FreeCalypso > hg > gsm-codec-lib
view libgsmefr/pstfilt2.c @ 242:f081a6850fb5
libgsmfrp: new refined implementation
The previous implementation exhibited the following defects,
which are now fixed:
1) The last received valid SID was cached forever for the purpose of
handling future invalid SIDs - we could have received some valid
SID ages ago, then lots of speech or NO_DATA, and if we then get
an invalid SID, we would resurrect the last valid SID from ancient
history - a bad design. In our new design, we handle invalid SID
based on the current state, much like BFI.
2) GSM 06.11 spec says clearly that after the second lost SID
(received BFI=1 && TAF=1 in CN state) we need to gradually decrease
the output level, rather than jump directly to emitting silence
frames - we previously failed to implement such logic.
3) Per GSM 06.12 section 5.2, Xmaxc should be the same in all 4 subframes
in a SID frame. What should we do if we receive an otherwise valid
SID frame with different Xmaxc? Our previous approach would
replicate this Xmaxc oddity in every subsequent generated CN frame,
which is rather bad. In our new design, the very first CN frame
(which can be seen as a transformation of the SID frame itself)
retains the original 4 distinct Xmaxc, but all subsequent CN frames
are based on the Xmaxc from the last subframe of the most recent SID.
author | Mychaela Falconia <falcon@freecalypso.org> |
---|---|
date | Tue, 09 May 2023 05:16:31 +0000 |
parents | 41d8e8f4058d |
children |
line wrap: on
line source
/************************************************************************* * * 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; }