FreeCalypso > hg > gsm-codec-lib
view libgsmefr/cod_12k2.c @ 85:db8e0b69a6bb
libgsmefr: pred_lt6.c compiles
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Sat, 26 Nov 2022 08:59:34 +0000 |
parents | d285877fc4cc |
children | d9ad0f5121e8 |
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/*************************************************************************** * * FILE NAME: cod_12k2.c * * FUNCTIONS DEFINED IN THIS FILE: * Coder_12k2 and Init_Coder_12k2 * * * Init_Coder_12k2(void): * Initialization of variables for the coder section. * * Coder_12k2(Word16 ana[], Word16 synth[]): * Speech encoder routine operating on a frame basis. * ***************************************************************************/ #include "gsm_efr.h" #include "typedef.h" #include "namespace.h" #include "basic_op.h" #include "sig_proc.h" #include "memops.h" #include "no_count.h" #include "codec.h" #include "cnst.h" #include "enc_state.h" #include "window2.tab" #include "vad.h" #include "dtx.h" /*-----------------------------------------------------------* * Coder constant parameters (defined in "cnst.h") * *-----------------------------------------------------------* * L_WINDOW : LPC analysis window size * * L_FRAME : Frame size * * L_FRAME_BY2 : Half the frame size * * L_SUBFR : Sub-frame size * * M : LPC order * * MP1 : LPC order+1 * * L_TOTAL : Total size of speech buffer * * PIT_MIN : Minimum pitch lag * * PIT_MAX : Maximum pitch lag * * L_INTERPOL : Length of filter for interpolation * *-----------------------------------------------------------*/ /* Spectral expansion factors */ static const Word16 F_gamma1[M] = { 29491, 26542, 23888, 21499, 19349, 17414, 15672, 14105, 12694, 11425 }; static const Word16 F_gamma2[M] = { 19661, 11797, 7078, 4247, 2548, 1529, 917, 550, 330, 198 }; /*************************************************************************** * FUNCTION: Init_Coder_12k2 * * PURPOSE: Initialization of variables for the coder section. * * DESCRIPTION: * - initilize pointers to speech buffer * - initialize static pointers * - set static vectors to zero * ***************************************************************************/ void Init_Coder_12k2 (struct EFR_encoder_state *st) { /* Static vectors to zero */ Set_zero (st->old_speech, L_TOTAL); Set_zero (st->old_exc, PIT_MAX + L_INTERPOL); Set_zero (st->old_wsp, PIT_MAX); Set_zero (st->mem_syn, M); Set_zero (st->mem_w, M); Set_zero (st->mem_w0, M); Set_zero (st->mem_err, M); Set_zero (st->ai_zero + MP1, L_SUBFR); Set_zero (st->hvec, L_SUBFR); /* set to zero "h1[-L_SUBFR..-1]" */ /* Initialize lsp_old [] */ st->lsp_old[0] = 30000; st->lsp_old[1] = 26000; st->lsp_old[2] = 21000; st->lsp_old[3] = 15000; st->lsp_old[4] = 8000; st->lsp_old[5] = 0; st->lsp_old[6] = -8000; st->lsp_old[7] = -15000; st->lsp_old[8] = -21000; st->lsp_old[9] = -26000; /* Initialize lsp_old_q[] */ Copy (st->lsp_old, st->lsp_old_q, M); return; } /*************************************************************************** * FUNCTION: Coder_12k2 * * PURPOSE: Principle encoder routine. * * DESCRIPTION: This function is called every 20 ms speech frame, * operating on the newly read 160 speech samples. It performs the * principle encoding functions to produce the set of encoded parameters * which include the LSP, adaptive codebook, and fixed codebook * quantization indices (addresses and gains). * * INPUTS: * No input arguments are passed to this function. However, before * calling this function, 160 new speech data samples should be copied to * the vector new_speech[]. This is a global pointer which is declared in * this file (it points to the end of speech buffer minus 160). * * OUTPUTS: * * ana[]: vector of analysis parameters. * synth[]: Local synthesis speech (for debugging purposes) * ***************************************************************************/ void Coder_12k2 ( struct EFR_encoder_state *st, Word16 ana[], /* output : Analysis parameters */ Word16 synth[] /* output : Local synthesis */ ) { /* handy pointers that were static vars in the original code */ Word16 *speech = st->old_speech + L_TOTAL - L_FRAME; Word16 *p_window = st->old_speech + L_TOTAL - L_WINDOW; Word16 *wsp = st->old_wsp + PIT_MAX; Word16 *exc = st->old_exc + PIT_MAX + L_INTERPOL; Word16 *zero = st->ai_zero + MP1; Word16 *h1 = st->hvec + L_SUBFR; Word16 *error = st->mem_err + M; /* LPC coefficients */ Word16 r_l[MP1], r_h[MP1]; /* Autocorrelations lo and hi */ Word16 A_t[(MP1) * 4]; /* A(z) unquantized for the 4 subframes */ Word16 Aq_t[(MP1) * 4]; /* A(z) quantized for the 4 subframes */ Word16 Ap1[MP1]; /* A(z) with spectral expansion */ Word16 Ap2[MP1]; /* A(z) with spectral expansion */ Word16 *A, *Aq; /* Pointer on A_t and Aq_t */ Word16 lsp_new[M], lsp_new_q[M];/* LSPs at 4th subframe */ Word16 lsp_mid[M], lsp_mid_q[M];/* LSPs at 2nd subframe */ /* Other vectors */ Word16 xn[L_SUBFR]; /* Target vector for pitch search */ Word16 xn2[L_SUBFR]; /* Target vector for codebook search */ Word16 res2[L_SUBFR]; /* Long term prediction residual */ Word16 code[L_SUBFR]; /* Fixed codebook excitation */ Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */ Word16 y2[L_SUBFR]; /* Filtered fixed codebook excitation */ /* Scalars */ Word16 i, j, k, i_subfr; Word16 T_op, T0, T0_min, T0_max, T0_frac; Word16 gain_pit, gain_code, pit_flag, pit_sharp; Word16 temp; Word32 L_temp; Word16 scal_acf, VAD_flag, lags[2], rc[4]; /*----------------------------------------------------------------------* * - Perform LPC analysis: (twice per frame) * * * autocorrelation + lag windowing * * * Levinson-Durbin algorithm to find a[] * * * convert a[] to lsp[] * * * quantize and code the LSPs * * * find the interpolated LSPs and convert to a[] for all * * subframes (both quantized and unquantized) * *----------------------------------------------------------------------*/ /* LP analysis centered at 2nd subframe */ scal_acf = Autocorr (p_window, M, r_h, r_l, window_160_80); /* Autocorrelations */ Lag_window (M, r_h, r_l); /* Lag windowing */ Levinson (st, r_h, r_l, &A_t[MP1], rc); /* Levinson-Durbin */ Az_lsp (&A_t[MP1], lsp_mid, st->lsp_old); /* From A(z) to lsp */ /* LP analysis centered at 4th subframe */ /* Autocorrelations */ scal_acf = Autocorr (p_window, M, r_h, r_l, window_232_8); Lag_window (M, r_h, r_l); /* Lag windowing */ Levinson (st, r_h, r_l, &A_t[MP1 * 3], rc); /* Levinson-Durbin */ Az_lsp (&A_t[MP1 * 3], lsp_new, lsp_mid); /* From A(z) to lsp */ if (st->dtx_mode) { /* DTX enabled, make voice activity decision */ VAD_flag = vad_computation (st, r_h, r_l, scal_acf, rc, st->ptch); tx_dtx (st, VAD_flag); /* TX DTX handler */ } else { /* DTX disabled, active speech in every frame */ VAD_flag = 1; st->txdtx_ctrl = TX_VAD_FLAG | TX_SP_FLAG; } /* LSP quantization (lsp_mid[] and lsp_new[] jointly quantized) */ Q_plsf_5 (st, lsp_mid, lsp_new, lsp_mid_q, lsp_new_q, ana, st->txdtx_ctrl); ana += 5; /*--------------------------------------------------------------------* * Find interpolated LPC parameters in all subframes (both quantized * * and unquantized). * * The interpolated parameters are in array A_t[] of size (M+1)*4 * * and the quantized interpolated parameters are in array Aq_t[] * *--------------------------------------------------------------------*/ Int_lpc2 (st->lsp_old, lsp_mid, lsp_new, A_t); if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { Int_lpc (st->lsp_old_q, lsp_mid_q, lsp_new_q, Aq_t); /* update the LSPs for the next frame */ Copy (lsp_new, st->lsp_old, M); Copy (lsp_new_q, st->lsp_old_q, M); } else { /* Use unquantized LPC parameters in case of no speech activity */ for (i = 0; i < MP1; i++) { Aq_t[i] = A_t[i]; move16 (); Aq_t[i + MP1] = A_t[i + MP1]; move16 (); Aq_t[i + MP1 * 2] = A_t[i + MP1 * 2]; move16 (); Aq_t[i + MP1 * 3] = A_t[i + MP1 * 3]; move16 (); } /* update the LSPs for the next frame */ Copy (lsp_new, st->lsp_old, M); Copy (lsp_new_q, st->lsp_old_q, M); } /*----------------------------------------------------------------------* * - Find the weighted input speech wsp[] for the whole speech frame * * - Find the open-loop pitch delay for first 2 subframes * * - Set the range for searching closed-loop pitch in 1st subframe * * - Find the open-loop pitch delay for last 2 subframes * *----------------------------------------------------------------------*/ A = A_t; move16 (); for (i = 0; i < L_FRAME; i += L_SUBFR) { Weight_Ai (A, F_gamma1, Ap1); Weight_Ai (A, F_gamma2, Ap2); Residu (Ap1, &speech[i], &wsp[i], L_SUBFR); Syn_filt (Ap2, &wsp[i], &wsp[i], L_SUBFR, st->mem_w, 1); A += MP1; move16 (); } /* Find open loop pitch lag for first two subframes */ T_op = Pitch_ol (wsp, PIT_MIN, PIT_MAX, L_FRAME_BY2); move16 (); lags[0] = T_op; move16 (); if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { /* Range for closed loop pitch search in 1st subframe */ T0_min = sub (T_op, 3); if (T0_min < PIT_MIN) { T0_min = PIT_MIN; move16 (); } T0_max = add (T0_min, 6); if (T0_max > PIT_MAX) { T0_max = PIT_MAX; move16 (); T0_min = sub (T0_max, 6); } } /* Find open loop pitch lag for last two subframes */ T_op = Pitch_ol (&wsp[L_FRAME_BY2], PIT_MIN, PIT_MAX, L_FRAME_BY2); if (st->dtx_mode) { lags[1] = T_op; move16 (); periodicity_update (st, lags); } /*----------------------------------------------------------------------* * Loop for every subframe in the analysis frame * *----------------------------------------------------------------------* * To find the pitch and innovation parameters. The subframe size is * * L_SUBFR and the loop is repeated L_FRAME/L_SUBFR times. * * - find the weighted LPC coefficients * * - find the LPC residual signal res[] * * - compute the target signal for pitch search * * - compute impulse response of weighted synthesis filter (h1[]) * * - find the closed-loop pitch parameters * * - encode the pitch delay * * - update the impulse response h1[] by including pitch * * - find target vector for codebook search * * - codebook search * * - encode codebook address * * - VQ of pitch and codebook gains * * - find synthesis speech * * - update states of weighting filter * *----------------------------------------------------------------------*/ /* pointer to interpolated LPC parameters */ A = A_t; move16 (); /* pointer to interpolated quantized LPC parameters */ Aq = Aq_t; move16 (); for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR) { if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { /*---------------------------------------------------------------* * Find the weighted LPC coefficients for the weighting filter. * *---------------------------------------------------------------*/ Weight_Ai (A, F_gamma1, Ap1); Weight_Ai (A, F_gamma2, Ap2); /*---------------------------------------------------------------* * Compute impulse response, h1[], of weighted synthesis filter * *---------------------------------------------------------------*/ Copy (Ap1, st->ai_zero, M+1); Syn_filt (Aq, st->ai_zero, h1, L_SUBFR, zero, 0); Syn_filt (Ap2, h1, h1, L_SUBFR, zero, 0); } /*---------------------------------------------------------------* * Find the target vector for pitch search: * *---------------------------------------------------------------*/ Residu (Aq, &speech[i_subfr], res2, L_SUBFR); /* LPC residual */ if ((st->txdtx_ctrl & TX_SP_FLAG) == 0) { /* Compute comfort noise excitation gain based on LP residual energy */ st->CN_excitation_gain = compute_CN_excitation_gain (res2); } else { Copy (res2, &exc[i_subfr], L_SUBFR); Syn_filt (Aq, &exc[i_subfr], error, L_SUBFR, st->mem_err, 0); Residu (Ap1, error, xn, L_SUBFR); /* target signal xn[] */ Syn_filt (Ap2, xn, xn, L_SUBFR, st->mem_w0, 0); /*--------------------------------------------------------------* * Closed-loop fractional pitch search * *--------------------------------------------------------------*/ /* flag for first and 3th subframe */ pit_flag = i_subfr; move16 (); /* set t0_min and t0_max for 3th subf.*/ if (i_subfr == L_FRAME_BY2) { T0_min = sub (T_op, 3); if (T0_min < PIT_MIN) { T0_min = PIT_MIN; move16 (); } T0_max = add (T0_min, 6); if (T0_max > PIT_MAX) { T0_max = PIT_MAX; move16 (); T0_min = sub (T0_max, 6); } pit_flag = 0; move16 (); } T0 = Pitch_fr6 (&exc[i_subfr], xn, h1, L_SUBFR, T0_min, T0_max, pit_flag, &T0_frac); move16 (); *ana = Enc_lag6 (T0, &T0_frac, &T0_min, &T0_max, PIT_MIN, PIT_MAX, pit_flag); } ana++; /* Incrementation of ana is done here to work also when no speech activity is present */ if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { /*---------------------------------------------------------------* * - find unity gain pitch excitation (adaptive codebook entry) * * with fractional interpolation. * * - find filtered pitch exc. y1[]=exc[] convolved with h1[] * * - compute pitch gain and limit between 0 and 1.2 * * - update target vector for codebook search * * - find LTP residual. * *---------------------------------------------------------------*/ Pred_lt_6 (&exc[i_subfr], T0, T0_frac, L_SUBFR); Convolve (&exc[i_subfr], h1, y1, L_SUBFR); gain_pit = G_pitch (xn, y1, L_SUBFR); move16 (); *ana = q_gain_pitch (&gain_pit); move16 (); } else { gain_pit = 0; move16 (); } ana++; /* Incrementation of ana is done here to work also when no speech activity is present */ if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { /* xn2[i] = xn[i] - y1[i] * gain_pit */ /* res2[i] -= exc[i+i_subfr] * gain_pit */ for (i = 0; i < L_SUBFR; i++) { L_temp = L_mult (y1[i], gain_pit); L_temp = L_shl (L_temp, 3); xn2[i] = sub (xn[i], extract_h (L_temp)); move16 (); L_temp = L_mult (exc[i + i_subfr], gain_pit); L_temp = L_shl (L_temp, 3); res2[i] = sub (res2[i], extract_h (L_temp)); move16 (); } /*-------------------------------------------------------------* * - include pitch contribution into impulse resp. h1[] * *-------------------------------------------------------------*/ /* pit_sharp = gain_pit; */ /* if (pit_sharp > 1.0) pit_sharp = 1.0; */ pit_sharp = shl (gain_pit, 3); for (i = T0; i < L_SUBFR; i++) { temp = mult (h1[i - T0], pit_sharp); h1[i] = add (h1[i], temp); move16 (); } /*--------------------------------------------------------------* * - Innovative codebook search (find index and gain) * *--------------------------------------------------------------*/ code_10i40_35bits (xn2, res2, h1, code, y2, ana); } else { build_CN_code (code, &st->L_pn_seed_tx); } ana += 10; move16 (); if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { /*-------------------------------------------------------* * - Add the pitch contribution to code[]. * *-------------------------------------------------------*/ for (i = T0; i < L_SUBFR; i++) { temp = mult (code[i - T0], pit_sharp); code[i] = add (code[i], temp); move16 (); } /*------------------------------------------------------* * - Quantization of fixed codebook gain. * *------------------------------------------------------*/ gain_code = G_code (xn2, y2); move16 (); } *ana++ = q_gain_code (st, code, L_SUBFR, &gain_code, st->txdtx_ctrl, i_subfr); /*------------------------------------------------------* * - Find the total excitation * * - find synthesis speech corresponding to exc[] * * - update filter memories for finding the target * * vector in the next subframe * * (update mem_err[] and mem_w0[]) * *------------------------------------------------------*/ for (i = 0; i < L_SUBFR; i++) { /* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */ L_temp = L_mult (exc[i + i_subfr], gain_pit); L_temp = L_mac (L_temp, code[i], gain_code); L_temp = L_shl (L_temp, 3); exc[i + i_subfr] = round (L_temp); move16 (); } Syn_filt (Aq, &exc[i_subfr], &synth[i_subfr], L_SUBFR, st->mem_syn, 1); if ((st->txdtx_ctrl & TX_SP_FLAG) != 0) { for (i = L_SUBFR - M, j = 0; i < L_SUBFR; i++, j++) { st->mem_err[j] = sub (speech[i_subfr + i], synth[i_subfr + i]); temp = extract_h (L_shl (L_mult (y1[i], gain_pit), 3)); k = extract_h (L_shl (L_mult (y2[i], gain_code), 5)); st->mem_w0[j] = sub (xn[i], add (temp, k)); } } else { Set_zero (st->mem_err, M); Set_zero (st->mem_w0, M); } /* interpolated LPC parameters for next subframe */ A += MP1; move16 (); Aq += MP1; move16 (); } /*--------------------------------------------------* * Update signal for next frame. * * -> shift to the left by L_FRAME: * * speech[], wsp[] and exc[] * *--------------------------------------------------*/ Copy (&st->old_speech[L_FRAME], &st->old_speech[0], L_TOTAL - L_FRAME); Copy (&st->old_wsp[L_FRAME], &st->old_wsp[0], PIT_MAX); Copy (&st->old_exc[L_FRAME], &st->old_exc[0], PIT_MAX + L_INTERPOL); return; }