/*---------------------------------------------------------------------------*\
FILE........: postfilter.c
AUTHOR......: David Rowe
DATE CREATED: 13/09/09
Postfilter to improve sound quality for speech with high levels of
background noise. Unlike mixed-excitation models requires no bits
to be transmitted to handle background noise.
\*---------------------------------------------------------------------------*/
/*
Copyright (C) 2009 David Rowe
All rights reserved.
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License version 2.1, as
published by the Free Software Foundation. This program is
distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or
FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public
License for more details.
You should have received a copy of the GNU Lesser General Public License
along with this program; if not, see .
*/
#include
#include
#include
#include
#include "defines.h"
#include "comp.h"
#include "dump.h"
#include "sine.h"
#include "postfilter.h"
/*---------------------------------------------------------------------------*\
DEFINES
\*---------------------------------------------------------------------------*/
#define BG_THRESH 40.0 /* only consider low levels signals for bg_est */
#define BG_BETA 0.1 /* averaging filter constant */
#define BG_MARGIN 6.0 /* harmonics this far above BG noise are
randomised. Helped make bg noise less
spikey (impulsive) for mmt1, but speech was
perhaps a little rougher.
*/
/*---------------------------------------------------------------------------*\
postfilter()
The post filter is designed to help with speech corrupted by
background noise. The zero phase model tends to make speech with
background noise sound "clicky". With high levels of background
noise the low level inter-formant parts of the spectrum will contain
noise rather than speech harmonics, so modelling them as voiced
(i.e. a continuous, non-random phase track) is inaccurate.
Some codecs (like MBE) have a mixed voicing model that breaks the
spectrum into voiced and unvoiced regions. Several bits/frame
(5-12) are required to transmit the frequency selective voicing
information. Mixed excitation also requires accurate voicing
estimation (parameter estimators always break occasionally under
exceptional conditions).
In our case we use a post filter approach which requires no
additional bits to be transmitted. The decoder measures the average
level of the background noise during unvoiced frames. If a harmonic
is less than this level it is made unvoiced by randomising it's
phases.
This idea is rather experimental. Some potential problems that may
happen:
1/ If someone says "aaaaaaaahhhhhhhhh" will background estimator track
up to speech level? This would be a bad thing.
2/ If background noise suddenly dissapears from the source speech does
estimate drop quickly? What is noise suddenly re-appears?
3/ Background noise with a non-flat sepctrum. Current algorithm just
comsiders scpetrum as a whole, but this could be broken up into
bands, each with their own estimator.
4/ Males and females with the same level of background noise. Check
performance the same. Changing Wo affects width of each band, may
affect bg energy estimates.
5/ Not sure what happens during long periods of voiced speech
e.g. "sshhhhhhh"
\*---------------------------------------------------------------------------*/
void postfilter(
MODEL *model,
float *bg_est
)
{
int m, uv;
float e, thresh;
/* determine average energy across spectrum */
e = 1E-12;
for(m=1; m<=model->L; m++)
e += model->A[m]*model->A[m];
assert(e > 0.0);
e = 10.0*log10f(e/model->L);
/* If beneath threhold, update bg estimate. The idea
of the threshold is to prevent updating during high level
speech. */
if ((e < BG_THRESH) && !model->voiced)
*bg_est = *bg_est*(1.0 - BG_BETA) + e*BG_BETA;
/* now mess with phases during voiced frames to make any harmonics
less then our background estimate unvoiced.
*/
uv = 0;
thresh = POW10F((*bg_est + BG_MARGIN)/20.0);
if (model->voiced)
for(m=1; m<=model->L; m++)
if (model->A[m] < thresh) {
model->phi[m] = (TWO_PI/CODEC2_RAND_MAX)*(float)codec2_rand();
uv++;
}
#ifdef DUMP
dump_bg(e, *bg_est, 100.0*uv/model->L);
#endif
}