Linked Questions

0
votes
2answers
744 views

FFT and IFFT : filtering frequencies

I am sampling at $30 Hz$. I have a sample of $N=150$. Here is an example of how my sample looks like, which starts at 54 and ends on 203 on the X Axis I am interested in observing frequencies ...
1
vote
3answers
3k views

FFT of any sinusoidal signal — how many point fft to take?

suppose I have to measure the DFT of an 60 hz sinusoidal signal and calculate upto 20 harmonics of this. Value of the 20th harmonic will be 20*60 = 1200 HZ. Then my sampling frequency should be >= 20*...
1
vote
2answers
796 views

Jack buffer size and FFT frequency resolution

In my actual Jack Audio setup (almost default values), Fs = 48kHz the buffer size is 1024 samples. I'm planning to do some filtering (lowpass, highpas, ...) and ...
8
votes
2answers
6k views

Which image filter can be applied to remove gridded pattern from corrupt jpegs?

I have around 1,400 jpegs that have been corrupted somehow and have lost the backup images. They all seem to have the same gridded pattern of lines over each (i.e. the gridding does not shift from ...
0
votes
0answers
684 views

How to implement a time-varying filter?

I'm working on a 10-second sound, sampled at 44.1 khz. I want to do filtering, and have a desired EQ (equalization) curve that varies over time, as suggested here (here $f0=250\ Hz$) How to ...
1
vote
0answers
263 views

Zero STFT bins (and not FFT bins)

Filtering by taking FFT, zeroing bins, and inverse FFT is a bad idea, as discussed here. But what about: take a STFT, (i.e. multiply the input signal by moving window function, and take FFT) zero ...
1
vote
1answer
226 views

Why do we not use square filters without windows? [duplicate]

This question may sound like "why do we use window functions", but it's not. So here it's: What we currently do: If we have a function $g(t)$ and its Fourier transform $G(f)$, and a window $W(f)$ and ...
2
votes
1answer
10k views

Delete frequency components with FFT and IFFT [duplicate]

I have a signal in Matlab defined by signal and t. As it is a noisy signal, I want to delete every component below a certain ...
0
votes
1answer
285 views

Beginner's question on LPF, Need for Sinc and windowing [duplicate]

As a student who has never constructed any real world DSP application, I am inclined to ask the following. Convolution in time domain is multiplication in frequency domain. For low pass filtering, ...
1
vote
2answers
227 views

MATLAB $\tt butter$ function

I suspect I have a very basic misconception so I'm trying to clarify that here: A signal should be filtered before it is sampled, correct? That is, I always sample an analogue signal at some sampling ...
0
votes
1answer
184 views

iZotope RX's FFT processing?

How does iZotope RX handle the FFT (i.e. how does it manage to perform those processes in the frequency domain without creating glitches) since its able to e.g. Apply another DSP process (even time-...
0
votes
2answers
2k views

High-pass filtering pre-processing before computing audio features

I want to compute acoustic features on a set of audio files. These features can be classical audio features as spectral flux, but also acoustic indices as proposed in the seewave R package ). I need,...
0
votes
2answers
3k views

What's a “Fourier filter”?

E.g. the constant Q-transform is built by adding so called "Fourier filters". What's a "Fourier filter"?
0
votes
2answers
1k views

Filter to add 3dB per octave?

If I want to add 3dB per octave to signal (e.g. to flatten out the power spectrum of an exponential sine sweep), is it really just as simple as... Take the Fourier transform of the time-domain signal,...
0
votes
3answers
258 views

What is the minimum number of time points recommended for accurate instantaneous phase estimates using the Hilbert transform

I have two time series. The two series look like they follow the same function, but one appears to lag slightly behind the other. I would like to determine if there are phase differences between the ...

15 30 50 per page