Skip to main content
New
Stack Overflow Jobs powered by Indeed: A job site that puts thousands of tech jobs at your fingertips (U.S. only). Search jobs
17 votes
Accepted

Why is the convolution of two sine waves a sinc function?

You are not convolving two sine waves, you are convolving two short snippets of sine waves. A snippet of a sine wave is the same as a real (infinite) sine wave multiplied with a rectangular window. ...
Hilmar's user avatar
  • 45.3k
2 votes
Accepted

FMCW Radar - What's wrong with the transmitted signal model?

There are two things you want to change: The frequency sweep you expect is at baseband so do not include the carrier as part of your signal. The sample rate needs to be at least $2B$ of the chirp's ...
Envidia's user avatar
  • 2,546
2 votes
Accepted

IFFT caused inverted phase

TL;DR You should remove the conjugation of $W$ on the left half of your frequency vector and add it to your right half. For more insight as to why this has to be done read the rest of the answer. ...
ZaellixA's user avatar
  • 1,314
2 votes
Accepted

Mixer and control systems

If we define "frequency domain" as the Fourier Transform: multiplication in one domain is equivalent to convolution in the other domain. So multiplication in time is the same as convolution ...
Hilmar's user avatar
  • 45.3k
2 votes
Accepted

Is a bandpass filter superfluous when the frequency band of interest after applying the FFT is below the Nyquist frequency?

There was some confusion from the OP with regards to the need to low-pass the data because of potential aliasing post sampling. These have been cleared now. For the OP’s purposes of checking if the ...
Jdip's user avatar
  • 6,265
1 vote

Linear convolution of a 100 sample time series and a 20 tap filter in the frequency domain

So the linear convolution correct length is as you say: $$ N_t + N_h - 1 $$ where $N_t$ is the length of $t$ (e.g. 100) and $N_h$ is the length of $h$ (e.g. 20). This equation: $$ {\tt FFT}^{-1} \left ...
Peter K.'s user avatar
  • 25.8k
1 vote

Linear convolution of a 100 sample time series and a 20 tap filter in the frequency domain

I have never heard of "20 tap filter". Is that just a FIR filter with 20 coefficients? yes. (some functions for filtering make it a bit confusing what is the number of coefficients and what ...
Marcus Müller's user avatar
1 vote
Accepted

Impulse response synthesis in the frequency domain

You can break down your transfer function into two different parts: $\frac{j \omega \rho}{4 \pi r}$ is a differentiator and $e^{-j k r}$ is a pure time delay. If we were sloppy, we could write the ...
Hilmar's user avatar
  • 45.3k
1 vote

Freq domain filtering in a sample by sample basis

The standard way of implementing an FIR filter in the frequency domain is the overlap-add (or overlap save). See for example: https://en.wikipedia.org/wiki/Overlap%E2%80%93add_method In your case this ...
Hilmar's user avatar
  • 45.3k
1 vote
Accepted

Frequency domain derivation of Hilbert transform of $\cos(\omega t)$

Since amplitude is not specified here, let's assume that an arrow in the figures on the right-hand side corresponds to the time-domain signal $\frac12 e^{\pm j\omega_0t}$. So the signal corresponding ...
Matt L.'s user avatar
  • 90.3k
1 vote

Calculation of frequency correlation function based on power delay profile

The PDP tells us the delay characteristics of a multipath channel, and the result in the frequency domain (as the Fourier Transform of the PDP) is telling the coherence between different frequencies ...
Dan Boschen's user avatar
  • 51.9k
1 vote

Estimating acoustic sound signal energy radiated

Can i use this data to find Pressure vs time domain ? If the data is in dBSPL and properly calibrated, then yes. Otherwise no. How do i then isolate Pressure vs time domain data by subtracting ...
Hilmar's user avatar
  • 45.3k
1 vote

Estimating acoustic sound signal energy radiated

Can i use this data to find Pressure vs time domain ? unless your "dB vs time" is already coming from a calibrated system, and dB refers to some specific physical sound pressure: no. In ...
Marcus Müller's user avatar
1 vote
Accepted

What does the frequency domain of bipolar Alternate Mark Inversion (AMI) look like?

Calculating the power spectrum of the AMI code is actually a quite complicated matter, but the fact that the power spectrum has a zero at DC seems intuitively clear, as already mentioned in RBJ's ...
Matt L.'s user avatar
  • 90.3k
1 vote

What does the frequency domain of bipolar Alternate Mark Inversion (AMI) look like?

Okay, so "Google is my friend" and I found a YouTube. Bits that are zero contribute nothing to the DC component. If the number of ones in a message is even, it's as positive as often as it ...
robert bristow-johnson's user avatar
1 vote

Filtering in time domain vs filtering in frequency domain

QUESTION : Why is the output signal shifted in time domain? Is this because multiplication in frequency domain corresponds to convolution in time domain? A windowed sync is non-causal. In order to be ...
Hilmar's user avatar
  • 45.3k
1 vote

How can I convert a 1/3rd octave spectrum into a linear spectrum for use as a frequency response?

This is difficult because a 3rd octave spectrum is missing a lot of information which needs to be filled in with "good" assumptions. For starters you need to determine who exactly that ...
Hilmar's user avatar
  • 45.3k
1 vote

How can I convert a 1/3rd octave spectrum into a linear spectrum for use as a frequency response?

There aren't a lot of possibilities for this. Probably (>95%) either ISO-3 or IEC 61260 bands were used. Some googling will give you the (nominal) band centers and edge frequencies. The difference ...
Max's user avatar
  • 2,368
1 vote

Does subtracting a phase from the frequency components of a DFT output result in rotation?

After several readings and experiments, I think that the best way to rotate data exploiting the phases is to use the DFT shifting theorem. By identifying the number of shifts required to align the ...
Franck Might's user avatar
1 vote

Routh's stability condition

Most of your problems with this stem from confusion about terms and mixing up "criterion" and "test". Words matter! And in engineering and math, precise wording matters, so maybe ...
Marcus Müller's user avatar
1 vote
Accepted

Can we do down sampling by taking fft, omitting samples on the ends of the spectrum, and then taking inverse fft?

Suppose that the the maximum frequency content of the signal is low enough that we do not have any concern about aliasing. ... Can we take the FFT of signal, then omit samples from 1023-th index up to ...
OverLordGoldDragon's user avatar
1 vote

Frequency domain equivalent to center oriented rotated convolution?

The Polar FFT is about data in polar coordinate system. The operation you describe is not shift invariant, hence it can not be represented as convolution. If the number of rotations is small (Let's ...
Royi's user avatar
  • 19.7k
1 vote

Generate an Ideal 2D Low Pass Filter in MATLAB

First, I will replace some of the coefficients with 0.5 and 0.25. In order to understand this, please have a look at: The ...
Royi's user avatar
  • 19.7k

Only top scored, non community-wiki answers of a minimum length are eligible