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I'm pretty new to Kalman filtering techniques, and have a problem that I'm trying to solve.
Imagine in a lake I have a splash generator that sends out sine waves across the lake. At various places throughout the lake I have sensors that can measure the waves. What I want to do is use the various sensors (maybe many) t... |
I can see from the literature that PDE-based image processing (especially inpainting) was a "hot topic" at one point. Why does it not seem to be an active area of research anymore, yet it looks like there are many unanswered seemingly important questions to be answered? If it is being done, what are some of the recent ... |
Background:
I am learning an interesting Fast CWT algorithm(PPCWT) by reading this paper published in 2019. The algorithm is summarized as below. The continuous wavelet transform of a signal can be calculated efficiently according to the last row of Eq.(1.4). To test this algorithm, I write a Cython module ppCWT_cymod.... |
I am looking for the optimum modulation scheme to build a half-duplex modem minimizing the required SNR at the expense of bandwidth efficiency. The key system parameters are described below:
Target bitrates: 15 - 100 Kbps
Available BW: 1.6 MHz
Power amplifier
operating at saturation level
Given that the maximum requi... |
Please let me ask you about my problem with Matlab code executing in R2022a release.
A signal has 3 harmonics with amplitudes 170, 220, and 150 in that order. Matlab's function abs(fft(X)) returns maximum amplitude for the first harmonic. What is the reason? Code is following:
SampFreq = 16000;
Segm = 1:2048;
Pitch = 4... |
What is the difference between stable, asymptotically stable, marginally stable, and unstable systems?
I am familiar with the first term (stable) and last term (unstable) but I often get very confused in understanding the other two (asymptotically stable and marginally stable).
Which of these two systems is stable in B... |
I am trying to upsample a signal using sinc interpolation. I have written a way to do this in python.
@staticmethod
def sinc_interp(x, s, u):
"""
Interpolates x, sampled at "s" instants
Output y is sampled at "u" instants ("u" for "upsampled")
from Matlab:
http://phaseportrait.blogspot.com/2008... |
I am new to wavelet transformation. I am learning it as a tool for signal processing. I have a time series that I want to analyze. I tried to learn wavelet transformation by applying it to a periodic function.
This is my code:
## Meta imports
import numpy as np
import scipy as sp
import matplotlib.pyplot as plt
from s... |
If GPS L1C/A receiver is experiencing a Doppler shift of $f_D$ and a Doppler rate of $\alpha_D$, assuming total integration duration of $T_{int}$, then how should I calculate the code phase estimate error due to this changing Doppler shift effect?
I understand that the code rate will change by a factor of : $$R_c\prime... |
I'm trying to write an online algorithm in Python to detect this below Down-Up-Down pattern in time-series.
It's not hard to do roughly if I calculate 3 contiguous non-overlapping moving averages, and just check that the first and last are both < -X, and the middle one is > 1.5*X.
But this has many downsides. For examp... |
I acquire an analog waveform and I expect it to have some local flatness (post-filtering). The waveform's exact shape is unknown but some rough characteristics (peak amplitude, duration, rise time, etc.) can be assumed to be known. The rough shape of the waveform can be assumed to be known (excluding any information ab... |
I am trying to get data from only 1 point of each rising edge and trying to relate it to 1 point of the sin wave at the same time. Ideally, the middle of the rising edge is where I want the sin wave data to be.
Context: I am sampling a sinewave with a square wave in ltspice, but ltpsice over samples the data resulting... |
In a book I'm reading, the author writes
For example, all real rational stable transfer matrices are analytic in the right-half plane
I'm not sure of the definitions of "real", "rational", and "stable" here. I'll mention what I think they mean, but would appreciate any feedback on whether they are right.
Given an $M ... |
I am sharing the IBE table 6.4.2.3-1 on TS 38101-1
For General emission requirement, there are three parts.The third part is (-57 dBm+10*log10(scs/15kHz))-P_RB. Where, P_RB is the average transmitted power normalized by the Allocated RB's in dBm. My Question is :
What is the significance of this third term as this on... |
So I work in this domain of biophysics that has to do with a light-based detection for measuring small movement of molecules (nanometer and piconewton scale) via a Quadrant Photodiode. This signal contains lots of information but is riddled with noise. One of the challenges is denoising this signal and while convention... |
I'm trying to write an online algorithm to detect when a time series goes sideways (i.e., its mean trend or mean drift is approximately zero) but is also unusually volatile. For example, in the red box below, it's not trending up or down, but it is highly volatile.
My thought was to use a low pass filter to strip out t... |
I am seeking to calculate the individual sample SNR from a given $E_{b}/N_{0}$. I am calculating this as follows:
$E_{s}/N_{0} = E_{b}/N_{0} + 10\log_{10}(M)$
$SNR = E_{s}/N_{0} - 10\log_{10}(n)$
Where $M$ and $n$ refer to the bits and samples per symbol, respectively.
I am encoding with $R_{c}=1/2$, resulting in a tot... |
I intend to buy the book by Oppenheim et al, "Signals & Systems" 2nd edition.
Now I also find (also by Oppenheim et al) "Signals, Systems & Inference".
Does anyone know, is this book the same but with slight changes, or is it completely different?
|
I was planning to build a data acquisition system to capture raw IQ data for a hobby project and the short-term goal is to gain practical experience on RF front ends and ADC's.
My background is mostly Wireless DSP implementations in MATLAB/C (and some basic GNU Radio experience using USRP B200).
Did anyone try somethin... |
As a followup to my True Peak detection question, I'm trying to implement a detection method by following this documentation using the Catmull-Rom interpolation method.
What I've done so far can be seen in my Desmos worksheet (includes sample of y -data which creates the issue in question).
I've (probably) got station... |
I have an audio recording where something went wrong and a copy of the audio was added with a few milliseconds delay. What technique can I use to remove that echo?
Here's a short snippet to explain: https://voca.ro/1fshI5WOr6oe
The echo occurred because the recording software (Jitsi) recorded both the browser tab's aud... |
I want to find the peak amplitude of an audio signal in a given frequency area.
My problem is that if the signal has a strong resonating frequency between 2 fft bins, then both bins are way below the actual amp.
I wrote this example python code, which:
Generates a sine with amp 1 at a frequency, which perfectly fits t... |
I am trying to figure out the difference between the FFT of an image, say, cameraman.tif in MATLAB and the FFT of image when the image is Gaussian blurred. What is the effect of the averaging filter on the FFT computation?
|
I am trying to plot the step response of an electronic circuit(differentiator) using MATLAB but I get an error:
Error using DynamicSystem/step (line 95)
Cannot simulate the time response of models with more
zeros than poles.
My code is below:
R=47*10^3%resistance value
C=1*10^-6% capacitor value
den=[-1]%denominator o... |
I have been trying to find the following inverse Fourier transform but without success:
$$
H(\omega) =
\begin{cases}
e^{-j \frac{\pi}{2}} & \omega \gt 0 \\
e^{j \frac{\pi}{2}} & \omega \lt 0 \\
\end{cases}
$$
I have tried using the inverse Fourier transform but of course the $e^{j \omega t}$ won't converge.
I have a... |
Let $f(t)$ and $g(t)$ be $2$ signals of duration $T$ and bandwidths $B$ and $B'$ respectively, with $B<B'$.
Consider $s(t)=f(t)+g(t)$. With Fourier transform, we have $S(\nu)=F(\nu)+G(\nu)$ and the bandwidth of $s$ is $B$.
Is it correct in digital communication to add signals that do not have the same bandwidth?
With... |
I am trying to create a sine wave in PLL output with Simulink.
But I get a square wave from the VCO block even though PFD, charge pump and loop filter outputs are correct. The VCO block says that only square wave is given as output in Matlab Help. It is written that only square wave is obtained, but theoretically PLL s... |
I am trying to generate a simple Gaussian pulse that has a 1 ns pulse width. However, when I generate the pulse, I realized that I did not meet the condition that calculates Rayleigh Bandwidth (1/Pulse width).My estimation is based on setting a threshold for the spectrum, which focuses on the higher magnitude values fr... |
The Fourier transform is derived from the Fourier series by considering a non-periodic signal, thinking of it as a infinitely long periodic signal, putting it into the Fourier series and making this series as a Riemann sum (the definition of integral) and thus, deriving the Fourier transform integral.
My question is, s... |
I am reading articles on LORA and noticed a problem. For example, if the carrier frequency is $f_0=470$ MHZ, the sampling frequency is defined by $f_s=1$ MHz. For me this is incorrect because normally we must have $f_s \geq 2f_0$.
Why do we make this choice of value for the sampling frequency?
|
Given a signal:
$$x(t) = -5\cos{(93.6\sqrt{2}\pi t - 2\pi/3)} + 3 \cos{(135.2\sqrt{2}\pi t + \pi/13)}$$
I am confused about the right process to take or if there is an easier way than following all the integration. I know I need the fundamental frequency, I need to find the coefficients, but am not sure how to proceed.... |
We have have a small data cleaning issue we would like to share.
Context: We have data from an accelerometer sensor worn at the hip. We set the device so that data are collected at a sampling rate of 100 Hz. We perform a 10-sec window analysis in order to extract several features of interest, some of them being obtaine... |
I want to implement a IIR Band Pass Filter without using MATLAB functions.
I get the coefficients as follows ($F_s$ = 1000, Band pass: 5-15Hz):
f1=5;
f2=15;
Wn=[f1 f2]*2/fs; ... |
When finding the power spectrum density (PSD) from the $N$-FFT of time-domain signal $x(t)$ of $N$ data points (assuming $N=2^n$, $n\in\mathbb{Z}^+$), we normalize the power of the FFT by the sampling rate $f_s$ times the number of data points $N$ like so
$$S[i]=\frac{2|X[i]|^2}{f_s\times N},\,\,\,i=0,\frac{fs}{N},\fra... |
My tutor does not explain it very well. Can someone please explain to me the reasoning and what null to null bandwidth actually is?
|
Defined as "negative DFT bins zero", when are such filters suboptimal for AM/FM extraction or related filtering? This answer reads,
[nulling] also has the worst performance compared to other methods such as windowing or using the optimized algorithms known as "least-squares" and "equiripple" (Parks-McLellan) that are ... |
I'm new to DSP and I'm struggling to solve a basic problem. I think I've solved it but I feel that something's off in my reasoning.
The problem involves investigating a signal of the form $e^{jk\pi n}$, where $n$ is an integer and $k$ is real. How do I confirm if this signal is periodic? I tried dividing the frequency ... |
What I want
I wish to design an LPF, preferably an FIR, but also open to IIR.
My constraints:
I care about phase non-linearity. Thus if I use an IIR, I need a filter similar to a Bessel filter.
My signal isn't too long, so I wish to have a short transient (short impulse response).
A seemingly contradictory requirement... |
Contextualizing
This question is inspired by the following video:
https://www.youtube.com/watch?v=-qgreAUpPwM&t=60s&ab_channel=3Blue1Brown
I own a sign, a drawing of a square with 200 points in total:
import numpy as np
import matplotlib.pyplot as plt
from scipy.integrate import quad, quad_vec
POINTS_EACH_SIDE = 50
x... |
The ideal integrator has differentiator has transfer function H(s)= -1/RCs while ideal differentiator has transfer function H(s)= -RCs
It is often said regarding above integrator that it has a zero at infinity similarly it is often said regarding above differentiator that it has a pole at infinity
But as far as given e... |
I’m looking for help to find a robust technique to remove a once per revolution variation in some vehicle test data. The data is collected by driving a vehicle around a circular path at increasing speed whilst measuring, amongst other things, roll angle and lateral acceleration. Due to the slight tilt in the track surf... |
I'm dealing with bunch of .asc(ascii) files that are the output of continous monitoring of various electronic equipments for certification purposes. We monitor various parameters of the equipments like voltage, current , temperature at different states(modes) of operation [sleep mode, minimal load , maximum load etc.].... |
I am doing radar sensor signal processing.
The radar sensor gives iq data corresponding to the range (sweeps).
I want to detect micro motion so that phase analysis is used.
Sweep is the distance measurement range. One sweep consists of 259 data points (L = 259).
Each sweep is noise-reducing lowpass filtered using
y(s)... |
I'm interested in oversampling the inputs to a digital controller to increase the SNR of the input process variable signal. I've read on this site and in articles like the one below that it is not accurate to apply the z-transform to an up-sampling or down-sampling operation along with frequency domain techniques becau... |
Suppose that we have four filters $H_0(z), H_1(z), F_0(z),$ and $F_1(z)$ forming a classic perfect-reconstruction 2-channel filter bank:
Will the perfect reconstruction still be achieved if we interchange $F_i(z)$ with $H_i(z)$?
My thinking is to check if the condition that the aliasing transfer function is equal to z... |
Disclaimer: I'm a beginner at all of this.
I am working with vibration data from accelerometers recorded at a sampling rate of 1600 Hz and the vibrations are induced by a source within the range of 20-40 Hz. I snip out about 60-80 seconds of data and use it for analysis. I have the following work flow to perform some b... |
I'm looking for some suggestions on detecting wifi using spectrograms (abs magnitude only) . My project involves a wifi driver which outputs a spectrogram whenever it scans the channel .
Can we use any of DSP techniques to detect WIFI in the spectrogram ? I have seen lot of implementations using deep learning based CNN... |
I am trying to get the estimation frequency from the FFT output in unity, I have read that Quinn's second estimator method is the most accurate method, and I found the algorithm for how to implement the method. The explanation is referring to the output array (samples) as complex numbers, And the FFT in Unity returns t... |
For time series, a simple high-pass filter is obtained by subtracting the previous value from each value:
$y(n) = x(n) - x(n-1)$
If I take a multi-period difference:
$y(n) = x(n) - x(n - a)$ where $a > 1$
Is this equivalent to a band-pass filter which attenuates frequencies significantly above and below $1 / a$? If not... |
I'm trying to get a good handle on the spectral theory of random signals, but there seem to be two different definitions of the power spectrum whose relation is not clear to me. On the one hand, we know that the autocovariance function of a stationary random signal has a spectral decomposition
$$
\begin{align*}
C_{XX}(... |
I am trying to tune the Active PF Correction converter current and voltage loop to get a sinusoidal input current. I can get to a voltage reference, but the current looks like this. Reading the literature, the advantage of using APRC is that the input current is sinusoidal and the inductor current should be half sinuso... |
I have 5 values, and each value is corrupted by Additive-Zero-Mean-Gaussian Noise with variance = 0.05.
Each value is in the range from 0 to 1. Is there anyway for me to remove the noise?
|
Prepping for exam and this is one of the practice problems:
I just want some clarification on some of the steps my professor took. This is the answer in the answer sheet
Only thing I dont understand is how the domain of Y is determined. Some help would be greatly appreciated.
|
The end goal of my project is to create an autotune program, But the problem I'm trying to solve right now concerns finding the pitch of someone singing a note. I have written some code that performs a Discrete Fourier Transform on a sample of 2048 values derived via my microphone. Heres the results of the code:
The to... |
This is two linear differential systems equating each other.
\begin{equation}
\frac{d^Ny(t)}{dt^N}+a_1\frac{d^{N-1}y(t)}{dt^{N-1}}+\cdots+a_{N-1}\frac{dy(t)}{dt}+a_Ny(t)=\\
b_{N-M}\frac{d^Mx(t)}{dt^M}+b_{N-M+1}\frac{d^{M-1}x(t)}{dt^{M-1}}+\cdots+b_{N-1}\frac{dx(t)}{dt}+b_Nx(t)
\end{equation}
In the textbook, it states ... |
To my understanding, there are two methods to do linear filtering. One is cross-correlation, and another is convolution. Convolution requires 'flipping' the kernel when you do the calculation.
I think that you can simply do cross-correlation without doing convolution. So I wonder why to choose convolution instead of co... |
I have a signal that is sampled at Fs and wish to apply a "16-point Sinc interpolation kernel". However, I am not sure what the 16-point refers to. Does it refer to having a kernel that is represented by 16 samples? How can I generate such a kernel in python?
EDIT:
I was hoping to resample the cosine signal using the c... |
I am using gradient descent on an adaptive IIR filter for the below system 1. At the moment I am just assuming the known system is not there and it works fine. However, occasionally when the known system has slower dynamics it does not perform very well. Are there any papers or methods for adapting a filter in this kin... |
I'm interested in developing audio software. I've read the books by Will Pirkle and there's something I'm struggling to understand about how non-linear functions affect the frequency spectrum of a signal. Particularly I'm concerned about avoiding aliasing noise.
In his book, Pirkle points out the problem: harmonics are... |
Is there a simple algorithm for computing the fractional discrete fourier transform analog to the FFT algorithm or even naive DFT matrix multiplication?
I am kind of new into signal processing. For my own understanding I built this tool in order to grasp an intuitive understanding for the relation between time and freq... |
I am quite new to the world of radar signal processing and there is this issue that I can't figure out for the life of me.
In Remote Monitoring of Human Vital Signs Using mm-Wave FMCW Radar (Section II.B) it is said that
"After DC compensation, the phase of each column in the range-slow time matrix is calculated by us... |
I am new at signal processing and was trying to understand the relation of the result of the FFT to the original signal.
I generated a signal with a sampling rate of 100 Hz within 1 second ($N=100$), consisting of a constant component and two sinusoids with a frequency of 5 Hz and 10 Hz, then applied FFT to it. For FFT... |
I want to qualify a speaker I use. I use that sound on my speaker to qualify it:
https://www.youtube.com/watch?v=qNf9nzvnd1k&t=4s
My first question is about the file itself. When I use Audacity the FFT analysis is that way:
I don't understand why the curve is that way. The magnitude of the input signal seems to be equ... |
I am new to signal processing and trying to build a gadget that hits an object and recognizes fractures and weaknesses in it using an accelerometer attached to a hammer. I have some initial results from manual tests and produced the waveforms below. The green wave is from a spot in the object that I know is intact. The... |
What is the difference about QPSK, BPSK and 16-QAM in spectrum? I just simulated all three, but I can't see the difference clearly.
|
I'm working on a voice changer. My plan is to make it so that it can change your voice in various different ways, but right now I'm just trying to make it change your voice to "chipmunk voice" or a high pitch voice. I have successfully programmed an algorithm that can perform a DFT and an inverse DFT on a recording of ... |
When I read the example
NR Intercell Interference Modeling in Matlab, I cannot find the process of interference. Can anyone tell me which function can calculate the interference?
|
The images above are tiles taken from a scanned painting. It's easy to see where there are tiny reflections scattered throughout. I wish to remove (or diminish) the tiny reflections somehow, across a few thousand (50k) such images. I want to preserve as much of the remaining detail as possible.
I am more concerned ... |
I have wandered into the realm of signal processing and am currently focusing on learning more about filtering signals. I am currently trying to understand how to think about and solve a practice problem from my theory book. Given the signal
which one of the following signals corresponds to the high-pass version of th... |
I have a known smooth complex signal $f(x)$:
It's added to an unknown complex signal $g(x)$ after being multiplied to an
unknown complex amplitude $A = u+iv$ (this image is just for illustration, in reality the amplitude of $f$ is ~1e3 greater):
How do I find $A$?
The method I've tried is minimizing the number and am... |
I did the cwt on a chirp signal :
import numpy as np
from scipy.signal import chirp
import matplotlib.pyplot as plt
T = 5
n = 1000
t = np.linspace(0, T, n, endpoint=False)
f0 = 1
f1 = 10
y = chirp(t, f0, T, f1, method='logarithmic')
plt.plot(t, y)
plt.grid(alpha=0.25)
plt.xlabel('t (seconds)')
plt.show()
from scipy... |
Using formula from Audio EQ Cookbook I implemented biquad IIR filters in Excel.
I can now plot every transfer function and total of 8 band EQ with shelvings and peakings.
My problem is, I can not implement any of the phase response formula in excel. I'm not very good at these kinds of functions and especially implemen... |
What does convolving an image with the filter [1 -2 1] do? I see that it's a form of edge detection but it's also very similar to the results I get when convolving with the derivative kernel [1 -1] so I don't really see the difference between them
|
Say I perform FFT on some data. If the underlying (measurement) sampling rate is not twice the highest frequency, I will almost assuredly get aliasing.
This limit on sampling we call the Nyquist limit. Now on the flip side, if the integration underlying measurment window is not greater than the wavelength of the lowest... |
I've been working on a phase vocoder program. The goal is to change the pitch of a recording of my voice. While doing research on how to change pitch, I came across this from a paper on phase vocoders written by Fischman,
"All that is required is to transpose each frame by an octave, which is done by doubling the freq... |
I have a signal $s(t)$ which is defined as $s(t)=f(t)+g(t)$ in the time domain. In the frequency domain, the signal $f$ is a baseband signal whereas the signal $g$ is a signal that occupies the right side of the bandwidth of signal $f$.
Thus is my questions:
1)Can we say that $g$ is baseband when it is not centered at... |
Fourier transform is an estimator of frequency. Because its an estimator, there is always some uncertainty associated with my coefficients as described by the Fourier/Gabor limit. I'm wondering how to generate a probability density of frequency from the transformed data. My intuition is that I like what they do in phy... |
Say I've got some data made from measurements with a too infrequent sampling rate; I know for certain there is aliasing. What I'm interested in is figuring out what frequencies are likely present despite aliasing.
Too start things out easy, lets say I know there are two frequencies present in the underlying signal, on... |
Is there any restriction on decreasing the bandwidth (beyond the sampling rate) that I should be aware of when filtering a signal with a bandpass? My intuition tells me that the quality of the result may compromised if the frequencies too close to each other but I can't see why. Is there any mathematical or rule of thu... |
Hij Vi Si the Desired signal
∑_(j=1, j≠i) Interfering signals
how can I model interference?
|
I have data collected from an experiment of shaking a 4-storey steel frame using a shaker and responses were recorded using accelerometer sensors. I realised that in the dataset some of the signals behave randomly and some behave as impulse response signals, as shown in the figures below. I was planning to investigate ... |
I experimentally transmit an OFDM signal through 2 meters channel. When I estimate the channel at the receiving end, I find the channel coefficients very small. An example of the channel I find is:
h = [0.0823 + 0.0167i 0.0017 + 0.0033i 0.0010 + 0.0023i 0.0008 + 0.0018i 0.0004 + 0.0011i 0.0001 + 0.0001i -0.0004 + 0.001... |
Question: $x[n]= 3\cos(0.3\pi n + 0.1) - \sin(0.11\pi n -\frac{\pi}{3})$ Find the fundamental period for this discrete signal.
My attempt: Write the two functions as:
$$\underbrace {3\cos\left(\frac{3}{10}\pi n + 0.1\right)}_{=f(n)} - \underbrace{\sin\left(\frac{11}{100}\pi n -\frac{\pi}{3}\right)}_{=g(n)}$$
The period... |
Background
I've been given the task of designing a filter.
The problem is constrained in terms of filter order, and as such, the specs initially conceived for the filter seem challenging, and we think we might need to release them somewhat. For those interested, here is a question I posted regarding the design requirem... |
I'm trying to grasp the "ah-ha!" moment for the DFT/FFT. One of the points I struggle with is: if the original time signal $x[n]$ is real, then the values of the DFT $X[k]$ are (correct me if I'm wrong) complex-conjugate between the intervals $[-N/2; 0]$ and $[0; N/2]$, or, equivalently, $[0, N/2]$ and $[N/2; N]$. Ther... |
While implementing/studying brickwall low pass filters for digital audio, I decided to run a unit impulse through FabFilter's Pro-Q 3 brickwall low pass filter to see what the resulting impulse response would look like.
When in "Linear Phase" mode (bottom half of image), the resulting IR sure enough looks exactly like ... |
Please can someone help me write a Matlab code that expresses the interference shown in the image.What is the formula for the interference in this case and how is it calculated?
The received signal of the first received antennas can be explained:
r1 = h11 x1+ h21x2 + v1
where h11 denotes the channel between T1T1 and R... |
I have a question about MSK modulation and the influence of doppler shift on this modulation.
For simplicity, suppose we consider the MSK signal in baseband: $s(t)=Ae^{j(d_k\frac{\pi t}{2T}+\phi_k)}$ where $T$ is the $k$-th bit period an $\phi_k$ the initial phase for the $k$-th bit.
With Doppler shift in baseband, we ... |
I want to define a threshold for maximal blocking artifacts in an image.
What is a good key value to measure blocking artifacts?
I have no reference image
non natural image
Example:
I looked into no reference image quality analysis like BRISQUE and PIQE, but they seem to be more focused on natural images.
Other meth... |
I'm working on a pitch shifting program. Everything works up to the point where I try to do the IDFT. Because I cut the DFT array past the Nyquist limit, when I run the IDFT, I don't get the same signal as I originally did. For example, If I pass a 1024 item buffer into my DFT code, it returns a 512 item array. If I do... |
I build a pam-2 modulation, then I make a pulse shaping with half sine (matched filter). Then I send it through the AWGN channel. At the end I do down sampling and demodulation. But i have problem with plotting BER. I don't understand what I'm doing wrong:
clc;
clear;
N=1e4;
N2 = 1e2;
M = 2;
range = 0:10;
error = zer... |
I am trying to apply the kaiser window during sinc interpolation.
The following is my sinc interpolation code:
def sincTrain(sig, interpolationFactor):
tn = np.arange(0,len(sig),1)
t = np.arange(0,len(sig),1/interpolationFactor)
sincTrain = np.zeros((len(t),len(sig)))
w = np.kaiser(len(t),2.5... |
I am using a sine wave signal (for test) with the following scenario:
Computer #1 --> AWG --> DC bias (2.2 Voltage) --> light source --> lens *** wireless channel 2 meters *** --> lens --> APD --> Oscilloscope --> Computer #2.
I am wondering why the received signal is not symmetric, I mean the negative part is clipped.... |
Consider the below given discrete signal which has been gathered via sampling of an analog current waveform with sampling period $T_s=100\,\mu s$.
I would like to evaluate the RMS value of its first order harmonic which has the frequency $f=58\,Hz$.
This task can be basically divided into two subtasks:
Finding the fi... |
I want to create a real-time sampler for woodwind instruments.
Because it is possible to slide from one note to another without stopping the sound I decided to do it as a synthesizer. I want to analyze the sample in the frequency domain (time-domain sample as the sum of sinus waves). This part can be heavy to compute, ... |
I have a signal $x(t)$ of bandwidth $W_1$ transmitted over carrier frequency $f_1$, and another signal $i(t)$ of bandwidth $W_2$ transmitted over carrier frequency $f_2$. The main lobe of both signals' spectrums don't overlap. They are transmitted on the same time. So the passband received signal can be expressed as
$$... |
Inspired by this question: Does a simple photograph contain more information than a complex painting?
Say that I have a discrete representation of some physical object - eg an image file. What is the true degree of "unpredictableness" or the size that a potential (lossless) compression scheme might never improve upon?
... |
I'm experimenting with MFSK in gnuradio, and seem to be quite successful simply using:
quadrature demod on the raw data to get the frequencies
RRC filter
clock sync on that
Constellation decoder with four custom constellation points all around the real axis
This seems to work for me. But is it a dead end? Will I get ... |
I am new to signal processing.
We were introduced to the sampling theorem and intuitively I am struggling to make sense of it:
There are 3 points I am struggling to comprehend:
The analog signal: the theorem says that you multiply it by a train of time shifted impulse functions. Is the analog signal assumed to be know... |
I have some vibration data (acceleration) on which I need to perform an FFT, integrate it and again do an FFT.
I read that for the nature of the data that I have, the input to the FFT must be first windowed with a Hann window in order to avoid spectral leakage in the frequency domain representation after I perform the ... |
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