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my circuit only need very low frequency of signal and cannot add anti alias filter in analog domain, so I adopt sample rate frequency hopping as follow:
1、use FFT to identify if there high value noise low frequency bin.
0.1fs/2<bins <0.2fs/2
2、use FFT to identify if there high value noise in high frequency bin.
0.2... |
Let me simplify the question: assuming a PAM-4 signal {-3,-1,1,3} with noise added. I focus on the BER for the signal 1. My question is: with same noise power, does Gaussian noise have the minimum SNR?
|
I am interested in adding a narrowband interfering signal to my spread spectrum signal.
In the waveform model, it's straightforward to generate a narrowband signal within the frequency range of the spread spectrum. However, I'm unsure about how to add a narrowband interfering signal to a spread spectrum signal in a bas... |
I have read the following quote from a different post:
If the noise is white (spread evenly across frequency), then the
power will be distributed equally to each bin, and as we increase the
number of bins, the amount in each bin will reduce accordingly (this
is the "FFT noise floor").
FFT is the discrete sampling of ... |
I need to check if MSK modulation keeps the phase continuous after DS spectrum spread. My thinking is that since we're expanding the bandwidth, it likely involves abrupt frequency changes so it won't keep the phase continous, but I'm not totally sure.
|
Let's say I've been feeding some digital IIR filter input samples $x[n]$ to get output samples $y[n]$.
If after some time $t$ the input becomes all 0s, i.e.
$$x[n] = 0, \; \; \forall n ≥ t$$
How can I find the total output energy from that point onwards (without just running the filter for ages):
$$\sum_{n =t}^\infty ... |
Im recording sound generated by a drilling machine against an object. I have the recording of following :
Ambient noise with drilling machine switched off.
Drilling machine on+ambient noise
Drilling machine on and drilling against a surface+ambient noise.
Recordings are done in terms of dB vs time.
With that I have f... |
My partner stitches, and uses sites like https://www.pixel-stitch.net/ in order to produce patterns.
These sites do two things: they downsample an image in space to produce fewer pixels, but they also sample the colors in an image to have fewer colors of thread to use.
My intuition is that the naive way to do both of t... |
Normally, when we are dealing with quantization noise, we assume quantization noise as uncorrelated, which is white.
In this case, we assume SNR = 6.02N + 1.76 where N = resolution
However, if quantization noise is correlated to input signal (ex : direct digital synthesis),it's likely that q noise is not white, rather ... |
Welch's method is a way to get better power spectral density (PSD) estimations than simple, naive periodograms. It has two main components:
Cutting the input into many segments and average their individual PSD to reduce variance.
Windowing each segment to reduce leakage from one frequency bin to the other.
My questio... |
I want to create a Comodulogram to assess PAC. I use the pactools library, which uses frequencies. Since I work with hourly data and want to look at multidien phases, it would be better to use periods instead of frequencies.
import numpy as np
import matplotlib.pyplot as plt
from pactools import Comodulogram, REFERENC... |
I am doing an undergraduate project where I want to first characterize an instrument based on its pitch and timbre and make an instrument recognition system afterward. Pitch characterization is pretty straightforward, I am planning to use the YIN algorithm to find the fundamental frequency of my samples. Now, timbre on... |
I have some mortgage data and I am trying to detect when a borrower refinances. To do so, I use a moving average filter to smooth my data and then I run a Bai and Perron (1998) structural break test on the borrowers monthly payments. I am wondering if it would make more sense to smooth the levels of the data or the mon... |
The beam pattern of a uniform linear array is
$$B(\theta) = \frac{1}{N} \frac{\sin(\frac{N}{2} \frac{2\pi d}{\lambda} \cos(\theta))}{\sin(\frac{1}{2} \frac{2\pi d}{\lambda} \cos(\theta))}$$
where $N$ is the number of elements, $d$ the interelement spacing and $\lambda$ the wavelength and $\theta$ the grazing angle.
I w... |
I have been reading "Understanding Digital Signal Processing" by Richard Lyons and I find the example in Section 3.11 on zero-padding a little confusing. Incidentally IMHO this is the best written DSP book (by far) and that is why I get concerned if I don't fully understand something.
In Figure 3-21 we see the DFT inpu... |
Suppose my data vectors $x$ consist of IID Cauchy values. Does LMS filter diverge for all values of step size $\alpha$?
I tried some simulations and it appears so. Wondering if this is well-known or how to show this.
|
I am a senior in high school and am currently trying to conduct an exploration of Fourier Analysis, specifically using the Discrete Fourier Transform to analyse a chord played on my piano. Basically, I'd play a chord of 3 notes, deduce the different frequencies of the sound wave, and estimate the chord played. Accordin... |
I have an digital system that needs an analytical signal of length (in time) T. The system is sampling the analog signal with time interval delta t such that T / delta t = 128. So I have 128 samples. I have a FIFO buffer to which, at each delta t, the new sample is stored.
The signal however is not analytical but real-... |
I know that
is the 1D Fourier Transform (FT), and the 2D FT is and 3D FT is , but I am not sure whether these expressions are in fact Fourier transform integral equations for a 1D signal, expanded into 2D and 3D. If they are, then I assume 4D would just be $$ G(u,v,w,\omega) = g(x,y,z,t)e^{-i2\pi \cdot (xu+yv+wz+\ome... |
Suppose there is a moving average process of order 1 that also includes a zeroth term thus:
\begin{equation}
Y_t=\theta\left(B\right)X_t = \left(\theta_0+\theta_1B\right)X_t
\end{equation}
where $X_t$ is white noise (normally distributed). Now I know that the PSD can be determined from:
\begin{equation}
f\left(\omega\r... |
Following figure is drawn as the block diagram of the digital modulator:
The output of the filter block (with the impulse response p(t)) is the convolution of the weighted impulse train with the impulse response of the filter, which in the time domain is equal to:
$\sum_{k} {a_k} p(t-kT)$
Considering that the convolut... |
I'm digitizing a zero-mean complex Gaussian white noise signal with certain variance, through independent I/Q baseband sampling (two ADCs).
The noise variance (power) depends on the thermal emission/signal of a certain microwave body, so out of the ADC I want to be able to compute the signal power to gain information o... |
I have heart rate data from a wearable sensor (no ECG). I want to perform Discrete Fourier Transform to see if some patterns are present. I have over 200 signals, as each individual has heart rate measurements for multiple days. How can I analyze these signals? And is it also to compare population group A with group B?... |
I was trying to compare ORB-Slam2 performace with ORB-Slam3 performance. Consider following comparison from ORB-Slam3 paper (columns with numbers correspond to different sequences from EuRoC dataset):
In 5 sequences (green), ORB-SLAM3 has better performance, in 4 (red), ORB-SLAM2 has better performance. One (blue) is ... |
Recently I met a question when I read the paper Optimized Degree Distributions for Binary and Non-Binary LDPC Codes in Flash Memory and Soft Information for LDPC Decoding in Flash: Mutual-Information Optimized Quantization.
Both papers give the expression of $I(X;Y)$ for 3 reads, the former is:
$$I(X;Y) = H(p_{14}, p_{... |
I am reading the Remez algorithm for designing FIR filters. In the first paragraph of section 15.3.1 of Andreas Antoniou's digital signal processing book, I have two questions.
extremals are to be located at the left-hand and right-hand band edges of each band. (why？)
B 1 + B 2 +···+ B J , should be divided intor + 1 ... |
I want to ask you a question about the waveform synthesis or more spesifically speech synthesis. Most of the state-of-the art papers use mel-spectrograms as their inputs, because it mimics the human ear in a best way. However, there is no inverse transformation that maps mel-spectrograms into waveforms without loosing ... |
Let $x(n)$ be a sequence of length $N = LM, n = 0,\dots,N-1$.$N$ and $M$ are integers.
The Discrete Fourier Transform (DFT) of $x(n)$ is given by
$X(k) = \sum_{n = 0}^{N-1}x(n)W^{nk}_N$
where $W_N = e^{\frac{-j 2 \pi}{N}}$.
Let $n = l + mL$ and $k = q + pM$
$0 \le l,p \le L-1$, $0 \le m,q \le M-1$
$X(p,q) = \sum_{m = 0... |
If the transmitter is on a satellite in a circular orbit at Low Earth Orbit with a QPSK modulated waveform at 1 megabaud centered at 12 GHz downlinking to the receiver, which is
stationary on the ground. How do we model the channel here?
I was thinking since we need to have a line of sight, Rician fading channel could ... |
I came across an interesting question from Strang & Nguyen$^\color{magenta}{\dagger}$ where it asks to design a $3$-point (or 4-point) high-pass filter with better performance than the $2$-point moving difference filter, which have the impulse response
$$ h(n) = 0.5\delta(n) - 0.5\delta(n-1)$$
Please see the question b... |
We know that the frequency-domain correlation function and the power delay profile appears as a Fourier transform pair, i.e.,
$$A_{c}(\Delta f) = \int_{-\infty}^{\infty}PDP(\tau) \cdot e^{-j2\pi \Delta f \tau} d\tau.$$
In practice, we can measure the PDP from the measurments and obtain the PDP in the discrete form. The... |
After studying digital communication courses, I wanted to implement these theories and create an application for sending and receiving a gray image via WiFi between two Raspberry Pi devices. The choice of WiFi modules was arbitrary; my goal is to understand the steps and their implementation.
So, I read an image where ... |
I have been studying the problem of room impulse responses (RIRs) interpolation for a couple of months. I am trying to use compressed sensing to reconstruct (at best) the sound field in the room with a better resolution, or (at least) get an RIR at some desired location where it was not measured before. "A better resol... |
My understanding of the Short Time Fourier Transform is that first I need to frame the signal and window it, after that apply an fft on these windowed frames. Using Librosa:
signal = np.array([1., 2., 3., 4., 5., 6., 7., 8., 9., 10.])
frame_length_ex = 4
hop_length_ex = 2
frames_ex = librosa.util.frame(signal, frame_le... |
i am trying to build a pulse radar using barker codes. i modulate the barker code and sine wave and added some noise as well. then i attached a matched filter whose parameters are shown in the image. and then made an attempt to demodulate, which i don't think is right.
can someone please help me with the matched filter... |
IEEE std1241-2010 specification part 4.2.3. step signal setup part shows like this design
What should the test setup described in section numbered be like? Is it enough to give one pulse and read it with ADC?
"Precision pulses and step signals can be used to measure both time domain parameters
(such as impulse respons... |
I have a full band audio signal (sampled at 48000Hz) and I want to process only the low frequencies with an FIR filter.
The maximum frequency that I need to process with the FIR is 3000Hz, i.e., an eighth of the Nyquist frequency of my original signal.
Computational cost of the final processing is critical to my applic... |
Suppose I have a discrete-time signal vector, for example, x(n)=[1,a1,a2,…,aN].
The signal is then transmitted by using the single carrier pulses,
constituting a single-carrier communication over a multipath channel.
Hence, at the receiver side, the received signal experiences also the zeros of the z-transform of the c... |
I would like to generate brickwall low pass filtered white noise. a cutoff of 1/n is fine for integer n. I suspect that I could mess around with samples that are convolved with the sinc function, but I hope there is a better way? preferably would like to keep the HF noise less than the dither level or if not possible t... |
I have a dataset that contains a lot of recordings sampled incorrectly. This incorrect resampling could be explained in two ways:
A recording S, which was recorded with a sampling rate of 16kHz, was upsampled to a sampling rate of 48kHz. This upsampled recording S was mixed with other recordings in a dataset containin... |
I would like to find periodicity and the oscillation frequency in a signal. I used this simple MATLAB code which works find for some signals, but for a harmonic polluted signal like the one given below it generates wrong results. Is there any way to estimate the frequency of oscillation for a signal like the one given ... |
Why window functions can't have both narrow main lobe and low-level side lobes?
i got a hint from my mentor that it's quite often expained in books about wavelets...
I thought it could be associated with uncertainty heisenberg principle but as understand this principle more about time-frequency uncertainty i.e. size of... |
In physics, a damped harmonic oscillator that is driven by a sinusoidal force has a steady state solution:
$$
x(t) = \frac{F_0}{m Z_m \omega} \sin(\omega t + \varphi)
$$
where
$$
Z_m = \sqrt{\left(2\omega_0\zeta\right)^2 + \frac{1}{\omega^2} (\omega_0^2 - \omega^2)^2}
$$
is the absolute value of linear response functio... |
I am aware that if we do not subtract the mean value from the white noise at the beginning (if the mean is not equal to 0), that its autocorrelation function will be triangle shaped and not a delta function. I wonder why that is? I'm also wondering why the signal mean is so often removed at the beginning of each proces... |
The documentation for Analog Device's sigmastudio shows some of the formulas behind their biquad IIR filter coefficients.
The usual suspects (HP, LP, peaking, ...) as described in RBJ's cookbook are all there. A couple more specialised filters (Butterworth & Bessel) are here as well.
However, what really caught my eye ... |
I am working towards establishing the error rate of the communication system relying on on-off keying. I have trouble understanding the estimation of bit error rate (BER) performance when the following formula
$$\mathrm{BER} \propto \mathrm{erfc}(\sqrt{\mathrm{SNR}}) $$
is used.
Suppose I have a received signal
$$x(t) ... |
I am working on the compression of a bmp image using discrete cosine transform.
First, I split the image into 4x4 blocks and then apply a DCT to each block.
I know that low frequency coefficents (the ones I want to keep for image compression) are grouped in the top left corner.
I want to apply a low pass filter to the ... |
What's the minimum nearest neighbour approximation and upper approximation of this?
Consider a modulation design with the signal space diagram illustrated in
the figure below. The signals are transmitted over an AWGN channel with
power spectral density $ N_0/2 = 5 × 10^{−6}$. The maximum signal energy is $25 × 10^{−4}$... |
I'm working on a vst plugin where a delay is used, the seek position of the delay sometimes changes gradually sometimes abruptly, when sudden changes occur sometimes a clicking sound is produced.
Some VSTs like GrossBeat (similar plugin) have an option to reduce clicking, what methods can they be using to reduce clicki... |
The graph includes a peak at zero frequency and some other frequencies. However, my professor says that there should not be a peak at zero. I contend that he is wrong. But my graph is also not right, it should show the second-largest peak at $50 \, \textrm{m/s}$.
This is my entire code:
%initialize all variables
close... |
I am working on an audio resampler that is realized as a polyphase FIR filter. While testing, I found that it does not suppress aliasing as well as I would have expected. For testing, a sine sweep is upsampled by a factor of 8, distorted by $\tanh(kx)$, and downsampled to its original rate. For $k \lessapprox 3.7$, the... |
I'd like to make a faster version of the Gaussian blur for Inkscape. Here's what I found various open source projects use:
librsvg: 3 box blurs when stdev > 2. Else, separable FIR filter.
Krita: FFT convolution or separable FIR. KisGaussianKernel::applyGaussian
Gimp (GEGL): IIR (aka. recursive filter)
Inkscape: FIR o... |
I need to acquire an analog signal via an ADC and send it via a digital communication line (CAN bus) with a given frequency $f_{send}=100Hz$. If I understand correctly this means that this constraint imposes a low pass filter on the acquired signal of $f_{send}$. My question is: what is the correct sampling rate to us... |
I'm an EE undergrad that struggles heavily with the intuition behind the Fourier Transform (most likely due to a shoddy mathematical foundation). Specifically:
From what I understand, the real part of the Fourier Transform is the Fourier Transform of the even part of the function in time domain, and the imaginary part... |
The following is my analysis. (The analysis is based on analysis from ESE 471: Example Union Bound M=8 Box QAM) Please correct me if I am wrong.
Suppose each symbol has a distance of $d$ from its nearest neighbour.
4 corner points has 2 neighbours.
$$4 \times 2 Q(\frac{d} {\sqrt{2 N_0}})$$
Four inner points has 4 neig... |
I have been working on a homography problem where the only known real world coordinates are the length between the parallel lines. For corn, the length between the rows is exactly 30 inches. Given an image of crop rows, I need to be able to find the distance between any two points in the real world. I am implementing t... |
There is a manual which presents a filter response. In the video they present a formula and a plot of the response. However when I tried to implement it in MATLAB I get a totally different plot.
Where did I go wrong implementing this formula?
Thanks.
clc
clear all
s=0:0.01:50;
H=1./((s/20.01).^4+2.6131*(s/20.01).^3+3.... |
I want to perform experiment(s) on a digital control system. I have several questions:
In analog control systems, the adder is usually an operational amplifier. However, since here the system will be discrete the adder will be a $3$-bit adder, right?
A discrete-time discrete-amplitude signal is defined only for time... |
I am trying to apply np.fft.fft on a 2D array of data of shape (98, 1024), where 98 is the number of trials and 1024 is the sampling rate in Hz. I am trying to find to get the Fourier coefficients at 500 msec (corresponds to the 512th sample) at a frequency of 10Hz. But in my understanding np.ftt.ftt returns an array o... |
SPEAKER AS RLC CIRCUIT
I read this article here which demonstrates a simulation of a speaker as a simple RLC circuit where the RLC components are in parallel:
MY GOAL
I am interested in creating a sample domain C++ script that can take audio input samples at a given sample rate and return the processed audio "output" ... |
I am trying to understand the concept of CRLB for a simple sinusoid in AWGN with a simulation in MATLAB.
According to Kay (p. 57) the CRLB for phase estimation is:
$var\{\phi\}=\frac{2(2N-1)}{SNR\cdot N(N+1)}$.
I generated a MATALB script that simulates 200 sinusoid signals with different amplitudes and fixed noise pow... |
E.g. the Symbol Sync block has a Loop Bandwidth, which (if I read the source right) ends up as a PI loop.
I know what PID does mathematically, and what FIR and IIR do, but how is one better than the other?
My non-expert guess assumed that an IIR or FIR would be better at tracking drifts, and discarding false samples (e... |
Consider a discrete-time signal consisting of a desired signal component and noise, as follows.
$$y_k=d_k + n_k, \qquad k\in\{0,1,\dots,K-1\}$$
This signal is applied to a linear time-invariant (LTI) filter with the goal of recovering the desired signal. In a "data-aided" scenario, where the desired signal is known a... |
I am using a raised cosine filter in my QPSK receiver and I see this IQ diagram of the signal, but it seems off to me. Can someone explain why this happens?
This is the MATLAB code I used to generate it
% Parameters
M = 4; % QPSK
symbolRate = 1e6; % 1 megabaud
fs = 2 * symbolRate; % Sampling frequency (oversampling)
n... |
Related to this.
I have been into exploring the world of digital control loop but I have encountered a very simple problem.
The unity negative feedback transforms a system which delays a signal by 1 clock input from :
$$\frac{1}{z}\rightarrow \frac{1}{z+1}.$$
However since the pole is exactly on the unit circle we cann... |
I'm building a Mode S and ADS-B receiver for TDOA calculation. I am planning to use an SDR with Zero IF to receive the downlink 1090MHz replies from aviation transponders.
Currently, I am modeling the work on Matlab and I would appreciate any suggestion or advice.
Let's start with how a transponder works.
The attached ... |
I send a QPSK signal over pluto sdr and receive it from rtlsdr. As you can see from the screenshot below SNR of the received signal is very high, and the signal is very distinguishable. Hence, it is very easy to find carrier frequency and bandwidth of the signal by monitoring the spectrum. I wonder if there is a robust... |
I am researching a dynamic system. It takes in a sine wave A*sin(2*pi*f*t) + D and outputs a non-isosceles triangle wave. Let's call input signal SP and output signal PV. I want to create amplitude and phase characteristics of this system. I have taken measurements of SP and PV for different fs. Amplitude A was the sam... |
I'm trying to deal with this question about multiplexing and choosing an appropriate modulation format. It's in two parts.
(i) You need to design a system for the transmission of surround audio channels (7+1). The maximum frequency of each channel is 22 KHz, however, you need to oversample the signal at twice the mini... |
I am trying to design a simple half-band complementary filter bank following this paper.
I don't have a Matlab license at the moment, so I am trying to implement it in Python, but I'm running into several issues.
The first issue is my calculation of the maximum and minimum cutoff frequencies of the signal. I follow the... |
I am doing a project for uni where i try to change the precieved gender of an audio using dsp techniques on matlab i started with pitch and am trying to move to formants now but i am not sure i fully understand the concepts.
I saw that changing pitch (i used a phacevocoder and then resampled the output) also effects th... |
As mentioned in the slide below (full slides located here, starting at PDF page 29), the apparent noise floor on a DFT plot depends on the number of sample points $N$ used to calculate the DFT. The next slide mentions one way of "correcting" for this on a given DFT plot, which is to normalize to the noise power per 1-H... |
I would like to know how to create an algorithm from the table below in order to distinguish between data that has artefacts and data that does not.
I'm trying to leverage video attributes like intensity, brightness, and colour to identify flickering or glitchy sections in a video using the data below, which was gather... |
I have a collection of discrete timeseries, which has already been smoothed temporally. However, the smoothing operation does not consider the boudary properly, and results in obvious distortion. I seek to find the correct smooth algorithm, then try to compensate it at the boundary in someway. Note that the original ti... |
The other day, I asked about how to model a receiver from 1090MHz ADS-B and Mode S signal, link to the post. With the support from Mbaz. I thought that it was not accurate that if I used a Zero-IF SDR to receive the signal, I should model the receiving chain from the baseband. Luckily, I borrowed a PlutoSDR from the s... |
im capturing ultrasonic waves using an analog MEMS mic and other components on a PCB. My signal is similiar broadband and non-stationary (see 2) and im trying to remove/reduce environmental noise. I have seen this post, but according to my research, the preferred way to do this is using the (dicrete) wavelet transform ... |
I played around with many microcontrollers before but never had a chance to do any kind of signal processing.
I'm trying to learn how to implement digital signal processing algorithms on the microcontrollers, but before the actual implementation of the algorithm I'm trying to understand the high-level overview of the w... |
I am trying to learn how to plot by hand Nyquist plots.
I took
$$G(s) = \frac{1}{s(s+1)(s+2)(s+3)}$$
I converted it to $G(jw)$ :
$$G(jw) = \frac{i w (-11 + w^2)}{(1 + w^2) (4 + w^2) (9 + w^2)} + \frac{6 - 6 w^2}{36 + 49 w^2 + 14 w^4 + w^6}$$
Now when I am computing the points that cut the axes we have:
$$Re(G(jw))=0 \i... |
In this single carrier QPSK receiver, is designed to receive QPSK data from a satellite in Low Earth Orbit. Since the satellite speed in this orbit is around 3000 m/s and our carrier frequency is 12 GHz, the doppler shift will be around 320 KHz.
Our sampling rate is 2 MSPS and our symbol rate is 1 MegaBaud. If I want t... |
I have time-series data with a sampling rate of 1 Hz, hence the Nyquist frequency is 0.5 Hz.
I am interested in the frequency band 0.01-0.1 Hz and I would like to apply a Periodogram to transform the time- into the frequency-domain to obtain the power spectrum of the data. Subsequently, I would cut the frequency band a... |
It seems impossible to find a precise answer to this question. I researched for quite some time but different sources come up different definitions, that sometimes disagree or contradict with each other.
So far, I got the following answers:
Level refers to the loudness or intensity of a signal. It is measured in decibe... |
I tested the function np.fft.rfft using the following code and determined the magnitude of a $50 \, \textrm{Hz}$ sine wave.
import numpy as np
import matplotlib.pyplot as plt
frequency = 50
f_sam = 192000 # sampling freq
dt = 1/f_sam # time step
t = np.arange(0, 1/frequency*2, dt)
# 5V 50Hz-sine wave
y = 5 * np... |
I have a three-phase electric signal and I've been asked to obtain its Clarke-Park transform (also called direct-quadrature-zero or dq0 transformation).
I'm using this Python library, and I've gotten the tutorial to work, which is the basic process I'm trying to replicate.
My code is quite simple.
# Each variable x_r, ... |
I was wondering if given a picture from a Raspberry Pi NoIr camera and a Raspberry Pi RGB camera, how can the two images be aligned. From what I understand the NoIR camera when used with the blue filter stores near IR values in the Red band. So how could the RGB image and the NoIR image be aligned? Can the IR band of t... |
On page 201 of https://stanford.edu/~dntse/Chapters_PDF/Fundamentals_Wireless_Communication_chapter5.pdf,
it is mentioned that
This observation suggests that the capacity result (5.89) holds for a much
broader class of fading processes. Only the convergence in (5.91) is needed.
This says that the time average should c... |
Let X be an N-point sequence in the frequency domain (where N is a power of two):
$$ X = (X[0], ..., X[N-1]) $$
where $X$ is the complex and conjugate symmetric forward DFT of the real sequence $x = (x[0], ..., x[N-1])$.
I am interested in using a single half-sized ($\frac{N}{2}$) FFT to compute the sequence $x_{h1} = ... |
As part of our product, we use a sensor/beacon to measure the inclination/angle of the product. We do this with a KKM 3-axis accelerometer (model: https://www.kkmcn.com/long-range-beacon-k5p). However, we face a big problem in the inaccuracy or lag of the sensor that controls the game on the mobile phone. The sensor is... |
I'm trying to improve my poor understanding of analogue filters. Similarly to this question, I've always wondered why libraries like scipy offer an option to design an analogue filter, and when I would want to use that mode.
Re-phrasing the other question more specifically: What do the (digital?) numbers of the filter ... |
I have been struggling to understand how to calculate the length of a sequence and also the minimal N to choose in order to avoid aliasing.
Most sources tell me to take the (last non-zero value - first non-zero value + 1 = N)
In this case, I calculated the length of x1[n] to be (3-0+1) =4 and x2[n] to be (3-1+1) =3
And... |
I am working on the simulation of FMCW radar, and I am trying to figure out how to plot the target range using the impulse response of the beat signal. I have successfully generated the chirps and the beat signal applied Hanning window to the beat signal and then applied FFT and IFFT on the beat signal to get the range... |
This a not a typical QPSK where $\alpha \neq \beta$
My book gives me the following
$$
P_e
\leq
Q(\frac{2\alpha} {\sqrt{2 N_0}})
+
Q(\frac{2\beta} {\sqrt{2 N_0}})
+
Q(\frac{2\sqrt{\alpha^2 + \beta^2}} {\sqrt{2 N_0}})
$$
The following is my analysis. (The analysis is based on analysis from ESE 471: Example Union Bound... |
I have gathered data using an FMCW Radar with a configuration of 64 chirps per frame and 64 samples per chirp. Now, I'm trying to simulate the data as if it were collected by a radar with a 32x32 configuration. Specifically, I am interested in applying cropping techniques, either on the raw data or the Range-Doppler ma... |
I have a signal sampled at $Fs$ [Hz], a big size FFT is used to detect some signals, then a small size IFFT is use to transform that portion of the spectrum in the time domain, my question is what would be the new sample time of the IFFT output vector related to the original sample time?
|
Compute the exact symbol error probability.
This is decision area.
For this rightmost points and leftmost points.
$$\bigl(1-Q(\frac{A} {\sqrt{2 N_0}}) \bigl)^2$$
For three points in the middle.
$$\left[\bigl(1-Q(\frac{A} {\sqrt{2 N_0}}) \bigl)^2 \right]^3$$
So the total error probability
$$P_e = 1- \bigl(1-Q(\frac{A}... |
I am trying to find out the algorithm for taylor window function but i have looked everywhere and i dont see any proper algorithm.
I read it is similar to Kaiser Window but thats all I know.
Can someone help me find the algorithm?
|
I do not know if this question fits this stack, but I think it might suit here because it is something related to the radar sensing processing.
The question is about the difference between the cooperative/collaborative radar system when localization a target. For the papers and textbooks I studied, I find they all use ... |
I have $2$ sensors allowing me to measure angles of arrival. These sensors are a distance $d$ ($d \simeq 10$ m) apart and the target to be located is at a distance $R$ with $d<<R$. In the literature we see the formulation for the angle of arrival: $\theta_i=\theta_{i0}+n_i $ with $\theta_{i0}$ the true angle for sensor... |
I want to perform a spectrum analysis of my own voice using the code below.
recObj = audiorecorder;
recDuration = 10;
disp("Begin speaking.")
recordblocking(recObj,recDuration);
disp("End of recording.")
playerObj = play(recObj);
disp('Properties of playerObj:');
myRecord = get(playerObj)
y = getaudiodata(recObj);
fi... |
I'm trying to learn the concept of Pulse-density Modulation.
The wiki page gives a good overview and pseudocode of encoding procedure, but it lacks the decoding part:
The process of decoding a PDM signal into an analog one is simple: one only has to pass the PDM signal through a low-pass filter. This works because the... |
I have a sampled audio signal that is corrupted with a noise pulse that occurs at regular intervals. The pulse is in the form of several periods of a sine wave, and I believe the frequency of that sine wave increases and decreases very slightly in a (seemingly) cyclic pattern.
Very roughly, I would say the pulses occur... |
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