--- 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 \

--- /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;
+}

--- /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

--- 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