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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How to interpolate/resample unequally spaced timeseries data in java?

I have a lot of movement data from a sensor of an android phone. The sensor-data is not equally spaced. The first column is the timestamp, the second to fourth are the values. 1476998233373060.00 -13.724.029 -0.32222483 0.6645355 1476998243378590.00…
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Sound is distorted after multiplying frequency spectrum by constant

I make a simple sound equalizer that operates in frequency domain and lets user to adjust frequencies in sound by using 4 sliders. The first one responsible for 0 - 5kHz, the fourth one for 15-20kHz. Steps are as follows: I read wav file and store…
mrJoe
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Explanation of Interpolate Hermite method

I've currently got this bounty running on how to resample audio data with the intention of increasing the pitch. Many solutions have been made and I have to admit I'm feeling a bit overwhelmed by the choices and information. I was directed to this…
Eric Brotto
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python SIGTERM handler is not invoked in multiprocessing

I replicated the following snippet to reproduce an issue I am facing while dealing with SIGTERM handler: a main process, with a SIGTERM handler. a thread spawned from main process a subprocess spawned from the above thread The intention is to…
Barun Sharma
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Display BPSK signal waveform in MATLAB

I want to display a signal waveform passed through AWGN channel,so I followed these block diagrams and referenced this website then finished this program. (http://drmoazzam.com/matlab-code-bpsk-modulation-and-demodulation-with-explanation/) I sent…
Ryan
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Perform FFT for every second on wav file with Python

I have code which performs FFT on a 5 second wav file. I'm not good at Python so I wrote very basic code which splits the wav file and calculates FFT for every second. Is there any more convenient way to do this? I'm also not sure if they show every…
Selim Turkoglu
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Drive an DAC from a stream that is clocked from another source?

My DAC is internally clocked at 48khz, I need to feed it from a stream that is also 48khz but is clocked from a different source. The two sources will drift over time and any buffering in between will inevitably overflow or underflow. Unfortunately,…
JeffV
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Continuous to discrete using bilinear interpolation in Python

I have a 1st order low-pass filter (LPF) in the frequency domain and I want to digitize it. I am comparing the frequency response graphs for testing, but I'm getting weird results... Though very basic, I couldn't get it right from reading the…
Malcolm Rest
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Swift 4: Detecting strongest frequency or presence of frequency in audio stream.

I am writing an application that needs to detect a frequency in the audio stream. I have read about a million articles and am having problems crossing the finish line. I have my audio data coming to me in this function via the AVFoundation Framework…
Gregg Harrington
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Curve-Fitting over an integral containing both a data array and function in python

I have a data set described by an integral with unknown constants that I am attempting to determine using python's curve_fit. However, the integrand contains a function being multiplied against a data set def integrand(tm, Pm, args): dt, alpha1,…
J.zendejas
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My note detection algorithm is failing on few cases?

I am using a simple approach to find out the musical note using FFT in python steps involved are: Reading the sound file(.wave) Detecting silence in the file(by computing square sum of squared elements of input falling within the window) Detecting…
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Scipy resample for downsampling, fourier method explanation

I am seeking explanation of what "Fourier method" is that is described for resampling in documentation of resample() method. I want to downsample an array. I know what decimation is and how it works, I also know what fourier transform is. I cannot…
chess
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How can I apply BandPass filter to EEG data being plotted in Processing 3?

I am trying to apply a 4-35Hz BandPass filter in processing. I explored the Sound library but I think it is strictly limited to audio files. Is there any library for fast implementation of signal filtering in Processing? I found this--…
Abstract123
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Total power of sine in matlab

i'm trying to use pwelch orperiodogram to get the power of a signal at different frequencies. As a test i tried a single sine: sqrt(2)*sin(x) It should has a power of one. However, my script shows different values in the plot and for the total…
Marvin Noll
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How to reconstruct the signal by applying thresholding on wavelet coefficients?

I want to reconstruct a 1D audio signal by thresholding the wavelet coefficients. First of all, I read the audio signal and normalised it. After that I added white Gaussian noise. Subsequently I calculated the maximum volume level for…