FreeCalypso > hg > gsm-codec-lib
view libtwamr/cod_amr.c @ 450:16bc3036cc84
LICENSE: acknowledge libtwamr origin from ETSI/3GPP
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Fri, 10 May 2024 01:48:20 +0000 |
parents | 93d6c6960a46 |
children |
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/* ***************************************************************************** * * GSM AMR-NB speech codec R98 Version 7.6.0 December 12, 2001 * R99 Version 3.3.0 * REL-4 Version 4.1.0 * ***************************************************************************** * * File : cod_amr.c * Purpose : Main encoder routine operating on a frame basis. * ***************************************************************************** */ #include "namespace.h" #include "cod_amr.h" /* ***************************************************************************** * INCLUDE FILES ***************************************************************************** */ #include "typedef.h" #include "basic_op.h" #include "no_count.h" #include "cnst.h" #include "memops.h" #include "qua_gain.h" #include "lpc.h" #include "lsp.h" #include "pre_big.h" #include "ol_ltp.h" #include "p_ol_wgh.h" #include "spreproc.h" #include "cl_ltp.h" #include "pred_lt.h" #include "spstproc.h" #include "cbsearch.h" #include "gain_q.h" #include "convolve.h" #include "ton_stab.h" #include "vad.h" #include "dtx_enc.h" /* ***************************************************************************** * LOCAL VARIABLES AND TABLES ***************************************************************************** */ /* ***************************************************************************** * PUBLIC VARIABLES AND TABLES ***************************************************************************** */ /* Spectral expansion factors */ static const Word16 gamma1[M] = { 30802, 28954, 27217, 25584, 24049, 22606, 21250, 19975, 18777, 17650 }; /* gamma1 differs for the 12k2 coder */ static const Word16 gamma1_12k2[M] = { 29491, 26542, 23888, 21499, 19349, 17414, 15672, 14105, 12694, 11425 }; static const Word16 gamma2[M] = { 19661, 11797, 7078, 4247, 2548, 1529, 917, 550, 330, 198 }; /* ***************************************************************************** * PUBLIC PROGRAM CODE ***************************************************************************** */ /* ************************************************************************** * * Function : cod_amr_reset * Purpose : Resets state memory * ************************************************************************** */ void cod_amr_reset (cod_amrState *st, Flag dtx, Flag use_vad2) { Word16 i; /* save DTX flag */ st->dtx = dtx; /*-----------------------------------------------------------------------* * Initialize pointers to speech vector. * *-----------------------------------------------------------------------*/ st->new_speech = st->old_speech + L_TOTAL - L_FRAME; /* New speech */ st->speech = st->new_speech - L_NEXT; /* Present frame */ st->p_window = st->old_speech + L_TOTAL - L_WINDOW; /* For LPC window */ st->p_window_12k2 = st->p_window - L_NEXT; /* EFR LPC window: no lookahead */ /* Initialize static pointers */ st->wsp = st->old_wsp + PIT_MAX; st->exc = st->old_exc + PIT_MAX + L_INTERPOL; st->zero = st->ai_zero + MP1; st->error = st->mem_err + M; st->h1 = &st->hvec[L_SUBFR]; /* 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->zero, L_SUBFR); Set_zero(st->hvec, L_SUBFR); /* set to zero "h1[-L_SUBFR..-1]" */ /* OL LTP states */ for (i = 0; i < 5; i++) { st->old_lags[i] = 40; } /* Reset lpc states */ lpc_reset(&st->lpcSt); /* Reset lsp states */ lsp_reset(&st->lspSt); /* Reset clLtp states */ cl_ltp_reset(&st->clLtpSt); gainQuant_reset(&st->gainQuantSt); p_ol_wgh_reset(&st->pitchOLWghtSt); ton_stab_reset(&st->tonStabSt); vad_reset(&st->vadSt, use_vad2); dtx_enc_reset(&st->dtx_encSt); st->sharp = SHARPMIN; } /*************************************************************************** * FUNCTION: cod_amr_first * * PURPOSE: Copes with look-ahead. * * INPUTS: * No input argument are passed to this function. However, before * calling this function, 40 new speech data 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 200). * ***************************************************************************/ int cod_amr_first(cod_amrState *st, /* i/o : State struct */ Word16 new_speech[]) /* i : speech input (L_FRAME) */ { Copy(new_speech,&st->new_speech[-L_NEXT], L_NEXT); /* Copy(new_speech,st->new_speech,L_FRAME); */ return 0; } /*************************************************************************** * FUNCTION: cod_amr * * PURPOSE: Main 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 argument are passed to this function. However, before * calling this function, 160 new speech data 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) * ***************************************************************************/ int cod_amr( cod_amrState *st, /* i/o : State struct */ enum Mode mode, /* i : AMR mode */ Word16 new_speech[], /* i : speech input (L_FRAME) */ Word16 ana[], /* o : Analysis parameters */ enum Mode *usedMode, /* o : used mode */ Word16 synth[] /* o : Local synthesis */ ) { /* LPC coefficients */ 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 *A, *Aq; /* Pointer on A_t and Aq_t */ Word16 lsp_new[M]; /* Other vectors */ Word16 xn[L_SUBFR]; /* Target vector for pitch search */ Word16 xn2[L_SUBFR]; /* Target vector for codebook search */ Word16 code[L_SUBFR]; /* Fixed codebook excitation */ Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */ Word16 y2[L_SUBFR]; /* Filtered fixed codebook excitation */ Word16 gCoeff[6]; /* Correlations between xn, y1, & y2: */ Word16 res[L_SUBFR]; /* Short term (LPC) prediction residual */ Word16 res2[L_SUBFR]; /* Long term (LTP) prediction residual */ /* Vector and scalars needed for the MR475 */ Word16 xn_sf0[L_SUBFR]; /* Target vector for pitch search */ Word16 y2_sf0[L_SUBFR]; /* Filtered codebook innovation */ Word16 code_sf0[L_SUBFR]; /* Fixed codebook excitation */ Word16 h1_sf0[L_SUBFR]; /* The impulse response of sf0 */ Word16 mem_syn_save[M]; /* Filter memory */ Word16 mem_w0_save[M]; /* Filter memory */ Word16 mem_err_save[M]; /* Filter memory */ Word16 sharp_save; /* Sharpening */ Word16 evenSubfr; /* Even subframe indicator */ Word16 T0_sf0 = 0; /* Integer pitch lag of sf0 */ Word16 T0_frac_sf0 = 0; /* Fractional pitch lag of sf0 */ Word16 i_subfr_sf0 = 0; /* Position in exc[] for sf0 */ Word16 gain_pit_sf0; /* Quantized pitch gain for sf0 */ Word16 gain_code_sf0; /* Quantized codebook gain for sf0 */ /* Scalars */ Word16 i_subfr, subfrNr; Word16 T_op[L_FRAME/L_FRAME_BY2]; Word16 T0, T0_frac; Word16 gain_pit, gain_code; /* Flags */ Word16 lsp_flag = 0; /* indicates resonance in LPC filter */ Word16 gp_limit; /* pitch gain limit value */ Word16 vad_flag; /* VAD decision flag */ Word16 compute_sid_flag; /* SID analysis flag */ Copy(new_speech, st->new_speech, L_FRAME); *usedMode = mode; move16 (); /* DTX processing */ if (st->dtx) { /* no test() call since this if is only in simulation env */ /* Find VAD decision */ if (st->vadSt.use_vad2) { vad_flag = vad2(st->new_speech, &st->vadSt.u.v2); vad_flag = vad2(st->new_speech+80, &st->vadSt.u.v2) || vad_flag; } else { vad_flag = vad1(&st->vadSt.u.v1, st->new_speech); } /* NB! usedMode may change here */ compute_sid_flag = tx_dtx_handler(&st->dtx_encSt, vad_flag, usedMode); } else { compute_sid_flag = 0; move16 (); } /*------------------------------------------------------------------------* * - Perform LPC analysis: * * * 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 */ lpc(&st->lpcSt, mode, st->p_window, st->p_window_12k2, A_t); /* From A(z) to lsp. LSP quantization and interpolation */ lsp(&st->lspSt, mode, *usedMode, A_t, Aq_t, lsp_new, &ana); /* Buffer lsp's and energy */ dtx_buffer(&st->dtx_encSt, lsp_new, st->new_speech); /* Check if in DTX mode */ test(); if (sub(*usedMode, MRDTX) == 0) { dtx_enc(&st->dtx_encSt, compute_sid_flag, &st->lspSt.qSt, &st->gainQuantSt.gc_predSt, &ana); Set_zero(st->old_exc, PIT_MAX + L_INTERPOL); Set_zero(st->mem_w0, M); Set_zero(st->mem_err, M); Set_zero(st->zero, L_SUBFR); Set_zero(st->hvec, L_SUBFR); /* set to zero "h1[-L_SUBFR..-1]" */ /* Reset lsp states */ lsp_reset(&st->lspSt); Copy(lsp_new, st->lspSt.lsp_old, M); Copy(lsp_new, st->lspSt.lsp_old_q, M); /* Reset clLtp states */ cl_ltp_reset(&st->clLtpSt); st->sharp = SHARPMIN; } else { /* check resonance in the filter */ lsp_flag = check_lsp(&st->tonStabSt, st->lspSt.lsp_old); } /*----------------------------------------------------------------------* * - Find the weighted input speech w_sp[] 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 * *----------------------------------------------------------------------*/ if (st->dtx && st->vadSt.use_vad2) { /* no test() call since this if is only in simulation env */ st->vadSt.u.v2.L_Rmax = 0; st->vadSt.u.v2.L_R0 = 0; } for(subfrNr = 0, i_subfr = 0; subfrNr < L_FRAME/L_FRAME_BY2; subfrNr++, i_subfr += L_FRAME_BY2) { /* Pre-processing on 80 samples */ pre_big(mode, gamma1, gamma1_12k2, gamma2, A_t, i_subfr, st->speech, st->mem_w, st->wsp); test (); test (); if ((sub(mode, MR475) != 0) && (sub(mode, MR515) != 0)) { /* Find open loop pitch lag for two subframes */ ol_ltp(&st->pitchOLWghtSt, &st->vadSt, mode, &st->wsp[i_subfr], &T_op[subfrNr], st->old_lags, st->ol_gain_flg, subfrNr, st->dtx); } } if ((sub(mode, MR475) == 0) || (sub(mode, MR515) == 0)) { /* Find open loop pitch lag for ONE FRAME ONLY */ /* search on 160 samples */ ol_ltp(&st->pitchOLWghtSt, &st->vadSt, mode, &st->wsp[0], &T_op[0], st->old_lags, st->ol_gain_flg, 1, st->dtx); T_op[1] = T_op[0]; } if (st->dtx && st->vadSt.use_vad2) { /* no test() call since this if is only in simulation env */ LTP_flag_update(&st->vadSt.u.v2, mode); } /* run VAD pitch detection */ if (st->dtx && !st->vadSt.use_vad2) { /* no test() call since this if is only in simulation env */ vad_pitch_detection(&st->vadSt.u.v1, T_op); } if (sub(*usedMode, MRDTX) == 0) { goto the_end; } /*------------------------------------------------------------------------* * 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 dealy * * - update the impulse response h1[] by including fixed-gain 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 * *------------------------------------------------------------------------*/ A = A_t; /* pointer to interpolated LPC parameters */ Aq = Aq_t; /* pointer to interpolated quantized LPC parameters */ evenSubfr = 0; move16 (); subfrNr = -1; move16 (); for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR) { subfrNr = add(subfrNr, 1); evenSubfr = sub(1, evenSubfr); /* Save states for the MR475 mode */ test(); test(); if ((evenSubfr != 0) && (sub(*usedMode, MR475) == 0)) { Copy(st->mem_syn, mem_syn_save, M); Copy(st->mem_w0, mem_w0_save, M); Copy(st->mem_err, mem_err_save, M); sharp_save = st->sharp; } /*-----------------------------------------------------------------* * - Preprocessing of subframe * *-----------------------------------------------------------------*/ if (sub(*usedMode, MR475) != 0) { subframePreProc(*usedMode, gamma1, gamma1_12k2, gamma2, A, Aq, &st->speech[i_subfr], st->mem_err, st->mem_w0, st->zero, st->ai_zero, &st->exc[i_subfr], st->h1, xn, res, st->error); } else { /* MR475 */ subframePreProc(*usedMode, gamma1, gamma1_12k2, gamma2, A, Aq, &st->speech[i_subfr], st->mem_err, mem_w0_save, st->zero, st->ai_zero, &st->exc[i_subfr], st->h1, xn, res, st->error); /* save impulse response (modified in cbsearch) */ if (evenSubfr != 0) { Copy (st->h1, h1_sf0, L_SUBFR); } } /* copy the LP residual (res2 is modified in the CL LTP search) */ Copy (res, res2, L_SUBFR); /*-----------------------------------------------------------------* * - Closed-loop LTP search * *-----------------------------------------------------------------*/ cl_ltp(&st->clLtpSt, &st->tonStabSt, *usedMode, i_subfr, T_op, st->h1, &st->exc[i_subfr], res2, xn, lsp_flag, xn2, y1, &T0, &T0_frac, &gain_pit, gCoeff, &ana, &gp_limit); /* update LTP lag history */ if ((subfrNr == 0) && (st->ol_gain_flg[0] > 0)) { st->old_lags[1] = T0; } if ((sub(subfrNr, 3) == 0) && (st->ol_gain_flg[1] > 0)) { st->old_lags[0] = T0; } /*-----------------------------------------------------------------* * - Inovative codebook search (find index and gain) * *-----------------------------------------------------------------*/ cbsearch(xn2, st->h1, T0, st->sharp, gain_pit, res2, code, y2, &ana, *usedMode, subfrNr); /*------------------------------------------------------* * - Quantization of gains. * *------------------------------------------------------*/ gainQuant(&st->gainQuantSt, *usedMode, res, &st->exc[i_subfr], code, xn, xn2, y1, y2, gCoeff, evenSubfr, gp_limit, &gain_pit_sf0, &gain_code_sf0, &gain_pit, &gain_code, &ana); /* update gain history */ update_gp_clipping(&st->tonStabSt, gain_pit); if (sub(*usedMode, MR475) != 0) { /* Subframe Post Porcessing */ subframePostProc(st->speech, *usedMode, i_subfr, gain_pit, gain_code, Aq, synth, xn, code, y1, y2, st->mem_syn, st->mem_err, st->mem_w0, st->exc, &st->sharp); } else { if (evenSubfr != 0) { i_subfr_sf0 = i_subfr; move16 (); Copy(xn, xn_sf0, L_SUBFR); Copy(y2, y2_sf0, L_SUBFR); Copy(code, code_sf0, L_SUBFR); T0_sf0 = T0; move16 (); T0_frac_sf0 = T0_frac; move16 (); /* Subframe Post Porcessing */ subframePostProc(st->speech, *usedMode, i_subfr, gain_pit, gain_code, Aq, synth, xn, code, y1, y2, mem_syn_save, st->mem_err, mem_w0_save, st->exc, &st->sharp); st->sharp = sharp_save; move16(); } else { /* update both subframes for the MR475 */ /* Restore states for the MR475 mode */ Copy(mem_err_save, st->mem_err, M); /* re-build excitation for sf 0 */ Pred_lt_3or6(&st->exc[i_subfr_sf0], T0_sf0, T0_frac_sf0, L_SUBFR, 1); Convolve(&st->exc[i_subfr_sf0], h1_sf0, y1, L_SUBFR); Aq -= MP1; subframePostProc(st->speech, *usedMode, i_subfr_sf0, gain_pit_sf0, gain_code_sf0, Aq, synth, xn_sf0, code_sf0, y1, y2_sf0, st->mem_syn, st->mem_err, st->mem_w0, st->exc, &sharp_save); /* overwrites sharp_save */ Aq += MP1; /* re-run pre-processing to get xn right (needed by postproc) */ /* (this also reconstructs the unsharpened h1 for sf 1) */ subframePreProc(*usedMode, gamma1, gamma1_12k2, gamma2, A, Aq, &st->speech[i_subfr], st->mem_err, st->mem_w0, st->zero, st->ai_zero, &st->exc[i_subfr], st->h1, xn, res, st->error); /* re-build excitation sf 1 (changed if lag < L_SUBFR) */ Pred_lt_3or6(&st->exc[i_subfr], T0, T0_frac, L_SUBFR, 1); Convolve(&st->exc[i_subfr], st->h1, y1, L_SUBFR); subframePostProc(st->speech, *usedMode, i_subfr, gain_pit, gain_code, Aq, synth, xn, code, y1, y2, st->mem_syn, st->mem_err, st->mem_w0, st->exc, &st->sharp); } } A += MP1; /* interpolated LPC parameters for next subframe */ Aq += MP1; } Copy(&st->old_exc[L_FRAME], &st->old_exc[0], PIT_MAX + L_INTERPOL); the_end: /*--------------------------------------------------* * Update signal for next frame. * *--------------------------------------------------*/ Copy(&st->old_wsp[L_FRAME], &st->old_wsp[0], PIT_MAX); Copy(&st->old_speech[L_FRAME], &st->old_speech[0], L_TOTAL - L_FRAME); return 0; }