# Linked Questions

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 ...
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*...
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 ...
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 ...
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 ...
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 ...
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 ...
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 ...
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, ...
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 ...
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-...
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,...
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"?
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,...
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 ...

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