how to measure SNR and power spectrum of a signal practically?

Thanks kaz

now I captured my 16 bit output by signal tap (the files of captured data and text file crated by signal tap were attached)

and also I found MATLAB code that plots the FFT spectrum of a desired test tone and will computes SNR, SINAD, THD and SFDR.

Matlab codes are as bellow: (also the pdf file contain the codes were attached)

does any body know how can I introduce my captured data file to Matlab code to plot fft for me?

%The following program code plots the FFT spectrum of a desired test tone. Test tone based on coherent sampling

criteria, and

%computes SNR, SINAD, THD and SFDR.

%Copyright Au/Hofner, Maxim Integrated Products, 120 San Gabriel Drive, Sunnyvale, CA94086

%This program is believed to be accurate and reliable. This program may get altered without prior notification.;

disp('HP16500C LA 100/110 State Card');

filename=input('Enter file name or press RETURN to accept data from LA via GPIB/HPIB ) : ');

if isempty(filename)

filename = 'listing';

end

fid=fopen(filename,'r');

numpt=input('Number of Points in FFT? ');

fclk=input('Sampling Frequency (MHz)? ');

numbit=input('ADC Resolution? ');

%Discard first 13 lines of the LA listing (LA header), as they don't contain valid data.

for i=1:13,

fgetl(fid);

end

[v1,count]=fscanf(fid,'%f',[2,numpt]);

fclose(fid);

v1=v1';

code=v1(:,2);

%Warning: ADC output may be clipping - reduce input amplitude

if (max(code)==2numbit-1) | (min(code)==0)

disp('WARNING: ADC OUTPUT MAYBE CLIPPING - CHECK INPUT AMPLITUDE!');

end

figure;

plot([1:numpt],code);

title('TIME DOMAIN')

xlabel('SAMPLES');

ylabel('DIGITAL OUTPUT CODE');

Dout=code-(2^numbit-1)/2; %Re-center the digitized sinusoidal input

Doutw=Dout;

Dout_spect=fft(Doutw);

Dout_dB=20×log10(abs(Dout_spect));

figure;

maxdB=max(Dout_dB(1:numpt/2));

plot([0:numpt/2-1].×fclk/numpt,Dout_dB(1:numpt/2)-maxdB );

grid on;

title('FFT PLOT');

xlabel('ANALOG INPUT FREQUENCY (MHz)');

ylabel('AMPLITUDE (dB ) ' );

a1=axis; axis([a1(1) a1(2) -100 a1(4)]);

fin=find(Dout_dB(1:numpt/2)==maxdB ) ; %Find the signal bin (DC represents bin=1)

span=max(round(numpt/200),5); %Determine span of input frequency on each side

spanh=2; %Search span for harmonic distortion components on each side

spectP=(abs(Dout_spect)).×

(abs(Dout_spect)); %Determine power level

Pdc=sum(spectP(1:span)); %Determine DC offset power level

Ps=sum(spectP(fin-span:fin+span)); %Determine signal power level

Fh=[]; %Vector storing frequency and power components of signal and

harmonics

Ph=[]; %HD1=signal, HD2=2nd harmonic, HD3=3rd harmonic, etc.

%Find the harmonic frequencies/power within the FFT plot

for har_num=1:10

tone=rem((har_num × (fin-1)+1)/numpt,1); %Note: tones > fSAMPLE are aliased back

if tone>0.5

tone=1-tone;

end

Fh=[Fh tone];

%For this method to work properly, make sure that the folded back high order harmonics do not overlap with DC and

signal

%components or lower order harmonics.

har_peak=max(spectP(round(tone×numpt)-spanh:round(tone×numpt)+spanh));

har_bin=find(spectP(round(tone×numpt)-spanh:round(tone×numpt)+spanh)==har_peak);

har_bin=har_bin+round(tone×numpt)-spanh-1; Ph=[Ph sum(spectP(har_bin-1:har_bin+1))];

end

Pd=sum(Ph(2:5)); %Total distortion power level

Pn=sum(spectP(1:numpt/2))-Pdc-Ps-Pd; %Extract noise power level

format;

A=(max(code)-min(code))/2numbit %Analog input amplitude in mV

AdB=20×log10(A) %Analog input amplitude in dB

SNR=10×log10(Ps/Pn) %SNR in dB

SNR=10×log10(Ps/Pn) %SINAD in dB

SINAD=10×log10(Ps/(Pn+Pd))

disp('THD - HD2 through HD5');

THD=10×log10(Pd/Ph(1)) %THD in dB

SFDR=10×log10(Ph(1)/max(Ph(2:10))) %SFDR in dB

disp('SIGNAL AND HARMONIC POWER (dB ) ' );

HD=10×log10(Ph(1:10)/Ph(1))

hold on;

plot(Fh(2)×fclk,0,'mo',Fh(3)×fclk,0,'cx',Fh(4)×fclk,0,'r+',Fh(5)×fclk,0,'g×',Fh(6)×fclk,0,'bs',Fh(7)×fclk,0,'bd',Fh(8)×fclk,0,'kv',

Fh(9)×fclk,0,'y^');

legend('SIGNAL','HD2','HD3','HD4','HD5','HD6','HD7','HD8','HD9');

hold off;

any guidance is really appropriated I dont know how to get fft plot ?