Questions tagged [signal-processing]

AKA digital signal processing (DSP). A signal is an information-carrying changing attribute of an entity, but in the digital sense, 'signal' refers to either received or transmitted streams/blocks of data, commonly representing real-world quantities such as audio levels, luminosity, pressure etc over time or distance. 'Processing' is the act of altering, analyzing or characterizing the data to retrieve/modify information inherent in the signal in question.

A signal is an information-carrying changing attribute of an entity, but in the digital sense, 'signal' refers to either received or transmitted streams/blocks of data, commonly representing real-world quantities such as audio levels, luminosity, pressure etc over time or distance.

These real-world quantities usually comes as analogue signals that are being sampled and quantized into a digital format, i.e. a sequence or higher-dimensional array of discrete numbers.

'Processing' is the act of altering, analyzing or characterizing the data to retrieve/modify information inherent in the signal in question.

NOTE: questions not related to the programming of signal processing implementations but the signal processing theory behind are off-topic for Stack Overflow and should be posted to the appropriate sister site, Signal Processing SE.

Common topics include:

  • digital filtering;
  • audio processing;
  • image processing;
  • sampling;
  • data compression;
  • spectral analysis;

Analysis tools commonly in practice are the Discrete Fourier Transform (DFT), especially its fast implementations (FFT); and wavelet transforms.

Common softwares used in the field include ; ; , especially with /; ///; ; and various languages.

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Is it possible to use 2 MICs in Android device independently?

I am working on a small audio processing project in Android using Java and Android Studio. I happen to request the usage of the 2 microphones built in the device, the default one is always going the be used, but the second microphone I was…
Galvar
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Reconstruct signal from FFT using Accord.net

I am trying to filter out signal noise using Fourier Transform. Using functions from Accord.net I was able to apply FFT on an input signal and reconstruct it. However I am unable get the correct phase of the signal. After hours of Googling, I…
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How to represent a square wave in python and how to convolve it?

I am trying to convolve a square wave with itself more than once and see the resulting graph. I know how to do convolution by my hands but I am not experienced in signal processing with Python. So my questions are: How can I represent a signal in…
Nope
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Comparison of Higher Order Ambisonics Signals

I'm working on a project with matlab where I have to compare two (Higher Order) Ambisonics impulse responses. I've calculated the FFT and plotted the smoothed energy of both signals over frequency: They seem to be correlated but I want to find a…
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spectrogram to power spectrum density

How do I compute the power spectrum density from the spectrogram? I found the following code but there is quite some difference when comparing to welch's method. Also I don't quite understand why I have to divide by 2 and nperseg? fs = 10e3 N =…
user307380
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Plotting Wigner-Ville distribution with pytftb on python 3

I am testing wigner ville distribution to see if it works for the estimation of original amplitude of a signal with noise. The pytftb provides a wigner ville function that works well with their examples. I use it like this: tfr =…
Danf
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Find equivalent digital filter

I'm trying to find an equivalent digital filter for a simple RC filter. The bode plots don't line up and I don't know why. ==================================================== import numpy as np import scipy.signal as sig import matplotlib.pyplot as…
dvc
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Using Matlab's sinc() in filter design?

In order to design a filter (windowing method), I first generate my sinc function, like so: L = 20; fc = 0.25; n = (1:L)'; my_sinc = sin(2*pi*fc*(n-L/2)) ./ (pi*(n-L/2)); my_sinc(L/2) = 2*fc; Then I apply window: win = blackman(L,…
Danijel
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Finding peaks in audio spectrogram

Introduction : I am working on audio fingerprinting and having some doubts regarding peak detection in the spectrogram, my input is a wav file with spectrogram as : The method I'm implementing is given here Problem : The peaks returning from the…
Shubham
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FFTW library: verifying fourier transform derivative property

I am trying to verify this relation with the fftw library: Therefore, I chose f to be a Gaussian, computed the Fourier Transform of its derivative and compared it with the one of the Fourier Transform of the Gaussian multiplied by ik. Here is what…
Krist
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How to quantify spatial distribution of signal in MATLAB

Let's assume I have 2 matrices, which are representing a signalling with 1 and 0 in a 2D space. A = [1,1,0,0;1,1,0,0;1,1,0,0]; A = 1 1 0 0 1 1 0 0 1 1 0 0 B = [1,0,1,1;1,0,0,1;1,0,0,0]; B = 1 0 1 …
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.bss vs .data: static q32 vs static int

The following code is compiled and linked for a DSP target processor: Test.c: static q32 a[15] = { Q31(0.1f),Q31(0.2f),Q31(0.3f),Q31(0.4f),Q31(0.5f), …
Danijel
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plot ECG signal in python from a .wav file

I plotted a an ECG signal form a .dat file using this code : import wfdb record = wfdb.rdsamp('mitdb/100', sampto=3000) annotation = wfdb.rdann('mitdb/100', 'atr', sampto=3000) print(type(record)) wfdb.plotrec(record, title='Record 100…
Abyr
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How to get a accurate estimation with Matlab function snr()?

I tied the Matlab function snr which is part of the signal processing toolbox since R2013b. As described in the Matlab documentation this function accepts the power spectral density (PSD) estimate as input. I though: 'This is cool! This means that I…
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Normalize the output signal to the same level as the input signal

I have a beamforming algorithm,and I use matlab to simulate data so that I can estiamte 'ratio of output signal-to-interference-plus-noise ratio (SINR) to input SINR (Like Matlab Does) . agCbf =…
马慧超
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