clf
clear
data=getspeech('0ef1s1t0.wav');
Fs=12500;
plot(data)
end1=input('end1?')
end2=input('end2?')
datar=data(end1:end2);
clear data
plot(datar)
[d,dsize]=size(datar);
deltaf=Fs/8192;
freq=-Fs/2:deltaf:Fs/2-deltaf;
dftdatar=abs(fftshift(fft(datar,8192)));
plot(freq,dftdatar)
axis([-Fs/2 Fs/2 0 max(dftdatar)])
xlabel('Analog Frequency (Hz)')
ylabel('Spectral Magnitude')
pause
soundsc(datar,Fs)
pause
plot(datar)
end1=input('end1?')
end2=input('end2?')
datar2=datar(end1:end2);
plot(datar2)
input('wideband spectrogram')
specgram(datar,512,12500,65,45)
title('wideband spectrogram')
input('narrowband spectrogram')
specgram(datar,1024,12500,650,550)
title('narrowband spectrogram')
Npoles=16;
[A,G]=lpc(datar2,Npoles);
e=conv(datar2,A);
Np=65; Ls=2049;
ex=zeros(1,Ls);
ex(1:Np:Ls)=1;
s(1:Npoles)=zeros(size(1:Npoles));
for n=Npoles:Ls,
s(n)=ex(n);
for m=1:Npoles-1;
s(n)=s(n)-A(m+1)*s(n-m);
end
end
input('totally synthesized sound')
soundsc(s,Fs)
%Here's where we show computational savings of
%using Levinson Durbin algorithm
%
rest=xcorr(datar2,'biased').';
[Y,I]=max(rest);
rvec=rest(I(1):I(1)+Npoles+1,1);
Rc=toeplitz(rvec(1:Npoles,1));
%
flopsTemp=flops;
aInv=-inv(Rc)*rvec(2:Npoles+1,1);
flopsInv=flops-flopsTemp;
%
flopsTemp=flops;
aLev=levinson(rvec,Npoles).';
flopsLev=flops-flopsTemp;
flopsInv-flopsLev