# HG changeset patch # User Mychaela Falconia # Date 1715053825 0 # Node ID 93d6c6960a46d6ba69f8b12768f574f96a797cd7 # Parent f17e42c630581aca0a5bb541e8129d2cf6ed1f87 libtwamr: integrate cod_amr.c diff -r f17e42c63058 -r 93d6c6960a46 libtwamr/Makefile --- a/libtwamr/Makefile Tue May 07 03:21:14 2024 +0000 +++ b/libtwamr/Makefile Tue May 07 03:50:25 2024 +0000 @@ -2,7 +2,7 @@ CFLAGS= -O2 OBJS= a_refl.o agc.o autocorr.o az_lsp.o b_cn_cod.o basicop2.o bgnscd.o \ bitno.o bits2prm.o c1035pf.o c2_11pf.o c2_9pf.o c3_14pf.o c4_17pf.o \ - c8_31pf.o c_g_aver.o calc_cor.o calc_en.o cbsearch.o cl_ltp.o \ + c8_31pf.o c_g_aver.o calc_cor.o calc_en.o cbsearch.o cl_ltp.o cod_amr.o\ convolve.o cor_h.o d1035pf.o d2_11pf.o d2_9pf.o d3_14pf.o d4_17pf.o \ d8_31pf.o d_gain_c.o d_gain_p.o d_plsf.o d_plsf_3.o d_plsf_5.o \ dec_gain.o dec_lag3.o dec_lag6.o dhf_check.o dhf_tables.o dtx_dec.o \ @@ -20,8 +20,8 @@ HDRS= a_refl.h agc.h autocorr.h az_lsp.h b_cn_cod.h basic_op.h bgnscd.h \ bitno.h bits2prm.h c1035pf.h c2_11pf.h c2_9pf.h c3_14pf.h c4_17pf.h \ c8_31pf.h c_g_aver.h calc_cor.h calc_en.h cbsearch.h cl_ltp.h cnst.h \ - cnst_vad.h convolve.h cor_h.h d1035pf.h d2_11pf.h d2_9pf.h d3_14pf.h \ - d4_17pf.h d8_31pf.h d_gain_c.h d_gain_p.h d_plsf.h dec_gain.h \ + cnst_vad.h cod_amr.h convolve.h cor_h.h d1035pf.h d2_11pf.h d2_9pf.h \ + d3_14pf.h d4_17pf.h d8_31pf.h d_gain_c.h d_gain_p.h d_plsf.h dec_gain.h\ dec_lag3.h dec_lag6.h dtx_common.h dtx_dec.h dtx_enc.h e_homing.h \ ec_gains.h enc_lag3.h enc_lag6.h ex_ctrl.h g_adapt.h g_code.h g_pitch.h\ gain_q.h gains_tab.h gc_pred.h gmed_n.h graytab.h hp_max.h int_defs.h \ diff -r f17e42c63058 -r 93d6c6960a46 libtwamr/cod_amr.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libtwamr/cod_amr.c Tue May 07 03:50:25 2024 +0000 @@ -0,0 +1,576 @@ +/* +***************************************************************************** +* +* 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; +} diff -r f17e42c63058 -r 93d6c6960a46 libtwamr/cod_amr.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libtwamr/cod_amr.h Tue May 07 03:50:25 2024 +0000 @@ -0,0 +1,165 @@ +/* +***************************************************************************** +* +* 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.h +* Purpose : Main encoder routine operating on a frame basis. +* +***************************************************************************** +*/ +#ifndef cod_amr_h +#define cod_amr_h "$Id $" + +/* +***************************************************************************** +* INCLUDE FILES +***************************************************************************** +*/ +#include "tw_amr.h" +#include "typedef.h" +#include "cnst.h" +#include "lpc.h" +#include "lsp.h" +#include "cl_ltp.h" +#include "gain_q.h" +#include "p_ol_wgh.h" +#include "ton_stab.h" +#include "vad.h" +#include "dtx_enc.h" + +/* +***************************************************************************** +* DEFINITION OF DATA TYPES +***************************************************************************** +*/ +/*-----------------------------------------------------------* + * Coder constant parameters (defined in "cnst.h") * + *-----------------------------------------------------------* + * L_WINDOW : LPC analysis window size. * + * L_NEXT : Samples of next frame needed for autocor. * + * L_FRAME : Frame size. * + * L_FRAME_BY2 : Half the frame size. * + * L_SUBFR : Sub-frame size. * + * M : LPC order. * + * MP1 : LPC order+1 * + * L_TOTAL7k4 : Total size of speech buffer. * + * PIT_MIN7k4 : Minimum pitch lag. * + * PIT_MAX : Maximum pitch lag. * + * L_INTERPOL : Length of filter for interpolation * + *-----------------------------------------------------------*/ +typedef struct { + /* Speech vector */ + Word16 old_speech[L_TOTAL]; + Word16 *speech, *p_window, *p_window_12k2; + Word16 *new_speech; /* Global variable */ + + /* Weight speech vector */ + Word16 old_wsp[L_FRAME + PIT_MAX]; + Word16 *wsp; + + /* OL LTP states */ + Word16 old_lags[5]; + Word16 ol_gain_flg[2]; + + /* Excitation vector */ + Word16 old_exc[L_FRAME + PIT_MAX + L_INTERPOL]; + Word16 *exc; + + /* Zero vector */ + Word16 ai_zero[L_SUBFR + MP1]; + Word16 *zero; + + /* Impulse response vector */ + Word16 *h1; + Word16 hvec[L_SUBFR * 2]; + + /* Substates */ + lpcState lpcSt; + lspState lspSt; + clLtpState clLtpSt; + gainQuantState gainQuantSt; + pitchOLWghtState pitchOLWghtSt; + tonStabState tonStabSt; + vadState vadSt; + Flag dtx; + dtx_encState dtx_encSt; + + /* Filter's memory */ + Word16 mem_syn[M], mem_w0[M], mem_w[M]; + Word16 mem_err[M + L_SUBFR], *error; + + Word16 sharp; +} cod_amrState; + +/* +******************************************************************************** +* DECLARATION OF PROTOTYPES +******************************************************************************** +*/ + +/* +************************************************************************** +* +* Function : cod_amr_reset +* Purpose : Resets state memory +* Returns : 0 on success +* +************************************************************************** +*/ +void cod_amr_reset (cod_amrState *st, Flag dtx, Flag use_vad2); + +/*************************************************************************** + * 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) */ +); + +/*************************************************************************** + * 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 */ +); + +#endif diff -r f17e42c63058 -r 93d6c6960a46 libtwamr/namespace.list --- a/libtwamr/namespace.list Tue May 07 03:21:14 2024 +0000 +++ b/libtwamr/namespace.list Tue May 07 03:50:25 2024 +0000 @@ -51,6 +51,7 @@ sid_sync sid_sync_reset sid_sync_set_handover_debt ton_stab_reset check_lsp check_gp_clipping update_gp_clipping +cod_amr cod_amr_reset cod_amr_first dtx_dec dtx_dec_reset dtx_dec_activity_update rx_dtx_handler dtx_enc dtx_enc_reset dtx_buffer tx_dtx_handler vad1 vad1_reset vad_complex_detection_update vad_tone_detection