0
votes
0answers
9 views

What could be causing these humps every 10 Hz on my PSD estimate data?

I am working with EEG recordings about 1,000,000 data pts long, recorded at 4,000 Hz. I'm generating PSD estimates for these recordings using the matlab "periodogram" function like so: ...
1
vote
1answer
10 views

Problem with Generalized Cross Correlation (PHAT filter)

I have to calculate time delay between two signals.I am using the GCC-PHAT method. First, I cut the signals into smaller parts (~0.5s) and then try to calculate the delay. When I display those two ...
0
votes
0answers
9 views
1
vote
1answer
22 views

DC component in accelerometer data - filter before or after XYZ Euclidean sum

I'm working on filtering accelerometer XYZ data. The aim is gait-pattern recognition via finding characteristic frequency points and feeding these to a artificial neural network. The trouble is, and ...
0
votes
1answer
21 views

Bandpass Filter Order vs Sample Numbers of Noisy Signal

I'm working on signal filtering in MATLAB. i wrote sample code that generate 50 Hz signal with some noisy frequency and use Kaiser bandpass filter to get only 50Hz frequency. this code work great! but ...
0
votes
0answers
14 views

Features of a non-periodic signal for comparison [on hold]

I am looking for comparing a signal (256 / 512 point discrete | non-periodic ) (obtained after extracting some features from the captured image) with a database of around 5000 signals (256 /512 point ...
0
votes
2answers
26 views

I want to know if there is a practical way to find out if multipath propagation is significant for audio in a given environment?

I am employing audio signals generated by a mechanical press for Blind Signal Separation. But, prior to algorithm application, I need to clarify if the signal strengths coming in multipath (except ...
0
votes
1answer
22 views

Complex Conjugate Sinusoids in Forward DFT

I hope this isn't such a dumb question, but I'm finally getting to grips with the inner workings of the DFT. What I'm having trouble understanding is why the basis complex sinusoids in the "forward" ...
0
votes
1answer
18 views

Specs for creating sample set?

I would like to create a set of audio samples for playback as a midi-controlled instrument. Thus: a sample for every key of the piano range, with multiple volume timbres for each note. I am able to ...
0
votes
1answer
18 views

Noisy signal filtering matlab

I'm currently working on rectifying a respiratory noisy signal shown below: I've already tried to filter the noise as you can see in the image below (full image): The red one is the noisy signal ...
0
votes
1answer
27 views

Accurate Image Resizing

I need to resize an image using bilinear interpolation and create an image pyramid. I will detect corners at the different levels of the pyramid and scale the pixel co-ordinates so that they are ...
0
votes
1answer
22 views

How are these pole zero plots created

Back around 5 BC, where C stands for the programming language, engineers would photograph their oscilloscope's display to record their test results. This appears to be such a photograph, taken from ...
0
votes
2answers
27 views

What does it mean for the DFT phase to be relative to a cosine wave?

The following paragraph from Understanding Digital Signal Processing got me puzzled: The answer is: The DFT phase at the frequency $mf_s/N$ is relative to a cosine wave at that same frequency of ...
2
votes
2answers
33 views

What do Skewness and Kurtosis represent?

I understand that the question could mean a lot of things but I am thinking specifically to image processing. For example, I know that the mean can be a basic texture feature that represents the ...
0
votes
0answers
17 views

Translating Bessel filter from python to C - odd behavior past the cutoff frequency

I recently translated the code from scipy.signal.bessel to C in order to implement a bessel filter in a piece of analysis ...
0
votes
1answer
30 views

Calculate Q factor of a Low Shelf and High Shelf filter

I'm developing an app that has an EQ using eq10q plugin. My eq has 10 bands. I'm using peaking filters for the middle bands and a low shelf filter for the first band and a high shelf filter for the ...
0
votes
1answer
21 views

Do I need the sampling frequency in order to gain periods out of FFT?

Currently I am using the transformInPlace() method from here. for a project at university. First of all I thought FFT would be the perfect solution for my ...
0
votes
2answers
45 views

Intuitive Interpretation of Fourier Transform in Beamforming

When applying Fourier transform in time domain signal will be carried to frequency domain. And when we apply Fourier transform in spatial domain signal will be carried to directivity domain. $$ ...
0
votes
0answers
8 views

Analytical expression for salient Blob detection in scale space

I am working on exact mathematical expression which can be obtained in image processing tasks.(I do not know much about image processing but my work is more mathmatical) In Gaussian scale space, we ...
1
vote
0answers
31 views

Flat top sampling to a step shape signal

Flat Top Sampling During transmission, noise is introduced at top of the transmission pulse which can be easily removed if the pulse is in the form of flat top. Here, the top of the samples are flat ...
0
votes
1answer
26 views

How frequency convolution theorem applied in this case?

I am referring derivation of modified periodogram. Following equation shows step that used frequency convolution and I am not able to understand why that $2\pi$ came in the next step? Can anyone ...
0
votes
1answer
28 views

Object detection in binary image

This task comes from tracking object on a steady background. So far I was able to remove the background and obtain binary masks like this: I need to get bounding rectangle of the toy without the ...
1
vote
1answer
41 views

Decoding final bits with viterbi

I'm writing a viterbi decoder and I'm rather confused about the correct process of decoding the final k bits for a kth order ...
1
vote
1answer
19 views

How to set threshold value for Hough Transform

I am new to Computer Vision, I am writing my own code for Hough Transform I have found the following algorithm: Algorithm: ...
5
votes
3answers
65 views

Acoustic Scenarios With Negative SNR

For all the practicing engineers out there: Where have you encountered a negative SNR? How negative, in dB, was it or estimated to be? Where would you expect to encounter a negative SNR? Note: ...
0
votes
3answers
40 views

Replace phases of image FFT

I have 2 images of the same scene, captured by different cameras. So they a bit displaced. As far as I can understood, phases of FFT contains info about image details. My goal is to take intensity ...
1
vote
1answer
28 views

Problems calculating Z-transform

I am trying to solve a class exercise in which I am given the following, in Laplace domain: $$G(s)=\dfrac{e^{-Ts}}{s+3}$$ $$H(s)=\dfrac{1}{s}$$ And I need to calculate $\dfrac{C(z)}{R(z)}$, which is ...
0
votes
0answers
23 views

How to understand the implementation codes in Debevec and Malik's HDR paper

The paper titled "Reconvering High Dynamic Range Radiance Maps from Photographs" from Debevec and Malik is a seminal paper in high dynamic imaging. In this paper, it proposed a method to estimate the ...
-1
votes
0answers
16 views

Plotting band-pass filtered signal of an audio in MATLAB [on hold]

I am new in MATLAB and try to plot some form of an audio file and I have managed to open and plot in time domain and FFT using the following code: Step I: Opening the Wave File ...
0
votes
1answer
15 views

How to set a Carrier Wave for DBPSK Modulators in gnuradio?

for a Project I need to build an Audio Modem in GNU Radio. I decided, that it would be the best to use DBPSK Modulation, because that is fast and relatively noise resistant. But as it seems, the ...
1
vote
1answer
33 views

Should the order of a filter be proportional to the sampling rate?

I have designed a fir filter with my desired frequency response at a specific sampling rate and filter order. If I were to redesign my filter for a new higher sampling rate, should I change the filter ...
0
votes
0answers
17 views

How to get real fft from complex fft calculation using butterfly Decimation in Time approach? [duplicate]

I have written a code to calculate fft using a butterfly structure algorithm. The input to the fft are sampled values of single frequency sinusoidal wave. I need to find real fft, using complex fft ...
0
votes
1answer
46 views

When does the convolution of $2$ signals equals zero?

For example, how can I determine if the convolution of $x(t)$ with $y(t)$ is equal to $0$?
1
vote
0answers
17 views

How do I find amplitude modulation effects in natural signals

It is relatively easy to discover overlaid (additive) signals in natural phenomena using wavelet decomposition, Fourier Transforms, CEEMD, etc, because addition in the time domain is the same as ...
0
votes
1answer
43 views

How many DFT bins must I use to get best precision when analyzing melody scale

I need to analyze some signals containing chords. How big DFT spectrum is needed to identify each possible musical note in such a signal sampled at 44100 Hz? I can see that 8192 bins gives some ...
0
votes
0answers
12 views

Edges in Content Based Image Retrieval (Java)

I have developed a programme that currently computes the standard deviation. I am trying to alter this algorithm so that it can extract the edges of an image. I have been told I can do so if I build ...
0
votes
0answers
26 views

What dictates the phase response of FIR filters when considering its zeros [on hold]

I assumed an even tap filter and i got a pole at $z = -1$ and an odd tapped filter (assuming the middle coefficient is zero) gave me poles at $z = -1$ and $z = +1$. This was confirmed by the theory ...
0
votes
0answers
24 views

Stopband Attenuation

Why is a stopband attenuation of -40dB preferred for the filter design & analysis. I understand at this magnitude response point, the o/p is just 0.01 of the O/P but how does it help in defining ...
0
votes
1answer
40 views

Output a sine wave from samples-limits?

Using a micro-processor of 48MHz, I am trying to output a sine wave from the DAC (digital to analog ), and I am not sure if my ...
3
votes
1answer
27 views

Can we determine if a filter is butterworth or chebyshev from its physical topology?

Can all analog filters be classified as Bessel, Elliptic, Butterworth or Chebyshev? Given a physical ladder topology of several stages of {L, C or LC} in {series or parallel}, is it possible to ...
1
vote
2answers
38 views

The relationship between downsampling and frequency resolution

I am having trouble understanding a concept that I have seen in many published papers. For example, in this paper (http://journals.plos.org/plosone/article?id=10.1371/journal.pone.0082748) they wrote ...
0
votes
1answer
12 views

Reconstruction of voice from distorted recordings

I would like to know how much of the sound of my voice getting through various types of walls can be reconstructed with easily obtainable software and algorithms. My plan is to place a microphone on ...
-2
votes
0answers
20 views

What is Stochastic process? [on hold]

I'm wondering if and how the understanding from Stochastic processes can be applied to the framework of Welfare Economics. I don't have a foundation in mathematics so please keep the explanations ...
0
votes
0answers
18 views

time correlation between two time varying signals [duplicate]

I have two time varying signals (sig1, sig2). I have to calculate the correlation between these two signals over time ( as shown in the figure).After all have to calculate a single number ( may be ...
3
votes
3answers
63 views

2nd order edge detectors more susceptible to noise?

I am reading about edge detection and I read that 2nd order detectors are more susceptible to noise. Is there a mathematical proof to this ?
0
votes
1answer
54 views

Plotting Power Spectrum (Matlab)

I want to make a plot of the power spectrum for a .wav sound file, over the frequency range from -2000 to 2000 Hz. Attempt: I used the following code, but the resulting plot is one sided. It doesn't ...
0
votes
1answer
21 views

Image averaging for impulse noise

I understand the mathematics behind the use of image averaging for gaussian noise. Why can't it be used for impulse noise ? Would it be because there is no way to prove that impulse noise is additive ...
1
vote
1answer
53 views

How can a signal have two maximum frequency components?

$x(t)$ can be exactly reconstructed from its samples at $\omega_s = 10 \textrm{ rad/sec}$. My conclusion is that the maximum frequency component in $x(t)$ is $5\textrm{ rad/sec}$. But I'm being told ...
0
votes
1answer
24 views

Behaviour of the poles in transfer function $H(s)$, given system properties

If a linear system is causal and its impulse response is an energy signal. What's the behaviour of the poles of $s$-domain transfer function $H(s)$?
0
votes
0answers
19 views

Problem with stft using matlab / python [on hold]

I am trying to translate the code from Python to Matlab. The original code is written by Assoc. Prof. Xavier Serra from UPF and is a part of the Sound analysis/synthesis tools for music applications: ...

15 30 50 per page