FreeCalypso > hg > gsm-codec-lib
view libgsmefr/pstfilt2.c @ 183:452c1d5a6268
libgsmefr BFI w/o data: emit zero output after decoder reset
In real-life usage, each EFR decoder session will most likely begin
with lots of BFI frames before the first real frame arrives. However,
because the spec-defined home state of the decoder is speech rather
than CN, our regular logic for BFI w/o data would have to feed
pseudorandom noise to the decoder (in the "fixed codebook excitation
pulses" part), which is silly to do at the beginning of the decoder
session right out of reset. Therefore, let's check reset_flag_old,
and if we are still in the reset state, simply emit zero output.
author | Mychaela Falconia <falcon@freecalypso.org> |
---|---|
date | Tue, 03 Jan 2023 00:12:18 +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; }