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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

Josef HoffmannJosef Hoffmann December 7, 20212 comments

Starting the calculation from the output makes deriving a passive network transfer function simple, and this post shows how to do it in MATLAB using a sixth-order low-pass example. The walkthrough uses tf('s') to build a symbolic H(s), extracts coefficients with tfdata, and shows numerical frequency-response plotting via freqs or direct j*omega evaluation, with code and component values to reproduce the results.


Sampling bandpass signals

Josef HoffmannJosef Hoffmann June 26, 20215 comments

Bandpass signals can be sampled at rates below the usual Nyquist limit, and this note shows how the band-limited spectrum appears in baseband after sampling. Using a simple example figure, it defines the center frequency fc = (fmax + fmin)/2 and bandwidth Δf = fmax - fmin, and highlights that choosing fs less than twice the signal's highest frequency violates the sampling theorem.


Simulink-Simulation of SSB demodulation

Josef HoffmannJosef Hoffmann June 13, 20211 comment

This post walks through Simulink models that implement SSB demodulation and modulation, using Richard Lyons' phasing method as a foundation. It shows practical models for simple carrier multiplication and for the phasing method with cosine and -sin paths plus Hilbert filtering, and it highlights sampling, decimation, filter choices, and delay alignment to make the techniques work in simulation.


The correct answer to the quiz of @apolin

Josef HoffmannJosef Hoffmann January 10, 2020

A compact Simulink model explains why certain DFT rows behave like negative-frequency bandpass filters, using dftmtx(8) rows as impulse responses. The demo shows that a 2 kHz tone with phase 0 or pi produces identical real parts and opposite imaginary parts, making a negative-frequency interpretation unnecessary. It also illustrates how a 6 kHz tone under 8 kHz sampling aliases to 2 kHz with opposite phase, visible in PSD plots.


Re: mitigating phase shift

Reply posted 3 years ago (11/24/2022)
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

Reply posted 3 years ago (11/17/2022)
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

Reply posted 3 years ago (11/16/2022)
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...
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...
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,...
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);...
Hi,thank you I didn't know that!Best regards
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

Reply posted 5 years ago (08/07/2021)
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

Reply posted 5 years ago (06/15/2021)
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...

Re: Single Sideband Demodulation

Reply posted 5 years ago (04/28/2021)
Hi Neilthanks for your answers. I send to Stephane my wish.RegardsJosef

Single Sideband Demodulation

New thread started 5 years ago
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

Reply posted 5 years ago (04/20/2021)
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

Reply posted 5 years ago (04/20/2021)
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

Reply posted 6 years ago (10/28/2020)
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

Reply posted 6 years ago (04/30/2020)
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

Reply posted 6 years ago (04/25/2020)
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

Reply posted 6 years ago (04/24/2020)
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...
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

Reply posted 6 years ago (04/05/2020)
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...

Re: Simplifications With Eulers Equation (DTFT, DTFS)

Reply posted 6 years ago (02/12/2020)
Euler_2.pdf

Re: Simplifications With Eulers Equation (DTFT, DTFS)

Reply posted 6 years ago (02/12/2020)
I send you the file Euler_1.png

Re: Simplifications With Eulers Equation (DTFT, DTFS)

Reply posted 6 years ago (02/12/2020)
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: ADC sampling rate matlab/simulink M-PSK

Reply posted 6 years ago (12/30/2019)
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 ?

Reply posted 7 years ago (11/03/2019)
The plot is only a section at the beginning of the FFT !

Re: Extract pitch of irregular signal ?

Reply posted 7 years ago (11/03/2019)
The second frequency is just the reflection of the first frequency in the second Nyquist interval !

Re: Extract pitch of irregular signal ?

Reply posted 7 years ago (11/02/2019)
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...
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

Reply posted 7 years ago (10/08/2019)
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

Reply posted 7 years ago (08/12/2019)
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

Reply posted 7 years ago (08/12/2019)
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...
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...
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...
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...
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...

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