# v_specsubm

## PURPOSE

V_SPECSUBM obsolete speech enhancement algorithm - use v_specsub instead

## SYNOPSIS

function [ss,po]=v_specsubm(s,fs,p)

## DESCRIPTION

    V_SPECSUBM obsolete speech enhancement algorithm - use v_specsub instead

     implementation of spectral subtraction algorithm by R Martin (rather slow)
     algorithm parameters: t* in seconds, f* in Hz, k* dimensionless
     1: tg = smoothing time constant for signal power estimate (0.04): high=reverberant, low=musical
     2: ta = smoothing time constant for signal power estimate
            used in noise estimation (0.1)
     3: tw = fft window length (will be rounded up to 2^nw samples)
     4: tm = length of minimum filter (1.5): high=slow response to noise increase, low=distortion
     5: to = time constant for oversubtraction factor (0.08)
     6: fo = oversubtraction corner frequency (800): high=distortion, low=musical
     7: km = number of minimisation buffers to use (4): high=waste memory, low=noise modulation
     8: ks = oversampling constant (4)
     9: kn = noise estimate compensation (1.5)
     10:kf = subtraction floor (0.02): high=noisy, low=musical
     11:ko = oversubtraction scale factor (4): high=distortion, low=musical

     Refs:
        (a) R. Martin. Spectral subtraction based on minimum statistics. In Proc EUSIPCO, pages 1182-1185, Edinburgh, Sept 1994.
        (b) R. Martin. Noise power spectral density estimation based on optimal smoothing and minimum statistics.
            IEEE Trans. Speech and Audio Processing, 9(5):504-512, July 2001.

## CROSS-REFERENCE INFORMATION

This function calls:

- v_irfft V_IRFFT Inverse fft of a conjugate symmetric spectrum X=(Y,N,D)
- v_rfft V_RFFT Calculate the DFT of real data Y=(X,N,D)

This function is called by:

## SOURCE CODE

    0001 function [ss,po]=v_specsubm(s,fs,p)
    0002 %V_SPECSUBM obsolete speech enhancement algorithm - use v_specsub instead
    0003 %
    0004 % implementation of spectral subtraction algorithm by R Martin (rather slow)
    0005 % algorithm parameters: t* in seconds, f* in Hz, k* dimensionless
    0006 % 1: tg = smoothing time constant for signal power estimate (0.04): high=reverberant, low=musical
    0007 % 2: ta = smoothing time constant for signal power estimate
    0008 %        used in noise estimation (0.1)
    0009 % 3: tw = fft window length (will be rounded up to 2^nw samples)
    0010 % 4: tm = length of minimum filter (1.5): high=slow response to noise increase, low=distortion
    0011 % 5: to = time constant for oversubtraction factor (0.08)
    0012 % 6: fo = oversubtraction corner frequency (800): high=distortion, low=musical
    0013 % 7: km = number of minimisation buffers to use (4): high=waste memory, low=noise modulation
    0014 % 8: ks = oversampling constant (4)
    0015 % 9: kn = noise estimate compensation (1.5)
    0016 % 10:kf = subtraction floor (0.02): high=noisy, low=musical
    0017 % 11:ko = oversubtraction scale factor (4): high=distortion, low=musical
    0018 %
    0019 % Refs:
    0020 %    (a) R. Martin. Spectral subtraction based on minimum statistics. In Proc EUSIPCO, pages 1182-1185, Edinburgh, Sept 1994.
    0021 %    (b) R. Martin. Noise power spectral density estimation based on optimal smoothing and minimum statistics.
    0022 %        IEEE Trans. Speech and Audio Processing, 9(5):504-512, July 2001.
    0023
    0024
    0025 %      Copyright (C) Mike Brookes 2004
    0026 %      Version: $Id: v_specsubm.m 10865 2018-09-21 17:22:45Z dmb $
    0027 %
    0028 %   VOICEBOX is a MATLAB toolbox for speech processing.
    0029 %   Home page: http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html
    0030 %
    0031 %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
    0032 %   This program is free software; you can redistribute it and/or modify
    0033 %   it under the terms of the GNU Lesser General Public License as published by
    0034 %   the Free Software Foundation; either version 3 of the License, or
    0035 %   (at your option) any later version.
    0036 %
    0037 %   This program is distributed in the hope that it will be useful,
    0038 %   but WITHOUT ANY WARRANTY; without even the implied warranty of
    0039 %   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
    0040 %   GNU Lesser General Public License for more details.
    0041 %
    0042 %   You can obtain a copy of the GNU Lesser General Public License from
    0043 %   https://www.gnu.org/licenses/ .
    0044 %
    0045 %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
    0046
    0047 if nargin<3 po=[0.04 0.1 0.032 1.5 0.08 400 4 4 1.5 0.02 4].'; else po=p; end
    0048 ns=length(s);
    0049 ts=1/fs;
    0050 ss=zeros(ns,1);
    0051
    0052 ni=pow2(nextpow2(fs*po(3)/po(8)));
    0053 ti=ni/fs;
    0054 nw=ni*po(8);
    0055 nf=1+floor((ns-nw)/ni);
    0056 nm=ceil(fs*po(4)/(ni*po(7)));
    0057
    0058 win=0.5*hamming(nw+1)/1.08;win(end)=[];
    0059 zg=exp(-ti/po(1));
    0060 za=exp(-ti/po(2));
    0061 zo=exp(-ti/po(5));
    0062
    0063 px=zeros(1+nw/2,1);
    0064 pxn=px;
    0065 os=px;
    0066 mb=ones(1+nw/2,po(7))*nw/2;
    0067 im=0;
    0068 osf=po(11)*(1+(0:nw/2).'*fs/(nw*po(6))).^(-1);
    0069
    0070 imidx=[13 21]';
    0071 x2im=zeros(length(imidx),nf);
    0072 osim=x2im;
    0073 pnim=x2im;
    0074 pxnim=x2im;
    0075 qim=x2im;
    0076
    0077 for is=1:nf
    0078    idx=(1:nw)+(is-1)*ni;
    0079    x=v_rfft(s(idx).*win);
    0080    x2=x.*conj(x);
    0081
    0082    pxn=za*pxn+(1-za)*x2;
    0083    im=rem(im+1,nm);
    0084    if im
    0085       mb(:,1)=min(mb(:,1),pxn);
    0086    else
    0087       mb=[pxn,mb(:,1:po(7)-1)];
    0088    end
    0089    pn=po(9)*min(mb,[],2);
    0090    %os= oversubtraction factor
    0091    os=zo*os+(1-zo)*(1+osf.*pn./(pn+pxn));
    0092
    0093    px=zg*px+(1-zg)*x2;
    0094    q=max(po(10)*sqrt(pn./x2),1-sqrt(os.*pn./px));
    0095    ss(idx)=ss(idx)+v_irfft(x.*q);
    0096
    0097 end
    0098 if nargout==0
    0099    soundsc([s; ss],fs);
    0100 end
    0101
    0102
    0103
    0104