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 come when I zero-pad a signal the fft's of the orignal signal and the zero padded signal are no longer concatenated around the same point?

I think i'm implementing the code wrong. I thoughts that the original and zero padded signal would be concatenated around the same point, with the same peaks. Is my understanding of this wrong or is my code the issue? clc;clear; N=257; %number of…
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Two versions of the same code return different results

I expect to get the same results when compressing these two MATLAB lines of code into one, but I don't! Code in 2 lines: [b,a]= butter(2,[0.4 0.6]) % Transfer function coefficients of the filter [A,B,C,D] = tf2ss(b,a) % State-space…
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One sample value in 16 bits 11025 mono WAVE data

I do recording of incoming raw wave data from mic. The FORMAT is supported by WebCamera and it's 16 bits per sample, 11025 samples per second, mono. The level of mic is on the very low.. just for purpose, to make sure the zero values are coming…
LolliPop
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Third octave frequency spectrum with python

I am trying to get a third-octave frequency spectrum of a time signal. The time signal is the acoustic pressure of rotational rotor noise which is harmonic. Its fundamental frequency is ff = n * N_b and for that reason, all frequencies should be…
Larzeb
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SciPy firwin2 has a low frequency error when designing arbitrary amplitude filter

I've been trying to design an arbitrary amplitude FIR filter which has a specific attenuation for each frequency. However I get an error at low frequency when comparing the results to my expected filter response as seen in the figure below. It…
Marcel
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How to convert sine wave form in to audio?

I have raw data: x-values y-values x being the time domain. Sample rate is 8000 Hz. I have plotted them in a sine wave, but can't seem to find any way to convert that into audio form. Can someone guide through this process?
Saif Ali
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Butterworth filter doesn't improve my neural signal

I have a signal from a (natural) neuron taken from a 16 bit ADC at 18.2 kHz. I'm trying to detect spikes in that signal. For those who are not familiar with this: when neurons fire, they give a specific signal: it starts with a steep rise (or fall,…
Nyos
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No points displayed in a graph

i was looking for a way to display the points in this graph
user10575790
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How to sound a signal in Python?

I have a signal in Python, and I want to hear it. This is possible?. The data is in the format numpy.ndarray. In Matlab one can use the command sound(data,f).
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Visualizing Audio File Waveform, Web Audio API

I'm having a lot of trouble trying to visualize the waveform of a audio file (.mp3, .wav) using canvas. Actually, pretty clueless on how to achieve this. I'm using the Web Audio API. I've been playing around with the code on here to achieve my goal:…
tkss
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normalizing mel spectrogram to unit peak amplitude?

I am new to both python and librosa. I am trying to follow this method for a speech recognizer: acoustic front end My code: import librosa import librosa.display import numpy as np y, sr = librosa.load('test.wav', sr = None) normalizedy =…
sabri
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arbitrarily weighted moving average (low- and high-pass filters)

Given input signal x (e.g. a voltage, sampled thousand times per second couple of minutes long), I'd like to calculate e.g. / this is not q y[3] = -3*x[0] - x[1] + x[2] + 3*x[3] y[4] = -3*x[1] - x[2] + x[3] + 3*x[4] . . . I'm aiming for variable…
Xpector
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Creating low-pass filter with PyAudio and scipy.signal

I've built a python class that has methods to read, record, save, replay, and display an audio file with PyAudio and Matplotlib. I now want to implement a simple low-pass filter. I've managed to put together a method that produces a low-pass filter…
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Add white-noise on image based on SNR

I would like to add white noise to an original image with different SNR levels, but not sure how to do. The original image is (256, 128) I am using acoustics package to add noise. original = cv2.imread(path) white =…
sealpuppy
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Comparing multiple signals for similarity

I have multiple (between 2 and 100) signals and need to determine when a significant number diverge from the rest. We're exploring machine learning techniques, but we also want tackle this as a signal processing problem and see where we get the best…
John Strong
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