There is no "output of the ADC". The input to the program is from the file test.dat. When test.dat is loaded, the data is placed in the variable (matrix) "test". It is then transposed and placed in variable "a".
No. I mean, assuming my converter is 7-bit and I simulated it in hspice, which nodes should I use as the ADC output and how should those nodes be arranged in the test.dat matrix to be ready for loading in progpam?
I do not understand what you said. Could you please explain more. I'm not strong in MATLAB programming. what does it mean?"it is expecting the input data as INTs."؟؟؟
Does this mean that "a" is a matrix containing arrays 0 and 1?
I do not have much trouble with MATLAB, Master.
Oh my god!!!
Or https://www.edaboard.com/threads/need-help-to-implement-this-matlab-code-to-calculate-sfdr.158379/
%Start MAX1448 Dynamic Performance Test Routine
<disp('HP16500C State Card');
filename=input('Type a:\filename or Press RETURN for HPIB Data Transfer: ');
if isempty(filename)
filename = 'listing';
end
fid=fopen(filename,'r');
numpt=input('Data Record Size (Number of Points)? ');
fclk=input('Sampling Frequency (MHz)? ');
%MAX1448 - 10-bit data converter
numbit=10;
%Discard first 13 lines from the data file, which do not contain data
for i=1:13,
fgetl(fid);
end
[v1,count]=fscanf(fid,'%f',[2,numpt]);
fclose(fid);
v1=v1';
code=v1:),2);
%Display a warning, when the input generates a code greater than full-scale
if (max(code)==2^numbit-1) | (min(code)==0)
disp('Warning: ADC may be clipping!!!');
end
%Plot results in the time domain
figure;
plot([1:numpt],code);
title('TIME DOMAIN')
xlabel('SAMPLES');
ylabel('DIGITAL OUTPUT CODE');
%Recenter the digital sine wave
Dout=code-(2^numbit-1)/2;
%If no window function is used, the input tone must be chosen to be unique and with
%regard to the sampling frequency. To achieve this prime numbers are introduced and the
%input tone is determined by fIN = fSAMPLE * (Prime Number / Data Record Size).
%To relax this requirement, window functions such as HANNING and HAMING (see below) can
%be introduced, however the fundamental in the resulting FFT spectrum appears 'sharper'
%without the use of window functions.
Doutw=Dout;
%Doutw=Dout.*hanning(numpt);
%Doutw=Dout.*hamming(numpt);
%Performing the Fast Fourier Transform
Dout_spect=fft(Doutw);
%Recalculate to dB
Dout_dB=20*log10(abs(Dout_spect));
%Display the results in the frequency domain with an FFT plot
figure;
maxdB=max(Dout_dB(1:numpt/2));
%For TTIMD, use the following short routine, normalized to —6.5dB full-scale.
%plot([0:numpt/2-1].*fclk/numpt,Dout_dB(1:numpt/2)-maxdB-6.5);
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) -120 a1(4)]);
%Calculate SNR, SINAD, THD and SFDR values
%Find the signal bin number, DC = bin 1
fin=find(Dout_dB(1:numpt/2)==maxdB);
%Span of the input frequency on each side
span=max(round(numpt/200),5);
%Approximate search span for harmonics on each side
spanh=2;
%Determine power spectrum
spectP=(abs(Dout_spect)).*(abs(Dout_spect));
%Find DC offset power
Pdc=sum(spectP(1:span));
%Extract overall signal power
Ps=sum(spectP(fin-span:fin+span));
%Vector/matrix to store both frequency and power of signal and harmonics
Fh=[];
%The 1st element in the vector/matrix represents the signal, the next element represents
%the 2nd harmonic, etc.
Ph=[];
%Find harmonic frequencies and power components in the FFT spectrum
for har_num=1:10
%Input tones greater than fSAMPLE are aliased back into the spectrum
tone=rem((har_num*(fin-1)+1)/numpt,1);
if tone>0.5
%Input tones greater than 0.5*fSAMPLE (after aliasing) are reflected
tone=1-tone;
end
Fh=[Fh tone];
%For this procedure to work, ensure the folded back high order harmonics do not overlap
%with DC or signal 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
%Determine the total distortion power
Pd=sum(Ph(2:5));
%Determine the noise power
Pn=sum(spectP(1:numpt/2))-Pdc-Ps-Pd;
format;
A=(max(code)-min(code))/2^numbit
AdB=20*log10(A)
SINAD=10*log10(Ps/(Pn+Pd))
SNR=10*log10(Ps/Pn)
disp('THD is calculated from 2nd through 5th order harmonics');
THD=10*log10(Pd/Ph(1))
SFDR=10*log10(Ph(1)/max(Ph(2:10)))
disp('Signal & Harmonic Power Components:');
HD=10*log10(Ph(1:10)/Ph(1))
%Distinguish all harmonics locations within the FFT plot
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('1st','2nd','3rd','4th','5th','6th','7th','8th','9th');
hold off;