ADC测试matlab代码

前面有做过ADC性能测试，测试方式是先使用ADC采集一个单频信号，然后利用matlab进行性能分析。

下面把matlab分析的代码记录下来：

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

 %fid=fopen('F:\pelican_ADC_test\vjtag_prj\data_analysis\single_tone.txt','r');
 %numpt=input('Number of  Points in FFT? ');
 %fclk=input('Sampling Frequency (MHz)? ');
 %numbit=input('ADC Resolution (bits)? ');
 close all;
 clear all;

 numpt=;
 fclk=/;
 % fclk=//;
 numbit=;

 % a=textread('dds_data_out.txt','%s')';%以字符形式打开文件
 % a=dlmread('sample_data.txt','%s')';%以字符形式打开文件
 a=dlmread('sample_data.txt','%s')';%以字符形式打开文件
 v1=a'; %16进制转化为10进制数，存储进v1矩阵
 % for i = :numpt
 %     v1(i) = bitxor(v1(i),);
 % end

 %v1 is a column vector(:numpt), and count is its length
 %v1=dlmread('F:\pelican_ADC_test\vjtag_prj\data_analysis\single_tone.txt','');
 %fclose(fid);
 %change v1 into a row vector
 code=v1';

 %Warning: ADC output may be clipping - reduce input amplitude
 if (max(code)==2^numbit-1) | (min(code)==0)
   disp('WARNING: ADC OUTPUT MAYBE CLIPPING - CHECK INPUT AMPLITUDE!');
 end

 figure;
 plot(code);
 title('TIME DOMAIN')
 xlabel('SAMPLES');
 ylabel('DIGITAL OUTPUT CODE');
 Dout=code-(2^numbit-1)/2; %Re-center the digitized sinusoidal input
 Voltage=Dout./((2^numbit-1)/2)*(0.5);
 figure;
 plot([1:numpt],Voltage);

 Doutw=(Dout').*blackmanharris(numpt);               %add Minimum -term Blackman-Harris window
 Dout_spect=fft(Doutw);
 Dout_dB=*log10(abs(Dout_spect));

 figure;
 maxdB=max(Dout_dB(:numpt/));     %numpt points FFT result in numpt/ points spectrum

 %计算距离满量程的幅度差
 max_voltage=max(Voltage);
 delta_amplitude=*log10(max_voltage/0.5);      %full scale voltage amplitude is .5v

 plot([:numpt/-].*fclk/numpt,Dout_dB(:numpt/)-maxdB+delta_amplitude);
 grid on;
 title('SINGLE TONE FFT PLOT');
 xlabel('ANALOG INPUT FREQUENCY (MHz)');
 ylabel('AMPLITUDE (dBfs)');
 a1=axis; axis([a1() a1() - a1()]);
 fin=find(Dout_dB(:numpt/)==maxdB); %Find the signal bin (DC represents bin=)
 DC_span=;              %default DC leakage bins are  bins
 signal_span = ;       %signal leakage bins are ± bins for minumun -term black-harris window
 spanh=;                %%default harmonic leakage bins are ± bins
 spectP=(abs(Dout_spect)).*(abs(Dout_spect));
 %Determine power level
 Pdc=sum(spectP(:DC_span)); %Determine DC offset power level
 Ps=sum(spectP(fin-signal_span:fin+signal_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=:
   tone=rem((har_num*(fin- )+)/numpt,);   %Note: tones > fSAMPLE are aliased back
   if tone>0.5
      tone=-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- ;     %make sure that the folded back high order harmonics do not overlap with DC and signal components or lower order harmonics
   Ph=[Ph sum(spectP(har_bin-:har_bin+))];
 end

 Pd=sum(Ph(:)); %Total distortion power level
 Pn=sum(spectP(:numpt/))-Pdc-Ps-Pd; %Extract noise power level
 format;
 A=(max(code)-min(code))/(^numbit) %Analog input amplitude in mV
 AdB=*log10(A) %Analog input amplitude in dB
 SNR=*log10(Ps/Pn) %SNR in dB
 SINAD=*log10(Ps/(Pn+Pd))  %SINAD in dB
 disp('THD - HD2 through HD9');
 THD=*log10(Pd/Ph()) %THD in dB
 SFDR=*log10(Ph()/max(Ph(:))) %SFDR in dB
 disp('SIGNAL AND HARMONIC POWER (dB)');
 HD=*log10(Ph(:)/Ph())

 hold on;
 plot(Fh()*fclk,-,'bo',Fh()*fclk,-,'bx',Fh()*fclk,-,'b+',Fh()*fclk,-,'b*',Fh()*fclk,-,'bs',Fh()*fclk,-,'bd',Fh()*fclk,-,'bv',Fh()*fclk,-,'b^');
 legend('SIGNAL','HD2','HD3','HD4','HD5','HD6','HD7','HD8','HD9');
 hold off;

输出结果

 A =

     0.5432

 AdB =

    -5.3006

 SNR =

    54.0005

 SINAD =

    53.4240

 THD - HD2 through HD9

 THD =

   -62.4784

 SFDR =

    63.0618

 SIGNAL AND HARMONIC POWER (dB)

 HD =

            -63.0618  -78.1190  -79.6691  -82.4058  -86.1153  -90.7795  -91.1868  -89.8460  -74.6853

会标记出谐波的位置。