/*---------------------------------------------------------------------------*\ 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 "defines.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 "codec2_internal.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, float ak[]); 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_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_1200(struct CODEC2 *c2, unsigned char * bits, short speech[]); void codec2_decode_1200(struct CODEC2 *c2, short speech[], const unsigned char * bits); 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. \*---------------------------------------------------------------------------*/ struct CODEC2 * CODEC2_WIN32SUPPORT codec2_create(int mode) { struct CODEC2 *c2; int i,l; c2 = (struct CODEC2*)malloc(sizeof(struct CODEC2)); if (c2 == NULL) return NULL; assert( (mode == CODEC2_MODE_3200) || (mode == CODEC2_MODE_2400) || (mode == CODEC2_MODE_1400) || (mode == CODEC2_MODE_1200) ); c2->mode = mode; for(i=0; iSn[i] = 1.0; c2->hpf_states[0] = c2->hpf_states[1] = 0.0; for(i=0; i<2*N; i++) c2->Sn_[i] = 0; c2->fft_fwd_cfg = kiss_fft_alloc(FFT_ENC, 0, NULL, NULL); make_analysis_window(c2->fft_fwd_cfg, c2->w,c2->W); make_synthesis_window(c2->Pn); c2->fft_inv_cfg = kiss_fft_alloc(FFT_DEC, 1, NULL, NULL); quantise_init(); c2->prev_Wo_enc = 0.0; 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/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(); if (c2->nlp == NULL) { free (c2); return NULL; } 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; return c2; } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_destroy AUTHOR......: David Rowe DATE CREATED: 21/8/2010 Destroy an instance of the codec. \*---------------------------------------------------------------------------*/ void CODEC2_WIN32SUPPORT codec2_destroy(struct CODEC2 *c2) { assert(c2 != NULL); nlp_destroy(c2->nlp); KISS_FFT_FREE(c2->fft_fwd_cfg); KISS_FFT_FREE(c2->fft_inv_cfg); free(c2); } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_bits_per_frame AUTHOR......: David Rowe DATE CREATED: Nov 14 2011 Returns the number of bits per frame. \*---------------------------------------------------------------------------*/ int CODEC2_WIN32SUPPORT codec2_bits_per_frame(struct CODEC2 *c2) { if (c2->mode == CODEC2_MODE_3200) return 64; if (c2->mode == CODEC2_MODE_2400) return 48; if (c2->mode == CODEC2_MODE_1400) return 56; if (c2->mode == CODEC2_MODE_1200) return 48; return 0; /* shouldn't get here */ } /*---------------------------------------------------------------------------*\ FUNCTION....: codec2_samples_per_frame AUTHOR......: David Rowe DATE CREATED: Nov 14 2011 Returns the number of bits per frame. \*---------------------------------------------------------------------------*/ int CODEC2_WIN32SUPPORT codec2_samples_per_frame(struct CODEC2 *c2) { if (c2->mode == CODEC2_MODE_3200) return 160; if (c2->mode == CODEC2_MODE_2400) return 160; if (c2->mode == CODEC2_MODE_1400) return 320; if (c2->mode == CODEC2_MODE_1200) return 320; return 0; /* shouldnt get here */ } void CODEC2_WIN32SUPPORT codec2_encode(struct CODEC2 *c2, unsigned char *bits, short speech[]) { assert(c2 != NULL); assert( (c2->mode == CODEC2_MODE_3200) || (c2->mode == CODEC2_MODE_2400) || (c2->mode == CODEC2_MODE_1400) || (c2->mode == CODEC2_MODE_1200) ); if (c2->mode == CODEC2_MODE_3200) codec2_encode_3200(c2, bits, speech); if (c2->mode == CODEC2_MODE_2400) codec2_encode_2400(c2, bits, speech); if (c2->mode == CODEC2_MODE_1400) codec2_encode_1400(c2, bits, speech); if (c2->mode == CODEC2_MODE_1200) codec2_encode_1200(c2, bits, speech); } void CODEC2_WIN32SUPPORT codec2_decode(struct CODEC2 *c2, short speech[], const unsigned char *bits) { assert(c2 != NULL); assert( (c2->mode == CODEC2_MODE_3200) || (c2->mode == CODEC2_MODE_2400) || (c2->mode == CODEC2_MODE_1400) || (c2->mode == CODEC2_MODE_1200) ); if (c2->mode == CODEC2_MODE_3200) codec2_decode_3200(c2, speech, bits); if (c2->mode == CODEC2_MODE_2400) codec2_decode_2400(c2, speech, bits); if (c2->mode == CODEC2_MODE_1400) codec2_decode_1400(c2, speech, bits); if (c2->mode == CODEC2_MODE_1200) codec2_decode_1200(c2, speech, bits); } /*---------------------------------------------------------------------------*\ 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[N]); pack(bits, &nbit, model.voiced, 1); Wo_index = encode_Wo(model.Wo); pack(bits, &nbit, Wo_index, WO_BITS); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, LPC_ORD); e_index = encode_energy(e); pack(bits, &nbit, e_index, E_BITS); encode_lspds_scalar(lspd_indexes, lsps, LPC_ORD); for(i=0; iprev_model_dec, &model[1]); 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); for(i=0; i<2; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fft_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma); apply_lpc_correction(&model[i]); } /* synthesise ------------------------------------------------*/ for(i=0; i<2; i++) synthesise_one_frame(c2, &speech[N*i], &model[i], &ak[i][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[N]); pack(bits, &nbit, model.voiced, 1); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, 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; ixq_dec, WoE_index); for(i=0; iprev_model_dec, &model[1]); 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); for(i=0; i<2; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fft_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma); apply_lpc_correction(&model[i]); } /* synthesise ------------------------------------------------*/ for(i=0; i<2; i++) synthesise_one_frame(c2, &speech[N*i], &model[i], &ak[i][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_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[N]); 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, 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*N]); pack(bits, &nbit, model.voiced, 1); /* frame 4: - voicing, joint Wo & E, scalar LSPs ------------------*/ analyse_one_frame(c2, &model, &speech[3*N]); pack(bits, &nbit, model.voiced, 1); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, 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; ixq_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(&model[3], &e[3], c2->xq_dec, WoE_index); for(i=0; iprev_model_dec, &model[1]); e[0] = interp_energy(c2->prev_e_dec, e[1]); interp_Wo(&model[2], &model[1], &model[3]); 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); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fft_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma); apply_lpc_correction(&model[i]); } /* synthesise ------------------------------------------------*/ for(i=0; i<4; i++) synthesise_one_frame(c2, &speech[N*i], &model[i], &ak[i][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_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[N]); 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, 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*N]); pack(bits, &nbit, model.voiced, 1); /* frame 4: - voicing, joint Wo & E, scalar LSPs ------------------*/ analyse_one_frame(c2, &model, &speech[3*N]); pack(bits, &nbit, model.voiced, 1); e = speech_to_uq_lsps(lsps, ak, c2->Sn, c2->w, 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; ixq_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(&model[3], &e[3], c2->xq_dec, WoE_index); for(i=0; iprev_model_dec, &model[1]); e[0] = interp_energy(c2->prev_e_dec, e[1]); interp_Wo(&model[2], &model[1], &model[3]); 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); } for(i=0; i<4; i++) { lsp_to_lpc(&lsps[i][0], &ak[i][0], LPC_ORD); aks_to_M2(c2->fft_fwd_cfg, &ak[i][0], LPC_ORD, &model[i], e[i], &snr, 0, 0, c2->lpc_pf, c2->bass_boost, c2->beta, c2->gamma); apply_lpc_correction(&model[i]); } /* synthesise ------------------------------------------------*/ for(i=0; i<4; i++) synthesise_one_frame(c2, &speech[N*i], &model[i], &ak[i][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....: 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, float ak[]) { int i; phase_synth_zero_order(c2->fft_fwd_cfg, model, ak, &c2->ex_phase, LPC_ORD); postfilter(model, &c2->bg_est); synthesise(c2->fft_inv_cfg, c2->Sn_, model, c2->Pn, 1); ear_protection(c2->Sn_, N); for(i=0; iSn_[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]; COMP Sw_[FFT_ENC]; COMP Ew[FFT_ENC]; float pitch, snr; int i; /* Read input speech */ for(i=0; iSn[i] = c2->Sn[i+N]; for(i=0; iSn[i+M-N] = speech[i]; dft_speech(c2->fft_fwd_cfg, Sw, c2->Sn, c2->w); /* Estimate pitch */ nlp(c2->nlp,c2->Sn,N,M,P_MIN,P_MAX,&pitch,Sw, c2->W, &c2->prev_Wo_enc); model->Wo = TWO_PI/pitch; model->L = PI/model->Wo; /* estimate model parameters */ two_stage_pitch_refinement(model, Sw); estimate_amplitudes(model, Sw, c2->W); snr = est_voicing_mbe(model, Sw, c2->W, Sw_, Ew, c2->prev_Wo_enc); //fprintf(stderr,"snr %3.2f v: %d Wo: %f prev_Wo: %f\n", // snr, model->voiced, model->Wo, c2->prev_Wo_enc); c2->prev_Wo_enc = model->Wo; } /*---------------------------------------------------------------------------*\ 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 erros, just reduces the worst of their damage. \*---------------------------------------------------------------------------*/ 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 1400 bit/s supported for now. Experimental method of sending voice/data frames for FreeDV. */ int CODEC2_WIN32SUPPORT codec2_get_spare_bit_index(struct CODEC2 *c2) { assert(c2 != NULL); if (c2->mode != CODEC2_MODE_1400) return -1; return 10; // bit 10 (11th bit) is v2 (third voicing bit) } /* Reconstructs the spare voicing bit. Note works on unpacked bits for convenience. */ int CODEC2_WIN32SUPPORT codec2_rebuild_spare_bit(struct CODEC2 *c2, int unpacked_bits[]) { int v0,v1,v3; assert(c2 != NULL); if (c2->mode != CODEC2_MODE_1400) return -1; v0 = unpacked_bits[0]; v1 = unpacked_bits[1]; v3 = unpacked_bits[11]; /* if either adjacent frame is voiced, make this one voiced */ unpacked_bits[10] = (v1 || v3); return 0; }