All Questions

-1
votes
0answers
8 views

Why do we need to write the noise as randn(2,1)./sqrt(2*SNR) instead of randn(1,1)/sqrt(2)?

I saw this code, about the error probability of on/off signal ,on the internet ...
0
votes
0answers
4 views

Spectrogram of chirp sequence radar signals

I am confused as to whether SFFT makes sense to process chirp sequence signals. The radar fires up a sequence of 100 linear ramps with that are only interrupted by the very short period in which the ...
0
votes
0answers
5 views

FIR band pass filter desiging using PSO technique

Can anyone help me in writing matlab code for optimization of filter coefficient of FIR band pass filter using PSO algorithm?
0
votes
0answers
6 views

What is a good book/tutorial that explains Viterbi decoder for convolutional codes?

I want to write my own code in MATLAB for Viterbi algorithm to implement a digital communication system in which BPSK is used along with channel coding (or FEC). The conventional books/tutorials ...
0
votes
0answers
3 views

Frequency scale in fft after zero filling (odd/even number of data)

If we have a spectroscopic data which consists of time versus light absorbance "A".There is another set of control data "B". Both data are sampled at the same rate ${Fs}$. The two sets are to be ...
-1
votes
0answers
5 views

Demodulation using orthogonal signaling

I am a beginner and I want to implement a demodulate function that takes in a signal (the received waveform), the number of bits per chunk k, and the length T in seconds of each signal (which is also ...
0
votes
0answers
6 views

Orthogonal Signal Modulation

I am a beginner and I want to implement a modulate function which takes the bitstring (the data), the number of bits per chunk k, and the length T in seconds of each signal (which is also the ...
0
votes
0answers
4 views

Kalman filter: error covariance growth for multiple consecutive update steps

I have a constant velocity model for position estimation: $x_{k+1} = \begin{bmatrix}1 & dt\\0 & 1\end{bmatrix}x_k + \begin{bmatrix}0\end{bmatrix}u_k$ and $x = \begin{bmatrix}p\\\dot{p}\end{...
0
votes
0answers
10 views

Deconvolution of two FIRs

Basic questions: What's the "correct" way to deconvolve two causal FIRs in the frequency domain (i.e. using the FFT), neither of which may be minimum phase but both may be considered to have stable ...
0
votes
0answers
4 views

Comparison Between Mahoney Filter and Kalman Filter in Euler Angles (Yaw, Pitch, Roll) Estimation

What are the advantages of Mahoney Filter over Kalman Filter for Euler Angle (Yaw, Pitch, Roll) estimation? Could anyone compare them for Angular State Estimation? Resources Mahoney Filter ...
0
votes
1answer
18 views

Is it possible to do better than Cramér Rao lower bound for different estimation methods

I’m new to learning about the Cramér Rao lower bound. Does the calculation of the CRLB imply that only particular estimation algorithms can be used (e.g. estimating delay and Doppler from a complex ...
0
votes
0answers
7 views

Parity Check matrix from degree and check nodes polynoms, LDPC codes

I'm trying to simulate LDPC codes which were optimized for the BEC channel. I generated the degree and check nodes polynomials according to chapter 3.15 in "Modern Coding Theory" by Richardson and ...
0
votes
0answers
17 views

Linear Combiner Based LMS Transfer Function

I am currently working on narrowband beamforming and looking to compute the transfer function for a linear combiner based LMS. While doing a quick search I have found the following article which ...
0
votes
1answer
18 views

Time Setting of $z$ and Laplace Transforms

I'm aware that the z-transform and the Laplace Transform have an analogous relationship but I want to be doubly-sure that the z-transform only works in discrete-time and that the Laplace transform ...
-2
votes
0answers
24 views

please help with this QUESTION [on hold]

Explain how the data rate over a channel can be increased, without increasing the bandwidth? What is the disadvantage of this approach?
0
votes
1answer
14 views

Differences in moving window rms due to sampling rate

I have a 10 second recorded signal band pass filtered from 100-500 Hz. The original sampling rate is 10 kS. Now I run a moving window RMS with a window length: 10 ms on the signal (10/1000 * sampling ...
0
votes
0answers
18 views

Intercepting a Bluetooth Signal

Does anybody know how to intercept a Bluetooth signal to disrupt it and or take over control of a speaker?
0
votes
1answer
7 views

Spectrogram power/magnitude normalization in analogy with image intensity normalization

In analogy with the 8-bit grayscale image color depth, in which the intensity of a sample pixel is normalized to a value between 0 and 255 by taking into account the minimum and maximum intensities ...
1
vote
2answers
48 views

Question About Sampling White Noise

Assuming that we have a continuous signal $v(t)$ $$v(t) = \int_0^t g(u)\, \mathrm du\tag1$$ where $g(t)$ is white noise.Then take the derivative of it. $$\frac{\mathrm d}{\,\mathrm dt} v(t) = g(t)\...
-1
votes
0answers
21 views

Is tanx(t) stable?

I am highly confused about the stability of tanx(t). If we provide a bounded input such as x[t]=u[t], the output is bounded I have confusion about stable Is the function BIBO stable and also is it ...
0
votes
1answer
17 views

confused in using modulation technique

I am given a question in which I have to draw the spectrum of the signal $m(t)$ and then draw the spectrum of this modulated signal. The signal is $$m(t)=\cos(1000\pi t)$$ Now in solution manual ...
0
votes
0answers
8 views

How do I interpret the Wavelet scattering transform frequency plane and tree diagrams?

I've been reading through this paper on "Invariant Scattering Convolution Networks" and am having a difficult time interpreting the tree diagrams and frequency plane diagrams shown respectively below. ...
0
votes
2answers
40 views

What is the interpretation of Fourier Transform containing only imaginary part?

The FT of a unit step function is taken as: $$ X(\omega) = \int_0^\infty e^{-j\omega t}dt = \frac{-1}{jw}e^{-j\omega t} \Biggr |_{0}^{\infty} = \frac{j}{\omega} $$ The transform only has the ...
0
votes
0answers
14 views

How to save a specific frequency range as a wave file?

When I use the bandpass filter on an audio file to retain, say, all frequencies between 1300-1400 Hz only, I end up with an audio spectrogram showing this bit and all the rest as black. Exactly as I ...
0
votes
1answer
29 views

Why is the ROC of Laplace transform independent of imaginary part of s?

An integral is defined as converging if it yields a finite value upon application of limits of integration. It is divergent otherwise. Now sticking to the mathematical notation of Laplace transform, ...
0
votes
1answer
22 views

What is the name of this filter?

I am having this simple function which acts on two state variable $s_t$ (states being $\{0,1\}$, boolean in sense), and I am evaluating it in discrete time steps. The function is $f_t=1$ if $s_t$ has ...
0
votes
0answers
9 views

Symmetric Capacity of a Binary Discrete Memoryless Channel

While reading this paper https://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=5075875, I found that the the symmetric capacity of BDMC $W$ is given by $$I(W) = \sum_{y\in Y}\sum_{x\in X}\...
0
votes
0answers
15 views

How to find the filter coefficient of matched filter to detect the pulse of a signal?

I would to understand how to find the coefficient of a matched filter. Let's consider we have a signal and the signal is buried in the following noise. we want to find the coefficient of a matched ...
0
votes
0answers
15 views

Calculating N for DFT

We want to compute an N-point DFT of a one-second duration compact disc (CD) audio signal x[n], whose sample rate is $f_s = 44.1Khz$ with a DFT sampling of 1 Hz. (a) What is the number of necessary x[...
0
votes
0answers
27 views

PLL carrier recovery: intermediate versus baseband

When looking at carrier recovery, a PLL is often applied to correct the frequency offset between the local oscillator and the transmitter after down conversion. I'm wondering what happens to the PLL ...
0
votes
0answers
6 views

How to match resolution of two or more signals

I have received the following 4 time-series signals and wanted to do some data analytics using these. Although the sampling is the same for all of these signals, 2 samples per second, problem is that ...
0
votes
0answers
7 views

Calculating noise floor of digitization system using FFT

I am trying to calculate self noise of a digitization system with differential inputs.For this purpose, i select a differential channel, set input range to +/-5 V, short both differential inputs and ...
2
votes
2answers
152 views

Do Fourier frequencies actually exist in real life in form of “instantaneous frequency”?

For me, this is a very awkward question to be asked, as at this point in my studies I'm supposed to be quite expert with elementary mathematical tools like Fourier transforms, but this has always ...
0
votes
1answer
9 views

MATLAB & Power Spectral Density - Error - Getting the same range of values regardless of sound type?

I wrote code [attached at end of question] to take .wav files of child vocalizations and tried to see if there are valid ranges for the formants F1:F2:F3 for the 5 ...
0
votes
0answers
12 views

Mscohere in Matlab-why would the coherence tends to 1 as nfft increases?

Does anyone have a clue as to why -using mscohere- the coherence becomes exactly one for all frequencies if nfft is set to about 190-200. I know it's standard practice to set it to a number that is a ...
0
votes
1answer
31 views

rate 1/128 channel code for 5 bits of information

I need to construct channel code for 5 bits of information for a wireless channel. Possible code rate could be as low as 1/128 (very reliable). At decoder I assume to use channel LLRs as decoder input ...
0
votes
0answers
12 views

Getting SNR of reconstructed signal in MATLAB

I am trying to run code to compare the efficiency of various basic sampling methods such as ideal sampling, natural sampling and so on. I want to then use average power to see how accurate the ...
0
votes
3answers
67 views

What is the precise definition of a nonlinear signal?

I tried to find the the precise definition of a nonlinear signal. However it is not easy. How can we define a nonlinear signal?
0
votes
0answers
16 views

Tetrahedral microphone array beamforming in MATLAB

I have a microphone array of 4 channels taken as channels [6,10,22,26] from Eigenmike spherical microphone array. I wish to do 3D beamforming and create $8\times 4$ beams, 4-elevations with 8 ...
2
votes
1answer
30 views

Odd-order Butterworth filter gain at cutoff frequency

Wikipedia gives the gain of an n-order Butterworth filter as $$G^2(\omega)=\left |H(j\omega)\right|^2 = \frac {{G_0}^2}{1+\left(\frac{j\omega}{\omega_c}\right)^{2n}}$$ here. If you look at the gain ...
0
votes
0answers
17 views

IFFT, negative frequencies and noncausal impulse

When using the IFFT to compute an approximation to the IDTFT, frequency range is 0 to 2$\pi$ versus $-\pi$ to $\pi$. This produces a different impulse sequence(causal I think). How do you calculate ...
0
votes
1answer
10 views

Fourier Transform of the Hilbert Transform of cos(t) (using Fourier time-shifting property)

If $x(t)=cos(t)=\frac{1}{2}e^{jt}+\frac{1}{2}e^{-jt}$, then $X(\omega)=\pi \delta(\omega-1)+\pi \delta(\omega+1)$. If $y(t)=cos(t-\frac{\pi}{2})=\frac{1}{2}e^{j(t-\frac{\pi}{2})}+\frac{1}{2}e^{-j(t-\...
1
vote
0answers
10 views

Understanding performance of Lucas Kanade Iterative Image Registration Technique

I am reading the paper: Lucas, B. D., & Kanade, T. (1981). An iterative image registration technique with an application to stereo vision. I am having some trouble understanding section 4.3 ...
0
votes
2answers
24 views

Does a phase rotation introduce a delay?

Suppose I have 2 complex time series, $x_1$ and $x_2$. If I measure the delay between $x_1$ and $x_2$, I get $t$. If I make $x_2 = x_2 e^{-j 2 \pi 10}$, and measure the delay between $x_1$ and $x_2$...
0
votes
0answers
6 views

Looking to simulate sound level meter (read microphone audio signal into 1/3 octave)

I am looking to pull audio from a microphone and analyse the 1/3 octave data. Can anyone suggest a software library to do this that I could then build a software interface on top of? Initially looking ...
0
votes
1answer
9 views

Comparisons of FIR Causal filters of type HRIR in Discrete Time and Frequency Domains

I have just implemented a Discrete Time Convolution between HRIR filter at a range of angles. I also implemented an Overlalp-and-Add method and FFT and IFFT to compute the same convolution in ...
1
vote
1answer
22 views

How to achieve a smoothed frequency phase response with no outliers

I have two sampled signals, which I then perform wavelet analysis on, by synthesising the Filterbank, perform the dft on each kernel, and the dft on the sampled signals, and multiply together. ...
0
votes
1answer
14 views

Correlation of vector signal

This is probably more of a mathematical question than a signal processing exclusive, but I hope somebody here knows an answer! I want to compute the autocorrelation of a signal, which is a vector (so ...
0
votes
0answers
11 views

Determining timing error for sampling

I am writing a small program for sampling a series of sensors on an Arduino. The sampling process is not 100% perfect, depending on how many sensors I am reading, the actual difference between two ...

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