FreeCalypso > hg > gsm-codec-lib
view libgsmefr/pstfilt2.c @ 282:9ee8ad3d4d30
frtest: rm gsmfr-hand-test and gsmfr-max-out utils
These hack programs were never properly documented and were written
only as part of a debug chase, in pursuit of a bug that ultimately
turned out to be in our then-hacky patch to osmo-bts-sysmo,
before beginning of proper patches in Osmocom. These hack programs
need to be dropped from the present sw package because they depend
on old libgsm, and we are eliminating that dependency.
author | Mychaela Falconia <falcon@freecalypso.org> |
---|---|
date | Sun, 14 Apr 2024 05:44:47 +0000 |
parents | 41d8e8f4058d |
children |
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/************************************************************************* * * FILE NAME: pstfilt2.c * * Performs adaptive postfiltering on the synthesis speech * * FUNCTIONS INCLUDED: Init_Post_Filter() and Post_Filter() * *************************************************************************/ #include "gsm_efr.h" #include "typedef.h" #include "namespace.h" #include "basic_op.h" #include "sig_proc.h" #include "memops.h" #include "no_count.h" #include "codec.h" #include "cnst.h" #include "dec_state.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 * *------------------------------------------------------------*/ /* 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 (struct EFR_decoder_state *st) { Set_zero (st->mem_syn_pst, M); Set_zero (st->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 ( struct EFR_decoder_state *st, 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], 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, st->res2, temp2, L_SUBFR); /* filtering through 1/A(z/0.75) */ Syn_filt (Ap4, st->res2, &syn_pst[i_subfr], L_SUBFR, st->mem_syn_pst, 1); /* scale output to input */ agc (st, &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; }