Linked Questions

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 ...

I'm currently using MATLAB's fdatool for filter design. Using that tool, I can easily design different kind of filters. For example, let's take a band-pass FIR filter with 10-40 Hz passband, and 5-10 ...

I'm very new in digital signal processing. I have multiple sensors and the way I filters the signals in post processing is: take FFT of the signals. put zero on out range of interesting frequency (...

I have collected a set of physiological signals through emg sensors. I will link at the end of the post an example (10 sec.) of a signal collected on zygomatic muscle. When I analyzed these data, I ...

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 ...

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, ...

Say I'd like to bandpass filter a signal with corner frequencies 1 kHz and 2 kHz (i.e., passes signals between 1 kHz and 2 kHz and suppresses other frequencies) using software like MATLAB. Intuition ...

Am trying to remove noise for certain frequency bands by attenuating freqeuncy bins in the FFT array. However this results in artifacts to be formed, where there are clicking sounds in between each ...

I am new to signal processing domain. When I run fft on data of length 1000, I get 1000 complex numbers. Now, if I want to extract the low frequency information or signal approximation, I take the ...

I have a signal, shown in the attachment (blue one). As you can see that the latter part of my signal has a higher dc offset. I want to bring the signal into the same level. I have tried eliminate the ...

I am trying to learn about signal processing and fft's applications within it. My understanding is that a classical use case for fft is to remove "noise" (ex. high frequencies that are not supposed to ...

I've worked a fair amount with EEG signals, though I've never had formal training in signal processing, so please excuse my ignorance. The problem is this: my signal has noise at many, many ...

I thought that DSP would be done by using FFT of portions of a signal, modify the samples that result from FFT (since they represent the spectrum of our signal + noise) and remove any unwanted signals ...

I don't quite understand why the textbooks say it is impossible to implement an ideal low pass filter. If I was to take the FFT of a discrete signal x[n], with Matlab's fft function I'd be returned ...

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 ...

In DSP book by Proakis and as well as in this pdf, it is mentioned that practical causal digital filters cannot have an infinitely sharp transition from Pass-band to Stop-band. Why is it so? Can you ...

I'm trying to remove noise from an audio file. This audio file contains speech as well as constant pink noise. I know that I have to use the Fourier transform to convert to the frequency domain then ...

I was explaining to someone how Fourier series work in context of constructing signals that are not everywhere differentiable, e.g. square waves, sawtooth waves, etc. When I mentioned the Gibbs ...

This is my first question at this site, please excuse me if I'm not asking the question properly. I have an acoustic signal of a test and I want to remove the "thin" peaks, because they don't really ...

I have a bunch of audio files all sampled at 44100 Hz sample frequency. I am trying to remove all the frequencies which are outside the human hearing range (I use the following as reference: Frequency ...

What do you think of this notch / band reject filter for a very very narrow band ? ...

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*...

Endolith's recent question on zeroing FFT bins got me thinking about a possible error in my workflow and I'd like to double check. Consider a signal $x_i[m],\ m\in\{1,\ldots,M\}$, i.e. $M$ samples ...

Connected with this topic , I would like to have some clarifications about the Fourier Transform and the Python tools that make it. Having my data set whose plot is showed in the above link, my array ...

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,...

E.g. the constant Q-transform is built by adding so called "Fourier filters". What's a "Fourier filter"?

I have a non-periodic signal collected from a force plate, it represents the foot/ground contact of a human when running. My aim is to calculate the signal power of each harmonic up to 50Hz. I will ...

I'm using a sensor with output that looks like the following figure. The cyclic noise is common to ask measurements made by the sensor. Is there a way to remove the noise without negatively effecting ...

My question involves proper FFT Technique for AM demodulation, in the special case where the modulation rate exceeds the carrier frequency. This question has a specific hobbyist application, the ...

I am currently working on functional connectivity analysis of EEG, and need to bandpass filter my data into different frequency bands (Delta, Theta, Alpha and Beta). An important thing is that the ...

In signal processing, I have heard of two terms: Using band-pass filter to extract some frequency bands. Using Discrete/Fast Fourier Transform (DFT/FFT) to extract some frequency bands. Are these ...

So here's the idea: you can generate a spectrogram from an audio file using shorttime Fourier transform (stft). Then some people have generated something called a "binary mask" to generate different ...

I have a real one dimensional signal s (light absorbance in a flow cell), which has significant noise and some periodic noise after performing a deconvolution of $S$ from $S_o$. Basically fft($S$) was ...

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 ...

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,...

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 ...

I don't really understand FFTs, and filtering neither. I hope I can explain my confusion in a way that is not too confusing for you to understand. consider the following example: I have an EEG signal,...

I have programmed a anti-aliasing filter that uses the fourier-transformation and makes a perfect cutoff on my desired cutoff frequency for downsampling without aliasing. I have read about other ...

I'm applying an ideal filter on the frequency domain. My code below: ...

I'am pretty new to this topic so i dont have much experience with DSP. I want to filter(highpass) a WAVE file, i have already programmed the FFT, and now want to filter the FFT vector via convolution (...

So if I were to have an $n$ length sample of a given sample rate and were to run it through an FFT, resulting in $n$ complex elements. How would I obtain the frequency bins, such that if I know the ...

If I want to amplify a certain frequency in a digital signal, can I just take the FFT, increase the real part of the value for the frequency I want (leaving the imaginary part alone), and then take ...

It has been so long since I studied this at University, so I'll better ask here if I'm doing something really wrong here. I'm using a USRP frontend for L1 band surveillance. I want a 20MHz bandwidth, ...

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 ...

I have a data signal (column vector) with N values taken over at specific sample rate (dx) at a fixed frequency. After taking the Fourrier transform of this signal and representing it in the Fourier ...

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 ...

The topic of zeroing frequency bins before IFFT has been discussed here : Why is it a bad idea to filter by zeroing out FFT bins? It was very helpful to understand why it is a bad idea. What do you ...

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 ...

first I need to mention I'm new to signal processing. here is the situation: I have an acceleration time-series derived from an accelerometer I wanted to imply a filtering method like high pass ...

I have a strongly aperiodic signal and I get weird artifacts after running a fft, using a low-pass filter, and then doing an inverse fft. I believe the artifacts are due to the way the signal is ...