instruction stringlengths 31 27k |
|---|
For an random signal $X$ (does not matter if it is correlated or uncorrelated), given lag $τ$. If white noise is added to this signal, then can the summation of them be a correlated signal? Also, even if these two are individually uncorrelated. Could somehow be related mutually?
|
I am a student working with digital circuits. I just started a term project with someone else working on an ADC. I need to design a IIR filter in verilog.
If the output of an ADC represents 1V by setting all output bits to '1' and 0V by all bits to '0', and I wanted to use a fixed point scheme for the filter coefficien... |
How does LBP work? To learn that, I decided to calculate by hand for the matrix but the problem is about the rotation pixel over the image based on the radius and neighbor of LBP. I do it clockwise and top most left then top most right, but the result is still different from the result in "skimage" lib.
With the examp... |
I am interested in the audio classification problem. You can think of the problem as ML, but my question is from a signal processing perspective. I am questioning the classification performance of MFCC in real-time (data is coming as streams) applications.
Let me summarize the process.After labeling the audio recording... |
Suppose I want to implement only via software on matlab a DFE.
I am studying the decision feedback equalizer but some slightly more practical concepts are not very clear to me from the theory.
Specifically:
How do I choose the number of taps (feedback and feedforward) to use in the equalizer? Are they chosen based on ... |
Sound attenuation can occur due to various phenomena. The most known are
Distance: For a point source the sound pressure decreases by 6dB with each doubling of the distance to the sound source.
Atmospheric absorption: Depending on the air pressure, humidity and tempreature sound attenutate increasingly faster if the f... |
I'm working on developing an audio device testing suite (for speakers and like). I'm looking for approaches to detecting two types of noise: static such as white or pink noise and artifacts such as crackling, 'wheezing' etc. I'm kind of new to the DSP field and was wondering how I should approach this task? The plan is... |
I am practicing Digital communication (Baseband demodulation/Detection) when I encounter this question.
An analog signal is PCM formatted and transmitted using binary waveforms over a
channel that is bandlimited to 100 kHz. Assume that 32 quantization levels are used
and that the overall equivalent transfer function is... |
I'm interested in identifying a square wave signal from recorded audio. This subject may be a complex problem so I would like to present the problem where I'm currently stuck at.
I recorded the audio at 44100 sampling rate, and applied FFT on the audio at the sampling rate, in ~0.1s intervals. For 10 seconds.
A 440Hz s... |
I don’t understand how this phenomenon appear in OQPSK carrier recovery loop. I found nothing on google. Is it a normal phenomenon in OQPSK loop or specific? Why does it get reversed I and Q? In the figure(b), why does the sampling point change?
|
The signal is an impulse repsonse. I am filtering the signal using a windowed sinc filter in both frequency and time domain. I am interested in knowing the differences between the two methods and artefacts present in the filtered signal.
FILTERING IN FREQUENCY DOMAIN:
I am multiplying a signal with a windowed sinc filt... |
Sorry about the wooly title, but I couldn't think of a concise way to describe this question.
I have a ZX Spectrum emulator project which can visualise the output from the sound chip example here. I think it's an accurate depiction, but I'm outputting the waves just as they are generated in real time from left to right... |
The Dirichlet kernel of order $N$ is defined as:
$$
D(\theta,N)=\frac{\sin(N\theta /2)}{\sin(\theta /2)}
$$
We know that the main lobe widths between zero crossings is: $4\pi/N$
I was wondering if there is a closed form mathematical expression for the FWHM?
Meaning I'm looking for $\theta_s$ s.t.:
$$
D(\theta_s,N)=\fra... |
I have an issue when I want to filter a signal with a 4th order Bessel-Thomson filter with a cutoff frequency at 18 GHz.
1. Generate data at 50 Gbps (T-spaced)
def prbs_generation(number_symbol=-1, size=15, oversampling_factor=100, bitrate=50, plot=False, save=False):
"""
Generate a PRBS
:param number_symbol: number of... |
I work with radar and I want to understand which spatial frequencies I can measure, i.e. the k-space coverage, given a set of coordinates of transmitter-receiver combinations. The ideal coverage for a reflective and transmissive mode of measurement can be found e.g. in this paper (figures 2 and 3), or here for a select... |
I hope this is the right place to ask this question since it is partially a note which might help others. Until recently, I always used the fft-algorithm implemented in the language of my choice if I wanted to calculate a spectrum. The main reason being it is easily available and is mentioned in every course on the sub... |
This is mine code for fir_filter_design in python. But I'm getting wrong output for low-pass, band-pass, and high-pass magnitude response.
def fir_filter_design(N, cutoff_frequency, window, Fs, ftype):
omega_c = 2 * np.pi * np.array(cutoff_frequency) / Fs
h = np.zeros(N)
for n in range(N):
omega =... |
Having read up on discrete convolution and how it is implemented, I've started to think about which "mode" is applicable to which situation. For two signals x1 and x2 of length M and N, respectively, NumPy and SciPy convolution functions (example doc) support modes:
"full": output of length M + N - 1
"same": output of... |
If given $x[n]$ is zero mean white Gaussian noise with variance $\sigma_x^2$, and a filter with a known impulse response $h[n]$, how would I find the mean, autocorrelation, and power spectrum of $y[n] = (x*h)[n]$?
I believe $\mu_y = \mu_x = 0$. I'm not sure how to find the other 2 components. I don't want to give actua... |
Is a PI-controller considered an LTI system?
Intuitively it seems that the integral part would break the time-invariant requirement requirement, because the output depends on how wound up the integrator is.
On the other hand, the output of an RC circuit also depends on the capacitor charge, so this would also seem to b... |
I have some noise datas which are having lower RMS values. But don't know why they are very loud when I am normalizing and playing them. After normalizing the audio remain still low rms values. But still the sound become louder. What are the possible reasons?
Here is the script how I am reading and playing the audio.
... |
I'm writing for code to detect human face by Local Binary Pattern algorithm in webcam, but with my limited knowledge in optimization code, I having to struggle in detect in each frame , I want to say that I used haar-cascade but my teacher say is prohibited. So I have to browse through each window ([45x45]) on every fr... |
thank you for considering this question. Do you know any reason why Daubechies wavelet can't be used for Continuous Wavelet Transfrom, but only for Discrete Wavelet Transform ? Matlab toolbox and Python packages don't allow for such transform, and I was wondering about theoritical rational here.
One surprising thing is... |
I need to find out the transfer function in Z from this diagram, how can I extract this information from this diagram
Obs: Doing the math, my H_z gave, H_z = 1 + 0.5z^-1 + 2z^-2 + 2z^-3 + 0.5z^-4 + z^-5, but I don't know if This H_z is correct for this diagram
|
I have a millimeter wave radar (IMAGEVK-74). I am new to signal processing and I need to generate images from the radar data.
The specifications of the radar are- it has a 20 transmitter 20 receivers antenna configuration giving 400 Tx-Rx pairs. It is working at multiple frequencies (21 currently) between 62-69 GHz. Th... |
Let $$x(t) = A \text{rect}_T({t-\tau})$$I calculated its fourier transform through the direct way: $$X(\omega) = Ae^{-i\omega \tau} \int_{-T/2}^{T/2} e^{-i \omega t} dt = Ae^{-i\omega \tau} \frac{\sin(T \omega/2)}{\omega /2}$$
but I want to see if there are other clever methods, infinite thanks to everyone!
|
I have C++ App which wich plot 3 plot
amplituede vs sample
amplitude vs frequncey(used FFT for calculation)
amplitude(in dB) vs frequncey
I want plot 2 with y axis lable represent in DB, without chaning the y value of the plot. That is the plot shape will stay same but the y axis lable need to convert to dB.
Is this... |
I have a bunch of images similar to the following:
As you can see, there are faint grains in a roughly ellipse-shaped area, along with a few bright ones. I need to identify the location and density of these. For example, in the marked area, in an ideal case the algorithm would detect approximately the following "blobs... |
The (modified) trigonometric functions $\{0, \cos(kx), \sin(kx)\}$ serves as a basis for periodic function. I have also seen (but not rigorously) that the Fourier transform can also be seen as an extension to the Fourier Series in the sense that we are expressing a signal $x(t)$ in terms of the uncountable basis $\{e^{... |
As the title says, I'm looking for a way to implement the point-wise multiplication of 2 signals somehow as a convolution.
Here's why:
In my system, I have an input signal $x[n]$ which has been sampled uniformly, however with periodically missing samples.
Best way to model this is as follows:
I defined a finite (Krone... |
I have small problem with cv in Python and rhombus. I can read frame from the camera and create contours...
img_BW = cv2.cvtColor(frame, cv2.COLOR_BGR2GRAY)
ret, thresh = cv2.threshold(img_BW, TR, 200, cv2.THRESH_BINARY)
cv2.imshow("ThreshPaint", thresh)
contours, hierarchy = cv2.findContours(image=thresh, mode=cv2.R... |
As far as I understand, the mel scale was developed empirically using something akin to "just noticeable differences" in a subject's perception of frequencies. The scale ends up being a log transformation of the frequency-amplitude space, which stretches and pulls the space so that "just noticeable differences" are the... |
I have a time domain data of some signal in volts vs time. Now I run this data through FFT and let's say I use the bin size 100Hz. I get an array of values: frequency, real and imaginary numbers. What is the physical value of the real and imaginary data in this case ? In other words I need to measure the signal value i... |
The steady-state error improves when the pole moves closer to the origin, as seen in lag or integral compensators with step input. Is there an intuitive explanation for this phenomenon?
|
Let
$$x(t) = A\sin(2 \pi f_0 t + \alpha)$$
its Fourier transform is given by $$ X(\omega) = \frac{A \pi}{i}(e^{ia}\delta(\omega-2\pi f_0) - e^{-ia}\delta(w+2\pi f_0)). $$
the Fourier series complex representation of a $T$-periodic is :
$$x(t) = \sum_{n=-\infty}^\infty c_n e^{(2 i \pi n)/T \cdot t}$$ thus its Fourier tr... |
how to set sound location left/right (in mono audio) and azimuth (up/down) of the wave? I know there are a lot of different libraries doing that, but I'm interested in some minimal examples to understand the method.
upd:
sound is 3d sphere wave, is it possible to somehow define/simulate that using periodic(sin) functi... |
$N$ is an even integer, $x[n]$ is a finite length signal over the interval $n \in [0,N-1]$, and $X[k]$ is the $N$-point DFT of $x[n]$. Analytically find the DFT of sequence below in terms of $X[k]$. DFT size is $2N$ and
So, I did this (note: below, in $Y[k]=X[k/2]$, $k$'s being odd creates indefiniteness.)
I found t... |
I have a synchronization chain that consists of
Coarse frequency synchronization block
Frame synchronization block
Symbol time synchronization block
Fine frequency synchronization block
PLL phase synchronization block
I would like to study the MSE after each synchronization block between the channel input and the out... |
i have an array contains the EEG signal values called inputSignal, and another array contains the importance degree or the attribution value of the corresponding element in inputSignal array ,
i want to visualize the eeg signal as a graph and the corresponding importance values as a background for it exactly the same a... |
Compressed sensing refers to the recovery of a high-dimensional but sparse vector $x\in\mathbb{R}^n$ from its linear measurement $y = Ax+\eta$, where $A\in\mathbb{R}^{m\times n}$ $(m<<n)$ is a measurement matrix and $\eta$ is the measurement noise with a known upper bound $||\eta||_2\leq \epsilon$.
Following is the mos... |
if we have a FIR with 100 coefficients, is there a method to "compress" the length? so for example, have a similar response with 50 coefficients?
I am ok with the response degrading within some spec
|
I transmit OFDM signal with bandwidth 20 MHz via ADALM-Pluto SDR device. To study the effect of the wireless channel, I used the same device for transmission and reception.
The maximum bandwidth of my ADALM-Plut SDR device is 20 MHz. I have read that due to the difference between the Local Oscillators, there will be so... |
I am trying to represent a filter response numerically using the formula shown in the end. I know its electromagnetic response.
Could you please help me understand how to make in Matlab such filter response photo shown below using the $Y_n$ and coefficients?
|
I have a planar array containing 16 transmitters and 88 receivers located on the same plane with dimension 30cm-by-30cm. I have an object of size 6cm-by-6cm. Object and antenna array plane are in parallel, and their distance is 15cm. GSAFT formula is the same as below:
This formula is extracted from the following arti... |
In MATLAB, I tried to convert the fft2-based multiplication using conv2.
Z = rand(100);
k = rand(100);
Fk= fft2(k);
V1 = real(ifft2(Fk.*ifftshift(Z)));
V2 = conv2(ifft2(ifftshift(Z)),k, 'same');
I want V1 and V2 to be the same, but they're not.
I need to avoid using the FFT while calculating V2.
|
I'm exploring the discretization of analog circuits and have come across the NODAL DK method as a potential technique. I'm curious if this method is suitable for circuits that incorporate Operational Transconductance Amplifiers (OTAs). Are there special considerations one should take into account when applying the NODA... |
I am currently employing a centered finite difference method to simulate the excitation of a circular membrane by a forcing term in Python, and the code is functioning as intended. To further enhance the simulation, I am considering the incorporation of a 3D Fourier transform. This would allow me to analyze the respons... |
At first is sampled data from the IMP23ABSU MEMS mic converted to V by multiplying each sample by the conversion factor $ \frac{1}{32786} $. Then i want to calculate the FFT and PSD after applying a passband with 20-80kHz range and windowing. Since im working with a lot of samples (around 17.538.000) it seems only righ... |
To find the frequency content of a noisy signal (PSD), there are two methods below:
#1 Take the fourier transform of its power signal (square the noisy signal)
#2 Find the autocorrelation function of the signal, then take the fourier transform
It seems that they achieve the same result. If so, what is the difference be... |
In this post, LJSilver mentioned that a PI compensator is not appropriate for a constantly changing reference, such as a sinusoidal waveform. In this scenario, a lag compensator is considered the better choice. Can anyone provide a more detailed explanation of why a PI compensator is not suitable here and why a lag com... |
So, the normalized power for unipolar NRZ scheme is $\frac{1}{2}V^2$. According to my book, the normalized power required for polar NRZ is half that of unipolar NRZ. So, the power should be $\frac{V^2}{4}$.
However, the normalized power for polar NRZ seems to me to be $\frac{1}{2}V^2+\frac{1}{2}(-V)^2=V^2$. Can someo... |
A function $f(\tilde{\phi}_i,\tilde{\theta}_j)$ defined on the sphere. The interpolation or filtering of the function is depending on $K$. I am confused on how to perform inverse-transformation using $\mathrm{fftw}$.
$$
\begin{align}
f\left(\tilde{\phi}_i,\tilde{\theta}_j\right) &= \sum_{m=-K}^{K} \tilde{f}^{m}\left(\t... |
I am trying to simulate tx, rx and if signals and from the if signal, compute the range and doppler of the target. In the function below, when i generate the range doppler heatmap, i am getting range values correctly in the plot but velocity always remains zero. Can anyone help me with why this is happening? I am consi... |
I've got the following FFT with a sampling frequency of 192 kHz that has around 17.000.000 samples. The signal is a hiss of escaping gas. This looks noisy and i want to determine the characteristic "hiss frequency" that should lie between ±40 kHz (according to other studies).
Since most of the noise in industrial area... |
I came across this research paper. In this they have mentioned power spectral density calculation of various file. I have read the paper and try to calculate and plot the graph, don't know it correct or not? Don't know how to compute frequency marginal, plot and next step?
A Signal Processing Approach for Cyber Data Cl... |
I'd like to plot an eye diagram with a 'hot' colormap that becomes clearer not when the signal has high values, but when the signal is dense at an instant in time, see picture below.
Here is what I've already done:
def prbs_generation(size=15, oversampling_factor=100, bitrate=50, Fc=18.75, snr=20, plot=True, save=Fal... |
I'm developing a Simulink model for noise according to the description of the algorithm of Accelerometer in this page: IMU simulation model.
My system takes the RandomWalk (defined in [unit], being unit [g] for accelerometer, [m] for a distance and so no), the NoiseDensity (in [unit/sqrt(SensorSampleRate[Hz])] ) and Bi... |
I'm building a vocoder on an FPGA, and now trying to mix $N$ signals that are each the product of some carrier signal in one of $N$ bands and an envelope of a modulator in that same band (essentially a digital VCA). But due to the width of these samples I'm getting clipping. In particular, both the carrier and envelope... |
The formula I know for the periodogram is
$$I\left(f\right)=\frac{1}{NT}\sum_{n=0}^{N-1}\left|x\left[n\right]\exp\left(-2\pi fjnT\right)\right|^2$$
but for this one question I have which states that $N$ number of noise samples $x=\left(x_{1},x_{2}....,x_{N}\right)$ with sampling period of $T=N\Delta t$ the periodogram ... |
Edit:
The source of my confusion is over the existence of multiple s-z mappings.
After researching these mappings and where they came from, I couldn't find why $z=e^{sT}$ can be ignored and replaced by the other mappings. Isn't $z=e^{sT}$ intrinsic to the very definition z-transform?
$$ G(s) = \int_{-\infty}^\infty {g(... |
I have to analyze beating cardiomyocytes. In order to do so, I have written a Matlab script to replicate the method found in Béland et al., Front. Physiol. (2022) (doi: doi.org/10.3389/fphys.2022.733706).
Briefly, for each time point, the algorithm calculates the module of the difference between the intensity of each c... |
Can we frame a mathematical model based on the clock frequency (CPU cycles), to compute the energy consumption per bit for a LDPC encoder?
|
I'm struggling to work out the relationship between Symbol Duration and Sampling Frequency in an MPSK link.
For instance, if the Sampling Frequency is 10MHz this means that a sample is taken every 10ms and this would suggest one bit every 10ms
If we have an Alphabet Size of 256, that requires log2(256) bits to encode, ... |
I am looking for an open source LDPC encoder available online as I just have to use it for encoding raw bits to LDPC-encoded data to estimate the energy consumption. Which is the best encoder from the point of view of lower energy consumption? Also, how to use C files in MATLAB? It will be helpful if the answer include... |
Based on IEEE 802.11a Standards, I am using two Training Sequence-Preamble for Synchronization and Channel estimation.
I have read in some papers that the channel estimation is performed by taking the mean of both received identical sequences of Long training Preamble (each of length 64 samples) and multiply the result... |
I have always used high-performance 32-bit microcontrollers with hardware FPU support. To calculate the coefficients of digital filters, I used the fdatool package in Matlab. I used single-precision floating point coefficients for digital filters. Next, in accordance with this figure, I implemented digital filters:
Mo... |
I am currently reading a paper titled 'Channel Estimation and Data Detection for Insufficient Cyclic Prefix MIMO-OFDM' published in IEEE Transactions on Vehicular Technology. As you can see below, the authors mentioned transforming the time-domain samples (2) into frequency-domain samples (3) using the Discrete Fourier... |
I have used BER vs. Tx-Gain in dB curve as one tool to measure my ofdm system's performance. Besides, I would like to measure the performance using the EVM.
How I can implement this EVM measurement in the same way as BER curve using matlab!
I am using several baseband modulation schemes like BPSK, QAM, 16QAM and 64QAM ... |
I was looking at this graph from The Scientist and Engineer's Guide to Digital Signal Processing, copyright ©1997-1998 by Steven W. Smith:
where basically it shows how differnt representations can contains the same information.
Unfortunately I don't understand how this Impulse Response will be created starting from a ... |
I have to find the Zero input response of the following system if $y[-2]= 0.5$
$$x[n] - 3y[n-1] - 4y[n-2] = 0$$
I am not sure if this is the correct way since every other example I have done had two initial conditions given.
My solution:
$-3y[n-1] - 4[n-2] = 0$ ( Homogeneous response equal to zero)
Let's set $y(n) = Ar... |
Wikipedia states that Bessel filters have a group delay within passband of $1/\omega_0$. While the inverse relationship to knee frequency makes sense, it ignores the effect the filter order has on the delay.
I wonder if there is analytic expression or an approximation for how the delay scales with order (and perhaps cr... |
I made ideal LPF, but result filter image doesn't exist.
Plz give me advice
I = rgb2gray(imread('testa.png'));
M = 512;
I = imresize(I, [M, M]);
[m, n] = size(I);
Ip = uint8(zeros(2*m, 2*n));
Ip(1:m,1:n) = double(I);
F = fft2(Ip);
D0 = round(M/10);
u = 0:2*M-1;
v = 0:2*M-1;
[u, v] = meshgrid(u,v);
D = sqrt( ... |
I determined the PSD of a BPSK modulated signal using the MATLAB pwelch function and default parameters. My code looks like the following:
N = 2000; %no of bits
Rb = 1000; %bit rate
k = 20; %samples per bit
Tb = 1/Rb;
fc1 = 3*Rb; %carrier frequency
... |
I am currently working on a project where i am to estimate a signal x_T using x_1 and x_2 with an RLS filter.
I have a problem where i don't quite get the results i am looking for. I think there is a problem with the filter coefficients. I don't know which ones i am to use.
This is what i have come up with that i need ... |
I have a feature detection algorithm that is called FAST - Feature Accelerated Segmented Test.
It's a very fast algorithm for finding feature points, e.g "corners" inside an image. Here is one example.
The problem with these points is that they are moving around very rapidly, like gaussian noise. Still the shape can b... |
Why is the output signal obtained through frequency response different from the other two.
A = [1, -1.1, 0.6];
B = [1.5, 0, 0];
N = 200;
dt = 0.001;
T = dt * N;
fs = 1 / dt;
t = 0:dt:(N-1)*dt;
u = zeros(1, N);
u(1) = 1;
h = filter(B, A, u);
s = zeros(1, N) + 1;
st = filter(B, A, s);
f1 = 20;
f2 = 100;
x = sin(2 * pi * ... |
Show for the DFT that if all x[n] are real, then X[N - k] is the complex conjugate of X[k] for k > 0. (i.e. if
X[k] is a + bi, then X[N - k] is a - bi)
|
I have been trying to recreate the welch method implemented in scipy.signal by hand but my results don't look close enough in my opinion. How come the scipy.signal.welch method is displaying much smoother results although the settings appear to be th same? What am I missing in my code?
Here is the difference:
import n... |
Let
$$\mathscr{F}\Big\{x_1(t)+x_2(t)\Big\}=X_1(f)+X_2(f)$$
I think that in general
$$\big|X_1(f)+X_2(f)\big|^2\leq\big|X_1(f)\big|^2+\big|X_2(f)\big|^2$$
but I was wondering if
$$X_1(f)X_2(f)=0,\qquad\forall f$$
it is true that
$$\big|X_1(f)+X_2(f)\big|=\big|X_1(f)\big|+\big|X_2(f)\big|$$
so one can find the Fourier tr... |
I have generated a large set of audio using Pyroomacoustics for room simulation, employing a circular microphone array and a single sound source.
I have successfully trained and tested a CRNN (Convolutional Recurrent Neural Network) model with this audio dataset to predict events and DOA estimation.
However, when using... |
I am trying to implement the backstepping techniques for quadrotor described in this article : Backstepping control for a quadrotor helicopter: Madani & Benallegue 2006
Since I am new to the control theory (and I know that my question concerns the most easiest part of this paper), I don't understand how can I generate ... |
Preparing for my exams, I stumbled upon this particular problem.
I tried to calculate it using following formulae:
$$ E = \int_{-\infty}^{\infty}|t|^2dt $$
Dividing it into two cases because of the absolute value of $t$, I got:
$$ E = -\int_{-\infty}^{0}t²dt + \int_{0}^{\infty}t²dt $$
As result for the previous integra... |
I'm trying to understand how to implement a high-pass filter using, quote:
a cascasded biquad direct form II IIR filter
with a cut-off of 8kHz.
DSP is a complete new field for me, and the terminologies are alien to me.
From reading the web, I roughly understand that:
A digital filter is roughly a function (a transfe... |
As part of my effort to understand how a high-pass filter can be implemented, I'm reading this section of HandWiki.
My question here is: How do the 2 simple formula presented there filter out at a cut-off at 8kHz? Do we just cleverly set the $a_{1,2}$ and $b_{0,1,2}$ coefficents? Do we not need another (set of) compone... |
I am trying to simulate an Inertial Measurement Unit (IMU) by generating sample noisy gyroscope and accelerometer data. Currently I am analyzing the simulated gyroscope( gyro)
data. Using the noise parameters for generating the gyro data, I computed the analytical expression for the Allan variance as a function of freq... |
I am a computer science student working on a school project involving RTL-SDR receivers, and I have a hard time grasping the concept of IQ data. Basically what I think I have learned from all of the videos I watched on the topic is that the I signal is the captured signal multiplied by some sine wave, while the Q signa... |
While preparing for a mid-term exam, I encountered negative amplitudes for the first time while convolving two signals.
I've already solved the problem, but my result and results from others conflict in the leading sign of the output signal.
Given the input signal $x(t) = u(t-4) - u(t-2)$ and the impulse response of t... |
The Channel Gain Estimation was obtained using Least Square method as follows:
Hk= mean(Rx_LTS ./ Tx_LTS, 2); k= -26 to 26
Where Rx_LTS is the received long training sequence (52 samples after removing the 12 zeros-subcarriers) and Tx_LTS is the known LTS (52 samples).
The plot of this linear channel estimate shows hi... |
I am experimenting with Dr. Schreier's DSM toolbox for MATLAB. I stumbled over a strange phenomenon when having a look at a bandpass DSM in demo-script 2. If I increased the amplitude of the test-tone in one of the demo scripts from 0.5 full-scale to 0.9 full-scale, the DSM would seemingly become unstable (a totally di... |
Any kind of references would be appreciable.
|
There are several thread regarding zero padding at the center and padding at the end.
I have a spectrum and want to interpolate it.
DFT of real sequence: $\left[X_0, X_1, X_2,\dots,-X_2, -X_1\right]$
Padding zeros: Now I can put zeros of required length from Nyquist till first -ve frequency bin.
My confusion is whether... |
Cross-posted from here
I encountered the following question in a Digital Image Processing examination:
Find the 2D DFT of $\frac{1}{2 \pi \sigma^2} e^{-\frac{(x - x_0)^2 + (y - y_0)^2}{2 \sigma^2}}$ where $x_0, y_0$ are integers and $x_0, y_0 \in \left[ -\frac{N}{2}, \frac{N}{2} \right]$.
We are considering an image w... |
I have some files from a spectrum analyzer in RMS mode equipped with a tri-axial antenna. I am supposed to compute the channel power evolution for two distinct LTE channels.
I have files containing the various measures which I parse and upload in MATLAB.
I was using this function to compute the channel power but I'm no... |
I am trying to compare two MEMS accelerometers in terms of their noise for ground motion monitoring applications. I would like to take some "clean" data (recorded by good quality instrumentation), add white noise to it so I can bring it to the noise level of the MEMS sensors I'm testing, and then run some tests on thes... |
Does the pole zero plot for $$H(z)=(1-z^{-1})^3(1+z^{-1})^3$$ have 6 poles at origin and 3 zeroes at 1? Just answer yes or no, not your homework question knowledge
|
I am building a second order modified biquad filter based on an implementation by Will Pirkle in "designing audio effect plugins in c++". However when I look at my implementation's frequency response, things look off: e.g., a +6 dB gain results in more boost than +12 dB. I am basically implementing what I see in the ... |
I'm interested in computing M, FIR filter taps / coefficients using a Wiener filter approach with known signal and noise timestreams. I know of two ways to compute the coefficients: one in the Fourier domain using FFTs which is more intuitive for me and the other in the time domain using a Toeplitz covariance matrix wh... |
Consider:
I'm looking at low-pass filters, and I see that the time domain representation of an "ideal" filter resembles the shape above whereas the frequency domain is a box. I also get the as you lower the cutoff, the main lobe in the middle gets wider.
So that implies that if you "slide over" this shape with a signa... |
A phase screen is a concept used in wave optics to model the statistical properties of wavefront aberrations caused by a medium. In the context of light propagation through atmospheric turbulence, for instance, the phase screen is used to simulate the random, turbulent fluctuations in the refractive index of the atmosp... |
Since STFT uses overlapped sections of the input signals and compute DFT, it resembles overlap-save method used for block convolution, instead the overlap-add method for block convolution uses non-overlapped sections of the input signal.
However, the following reconstruction process confuses me. What is the reason fo... |
Subsets and Splits
No community queries yet
The top public SQL queries from the community will appear here once available.