view libgsmefr/pstfilt2.c @ 581:e2d5cad04cbf

libgsmhr1 RxFE: store CN R0+LPC separately from speech In the original GSM 06.06 code the ECU for speech mode is entirely separate from the CN generator, maintaining separate state. (The main intertie between them is the speech vs CN state variable, distinguishing between speech and CN BFIs, in addition to the CN-specific function of distinguishing between initial and update SIDs.) In the present RxFE implementation I initially thought that we could use the same saved_frame buffer for both ECU and CN, overwriting just the first 4 params (R0 and LPC) when a valid SID comes in. However, I now realize it was a bad idea: the original code has a corner case (long sequence of speech-mode BFIs to put the ECU in state 6, then SID and CN-mode BFIs, then a good speech frame) that would be broken by that buffer reuse approach. We could eliminate this corner case by resetting the ECU state when passing through a CN insertion period, but doing so would needlessly increase the behavioral diffs between GSM 06.06 and our version. Solution: use a separate CN-specific buffer for CN R0+LPC parameters, and match the behavior of GSM 06.06 code in this regard.
author Mychaela Falconia <falcon@freecalypso.org>
date Thu, 13 Feb 2025 10:02:45 +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;
}