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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Audiokit Cross Fade sound for loop

I am reproducing a sound through AudioKit player. It is a rain sound of 1 minute. I would like to reproduce it for long timing using loop property = true. I would like to reach the best possibile experience for my user, but at the moment they can…
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Why does scipy.signal.welch suppress zero frequency?

I try to use whelch method and I found that zero frequency is abnormal import numpy as np import scipy.signal as signal import matplotlib.pyplot as plt n = 100000 s = np.ones(n) f, psd = signal.welch(s, return_onesided=False) plt.plot(f,…
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Listening AVAudioPCMBuffer

I've implemented installTap method, which provides me audio buffer float samples. I've filtered them by my C++ DSP library. I want to "send" this buffer to headphones/speaker. I've did AVAudioPCMBuffer again from samples. Anyone know how to do that?…
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Strange behaviour of cwt at the end and the beginning in matlab

I have been experiencing with the continous wavelet transform (CWT) in matlab R2016a, and found something strange. When I do a CWT to a constructed signal, and plot the logarithm of the absolute value of the coefficients, the figure shows, that at…
Horváth Dávid
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Need explanation how specgram function work in python (matplotlib - MATLAB compatible functions)

I'm working on converting my code from python to objective c. Inside matplotlib.mlab.specgram function I see 3 important functions before fft : result = stride_windows(x, NFFT, noverlap, axis=0) result = detrend(result, detrend_func, axis=0) …
hoangpx
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Play sound in octave with pulseaudio

I would like to play sound in octave with pulseaudio. I saw this post but the octave audio management is changed in the new releases and it is not longer possible to use paplay like say that post, I have the version 4.2.1-4 Now when I try to use…
Andrea993
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How to capture the microphone buffer raw data?

I'm needing to capture the raw data (every few miliseconds) that the microphone provides. For preference on Python, but it can be in C/C++ too. I'm using Linux/macOS. How do I capture the audio wave (microphone input) and what kind of data it will…
denisb411
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Frequency domain phase shift, amplitude, hope size and non-linearity

I am trying to implement a frequency domain phase shift but there are few points on which I am not sure. 1- I am able to get a perfect reconstruction from a sine or sweep signal using a hanning window with a hop size of 50%. Nevertheless, how…
papaya
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Why does python's scipy.signal.dimpulse introduce delay in impulse response?

Consider a simple linear, time invariant system of the form: y_k = c*y_k-1 + (1-c)x_k The impulse response of this system can be computed by either dimpulse or by applying lfilter to a vector composed of a one followed by zeros: import scipy.signal…
rhz
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Apply frequency sampling filter on audio file

I am trying to apply a double-bandstop filter using frequency sampling from a given data file. The method I am using is as follows Inverse fft (ifft) of the given frequency sampling file Circular shifting of the real values given from step…
con
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Error while importing scikits.talkbox

I want to use scikits.talkbox, but i get the following error while import scikits.talkbox. Traceback (most recent call last): File "/home/seref/Desktop/machine learning codes/MFCC/main.py", line 3, in from scikits.talkbox.features.mfcc…
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What is the most efficient way to filter (smooth) continuous streaming data

I am in the process of making my own system monitoring tool. I'm looking to run a filter (like a Gaussian filter or similar) on a continuous stream of raw data that i'm receiving from a device (My cpu % in this case). The collection of data values…
HexxNine
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Scipy Spectrogram vs Matlab Spectrogram

As in my last question AttributeError: 'module' object has no attribute 'spectrogram', I'm writing a little python script which reads a Wav Audio File, computes the spectrogram and writes the corresponding data back into a Wav File again. Here's a…
Nico
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Implementing an FFT using vDSP

I have data regarding the redness of the user's finger that is currently quite noisy, so I'd like to run it through an FFT to reduce the noise. The data on the left side of this image is similar to my data currently. I've familiarized myself with…
fi12
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Using writehtk for feature extraction (Speaker Identification)

I am working on Speaker Identification and found this post on stackoverflow.com very useful. Although the code is working fine, I just had a small doubt: Code given in answer fRate = 0.010 * fs; .... writehtk(featureFilename, mfc', 100000,…
Sonal
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