->phase_fft_inv_cfg, indexes, c2->user_rate_K_vec_no_mean_, c2->post_filter_en); for(i=0; in_samp*i], &model[i], &HH[i][0], 1.5); } } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_energy_700c AUTHOR......: Jeroen Vreeken DATE CREATED: Jan 2017 Decodes energy value from encoded bits. \*---------------------------------------------------------------------------*/ float codec2_energy_700c(struct CODEC2 *c2, const unsigned char * bits) { int indexes[4]; unsigned int nbit = 0; assert(c2 != NULL); /* unpack bits from channel ------------------------------------*/ indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[2] = unpack_natural_or_gray(bits, &nbit, 4, 0); indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0); float mean = newamp1_energy_cb[0].cb[indexes[2]]; mean -= 10; if (indexes[3] == 0) mean -= 10; return POW10F(mean/10.0); } float codec2_energy_450(struct CODEC2 *c2, const unsigned char * bits) { int indexes[4]; unsigned int nbit = 0; assert(c2 != NULL); /* unpack bits from channel ------------------------------------*/ indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0); //indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[2] = unpack_natural_or_gray(bits, &nbit, 3, 0); indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0); float mean = newamp2_energy_cb[0].cb[indexes[2]]; mean -= 10; if (indexes[3] == 0) mean -= 10; return POW10F(mean/10.0); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_get_energy() AUTHOR......: Jeroen Vreeken DATE CREATED: 08/03/2016 Extract energy value from an encoded frame. \*---------------------------------------------------------------------------*/ float codec2_get_energy(struct CODEC2 *c2, const unsigned char *bits) { assert(c2 != NULL); assert( ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode)) || ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) ); MODEL model; float xq_dec[2] = {}; int e_index, WoE_index; float e; unsigned int nbit; if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_3200, c2->mode)) { nbit = 1 + 1 + WO_BITS; e_index = unpack(bits, &nbit, E_BITS); e = decode_energy(e_index, E_BITS); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_2400, c2->mode)) { nbit = 1 + 1; WoE_index = unpack(bits, &nbit, WO_E_BITS); decode_WoE(&c2->c2const, &model, &e, xq_dec, WoE_index); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1600, c2->mode)) { nbit = 1 + 1 + WO_BITS; e_index = unpack(bits, &nbit, E_BITS); e = decode_energy(e_index, E_BITS); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1400, c2->mode)) { nbit = 1 + 1; WoE_index = unpack(bits, &nbit, WO_E_BITS); decode_WoE(&c2->c2const, &model, &e, xq_dec, WoE_index); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1300, c2->mode)) { nbit = 1 + 1 + 1 + 1 + WO_BITS; e_index = unpack_natural_or_gray(bits, &nbit, E_BITS, c2->gray); e = decode_energy(e_index, E_BITS); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_1200, c2->mode)) { nbit = 1 + 1; WoE_index = unpack(bits, &nbit, WO_E_BITS); decode_WoE(&c2->c2const, &model, &e, xq_dec, WoE_index); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700, c2->mode)) { nbit = 1 + 5; e_index = unpack_natural_or_gray(bits, &nbit, 3, c2->gray); e = decode_energy(e_index, 3); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700B, c2->mode)) { nbit = 1 + 5; e_index = unpack_natural_or_gray(bits, &nbit, 3, c2->gray); e = decode_energy(e_index, 3); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode)) { e = codec2_energy_700c(c2, bits); } if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode)) { e = codec2_energy_450(c2, bits); } return e; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_encode_450 AUTHOR......: Thomas Kurin and Stefan Erhardt INSTITUTE...: Institute for Electronics Engineering, University of Erlangen-Nuremberg DATE CREATED: July 2018 450 bit/s codec that uses newamp2 fixed rate VQ of amplitudes. Encodes 320 speech samples (40ms of speech) into 28 bits. The codec2 algorithm actually operates internally on 10ms (80 sample) frames, so we run the encoding algorithm four times: frame 0: nothing frame 1: nothing frame 2: nothing frame 3: 9 bit 1 stage VQ, 3 bits energy, 6 bit scalar Wo/voicing/plosive. No spare bits. If a plosive is detected the frame at the energy-step is encoded. Voicing is encoded using the 000000 index of the Wo quantiser. Plosive is encoded using the 111111 index of the Wo quantiser. The bit allocation is: Parameter frames 1-3 frame 4 Total ----------------------------------------------------------- Harmonic magnitudes (rate k VQ) 0 9 9 Energy 0 3 3 log Wo/voicing/plosive 0 6 6 TOTAL 0 18 18 \*---------------------------------------------------------------------------*/ void codec2_encode_450(struct CODEC2 *c2, unsigned char * bits, short speech[]) { MODEL model; int indexes[4], i,h, M=4; unsigned int nbit = 0; int plosiv = 0; float energydelta[M]; int spectralCounter; assert(c2 != NULL); memset(bits, '\0', ((codec2_bits_per_frame(c2) + 7) / 8)); for(i=0; in_samp]); energydelta[i] = 0; spectralCounter = 0; for(h = 0;h<(model.L);h++){ //only detect above 300 Hz if(h*model.Wo*(c2->c2const.Fs/2000.0)/M_PI > 0.3){ energydelta[i] = energydelta[i] + 20.0*log10(model.A[10]+1E-16); spectralCounter = spectralCounter+1; } } energydelta[i] = energydelta[i] / spectralCounter ; } //Constants for plosive Detection tdB = threshold; minPwr = from below this level plosives have to rise float tdB = 15; //not fixed can be changed float minPwr = 15; //not fixed can be changed if((c2->energy_prev)((c2->energy_prev)+tdB)){ plosiv = 1; } if(energydelta[0](energydelta[0]+tdB)){ plosiv = 2; } if(energydelta[1](energydelta[1]+tdB)){ plosiv = 3; } if(energydelta[2](energydelta[2]+tdB)){ plosiv = 4; } if(plosiv != 0 && plosiv != 4){ analyse_one_frame(c2, &model, &speech[(plosiv-1)*c2->n_samp]); } c2->energy_prev = energydelta[3]; int K = 29; float rate_K_vec[K], mean; float rate_K_vec_no_mean[K], rate_K_vec_no_mean_[K]; if(plosiv > 0){ plosiv = 1; } newamp2_model_to_indexes(&c2->c2const, indexes, &model, rate_K_vec, c2->n2_rate_K_sample_freqs_kHz, K, &mean, rate_K_vec_no_mean, rate_K_vec_no_mean_, plosiv); pack_natural_or_gray(bits, &nbit, indexes[0], 9, 0); //pack_natural_or_gray(bits, &nbit, indexes[1], 9, 0); pack_natural_or_gray(bits, &nbit, indexes[2], 3, 0); pack_natural_or_gray(bits, &nbit, indexes[3], 6, 0); assert(nbit == (unsigned)codec2_bits_per_frame(c2)); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_decode_450 AUTHOR......: Thomas Kurin and Stefan Erhardt INSTITUTE...: Institute for Electronics Engineering, University of Erlangen-Nuremberg DATE CREATED: July 2018 \*---------------------------------------------------------------------------*/ void codec2_decode_450(struct CODEC2 *c2, short speech[], const unsigned char * bits) { MODEL model[4]; int indexes[4]; int i; unsigned int nbit = 0; assert(c2 != NULL); /* unpack bits from channel ------------------------------------*/ indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0); //indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[2] = unpack_natural_or_gray(bits, &nbit, 3, 0); indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0); int M = 4; COMP HH[M][MAX_AMP+1]; float interpolated_surface_[M][NEWAMP2_K]; int pwbFlag = 0; newamp2_indexes_to_model(&c2->c2const, model, (COMP*)HH, (float*)interpolated_surface_, c2->n2_prev_rate_K_vec_, &c2->Wo_left, &c2->voicing_left, c2->n2_rate_K_sample_freqs_kHz, NEWAMP2_K, c2->phase_fft_fwd_cfg, c2->phase_fft_inv_cfg, indexes, 1.5, pwbFlag); for(i=0; in_samp*i], &model[i], &HH[i][0], 1.5); } } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_decode_450pwb AUTHOR......: Thomas Kurin and Stefan Erhardt INSTITUTE...: Institute for Electronics Engineering, University of Erlangen-Nuremberg DATE CREATED: July 2018 Decodes the 450 codec data in pseudo wideband at 16kHz samplerate. \*---------------------------------------------------------------------------*/ void codec2_decode_450pwb(struct CODEC2 *c2, short speech[], const unsigned char * bits) { MODEL model[4]; int indexes[4]; int i; unsigned int nbit = 0; assert(c2 != NULL); /* unpack bits from channel ------------------------------------*/ indexes[0] = unpack_natural_or_gray(bits, &nbit, 9, 0); //indexes[1] = unpack_natural_or_gray(bits, &nbit, 9, 0); indexes[2] = unpack_natural_or_gray(bits, &nbit, 3, 0); indexes[3] = unpack_natural_or_gray(bits, &nbit, 6, 0); int M = 4; COMP HH[M][MAX_AMP+1]; float interpolated_surface_[M][NEWAMP2_16K_K]; int pwbFlag = 1; newamp2_indexes_to_model(&c2->c2const, model, (COMP*)HH, (float*)interpolated_surface_, c2->n2_pwb_prev_rate_K_vec_, &c2->Wo_left, &c2->voicing_left, c2->n2_pwb_rate_K_sample_freqs_kHz, NEWAMP2_16K_K, c2->phase_fft_fwd_cfg, c2->phase_fft_inv_cfg, indexes, 1.5, pwbFlag); for(i=0; in_samp*i], &model[i], &HH[i][0], 1.5); } } /*---------------------------------------------------------------------------* \ FUNCTION....: synthesise_one_frame() AUTHOR......: David Rowe DATE CREATED: 23/8/2010 Synthesise 80 speech samples (10ms) from model parameters. \*---------------------------------------------------------------------------*/ void synthesise_one_frame(struct CODEC2 *c2, short speech[], MODEL *model, COMP Aw[], float gain) { int i; //PROFILE_VAR(phase_start, pf_start, synth_start); //#ifdef DUMP //dump_quantised_model(model); //#endif //PROFILE_SAMPLE(phase_start); if ( CODEC2_MODE_ACTIVE(CODEC2_MODE_700C, c2->mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_450, c2->mode) || CODEC2_MODE_ACTIVE(CODEC2_MODE_450PWB, c2->mode) ) { /* newamp1/2, we've already worked out rate L phase */ COMP *H = Aw; phase_synth_zero_order(c2->n_samp, model, &c2->ex_phase, H); } else { /* LPC based phase synthesis */ COMP H[MAX_AMP+1]; sample_phase(model, H, Aw); phase_synth_zero_order(c2->n_samp, model, &c2->ex_phase, H); } //PROFILE_SAMPLE_AND_LOG(pf_start, phase_start, " phase_synth"); postfilter(model, &c2->bg_est); //PROFILE_SAMPLE_AND_LOG(synth_start, pf_start, " postfilter"); synthesise(c2->n_samp, c2->fftr_inv_cfg, c2->Sn_, model, c2->Pn, 1); for(i=0; in_samp; i++) { c2->Sn_[i] *= gain; } //PROFILE_SAMPLE_AND_LOG2(synth_start, " synth"); ear_protection(c2->Sn_, c2->n_samp); for(i=0; in_samp; i++) { if (c2->Sn_[i] > 32767.0) speech[i] = 32767; else if (c2->Sn_[i] < -32767.0) speech[i] = -32767; else speech[i] = c2->Sn_[i]; } } /*---------------------------------------------------------------------------*\ FUNCTION....: analyse_one_frame() AUTHOR......: David Rowe DATE CREATED: 23/8/2010 Extract sinusoidal model parameters from 80 speech samples (10ms of speech). \*---------------------------------------------------------------------------*/ void analyse_one_frame(struct CODEC2 *c2, MODEL *model, short speech[]) { COMP Sw[FFT_ENC]; float pitch; int i; //PROFILE_VAR(dft_start, nlp_start, model_start, two_stage, estamps); int n_samp = c2->n_samp; int m_pitch = c2->m_pitch; /* Read input speech */ for(i=0; iSn[i] = c2->Sn[i+n_samp]; for(i=0; iSn[i+m_pitch-n_samp] = speech[i]; //PROFILE_SAMPLE(dft_start); dft_speech(&c2->c2const, c2->fft_fwd_cfg, Sw, c2->Sn, c2->w); //PROFILE_SAMPLE_AND_LOG(nlp_start, dft_start, " dft_speech"); /* Estimate pitch */ nlp(c2->nlp, c2->Sn, n_samp, &pitch, Sw, c2->W, &c2->prev_f0_enc); //PROFILE_SAMPLE_AND_LOG(model_start, nlp_start, " nlp"); model->Wo = TWO_PI/pitch; model->L = PI/model->Wo; /* estimate model parameters */ two_stage_pitch_refinement(&c2->c2const, model, Sw); //PROFILE_SAMPLE_AND_LOG(two_stage, model_start, " two_stage"); estimate_amplitudes(model, Sw, c2->W, 0); //PROFILE_SAMPLE_AND_LOG(estamps, two_stage, " est_amps"); est_voicing_mbe(&c2->c2const, model, Sw, c2->W); //PROFILE_SAMPLE_AND_LOG2(estamps, " est_voicing"); #ifdef DUMP dump_model(model); #endif } /*---------------------------------------------------------------------------*\ FUNCTION....: ear_protection() AUTHOR......: David Rowe DATE CREATED: Nov 7 2012 Limits output level to protect ears when there are bit errors or the input is overdriven. This doesn't correct or mask bit errors, just reduces the worst of their damage. \*---------------------------------------------------------------------------*/ static void ear_protection(float in_out[], int n) { float max_sample, over, gain; int i; /* find maximum sample in frame */ max_sample = 0.0; for(i=0; i max_sample) max_sample = in_out[i]; /* determine how far above set point */ over = max_sample/30000.0; /* If we are x dB over set point we reduce level by 2x dB, this attenuates major excursions in amplitude (likely to be caused by bit errors) more than smaller ones */ if (over > 1.0) { gain = 1.0/(over*over); //fprintf(stderr, "gain: %f\n", gain); for(i=0; i= 0.0) && (beta <= 1.0)); assert((gamma >= 0.0) && (gamma <= 1.0)); c2->lpc_pf = enable; c2->bass_boost = bass_boost; c2->beta = beta; c2->gamma = gamma; } /* Allows optional stealing of one of the voicing bits for use as a spare bit, only 1300 & 1400 & 1600 bit/s supported for now. Experimental method of sending voice/data frames for FreeDV. */ int codec2_get_spare_bit_index(struct CODEC2 *c2) { assert(c2 != NULL); switch(c2->mode) { case CODEC2_MODE_1300: return 2; // bit 2 (3th bit) is v2 (third voicing bit) break; case CODEC2_MODE_1400: return 10; // bit 10 (11th bit) is v2 (third voicing bit) break; case CODEC2_MODE_1600: return 15; // bit 15 (16th bit) is v2 (third voicing bit) break; case CODEC2_MODE_700: return 26; // bits 26 and 27 are spare break; case CODEC2_MODE_700B: return 27; // bit 27 is spare break; } return -1; } /* Reconstructs the spare voicing bit. Note works on unpacked bits for convenience. */ int codec2_rebuild_spare_bit(struct CODEC2 *c2, int unpacked_bits[]) { int v1,v3; assert(c2 != NULL); v1 = unpacked_bits[1]; switch(c2->mode) { case CODEC2_MODE_1300: v3 = unpacked_bits[1+1+1]; /* if either adjacent frame is voiced, make this one voiced */ unpacked_bits[2] = (v1 || v3); return 0; break; case CODEC2_MODE_1400: v3 = unpacked_bits[1+1+8+1]; /* if either adjacent frame is voiced, make this one voiced */ unpacked_bits[10] = (v1 || v3); return 0; break; case CODEC2_MODE_1600: v3 = unpacked_bits[1+1+8+5+1]; /* if either adjacent frame is voiced, make this one voiced */ unpacked_bits[15] = (v1 || v3); return 0; break; } return -1; } void codec2_set_natural_or_gray(struct CODEC2 *c2, int gray) { assert(c2 != NULL); c2->gray = gray; } void codec2_set_softdec(struct CODEC2 *c2, float *softdec) { assert(c2 != NULL); c2->softdec = softdec; } void codec2_open_mlfeat(struct CODEC2 *codec2_state, char *filename) { if ((codec2_state->fmlfeat = fopen(filename, "wb")) == NULL) { fprintf(stderr, "error opening machine learning feature file: %s\n", filename); exit(1); } } #ifndef __EMBEDDED__ void codec2_load_codebook(struct CODEC2 *codec2_state, int num, char *filename) { FILE *f; if ((f = fopen(filename, "rb")) == NULL) { fprintf(stderr, "error opening codebook file: %s\n", filename); exit(1); } //fprintf(stderr, "reading newamp1vq_cb[%d] k=%d m=%d\n", num, newamp1vq_cb[num].k, newamp1vq_cb[num].m); float tmp[newamp1vq_cb[num].k*newamp1vq_cb[num].m]; int nread = fread(tmp, sizeof(float), newamp1vq_cb[num].k*newamp1vq_cb[num].m, f); float *p = (float*)newamp1vq_cb[num].cb; for(int i=0; inse) return codec2_state->se/codec2_state->nse; else return 0; } float *codec2_enable_user_ratek(struct CODEC2 *codec2_state, int *K) { codec2_state->user_rate_K_vec_no_mean_ = (float*)malloc(sizeof(float)*NEWAMP1_K); *K = NEWAMP1_K; return codec2_state->user_rate_K_vec_no_mean_; } void codec2_700c_post_filter(struct CODEC2 *codec2_state, int en) { codec2_state->post_filter_en = en; } void codec2_700c_eq(struct CODEC2 *codec2_state, int en) { codec2_state->eq_en = en; codec2_state->se = 0.0; codec2_state->nse = 0; }