### Josef Hoffmann (@josefsepp)

I am Professor at the University of Applied Sciences in Karlsruhe Germany. I am interested in Signal Processing in general and especially to wavelet transformation

## Determination of the transfer function of passive networks with MATLAB Functions

With MATLAB functions, the transfer function of passive networks can be determined relatively easily. The method is explained using the example of a passive low-pass filter of the sixth order, which is shown in Fig.1

Fig.1 Passive low-pass filter of the sixth order

If one tried, as would be logical, to calculate the transfer function starting from the input, it would be quite complicated. On the other hand, if you start from the output, the determination of this function is simple...

## Sampling bandpass signals

Sampling bandpass signals 1.1 Introduction

It is known [1], [3] that bandpass signals can be sampled with a sampling frequency which is lower than the sampling frequency according to the sampling theorem.

Fig. 1 shows an example of how the spectrum of a bandpass signal sampled with \$f_s\$ (Fig. 1a) arises in the baseband with \$−f_s / 2 ≤ f < f_s/2\$. The bandpass signal is assumed to have a center frequency \$f_c = (f_{max} + f_{min})/2\$ and bandwidth \$\Delta f...

June 13, 20211 comment
≥≥≥ Simulink-Simulation of SSB demodulation or modulation from the article “Understanding the ‘Phasing Method’ of Single Sideband Demodulation” by Richard Lyons Josef Hoffmann

The article “Understanding the ‘Phasing Method’ of Single Sideband Demodulation” by Richard Lyons is a very good description of this topic. The block representation from the figures are clear and easy to understand. They are predestined for a simulation in Simulink. The simulation can help...

## The correct answer to the quiz of @apolin

January 10, 2020

The correct answer to the @apolin quiz can be easily explained using the following Simulink model:

In MATLAB you have to initialize the two filters:

h = dftmtx (8);

h1 = h (3, :); % The filter of the quiz

h2 = h (7, :); % The mirrored filter

The impulse responses of the filters h1, h2 are complex and the responses to a broadband random signal are also complex. The two spectrum analyzer blocks then show the PSD, typical for analytical...

## Re: mitigating phase shift

Hi,The amount of 0.999 of the poles causes the poles to be very close to the unit circle in the z complex plane. In implementation, the poles can go outside the...

## Re: Power Spectral Density PSD° of OFDM signals

If the complex signal is equal to xi +j*xq, then you need to form Xk = FFT(xi+j*xq) first and then use your formula to get the PSD.

## Re: Power Spectral Density PSD° of OFDM signals

Hello,I have created a simulink model that can be used to answer your questions very easily and clearly.A complex signal corresponding to QPSK is formed from two...

## Re: Noise floor is varying with respect to Decimation rate.

Himy opinion is that decimation cannot work here. If you want to get the analog signal from 65MHz down to 2MHz in baseband, then you may simply need to sample the...

## Re: Phase brake in tonal signal and frequency estimation

HiI have expanded my program with the following MATLAB lines and the amplitude spectrum only shows one pick.s = [x1 x2];S = fft(s)/(2*1001);figure(3);clf;plot(0:2001,...

## Re: Phase brake in tonal signal and frequency estimation

Hi,I did not receive your problem with these few MATLAB lines:% Script test_1.mx1 = sin(2*pi*100*(0:1000)/1000);phi = pi/3;x2 = sin(2*pi*100*(0:1000)/1000 + phi);figure(1);...

## Re: How to estimate the real channel response from its PSD?

Hi,thank you I didn't know that!Best regards

## Re: How to estimate the real channel response from its PSD?

HiI think I found the bug. The option 'centered' in the command pwelch has to be replaced with 'twosided'. This gives the correct symmetry of the psd, which results...

## Re: Help with polynomial zeros

Hello try to use the function poly:>> help poly poly Convert roots to polynomial.    poly(A), when A is an N by N matrix, is a row vector with    N+1...

## Re: FFT spectrum shift after time domain decimation

Hi,The sketch shows how the spectra of the signals change through the filtering and downsample.In the illustration a) the original spectrum is shown. It contains...

## Single Sideband Demodulation

After watching the very good blog 'Phasing Method of Single Sideband Demodulation' authorRick Lyons, I decided to simulate some of the arrangements shown (such as...

## Re: Impulse response in OFDM system using IFFT

It is not a phase correction but a multiplication with a complex oscillation. The link that is mentioned contains the same error.

## Re: Impulse response in OFDM system using IFFT

In my opinion, the formula you use is incorrect. With ifft () you generate a time sequence that you have to multiply by the complex oscillation exp (-2j pi fc t)....

## Re: DC Motor control system sampling rate

Hello,Attached is a PDF file with a text in German and a translation in English. I hope it helps to understand how to describe a DC motor.RegardsDC_motor.pdf

## Re: Undersampling FM radio frequencies

HelloYou can subsample so that your bandpass signal is shifted in the baseband. In a Simulink model (undersample_100.slx) a sine signal of 100.01 MHz is shifted...

## Re: Matlab, properly using IFFT, FIR Filter Desing

Hello,the following MATLAB-Script might help you:%################################################% Script sampled_mag_1.m, in which the impulse response of % a...

## Re: Matlab, properly using IFFT, FIR Filter Desing

Hi,i think you have a time aliasing due to the sampling of the DTFT and therefore the reconstructed signal via the inverse DTFT is not the expected signal.With the...

## Re: Evaluating RF ADC architectures - Sub-Sampling or Nyquist ?

Hello,with the following pictureI will try to answer you. For a channel with a sampling frequency of 5.4 GHz, a bandwidth of BW = 2.7 GHz in the vicinity of 8 GHz...

## Re: DF2T IIR SOS

Hello Mr. Fragavetti,here is a MATLAB script that I used to edit an audio file. You need to change the script for your needs.Always look at the time signals. Instead...

Euler_2.pdf

## Re: Simplifications With Eulers Equation (DTFT, DTFS)

I send you the file Euler_1.png

## Re: Simplifications With Eulers Equation (DTFT, DTFS)

Hi there,you have to find a suitable factor so that you can always use the Euler formula, e.g. here:Euler_1.png

## Re: Any info on the BER performance of a communication system involving a DUC(Digital Up-Converter) and DDC(Digital Down-Converter)?

...

Here is a Simulink model that can be used to answer your question.Random numbers between -1 and +1 with a period of 1/100 s are generated from a Random Number block....

## Re: Extract pitch of irregular signal ?

The plot is only a section at the beginning of the FFT !

## Re: Extract pitch of irregular signal ?

The second frequency is just the reflection of the first frequency in the second Nyquist interval !

## Re: Extract pitch of irregular signal ?

Hello,I tried to analyze your wav file with a wavelet transform. With two stages, the approximation coefficients are freed from interference, giving the pitch frequency...

## Re: Increasing frequency of a sinusoids after FFT of a signal

Hello,the idea of djmaguire is very good. For the frequency of 1700.5 HZ you have to increase the resolution of the FFT to 0.5 Hz / Bin, so you need a data block...

## Re: FFT Channelizer and PFB

Please try to use the FFT for analysis and synthesis or for both the iFFT. I use this in my textbook and it works.

## Re: Average Impulse Response from multiple measurements

An example of such identification:% Skript identif_2.m, clear;% ------ SignalsN = 1024;%x = sign(randn(1,N));x = randn(1,N);% Systemnord = 64;h = fir1(nord, 0.2); ...

## Re: Average Impulse Response from multiple measurements

Hello,You can determine the autocorrelation of the excitation and the cross correlation input / output for each data block. You can then average these and improve...

## Re: Understanding the Nyquist frequency when doing complex FFT

I changed figure(3):figure(3);subplot(211), plot(tk, real(yk));title('Continuous signal'), grid on;subplot(212), plot(tk, y_rec);title('The samples and with sinc-functions...

## Re: Understanding the Nyquist frequency when doing complex FFT

To be sure that the samples are correct one has to reconstruct the continuous signal with the sinc function from these samples:function mist1N = 64; %number of samplest_vec...

## Re: Understanding the Nyquist frequency when doing complex FFT

try this !!!N = 64; %number of samplest_vec =(0:1/N:1-(1/N)); % time valuestk = (0:1/512:1-1/512); % continuos timeA3=1;f3=30; % signal frequencyy = (A3*exp(j*2*pi*f3*t_vec));yk...

## Re: Polyphase decimation filter: Simulink implementation of commutator variant not as expected

Hello,Here's another solution with a block Delay Line:polyphase_solution_2.slxpolyphase_solution2.mThe delay block with a delay equal to one is necessary because...

## Re: Polyphase decimation filter: Simulink implementation of commutator variant not as expected

...

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