view code Kat/spsmodel.m @ 1:881c3acf1164

matlab code Kat

function [y,yh,ys] = spsmodel(x,fs,w,N,t,maxnS,stocf)
%=> analysis/synthesis of a sound using the sinusoidal harmonic model
% x: input sound, fs: sampling rate, w: analysis window (odd size),
% N: FFT size (minimum 512), t: threshold in negative dB,
% maxnS: maximum number of sinusoids,
% stocf: decimation factor of mag spectrum for stochastic analysis
% y: output sound, yh: harmonic component, ys: stochastic component
M = length(w); % analysis window size
Ns = 1024; % FFT size for synthesis
H = 256; % hop size for analysis and synthesis
N2 = N/2+1; % half-size of spectrum
soundlength = length(x); % length of input sound array
hNs = Ns/2; % half synthesis window size
hM = (M-1)/2; % half analysis window size
pin = max(N2,1+hM) % initialize sound pointer to middle of analysis window
pend = soundlength-max(N2,hM); % last sample to start a frame
fftbuffer = zeros(N,1); % initialize buffer for FFT
yh = zeros(soundlength+Ns/2,1); % output sine component
ys = zeros(soundlength+Ns/2,1); % output residual component
w = w/sum(w); % normalize analysis window
sw = zeros(Ns,1);
ow = triang(2*H-1); % overlapping window
ovidx = Ns/2+1-H+1:Ns/2+H; % overlap indexes
sw(ovidx) = ow(1:2*H-1);
bh = blackmanharris(Ns); % synthesis window
bh = bh ./ sum(bh); % normalize synthesis window
wr = bh; % window for residual
sw(ovidx) = sw(ovidx) ./ bh(ovidx);
sws = H*hanning(Ns)/2; % synthesis window for stochastic
lastysloc = zeros(maxnS,1); % initialize synthesis harmonic locations
ysphase = 2*pi*rand(maxnS,1); % initialize synthesis harmonic phases
fridx = 0;





while pin<pend
    %-----analysis-----%
    xw = x(pin-hM:pin+hM).*w(1:M); % window the input sound
    fftbuffer = fftbuffer*0; % reset buffer;
    fftbuffer(1:(M+1)/2) = xw((M+1)/2:M); % zero-phase window in fftbuffer
    fftbuffer(N-(M-1)/2+1:N) = xw(1:(M-1)/2);
    X = fft(fftbuffer); % compute the FFT
    mX = 20*log10(abs(X(1:N2))); % magnitude spectrum
    pX = unwrap(angle(X(1:N/2+1))); % unwrapped phase spectrum
    ploc = 1 + find((mX(2:N2-1)>t) .* (mX(2:N2-1)>mX(3:N2)) ...
        .* (mX(2:N2-1)>mX(1:N2-2))); % find peaks
    [ploc,pmag,pphase] = peakinterp(mX,pX,ploc); % refine peak values
    % sort by magnitude
    [smag,I] = sort(pmag(:),1,'descend');
    nS = min(maxnS,length(find(smag>t)));
    sloc = ploc(I(1:nS));
    sphase = pphase(I(1:nS));
    if (fridx==0)
        % update last frame data for first frame
        lastnS = nS;
        lastsloc = sloc;
        lastsmag = smag;
        lastsphase = sphase;
    end
    % connect sinusoids to last frame lnS (lastsloc,lastsphase,lastsmag)
    sloc(1:nS) = (sloc(1:nS)~=0).*((sloc(1:nS)-1)*Ns/N); % synth. locs
    lastidx = zeros(1,nS);
    for i=1:nS
        [dev,idx] = min(abs(sloc(i) - lastsloc(1:lastnS)));
        lastidx(i) = idx;
    end
    ri= pin-hNs; % input sound pointer for residual analysis
    xr = x(ri:ri+Ns-1).*wr(1:Ns); % window the input sound
    Xr = fft(fftshift(xr)); % compute FFT for residual analysis
    Xh = genspecsines(sloc,smag,sphase,Ns); % generate sines
    Xr = Xr-Xh; % get the residual complex spectrum
    mXr = 20*log10(abs(Xr(1:Ns/2+1))); % magnitude spectrum of residual
    mXsenv = decimate(max(-200,mXr),stocf); % decimate the magnitude spectrum
    % and avoid -Inf
    %-----synthesis data-----%
    ysloc = sloc; % synthesis locations
    ysmag = smag(1:nS); % synthesis magnitudes
    mYsenv = mXsenv; % synthesis residual envelope
    
    
%-----transformations-----%

%-----time mapping-----%
outsoundlength = soundlength*2;
TM=[ 0 soundlength/fs; % input time (sec)
0 outsoundlength/fs ]; % output time (sec)

%%-----frequency stretch-----%
%fstretch = 1.1;
%ysloc = ysloc .* (fstretch.^[0:length(ysloc)-1]');
   
%-----synthesis-----%
    if (fridx==0)
        lastysphase = ysphase;
    end
    if (nS>lastnS)
        lastysphase = [ lastysphase ; zeros(nS-lastnS,1) ];
        lastysloc = [ lastysloc ; zeros(nS-lastnS,1) ];
    end
    ysphase = lastysphase(lastidx(1:nS)) + 2*pi*(lastysloc(lastidx(1:nS))+ysloc)/2/Ns*H; % propagate phases
    lastysloc = ysloc;
    lastysphase = ysphase;
    lastnS = nS; % update last frame data
    lastsloc = sloc; % update last frame data
    lastsmag = smag; % update last frame data
    lastsphase = sphase; % update last frame data
    Yh = genspecsines(ysloc,ysmag,ysphase,Ns); % generate sines
    mYs = interp(mYsenv,stocf); % interpolate to original size
    roffset = ceil(stocf/2)-1; % interpolated array offset
    mYs = [ mYs(1)*ones(roffset,1); mYs(1:Ns/2+1-roffset) ];
    mYs = 10.^(mYs/20); % dB to linear magnitude
    pYs = 2*pi*rand(Ns/2+1,1); % generate phase spectrum with random values
    mYs1 = [mYs(1:Ns/2+1); mYs(Ns/2:-1:2)]; % create complete magnitude spectrum
    pYs1 = [pYs(1:Ns/2+1); -1*pYs(Ns/2:-1:2)]; % create complete phase spectrum
    Ys = mYs1.*cos(pYs1)+1i*mYs1.*sin(pYs1); % compute complex spectrum
    yhw = fftshift(real(ifft(Yh))); % sines in time domain using inverse FFT
    ysw = fftshift(real(ifft(Ys))); % stochastic in time domain using IFFT
    yh(ri:ri+Ns-1) = yh(ri:ri+Ns-1)+yhw(1:Ns).*sw; % overlap-add for sines
    ys(ri:ri+Ns-1) = ys(ri:ri+Ns-1)+ysw(1:Ns).*sws; % overlap-add for stochastic
    pin = pin+H; % advance the sound pointer
    fridx = fridx+1;
end
y= yh+ys; % sum sines and stochastic