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changeset 401:59655481e45b

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libtwamr: integrate pstfilt.c
author Mychaela Falconia <falcon@freecalypso.org>
date Mon, 06 May 2024 19:44:15 +0000
parents ffd48f0a2ab5
children a8b73b1c5b19
files libtwamr/Makefile libtwamr/namespace.list libtwamr/pstfilt.c libtwamr/pstfilt.h
diffstat 4 files changed, 291 insertions(+), 7 deletions(-) [+]
line wrap: on
line diff
--- a/libtwamr/Makefile	Mon May 06 19:34:59 2024 +0000
+++ b/libtwamr/Makefile	Mon May 06 19:44:15 2024 +0000
@@ -11,10 +11,10 @@
 	inv_sqrt.o lag_wind.o levinson.o log2.o lpc.o lsfwt.o lsp.o lsp_avg.o \
 	lsp_az.o lsp_lsf.o lsp_tab.o mac_32.o oper_32b.o ph_disp.o pitch_fr.o \
 	post_pro.o pow2.o pre_big.o pre_proc.o pred_lt.o preemph.o prm2bits.o \
-	prmno.o q_gain_c.o q_gain_p.o q_plsf.o q_plsf3_tab.o q_plsf5_tab.o \
-	q_plsf_3.o q_plsf_5.o qgain475.o qgain795.o qua_gain.o qua_gain_tab.o \
-	reorder.o residu.o s10_8pf.o set_sign.o sqrt_l.o syn_filt.o tls_flags.o\
-	weight_a.o window.o
+	prmno.o pstfilt.o q_gain_c.o q_gain_p.o q_plsf.o q_plsf3_tab.o \
+	q_plsf5_tab.o q_plsf_3.o q_plsf_5.o qgain475.o qgain795.o qua_gain.o \
+	qua_gain_tab.o reorder.o residu.o s10_8pf.o set_sign.o sqrt_l.o \
+	syn_filt.o tls_flags.o weight_a.o window.o
 HDRS=	namespace.h
 LIB=	libtwamr.a
 
--- a/libtwamr/namespace.list	Mon May 06 19:34:59 2024 +0000
+++ b/libtwamr/namespace.list	Mon May 06 19:44:15 2024 +0000
@@ -28,9 +28,8 @@
 Lag_window Levinson Levinson_reset
 Lsf_lsp Lsp_lsf Reorder_lsf Lsf_wt Lsp_Az
 Pitch_fr Pitch_fr_reset
-Pre_Process Pre_Process_reset
-Post_Process Post_Process_reset
-Pred_lt_3or6
+Pre_Process Pre_Process_reset Pred_lt_3or6
+Post_Filter Post_Filter_reset Post_Process Post_Process_reset
 Q_plsf_reset Q_plsf_3 Q_plsf_5 Qua_gain
 Residu Syn_filt Weight_Ai
 
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libtwamr/pstfilt.c	Mon May 06 19:44:15 2024 +0000
@@ -0,0 +1,217 @@
+/*************************************************************************
+*
+*      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             : pstfilt.c
+*      Purpose          : Performs adaptive postfiltering on the synthesis
+*                       : speech
+*
+********************************************************************************
+*/
+
+/*
+********************************************************************************
+*                         MODULE INCLUDE FILE AND VERSION ID
+********************************************************************************
+*/
+#include "namespace.h"
+#include "pstfilt.h"
+ 
+/*
+********************************************************************************
+*                         INCLUDE FILES
+********************************************************************************
+*/
+#include "tw_amr.h"
+#include "typedef.h"
+#include "basic_op.h"
+#include "memops.h"
+#include "weight_a.h"
+#include "residu.h"
+#include "syn_filt.h"
+#include "preemph.h"
+#include "no_count.h"
+#include "cnst.h"
+
+/*
+********************************************************************************
+*                         LOCAL VARIABLES AND TABLES
+********************************************************************************
+*/
+/*---------------------------------------------------------------*
+ *    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) */
+
+/* Spectral expansion factors */
+static const Word16 gamma3_MR122[M] = {
+  22938, 16057, 11240, 7868, 5508,
+  3856, 2699, 1889, 1322, 925
+};
+
+static const Word16 gamma3[M] = {
+  18022, 9912, 5451, 2998, 1649, 907, 499, 274, 151, 83
+};
+
+static const Word16 gamma4_MR122[M] = {
+  24576, 18432, 13824, 10368, 7776,
+  5832, 4374, 3281, 2461, 1846
+};
+
+static const Word16 gamma4[M] = {
+  22938, 16057, 11240, 7868, 5508, 3856, 2699, 1889, 1322, 925
+};
+
+/*
+********************************************************************************
+*                         PUBLIC PROGRAM CODE
+********************************************************************************
+*/
+
+/*************************************************************************
+*
+*  Function:   Post_Filter_reset
+*  Purpose:    Initializes state memory to zero
+*
+**************************************************************************
+*/
+void Post_Filter_reset (Post_FilterState *state)
+{
+  Set_zero (state->mem_syn_pst, M);
+  Set_zero (state->res2, L_SUBFR);
+  Set_zero (state->synth_buf, L_FRAME + M);
+  agc_reset(&state->agc_state);
+  preemphasis_reset(&state->preemph_state);
+}
+
+/*
+**************************************************************************
+*  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
+*
+**************************************************************************
+*/
+int Post_Filter (
+    Post_FilterState *st, /* i/o : post filter states                        */
+    enum Mode mode,       /* i   : AMR mode                                  */
+    Word16 *syn,          /* i/o : synthesis speech (postfiltered is output) */
+    Word16 *Az_4          /* i   : interpolated LPC parameters in all subfr. */
+)
+{
+    /*-------------------------------------------------------------------*
+     *           Declaration of parameters                               *
+     *-------------------------------------------------------------------*/
+
+    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;
+    Word16 *syn_work = &st->synth_buf[M]; move16 ();
+
+    /*-----------------------------------------------------*
+     * Post filtering                                      *
+     *-----------------------------------------------------*/
+
+    Copy (syn, syn_work , L_FRAME);
+
+    Az = Az_4;
+
+    for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
+    {
+       /* Find weighted filter coefficients Ap3[] and ap[4] */
+
+       test (); test ();
+       if (sub(mode, MR122) == 0 || sub(mode, MR102) == 0)
+       {
+          Weight_Ai (Az, gamma3_MR122, Ap3);
+          Weight_Ai (Az, gamma4_MR122, Ap4);
+       }
+       else
+       {
+          Weight_Ai (Az, gamma3, Ap3);
+          Weight_Ai (Az, gamma4, Ap4);
+       }
+       
+       /* filtering of synthesis speech by A(z/0.7) to find res2[] */
+       
+       Residu (Ap3, &syn_work[i_subfr], st->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 (&st->preemph_state, st->res2, temp2, L_SUBFR);
+
+       /* filtering through  1/A(z/0.75) */
+
+       Syn_filt (Ap4, st->res2, &syn[i_subfr], L_SUBFR, st->mem_syn_pst, 1);
+
+       /* scale output to input */
+
+       agc (&st->agc_state, &syn_work[i_subfr], &syn[i_subfr],
+            AGC_FAC, L_SUBFR);
+       
+       Az += MP1;
+    }
+
+    /* update syn_work[] buffer */
+
+    Copy (&syn_work[L_FRAME - M], &syn_work[-M], M);
+
+    return 0;
+}
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libtwamr/pstfilt.h	Mon May 06 19:44:15 2024 +0000
@@ -0,0 +1,68 @@
+/*
+********************************************************************************
+*
+*      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             : pstfilt.h
+*      Purpose          : Performs adaptive postfiltering on the synthesis
+*                       : speech
+*
+********************************************************************************
+*/
+#ifndef pstfilt_h
+#define pstfilt_h "$Id $"
+ 
+/*
+********************************************************************************
+*                         INCLUDE FILES
+********************************************************************************
+*/
+#include "tw_amr.h"
+#include "typedef.h"
+#include "cnst.h"
+#include "preemph.h"
+#include "agc.h"
+/*
+********************************************************************************
+*                         DEFINITION OF DATA TYPES
+********************************************************************************
+*/
+typedef struct{
+  Word16 res2[L_SUBFR];
+  Word16 mem_syn_pst[M];
+  preemphasisState preemph_state;
+  agcState agc_state;
+  Word16 synth_buf[M + L_FRAME];  
+} Post_FilterState;
+
+/*
+********************************************************************************
+*                         DECLARATION OF PROTOTYPES
+********************************************************************************
+*/
+ 
+void Post_Filter_reset (Post_FilterState *st);
+/* reset post filter (i.e. set state memory to zero)
+   returns 0 on success
+ */
+
+int Post_Filter (
+    Post_FilterState *st, /* i/o : post filter states                        */
+    enum Mode mode,       /* i   : AMR mode                                  */
+    Word16 *syn,          /* i/o : synthesis speech (postfiltered is output) */
+    Word16 *Az_4          /* i   : interpolated LPC parameters in all subfr. */
+);
+/* filters the signal syn using the parameters in Az_4 to calculate filter
+   coefficients.
+   The filter must be set up using Post_Filter_init prior to the first call
+   to Post_Filter. Post_FilterState is updated to mirror the current state
+   of the filter
+ 
+   return 0 on success
+ */
+ 
+#endif