--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libgsmefr/pstfilt2.c	Fri Nov 25 16:18:21 2022 +0000
@@ -0,0 +1,182 @@
+/*************************************************************************
+ *
+ *  FILE NAME:   pstfilt2.c
+ *
+ * Performs adaptive postfiltering on the synthesis speech
+ *
+ *  FUNCTIONS INCLUDED:  Init_Post_Filter()  and Post_Filter()
+ *
+ *************************************************************************/
+
+#include "typedef.h"
+#include "basic_op.h"
+#include "sig_proc.h"
+#include "count.h"
+#include "codec.h"
+#include "cnst.h"
+
+/*---------------------------------------------------------------*
+ *    Postfilter constant parameters (defined in "cnst.h")       *
+ *---------------------------------------------------------------*
+ *   L_FRAME     : Frame size.                                   *
+ *   L_SUBFR     : Sub-frame size.                               *
+ *   M           : LPC order.                                    *
+ *   MP1         : LPC order+1                                   *
+ *   MU          : Factor for tilt compensation filter           *
+ *   AGC_FAC     : Factor for automatic gain control             *
+ *---------------------------------------------------------------*/
+
+#define L_H 22  /* size of truncated impulse response of A(z/g1)/A(z/g2) */
+
+/*------------------------------------------------------------*
+ *   static vectors                                           *
+ *------------------------------------------------------------*/
+
+ /* inverse filtered synthesis */
+
+static Word16 res2[L_SUBFR];
+
+ /* memory of filter 1/A(z/0.75) */
+
+static Word16 mem_syn_pst[M];
+
+ /* Spectral expansion factors */
+
+const Word16 F_gamma3[M] =
+{
+    22938, 16057, 11240, 7868, 5508,
+    3856, 2699, 1889, 1322, 925
+};
+const Word16 F_gamma4[M] =
+{
+    24576, 18432, 13824, 10368, 7776,
+    5832, 4374, 3281, 2461, 1846
+};
+
+/*************************************************************************
+ *
+ *  FUNCTION:   Init_Post_Filter
+ *
+ *  PURPOSE: Initializes the postfilter parameters.
+ *
+ *************************************************************************/
+
+void Init_Post_Filter (void)
+{
+    Set_zero (mem_syn_pst, M);
+
+    Set_zero (res2, L_SUBFR);
+
+    return;
+}
+
+/*************************************************************************
+ *  FUNCTION:  Post_Filter()
+ *
+ *  PURPOSE:  postfiltering of synthesis speech.
+ *
+ *  DESCRIPTION:
+ *      The postfiltering process is described as follows:
+ *
+ *          - inverse filtering of syn[] through A(z/0.7) to get res2[]
+ *          - tilt compensation filtering; 1 - MU*k*z^-1
+ *          - synthesis filtering through 1/A(z/0.75)
+ *          - adaptive gain control
+ *
+ *************************************************************************/
+
+void Post_Filter (
+    Word16 *syn,    /* in/out: synthesis speech (postfiltered is output)    */
+    Word16 *Az_4    /* input: interpolated LPC parameters in all subframes  */
+)
+{
+    /*-------------------------------------------------------------------*
+     *           Declaration of parameters                               *
+     *-------------------------------------------------------------------*/
+
+    Word16 syn_pst[L_FRAME];    /* post filtered synthesis speech   */
+    Word16 Ap3[MP1], Ap4[MP1];  /* bandwidth expanded LP parameters */
+    Word16 *Az;                 /* pointer to Az_4:                 */
+                                /*  LPC parameters in each subframe */
+    Word16 i_subfr;             /* index for beginning of subframe  */
+    Word16 h[L_H];
+
+    Word16 i;
+    Word16 temp1, temp2;
+    Word32 L_tmp;
+
+    /*-----------------------------------------------------*
+     * Post filtering                                      *
+     *-----------------------------------------------------*/
+
+    Az = Az_4;
+
+    for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
+    {
+        /* Find weighted filter coefficients Ap3[] and ap[4] */
+
+        Weight_Ai (Az, F_gamma3, Ap3);
+        Weight_Ai (Az, F_gamma4, Ap4);
+
+        /* filtering of synthesis speech by A(z/0.7) to find res2[] */
+
+        Residu (Ap3, &syn[i_subfr], res2, L_SUBFR);
+
+        /* tilt compensation filter */
+
+        /* impulse response of A(z/0.7)/A(z/0.75) */
+
+        Copy (Ap3, h, M + 1);
+        Set_zero (&h[M + 1], L_H - M - 1);
+        Syn_filt (Ap4, h, h, L_H, &h[M + 1], 0);
+
+        /* 1st correlation of h[] */
+
+        L_tmp = L_mult (h[0], h[0]);
+        for (i = 1; i < L_H; i++)
+        {
+            L_tmp = L_mac (L_tmp, h[i], h[i]);
+        }
+        temp1 = extract_h (L_tmp);
+
+        L_tmp = L_mult (h[0], h[1]);
+        for (i = 1; i < L_H - 1; i++)
+        {
+            L_tmp = L_mac (L_tmp, h[i], h[i + 1]);
+        }
+        temp2 = extract_h (L_tmp);
+
+        test (); 
+        if (temp2 <= 0)
+        {
+            temp2 = 0;          move16 (); 
+        }
+        else
+        {
+            temp2 = mult (temp2, MU);
+            temp2 = div_s (temp2, temp1);
+        }
+
+        preemphasis (res2, temp2, L_SUBFR);
+
+        /* filtering through  1/A(z/0.75) */
+
+        Syn_filt (Ap4, res2, &syn_pst[i_subfr], L_SUBFR, mem_syn_pst, 1);
+
+        /* scale output to input */
+
+        agc (&syn[i_subfr], &syn_pst[i_subfr], AGC_FAC, L_SUBFR);
+
+        Az += MP1;
+    }
+
+    /* update syn[] buffer */
+
+    Copy (&syn[L_FRAME - M], &syn[-M], M);
+
+    /* overwrite synthesis speech by postfiltered synthesis speech */
+
+    Copy (syn_pst, syn, L_FRAME);
+
+    return;
+}