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I have a task in image processing, which need to detect vertical lines from matrix. For example, there is a pair of white and black vertical lines segmenting the picture below. I need to find it. So I want to find the lines labeled in red.
Actually, this matrix consists only -1(black), 0(gray) and 1(white), which is ... |
I am looking for a comprehensive explanation of Automatic Gain Control (AGC) for Radio Frequency (RF). I am unable to find any good resource online. Can anyone provide some pointers or an explanation of various blocks and types for AGC in RF?
Thanks.
|
My question is related to a theoretical part of the book: Digital Processing of Synthetic Aperture Radar Data, Algorithms and Implementation by I.G. Cumming et al. [Chapter 4. Equation 4.25 (for those who know the book)].
Basically what I undertand that is said there, is that if I have my transmit pulse waveform $s_{p... |
I am wondering about physical interpretation of energy / power for complex signals. E.g. for signal energy definition:
\begin{equation}
E=\int\limits_{-\infty}^{\infty}|x(t)|^2 \, \mathrm{d}t
\end{equation}
For $x(t)$ real, when representing voltage (or current) waveform, above definition would give energy dissipated a... |
I'm trying to understand how oversampling can help to reduce the signal to noise ratio, so I looked this document, and at page 6 it is said that:
Along with the input signal, there will be a noise signal
(present in all frequencies as white noise) that will fold or alias into the measured frequency band of interest (f... |
I have a real world system of a rotating back and forth motion and the time domain data suggests that the data is a random signal (not a deterministic signal). I recorded discrete data points at a constant sampling rate in my lab notebook from raw data recorded by a computer program at a sampling rate where I don't ha... |
In short, I am coding an optics simulation for an experiment I am running. There is a point where I perform a Fourier transform on my incoming signal and want to extract the phase; however, I get unexpected results.
More specifically, I want to unwrap the phase from the signal below:
To extract the phase I use the fol... |
For computing the Harris corner detector, I want to calculate $I_{xy}$ and $I_{xy}$. Obtaining $I_x$ and $I_y$ are clear for me. But what about the second order derivatives?
This and this and this one posts say I have to do convolution operation twice:
$$
\frac{\partial^2}{\partial x \partial y} f = \frac{\partial}{\pa... |
I am reading Digital Communication Systems by Simon Haykin and I am stuck at one point.
Consider a two-dimensional signal space that has a message constellation of four points, given by $s_1,s_2,s_3,s_4$. We need to calculate pairwise error probability in the presence of additive white Gaussian noise ($w$). The signal ... |
In this paper they preprocess a cent spectrum of an audio file.
For comparison of audio files they create the Logarithmic Fluctuation Pattern(LFP)(p.3).
For this I have to use a FFT on each frequency band of a block and then only take into account the amplitude modulations up to 600 bpm. (See screenshot)
I want to im... |
This is simulation data flow:
/*
Full wideband FX-lms schmatic
.....................................................................................
disturbance ----- noise
----------------------------------->| P |----------------------
| ... |
In a sigma-delta ADC, the sigma-delta modulator outputs a digital stream of 0's and 1's which are then low pass filtered and decimated. Where does the bit resolution of ADC come into the picture? What is the input to the low pass filtering operation? Remember the output of sigma-delta modulator is a stream of 0's and 1... |
Most auroral pictures are post-processed and enhance the aurora to such an extent that you don't see the background stars. Suppose I have a photo of the night sky with an aurora, taken with my phone or any camera. What method should I use to do the opposite - fade the actual aurora as much as possible and enhance the b... |
What will happen when sampling rate of an audio signal is changed? Does it changes signals duration or signals total sample or both?
|
Is there a difference between shifting a lowpass FIR filter to an arbitrary frequency offset versus creating an equivalent bandpass filter, such that the passband and cutoff frequencies are identical? What are the pros/cons of doing it one way vs the other?
from scipy import signal
from scipy.signal import firwin
impor... |
I am working on optical coherence tomography algorithm. In main pipeline I should use nonuniform resampling to eliminate transformation of optical signal in IR-spectrometer (that signal disperses by prism\diffraction grating almost linearly in $\lambda$-space, but in order to use further FFT we need to have signal line... |
I have daily temperature data from some place on Earth. Apparently, the sampling frequency f_s = 1/day.
I would like to use these data to demonstrate (visually) the use of low-pass, high-pass, and maybe band-pass filters in the context of Fourier transform and convolution via multiplication. For those, one typically ne... |
Whilst trying to compare the performance of three blind source separation methods, is it relevent to plot the parcimony of a signal in function of its SDR?
Thank you...
|
Can someone please provide an overview of sigma delta DAC? In particular, I would like to know how does the sigma delta modulator, which is digital, works for DAC. What is difference between sigma delta modulator for DAC and ADC?
|
I'm looking to composite an opaque object (e.g. 3D art) into video footage of semitransparent fog/clouds before a landscape background so that the object seems immersed in the clouds but occludes the background. Ideally this would mean separating the footage with the clouds and landscape into two layers.
What is the st... |
As the question states, is it possible to predict the peak value of a time-domain signal given its frequency-domain spectrum?
Since the time-domain signal is just the sum of the individual sinusoids making up the frequency domain spectrum (i.e. inverse Fourier transform), I have been reading some answers on estimating ... |
If I have two audio signals, say $A_{1}$ (music) and $A_{2}$ (Speech), I want to calculate the energy of the two signals for their relative comparisons. One method I can use is by adding the square of audio samples over its length as follows:
$$ E = \sum_{n = 0}^{N} A \left[ n \right]^{2} \tag{1} \label{1} $$
My doubt ... |
If I have a sampled cosine wave signal with Fc = 80kHz and Fs = 200kHz (so the Nyquist region is between 100kHz and -100kHz) and I want to preform a decimation by the factor of 2.
Will my input signal be outside of the new Nyquist region (50kHz to -50kHz) and I will see only the Fc - Fs = -20kHz product of the signal o... |
I'm slightly confused by various statements around energy / power of PCM audio. Specifically, I came across two seemingly contradicting statements in these two answers:
Answer 1 states: [...] you can formally sum the squares of the samples and call it energy [...].
$ E = \sum\limits_{n=0}^{N-1} s[n]^2 $
Answer 2 stat... |
First year student so please excuse my lack of knowledge.
As i understand i need to use convolution which is:
$$ y^{out}(t) = u^{in}(t) * h(t) $$
Or maybe my thoughts are wrong so please correct me.
But still i do not understand how to draw magnitude spectrum of output signal.
I did find images of convolution but still... |
I have a uniform PCM whose bandwidth is W = 1Hz and n = 12 bits, i.e. the transmission bandwidth B_T = n*W = 12Hz.
Given this features of the signal, I need to calculate the minimum transmission badnwidth B_T if the same PCM signal is transmitted using m-ary PPM with M=4 and M=8.
I don't yet understand how the choice o... |
I'm doing gesture recognition based on accelerometer and gyroscope data (based on a smartwatch). I recorded the following two gestures (a) rotation and (b) double-tap.
I now have difficulties determining the optimal cut off frequency for the low pass filter in the frequency domain. As you can see in the plots, for the ... |
In some textbooks and websites, circular shift operation is done before doing FFT of windowed data, in the explaination, circular shift is to ensure zero-phase. But in other textbooks and websites(I would say most), circular shift operation is ignored in the implementation. So what is the correct way to do STFT and Why... |
I found, as rule of thumb, that a system is causal and stable when it poles lies inside the unit circle.
However, more generally we should argue with region of convergence here, like in this example of Wikipedia:
https://en.wikipedia.org/wiki/Z-transform#Region_of_convergence
It's hard to see for me, how region of conv... |
I would like to understand why the second frequency component usually has the greater magnitude within the range [1, (N/2)], i.e why (I remove the DC component and the complex conjugates freq after N/2) when we're computing the FFT of a positive real-valued signal. Can I have a mathematical explanation of this phenome... |
Suppose I have a system transmitting at $8192 \ b/s$, using raised cosine with roll-off $\alpha = 0.5$. I am really confused if the symbol rate and the bit rate are the same for this case, i.e. $T_s = T_b = \frac{1}{8192} s$.
|
I'm currently reading a paper (Page 23 of the PDF) about the application of the Fourier transform to standardize some climatic data to easily compare them.
I have the following text :
2.3 Standardising dates
:
If we subtract the phase angle of the resultant of the first frequency
vectors for rainfall and temperature ... |
Background
I recently asked this question over on Electrical Engineering Stack Exchange. On the advice of some commenters there, I've broken off those pieces which are appropriate for asking as smaller questions here.
I'm attempting to write a physics simulation code, one portion of which involves the simulation of the... |
Background
I recently asked this question over on Electrical Engineering Stack Exchange. On the advice of some commenters there, I've broken off those pieces which are appropriate for asking as smaller questions here. Namely, I'm asking separately about modeling the signal chain, and a more accurate theoretical model o... |
I am attempting to implement a filter described in the paper Functional Count-Comparison Model for Binaural Decoding by Pulkki et al 2009:
Here is Figure 9:
However, I am quite certain that the equation for n is incorrect since the cosine is squared and there is a fourth root of n/fs. This equation does not produce a... |
We have the need to calculate the overlapping Allan deviation on possibly some billion data points (for long-running measurements). While alantools.oadev() is working for a certain workflow (up to 0.5 billion data points on a local computer with 16 GB RAM), it does not scale well. Are there any alternative implementati... |
The DFT of
$$
\cos(2\pi f t + \phi)
$$
peaks at $k=\pm f$ if $t = \frac{1}{N}[0, 1, ..., N - 1]$ (for integer $f$, & within Nyquist). What about other $t$? What if we double the sampling rate or duration?
How would one plug it into a sine DFT closed form solution? Given x = cos(2*pi*f*t + phi), would it be sine_dft(N, ... |
The polynomial here is impulse coefficients of minimum phase FIR filter, or it's impulse response.
A code somewhere tries to finding roots of polynomial on the circle. It have roots of polynomial ru and it's derivatives root rdu.
The algorithm to finding them uses weird merit to increase unit circle tolerance and that ... |
I have a buffer of N samples from which I'm playing a loop of L samples (L <= N) starting from sample S. The reading position is calculated as (S + P) % N where P is current playhead position which is advanced as P = (P + 1) % L. Adjusting S during playback I'm getting clicks and I'm looking for the ways to eliminate t... |
I am trying to determine stationarity from a somewhat stochastic process. Every 10 minutes, I collect a set of 50 FFTs, i.e., 1 trial over $50$ seconds, so an FFT occurs every time second. I understand the definition of weak-stationarity with the covariance only being dependent upon the lag interval, the mean and varia... |
A closed form solution to
$$
X[k] = \texttt{DFT}\{\cos(2\pi f t + \phi)\}
$$
confirmed many known properties of a finite sine's spectrum, also revealed new ones. Can the same be done for $|X|$, or $|X|^2$? Is there an expression for $|X|$ that's useful for analysis (not limited to one based on what's linked)?
For examp... |
I have a signal processing chain which goes:
Every 256 samples, take 512 samples of input buffer
Apply a Hann window to the samples
Copy the windowed samples into the real components of a complex array
FFT the complex array, obtaining complex spectral analysis data
Modify the spectral analysis data
... |
Inspired by this post, I tried to reproduce the procedure described in the answer in Python considering rectangular pulse:
import numpy as np
%matplotlib inline
import matplotlib.pyplot as plt
t = np.arange(-1, 1, 1e-3)
N = len(t)
x = np.ones(N)
plt.plot(t, x)
and exponential step response:
A = 1
tau = 100e-3
def ... |
I have obtained daily close price stock data ( and transformed them to logarithmic returns series data) from a Stock Exchange between 2 financial years, and I wish to generate a power spectral density to study its behaviour.
I work on Python , I went to the documentation of Scipy.Signal.Welch : https://docs.scipy.org/d... |
Is there any previous work describing the implementation of an Error Vector Magnitude (EVM) measurement system using GNU Radio?
The system in question is made of an SDR HW platform, e.g. USRP, providing digital I/Q signals, used as a source in GNU Radio sketch. The sketch goal shall be to decode symbols of a given modu... |
I need help understanding why FFT-based computation of convolution on a finite domain, between two non-periodic functions often gets it wrong. Specific questions:
(1) why does FFT-based computation have error when interval [a, b] is short in the specific example below but no error when interval [a, b] is wider ;
(2) ho... |
Let $$\dot{x} = Ax+Bu$$ $$y = Cx + Du$$ be a linear ODE with $x(0)=0$. Here, I am assume $A$ is invertible.
As you can see, the relation $$H:u(.) \mapsto y(.),$$ where $(u(.),y(.))$ is a solution to the ODE above, forms an LTI system. If I set $u(t) = e^{st}$, for some $s \in \mathbb{C} - \text{spec}(A)$, then the outp... |
Suppose you have a function that can be described as $$f(s) = \sum_{n=0}^{\infty} a_n e^{f_n s}$$ where each $f_n$ is a complex number. I am looking for a transform $T$ to act on $f$ which produces a function $T(f): \mathbb{C} \rightarrow \mathbb{C}$ such that if $f_n$ was one of the frequencies in our original $f(s)$... |
I am trying to use MUSIC algorithm to detect individual frequencies of an incident signal. I read the original paper and some interesting books that explain the algorithm. I used the following equation for estimating pseudospectrum.
The above equation (Eq. 8.162) is given in Statistical Digital Signal Processing and M... |
I have simulated 3 scatterers for FMCW radar signal processing using Range Migration Algorithm.
Background:
The Range Migration algorithm steps are as follows:
Applying Azimuth FFT on the raw data
Apply "Matched Filter"
Apply "Stolt Interpolation"
Apply 2D Inverse Fourier Transform on the resulting signal
Signal Mode... |
I'm facing confusion about the definition of the convolution between two discrete periodic signals.
Basically, the definition of convolution between s and t is defined as :
In the operation, while the index of s is increasing ([0, K-1]), we have the index of t doing the opposite [n, n-K+1].
But here for example, the c... |
I am trying to implement a quadratic chirp on a gaussian pulse which has a 4 ps fwhm.
I convert the pulse to the frequency domain using an fft and then encode a chirp with the following function:
def quadChirp(ffreqs, signal, chirp_rate):
"""
Introduces quadratic chirp to a given signal (input must be in the f... |
I am curious to know if this problem is solved. The idea is quite simple. For a given audio (or video with audio) we limit the loudness to a specific value without using a cutoff or saturation. Let me explain e.g we have an audio where a person sometimes speaks in normal voice and sometimes shouts and somtimes mumbles ... |
Lets take an algorithm as an example, the DCT-II used in JPEG. The formula is as follows:
Now the thing is that, the values in M are not integers and many are even less than 1. We have two options: 1) Fixed point maths and 2) Floating point maths. We can also see that the data input is multiplied with M and also its ... |
I am trying to implement a deep learning classification solution for (1 second long) audio signals recorded at a very high sample rate, i.e. 2 MHz. All the material and tutorials that I could find work mainly in the human audible range so they are mostly relevant for relatively low sample rates.
Since I am new to audio... |
In sound anomaly detection, I see a processing step that I am not able to understand. We derive the Mel spectrogram the standard way. The signal shape is (say) 64 Mels by (say) 313 time bins. So far so good. Now the extra step is as follows.
##https://github.com/MIMII-hitachi/mimii_baseline/blob/master/baseline.py
##lo... |
I am testing some Hilbert transform theory in practice. On the practical side I use Python and a soundcard to generate baseband I and Q signals. And an IQ modulator to upconvert with 250kHz. Lastly a spectrum analyzer to inspect the result.
I was under the impression that if you take a real signal, calculate its Hilber... |
As part of an image preprocessing step, I need to generate vertical and horizontal motion blur with a filter that's 30 pixels long:
def motion_blur(image, filter_length):
horizontal_kernel = np.ones((filter_length, 1)) / filter_length
# create blurred images
blurred_vertical = cv2.filter2D(image, ddepth=-1... |
I am using OpenAL and a loopback driver to grab PCM samples of whatever audio is being played on the computer in realtime. The PCM data is 1024 samples of 16 bytes (C# short array), and I find that data does sometimes approach the min/max values (+/-32767).
However, I am also using FFT to generate frequency magnitude v... |
I am testing about injection sine wave current to r,c impedance and measurement R + jX see fig.1 .
When scope is Va .
When sine wave multiple (Va*Vb).
I put average to find DC of this signal.Is should be closing 4kV but it is loss some DC gain about 2 times.
What is equation to calculate this DC gain.
|
I have a signal that looks like this:
Red line is the function I'd like to obtain somehow. Everything <= -60 is basically absent data (i.e. not low signal).
If I understand correctly, Hilbert transform won't work for me since the signal is aperiodic. I tried spline interpolation but it doesn't produce anything even re... |
I'm working on algorithms for a camera system (which go down to sub-pixel precision). Until now, I just worked on a recorded dataset, but I'm now moving to real cameras.
My question is how to find the "perfect" sharpness? Down to about pixel level, it's rather easy : I printed some patterns (an USAF resolution chart, c... |
I am looking for a mathematical proof for the energy correction factor in conteext of windowed discrete fourier transform.
In Spectrum and spectral density estimation by the Discrete Fourier transform (DFT), including a comprehensive list of window functions and some new flat-top windows., the author defines (page 12) ... |
Is there any procedure described, or some example code (in Matlab) to calculate SNR from the sampled real signal?
|
I am working on analyzing data obtained from Fiber Optic - Distributed Acoustic Sensor (DAS) to estimate the properties of acoustic waves that impinge FO cable, such as velocity of acoustic waves. FO-DAS provides real-time estimation of acoustic energy along the fiber (i.e., data over distance and time). Recently, I re... |
I am currently studying for my last exam.
I wanted to ask why it is necessary or useful to perform the autocorrelation of a signal with a lagged version of itself
Why does it have to be lagged?
|
I'm working on machine learning model and tried to downsample WAV files with Python script, but it didn't work, so I made the simple version - script "resamples" WAV file from 44100Hz to the same 44100Hz sample rate. Original and output files seems identical while listening to, they are both 16-bit depth, 44100Hz sampl... |
Hi I am a newbie to the field so please don't mind a rookie question. I have read multiple articles and want to ensure my understanding is correct.
I have following two images. In the second image, the object of interest hasn't moved , its just the camera. Is that why its called rotation? How would translation look lik... |
Before I start, let me note that I have 0 experience with signal processing, so please bear with me:
My System
My system can be represented as an $m \times n$ matrix $X$ (input) where each
column represents a "channel" and each row represents "time". A given input
$x_{i,j}$ (time $i$ and channel $j$) produces:
A signa... |
I have logged gyroscope sensor data for roll(x), pitch(y) and yaw(z) axis and i want to plot their step response with Python to be able to tune the pid controller of my quad better.
1. I have found example python code for creating transfer function and taking step response. However, i don't know how to determine the tr... |
I have a problem in signals and systems to solve which is basically math
So suppose we have two functions x(t) = H(t) which is the Heaviside unit step function as input and an impulse response
$$h(t)=e^{-\left | t \right |} + e^{-\left | 2t \right |}$$
which is $$h(t)=(e^t+e^{2t})H(-t)+(e^{-t}+e^{-2t})H()t)$$
We want t... |
I designed a board with an STM32H7, an ADC, and a TLV320DAC32. I'm in the stages of just getting a straight audio passthrough working. I am now getting signal output but it's all jumbled, and not at all representative of the input waveform. After lots of reading and messing about, I believe I have the DAC configured co... |
If we want to calculate a dc value of any given signal, should the dc value be RMS or average value?
|
I'm following a guide about signal processing, but since I'm a fresher to the domain, the guide just stops at a point where only a function that could return the spectrogram values is written. So, my problem is:
Since I'm not good on the domain yet, I'm stuck where I don't understand how to plot the spectrogram from t... |
I have a signal with multiple frequencies. The frequency changes depending on the amount of air flowing through the fuel snap. The signal resembles a square wave ranging from $0.4$ to $4.5$ Volts and was measured over a duration of $30$ seconds, resulting in $1,200,00$ values. I am attempting to analyze the signal, but... |
I'm not a specialist in this field and learnt FFT fairly recently.
I want to calculate phase difference of between 1 MHz signals sampled at 4 MHz. The sample counts are both 144. I think the magnitudes and the frequencies are all correct. But phase shifts are off.
So, I decided to go back and start with simpler sets of... |
This question is about the ideal digital clock. Lets consider an ideal clock with 0 raise time and 0 fall time i.e. perfect square wave. As shown in the diagram below. Let the original clock period be T
Now suppose we want to introduce jitter which is periodic in nature. Each clock edge will move wrt previous edge aan... |
I'm trying to design the highpass filter in Python for my accelerometer data in order to integrate them. I need to preserve low frequencies. For example, data with a sampling frequency of 2 kHz need to have a highpass filter with a cutoff frequency of 1 Hz, maybe even 0.5 Hz. Unfortunately, scipy signal filters (I trie... |
In this paper they preprocess an audio file with a STFT and then apply two equations to the computed matrix.
I tried implementing all of it in python and used librosa for the STFT.
The issue I encountered is, that the matrix includes a lot of low values or zeros.
By the 2. equations including log those values get tra... |
I am working with Bluetooth specification 5.1 where the advertisement packets can send a constant tone extension (CTE) over the baseband signal to estimate the angle of arrival with an antenna array.
Due to inaccuracy of transciever and reciever clocks for bluetooth devices, the CTE is not exact.. This leads to a phase... |
I'd like to display the freq response of a Sallen Key filter type (in the specific case, this simple resonant filter).
on biquad, for example, I calculate magnitude and phase from zeros and poles, such as (in cpp):
// zeros
std::complex<double> resZeros(0.0, 0.0);
resZeros += mCoefficients.mA0 * std::exp(std::complex<d... |
Floating point representation encodes a binary number using a mantissa $a$ and exponent $b$ according to $(1+a)\cdot 2^b$, where $a$ is an unsigned fractional fixed point number such that $1\le 1+a < 2$ (in decimal form) and $b$ is a signed integer.
Thus for any value $x$ encoded in floating point, we can make an appr... |
Suppose I'm transmitting two streams at the same time in blocks each of length $N$ samples each sample of duration $T_s$ seconds. Block $t$ of stream $i$ is $\mathbf{x}_i[t]=\left[x_{i,0}[t],\,\,\,x_{i,1}[t],\cdots,\,\,\,x_{i,N-1}[t]\right]^T$. The second stream is shifted by $\Delta\tau=LT_s+\epsilon T_s$ seconds, whe... |
I am trying to measure the frequency response of different audio canals, e.g. tubes, hollow pipes etc.
My experimental setup looks like this: I generate an audio signal with a loudspeaker (Visaton WS 17 E 4 Ohm, datasheet: Datasheet loudspeaker which then passes through my device under test (e.g. the tube) and is then ... |
In my research work, I came across the concept in the paper
Y. Hu, P. Wang, Z. Lin, M. Ding and Y. -C. Liang, "Machine Learning Based Signal Detection for Ambient Backscatter Communications," 2019 IEEE International Conference on Communications (ICC), Shanghai, China, 2019, pp. 1-6, doi: 10.1109/ICC.2019.8761796.,
eq. ... |
I'm using Octave code to extract minimum phase, but this minimum phase has became inverted phase acording to original signal, What is goin wrong?
Resault here I have shifted main signal about 48 sample to matches it's minimumphase component(orange) and obviously these are out of phase:
Note:
The coefficients length is... |
The uncertainty principle states that if you have a signal which is very concentrated in time, then its Fourier transform will be rather outspread and vice versa. However, I don't really understand how this implies that in general we can't find a function that gives us the frequency spectrum for a particular point in t... |
The problem is question 6.25 from Proakis, Communication Systems Engineering (2001)
Design a ternary Huffman code, using 0, 1, 2 as letters, for a source with output
alphabet probabilities given by {0.05, 0.1, 0.15, 0.17, 0.18, 0.22, 0.13}. What is
the resulting average code word length? Compare the average code word ... |
I was working/studing a basic Resonant Filter, using the Paul Kellet one as test case: https://www.musicdsp.org/en/latest/Filters/29-resonant-filter.html
The freq is feed in normalized value 0..1.
So the question: how can I convert these values into hz? Getting the mapping function from/to normalized/hz?
I Guess 0.14 i... |
I want to add a delay to the messages in GNURadio. For example, when I click on a button, some action will not be triggered immediately but after specific delay.
I think this behavior can be accomplished in the following way:
Button (produce message) -> PDU -> Tagged stream -> Delay -> PDU -> Target block
The problem ... |
I am trying to understand the radar detection problem in the form of the generalized likelihood ratio test and am having a little trouble with understanding the noise distributions. Perhaps this will be easiest understood by my explaining my thought process and somebody pointing out where things are going wrong!
In the... |
I'm trying to estimate RT60 based on reverberant sound. In order to generate reverberant sound, I am required to give dimensions of a room so that RIR is generated and is convolved with clean sound.
While reading few papers, they had different room dimensions and RIRs ranging from 0.1s to 1.5s.This had me puzzled. Now... |
This is probably a basic question but I didn't find any answer online.
Suppose we configure a SDR (least say, with a maximum sample rate of 100 Msamples/s) with a low sample rate (e.g. 1 Msample/s).
Does the SDR actually samples at 1Msample/s and sends only that (e.g. to GNU-radio)?
Or does the SDR sample at 100Msamp... |
I created an algorithm that approximates the tiny image texture by a series of sinusoids. It is NOT Fourier or discrete cosine because the boundary is highly irregular.
I have each wave's frequency, amplitude, direction (angle relative to base vectors), and shift. Each texture is described by something like three waves... |
I have two function, But I don't know which one is correct to convert my Transfer Function(tf) to secound order sections(sos).
I'm using GNU octave, and it have intrinsic tf2cas, also in github function dir2casis existing. I wonder to use which, since they have different response.
Octave intrinsic tf2cas weirdly does n... |
I have filter of order 2000 IIR all-pole. I want to implement it via cascade IIR filter. Is there any way to reduces poles order and increase zeroes as its cost. To balance zero and poles. Since all pole cascade have lots of zero coefficient because of lack of zero, in this way I want to gain more computational and may... |
I have data in the frequency domain. How can I apply the low pass filter such that it still removes the high frequency information?
|
I have opened the file in Matlab and am using MNE Python too. This is a processed EEG file in Matlab data format instead of raw EEG (Screenshot 1).
Screenshot 1: struct of the 801_1_PD_REST
Now the problem is that inside the Matlab data file, the values inside EEG are very large (see screenshot 2), and what are their ... |
The question is to find the CTFT of $$x(t) = e^{-t}u(t)\cdot \sum_{n=-\infty}^{\infty} \delta\left(t-\frac{n}{2}\right)$$
Now I know that multiplication in time means convolution in the Fourier Domain but how can I find the CTFT of $\sum_{n=-\infty}^{\infty} \delta(t-\frac{1}{2}n)$ since I am not allowed to directly us... |
I'm trying to implement the Rayleigh flat fading channel in MATLAB using SRRC (Square-root raised cosine) pulse shape. The modulation scheme is BPSK. Here is my code:
EbNo_vec = -4:2:15; % Eb/N0 vector
number_of_packets = 3e3; % Number of packets for each Eb/N0
packet_len = 501; % Length of each packet
span = 30; % Spa... |
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