matlab程序求注释，语音增强的，减谱法.....
function [ss,po]=specsubm(s,fs,p)
%利用频谱相减（spectral subtraction）增强 [SS,PO]=(S,FS,P)
%
% 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
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
ns=length(s);
ts=1/fs;
ss=zeros(ns,1);
ni=pow2(nextpow2(fs*po(3)/po(8)));
ti=ni/fs;
nw=ni*po(8);
nf=1+floor((ns-nw)/ni);
nm=ceil(fs*po(4)/(ni*po(7)));
win=0.5*hamming(nw+1)/1.08;win(end)=[];
zg=exp(-ti/po(1));
za=exp(-ti/po(2));
zo=exp(-ti/po(5));
px=zeros(1+nw/2,1);
pxn=px;
os=px;
mb=ones(1+nw/2,po(7))*nw/2;
im=0;
osf=po(11)*(1+(0:nw/2).'*fs/(nw*po(6))).^(-1);
imidx=[13 21]';
x2im=zeros(length(imidx),nf);
osim=x2im;
pnim=x2im;
pxnim=x2im;
qim=x2im;
for is=1:nf
idx=(1:nw)+(is-1)*ni;
x=rfft(s(idx).*win);
x2=x.*conj(x);
pxn=za*pxn+(1-za)*x2;
im=rem(im+1,nm);
if im
mb(:,1)=min(mb(:,1),pxn);
else
mb=[pxn,mb(:,1:po(7)-1)];
end
pn=po(9)*min(mb,[],2);
%os= oversubtraction factor
os=zo*os+(1-zo)*(1+osf.*pn./(pn+pxn));
px=zg*px+(1-zg)*x2;
q=max(po(10)*sqrt(pn./x2),1-sqrt(os.*pn./px));
ss(idx)=ss(idx)+irfft(x.*q);
end
if nargout==0
soundsc([s; ss],fs);
end
不外乎是，分帧，加窗，估计噪音，傅里叶变换，反傅里叶变换等。
%输入参数s 语音数据，fs 采样频率，p 下面有说明，共11个，可不输入，有默认值
%“过度减法(oversubtraction)”作减法的时候,保留一小部分原来的背景噪音,用这部分背景噪音来掩盖住音乐噪音的谱峰,从而消除了令人不悦的音乐噪音。
%通过给的参数p，估计噪音，做谱减法。从而消除噪音。
function [ss,po]=specsubm(s,fs,p)
%利用频谱相减（spectral subtraction）增强 [SS,PO]=(S,FS,P)
%
% 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
%检查函数的输入参数，如果输入少于三个，po为默认值，po的参数上面有说明
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
ns=length(s);
ts=1/fs;
ss=zeros(ns,1);
ni=pow2(nextpow2(fs*po(3)/po(8)));
ti=ni/fs;
nw=ni*po(8);
nf=1+floor((ns-nw)/ni);
nm=ceil(fs*po(4)/(ni*po(7)));
win=0.5*hamming(nw+1)/1.08;win(end)=[];
zg=exp(-ti/po(1));
za=exp(-ti/po(2));
zo=exp(-ti/po(5));
px=zeros(1+nw/2,1);
pxn=px;
os=px;
mb=ones(1+nw/2,po(7))*nw/2;
im=0;
osf=po(11)*(1+(0:nw/2).'*fs/(nw*po(6))).^(-1);
imidx=[13 21]';
x2im=zeros(length(imidx),nf);
osim=x2im;
pnim=x2im;
pxnim=x2im;
qim=x2im;
for is=1:nf
idx=(1:nw)+(is-1)*ni;
x=rfft(s(idx).*win);
x2=x.*conj(x);
pxn=za*pxn+(1-za)*x2;
im=rem(im+1,nm);
if im
mb(:,1)=min(mb(:,1),pxn);
else
mb=[pxn,mb(:,1:po(7)-1)];
end
pn=po(9)*min(mb,[],2);
%os= oversubtraction factor
os=zo*os+(1-zo)*(1+osf.*pn./(pn+pxn));
px=zg*px+(1-zg)*x2;
q=max(po(10)*sqrt(pn./x2),1-sqrt(os.*pn./px));
ss(idx)=ss(idx)+irfft(x.*q);
end
if nargout==0
soundsc([s; ss],fs);
end
