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I want to speed up my blob detection function by computing the second order derivatives of the input image using the integral image and some box filters.
Initially, I used the following method, which provides correct results.
def smooth_gradient(image, sigma, deg):
# define the filters according to the arguments
... |
I have a DAC which is assumed to be nonlinear, such that it produces unwanted harmonic distortions at integer multiples of the input frequencies. (EDIT: Any other nonlinear distortions, such as intermodulation products, are assumed to be negligible).
If the harmonics fall outside the bandwidth of the analog bandpass fi... |
In FMCW, we have range resolution calculated as delta_d = c/(2B), and dmax = fs * c/(2*S) where S is the slope.
my understanding is numRangeBins = dmax / delta_d. However, in the post (range-doppler map to range-velocity map conversion), it says:
rangeAxis = (1:numRangeBins)*(c/(2*fs)) % Range bin size is c/(2*fs).
f... |
Let N be an even integer, and $x[n]$ be a finite length signal over the interval $n\in [0,N-1]$; let $X[k]$ be the N-pt DFT of $x[n]$.
Find the 2N-Pt DFT of $x_1[n]$ in terms of $x[n]$, where:
$$x_1[n] = \begin{cases}
x[\frac{n}{2}], & \text{if } n \text{ is even}\\
0 & \t... |
In automatic speech recognition the spectrum $f$ is interpreted as the convolution of elicitation $e$ and impulse response $h$.
By applying the cepstrum $cep(f) = FT^{-1}(log(FT(f)))$ this convolution is turned into an addition:
\begin{align}
f &= e &* h\\
FT... |
Can feedback stabilize a unstable system?I guess so since we may get rid of any poles in the positive complex plane by feedback but I am not sure.
|
I am trying to model a Kalman Filter for an IMU (inertial measurement unit) with the method described by Zhou (2004)
and Filippeschi (2017, pp.11-12).
In this method, the state vector is:
$$
X = \begin{bmatrix}g_x\\g_y\\g_z\\m_x\\m_y\\m_z\end{bmatrix}
$$
Where $g$ is the gravity vector and $m$ is the magnetic vector.
T... |
I'd like to implement a lowpass with arbitrary slope. I'm currently cascading biquads in order to get a higher order, hence higher slope. But is it possible to do the opposite - lower the slope of one biquad? Normally one would be 12db/oct, so anything lower than that. Sort of mixing with dry signal, but without the ph... |
Based on this post here, I tried to calculate the phase difference between sine waves. Here is my code.
Fs = 64*10**6
N = 2048
f = 100000
t = np.linspace(0, (N-1)/Fs,N)
order = 3
amp = (2**(12)-1)/2
def butter_lowpass(cutoff, fs, order):
return butter(order, cutoff, fs=fs, btype='low', analog=False)
def butter_low... |
I'm working with some IQ data from the paper A Wideband Signal Recognition Dataset. Specifcally I've been trying to plot the constellation diagram for some the modulations within the dataset, however, when plotting IvQ I get something like this:
The meta data corresponding to this IQ data/plot looks like this:
"core... |
Consider the MIMO system which has $N_t$ antennas at transmitter and $N_r$ antennas at receiver and uses Generalized space shift keying (GSSK) modulation. The received signal is given by:
$$Y = H X + N_w\tag 1$$
where, $H$ is the channel of dimension $N_r \times N_t$, $N_w$ is AWGN of dimension $N_r \times L$, $X = [\m... |
The adjoint of the DFT is the IDFT. But suppose we don't need to calculate a full DFT (just a subset of frequencies), and so are using the Goertzel algorithm instead. What is the adjoint of the Goertzel algorithm?
One way to think about this might be to model the Goertzel operation as a full FFT followed by truncation ... |
I have an audio file which was created by substituting multiple speech segments into an original utterance.
I want to find the timestamps where such substitutions happened, so I guess that one thing I can do is looking for the abrupt changes in the audio waveform.
By visualizing the waveform, I have something like bel... |
I have 2 time-shifted signals,$f(t-t_0)$ and $g(t-t_0)$. Assuming that $y=f(t)*g(t)$, which is the convolution of the signals.
By definition, I have $f(t-t_0)*g(t-t_0)=\int_{-\infty}^{+\infty}{f(\tau-t_0)g(t-t_0-\tau)d\tau}$
Substitute $\tau-t_0$ with $t_1$ then I have
$f(t-t_0)*g(t-t_0)=\int_{-\infty}^{+\infty}{f(t-2t... |
I am new to SDR. But I have an USRP X310. Installed UHD and GNU Radio from source on Ubuntu 20.04. Connected to the USRP with SFP+ through the port 1 and powered the USRP. But the LED of the ethernet port doesn’t light up at all. And it seems like the computer also doesn’t see the device.
Checked with other cables and ... |
Let's assume that there are two convolutions:
$y_1 = (h_2[n]\cdot x[n])*(\bar{h_1}[n]\cdot\bar{x}[n])$
$y_2 = (h_3[n]\cdot x[n])*(\bar{h_2}[n]\cdot\bar{x}[n])$
where "$\bar{x}$" is complex conjugate, $\cdot$ is element-by-element multiplication, and $*$ is linear convolution, that is $(f*g)[n]=\sum_mf[m]g[n-m]$.
I am l... |
I am new to signal processing and I want to implement an LPF using SciPy. In order to do so, I used the following python code from: here
The code itself looks like:
import numpy as np
from scipy.signal import butter, lfilter, freqz
import matplotlib.pyplot as plt
def butter_lowpass(cutoff, fs, order=5):
nyq = 0.5... |
I saw many answers for such questions by writing down equations such that the symbol is formed of n bits so the duration of the symbol in QPSK must be n multiple of the bit duration.
I am not convinced as from my understanding, QPSK is making use of the phase to represent two bits together using same bit symbol we used... |
I have a signal $3.5 \cdot 10^{-5}s$ long as shown below.
It was unevenly sampled so I did cubic spine interpolation.
My signal as shown in time domain about 0.9V and 1MHz frequency.
I want to make an accurate FFT picture with $F_s=10\texttt{Mhz}$ so $T_s=10^{-7}s$, hence
$N = 3.5*10^{-5}/10^{-7}=350$
I want an accurat... |
I'm trying to frequency shift a sine wave at 50kHz by using a complex exponential at 15kHz. I should be getting an FFT with a peak at 65kHz.
Instead I'm getting a strange looking peak at 52.4kHz. Anyone know why this is?
This is probably a simple issue but I've been stuck for too long.
Here is my Matlab code:
clc,cle... |
Let $y(t) = (h * x)(t) + n(t)$ be some observed signal where $h(t)$ is some filter / impulse response, $x(t)$ is some input signal we are interested in, and $n(t)$ is noise.
In order to recover $x$ given $y$ and $h$, I can define a regularized inverse filter, (in Fourier domain)
$$G(\nu) = \frac{H^*(\nu)}{|H(\nu)|^2 + ... |
I have a time-series data file in python and I am asked to apply a Hanning window $w_n=1-\cos\Big(\frac{2\pi n}{N}\Big), 0\leq n\leq N=\text{file length}$ to the data given. I am very new to this subject so I have no idea how to apply this window function to the data. Any instruction? Thanks in advance.
|
In my digital communication, class we have covered the various modulation schemes, namely QAM and its derivatives (B(A/P)SK, PAM and (Q/M)PSK) as well as BFSK. These two schemes have one or at most two basis signals. There are many orthonormal sets whose size is larger than two. Why not modulate amplitude and phase wit... |
I am looking at the H(f) which is the frequency response of an LTI system. What kind of relationship should I expect between the real and imaginary part of this frequency response? Here is an example of the real and imaginary part from experimental measurement. Is there any way to prove if the system is LTI or not?
|
I am confused about a solution because there is feedback.
Let's introduce a delay parameter $k$ and rewrite the system equation as:
$$ y \left[ n \right] = y \left[ n - 4 - k \right] + x \left[ n - 4 - k \right] $$
Now, we can see that the output $y \left[ n \right]$ depends on the input $x \left[ n - 4 - k \right]$ an... |
I currently researching the phase (between real and imaginary part) of the fourier transformation. It seems that one can use it for detection.
For the signal processing I use two fft transforms. One to compress range and one to compress azimuth. Then I want to apply detection. At this point calibration and corrections ... |
Is it possible to calculate the output level change (dB) filter does?
Lets say you have a peaking EQ filter with +12dB gain boost and Q = 1/sqrt(2). I've tried by just summing the magnitude responses (dB) at every sample and then divided result by number of samples in calculation (fs/2):
pkg load signal
pkg load contro... |
For the traditional received signal model $y=hx+n$, the channel coeficient is usually considered as complex number.
But when analyzing the motivation for the Alamouti space time coding scheme here using real channel coeficient, the motivation become very easy to understand and extremely simple.
This just motivate me to... |
Thanks in advance for the help. I don't post on here often, so I hope I can convey my question properly.
I think I understand what the Allan deviation (same as IEEE Allan variance?) does. You take a signal, divide it into sections, average the sections, then basically find the std of the differences between those ave... |
As a continuation of the question Least Squares Solution Using the DFT vs Wiener Hopf Equations raised by Dan Boschen.
The question is, given the model:
$$ \boldsymbol{y} = \boldsymbol{h} * \boldsymbol{x} + \boldsymbol{n} $$
Where $ \boldsymbol{y} $ and $ \boldsymbol{x} $ are known discrete signals (Here as a vectors) ... |
I have been trying to implement PLL in a STM32 microcontroller. Starting with the first step, I want to implement a phase detector. I tried to first simulate in Python so that I can later use in STM32.
So far, I used a analog multiplier getting idea from this post. It worked kind of okayish for wide range of phase rang... |
My understanding is that if a single tone at frequency F is inputted to the frequency doubler, one should observe at its output in the spectrum the same tone but now at a frequency 2F.
However, what happens if this signal is no longer a tone? To make it simpler and to minimize the impact of possible modulation, let's c... |
I am trying to calculate the area under my FIR filter kernel using trapz but I'm getting weird results. The reason I want to know the area is eventually I would like to loop through filter orders in a small range and pick the order which most closely matches my ideal filter window.
When you run this code in Matlab the ... |
Let's say you are given a frequency domain specification for a filter design.
As a simple example, let's use an ideal (brickwall) LPF. Now the impulse response of an ideal LPF is a $\operatorname{sinc}(\cdot)$ function which is infinite length and not causal. So to make it feasible you truncate it and shift it to the r... |
I am trying to detect swallows from recordings taken from hospital. I manually labelled the recordings on the Praat. Now the valid labels are silence, swallows and nonswallows(noise, enviromenment noise etc.) After I construct my dataset and train with SVM(using libSVM library) I observed that the amplitudes variations... |
I am trying to figure out if
$$y[n]-2.56y[n-1]+2.22y[n-2]-0.65y[n-3]=x[n]+x[n-3]$$
is either IIR or FIR, causal or non-causal, stable or unstable (BIBO).
I got the transfer function from the coefficient
$$H(z)=\frac{1+z^{-3}}{1-2.56z^{-1}+2.22z^{-2}-0.65z^{-3}}$$
I saw this post and tried to do pole/zero cancelation bu... |
I'm looking for a simple algorithm where given a RIR sample file I can find the beginning of the impulse and the last point of the tail.
If possible, would love to implement it in python.
|
Consider the following super-heterodyne receiver
The carrier frequency of the local oscillator is chosen to be
$$ f_{LO} = f_c + f_{IF} \tag{1} $$
where $f_c$ is the transmitted carrier frequency and $f_{IF}$ is the intermediate frequency. My goal is the derive the constraint of the RF filter bandwidth such that it re... |
I have been reading about kalman filters for IMU and I am confused on the error state formulation.
Reading this paper, starting on page 63 and going onto 64 the paper addresses the reset of the error state.
https://arxiv.org/pdf/1711.02508.pdf
My confusion is what is meant by line 284? 285 just looks like it sets the ... |
I have the following wave:
I know that it is demodulated by an envelope detector whose source resistance is 0Ohm and its load resistance is 250Ohm. Now, I need to calculate a C capacitor in order that distortion is negligible for frequencies up to and inlcuding the eleventh harmonic. I know, that in order to have a de... |
Assuming I have two sensor feeds, is there an algorithm that gives an estimate of how likely it is that they point at/derive data from the same scene (possibly from different angles, with different modalities, filters and perspectives)?
Perhaps a measure of how easily the data streams could be related/reduced to a comm... |
Let $\mathbf{A} \in \mathbb{R}^{n \times n}$. I'm working in a problem where I have a black-box algorithmic solution to compute the products $\mathbf{A}\mathbf{x}$ and $\mathbf{A}^T \mathbf{x}$ given an input $\mathbf{x} \in \mathbb{R}^n$. Knowing that I can't access the value of $\mathbf{A}$ directly, what are some ex... |
Looking at the EKF formulation for Gyroscopes in more depth I'm wondering, why can't we use Euler Angles in the Error State EKF framework for error and covariance propagation instead of Quaternions? Wouldn't this completely solve the quaternion normalization issue(it's over-parametrized).
Basically my idea is to let th... |
I have measurements of a transfer function $H(f)$ in the frequency domain between 1-6 GHz. I also have an input signal, which is pulsed chirp $x(t)$. The goal is to build the time-domain response of the system. To that end, I take a Fourier transform of the initial signal which gives me $X(f)$. Then I build the $H(f)$ ... |
I am trying to solve the below problem:
To begin with the frequency response of the ideal low pass filter with $\omega_c= \pi/4$ is given by $$
H(\omega) =
\begin{cases}
1 & \text{if -$\pi/4$ $\le$ |$\omega$| $\le$ $\pi/4$;}\\
0 & \text{otherwise.}
\end{cases}
$$
Proceeding to solve , we have
$$w(n) = h(n) * \left[... |
I've been given these signals:
$$S1(t) = 1$$
$$S2(t) = t$$
$$S3(t) = t^2$$
I have to find 3 basis functions using Gram - Schmidt algorithm on the interval $|t| \le 1$. Now, I know the procedure and formulas but I am not sure how to treat $|t| \le 1$. Will it only affect the limits of the integral?
Thank you in advance!... |
Suppose that for a system $S$ if we have $t_{2} = t_{1}+t_{0}\rightarrow h(t,t_{2}) =h(t,t_{1})+h(t,t_{0}) $ .Then if we take the double Laplace transform to$t,t_{2}$ we will get:
$$L_{t_{2}}(L_{t}(h(t,t_{2})) = L_{t_{2}}(L_{t}(h(t,t_{1}))+L_{t_{2}}(L_{t}(h(t,t_{0})).$$
But
\begin{align*}
t_{2} &= t_{1}+t_{0}\\
&\right... |
I am trying to translate a signal to baseband by multiplying by the complex exponential. The issue is when I do the calculation in MATLAB the signal seems to disappear. I originally thought the signal was moved out of the axis scale but even when shifting down to a few kHz the signal disappears.
Here is my Matlab code.... |
I am working on finding the peaks in EKGs using Matlab. I have a couple of hundred of recordings. The problem is that some of them have their peaks inverted (so they are negative.) The goal is to be able to find the peaks without having to visually inspect the files to determine if they are reversed or not. I want to ... |
I am currently using the firls function in Matlab to create my FIR filter kernel. This works well but I typically have to play with the filter order to get the gain to 1 and the shape to be somewhat Gaussian in the frequency domain.
I would like to write an algorithm that finds the best filter kernel given a range of p... |
This is a follow-up question to Allan deviation to determine averaging time
I was under the impression that with DC signals I couldn't capture or average a signal forever to get better SNR, because of 1/f or pink noise. I thought the Allan deviation was a good way to determine averaging time.
I have now convinced myse... |
Referencing the following exercise by Dan: $9\rm V$ Battery with $45^\circ$ phase, I'm curious on what this would be used for in real world applications?
|
I'm using laplacian filter in opencv.
I usually use laplacian filter with kernel size 3 which is default setting in opencv.
I'm just curious what happen(intuitively) when I choose bigger size kernel?
I think if I take small size such as 3x3, this gives me 3x3 matrix. So the result is "how much change in pixel intensity... |
Suppose a wideband signal is sent and received by a RX. If different frequency components travel in different speed then there exists group delay in the signal, right? Not consider the multipath effect causing signal spread. Is there any algorithm that can calculate and compensate the group delay caused by channel? If ... |
I have the same weighted least square form weighted least square composed of diagonal weighting matrix $ W $, IDFT matrix $F({\omega})$, desired response vector $D({\omega})$ and FIR Filter coefficients $h$,
$$
min_{h} \sqrt{W} \{ F({\omega})h - D({\omega}) \}
$$
The closed form of the first problem is $(\sqrt{W} F({... |
Let's say I have a discrete-time signal $y[n]$ that is passed through an Ideal Lowpass Filter (impulse response $h[n]$) to get $y_1[n]$. The filter has cutoff frequencies at $\pm \pi/N$ rad.
Now this $y_1[n]$ is downsampled by a factor of $N$ to obtain $z[n]$. I know that $z[n] = y_1[nN] = \sum_{k=-\infty}^{\infty} h[k... |
For an Music Analysis Project I'm doing, I want to transform a song like it is done in this paper by K. Seyerlehner et al. http://dafx10.iem.at/proceedings/papers/SeyerlehnerWidmerPohle_DAFx10_P31.pdf
I'm totally new to Signal Processing, so I tried to look up everything but I'm stuck at the part, where I have to map t... |
What I want to ask you comes from a place where I can play with all the parameters as much as I want (I can measure as long as I want and as fast as I want, as well as change sensor/ADC).
The signal of origin will be a pure sine wave of known frequency (I make it) and maximum amplitude. As it travels through space it w... |
I am trying to measure the maximum output power of an SDR transmitting a 20 MHz LTE signal with a spectrum analyzer.
Adjusting the resolution bandwidth (RBW) changes the power spectral density (PSD) of my signal, which seems to suggest there is a sweep spot for the RBW.
I have been doing some reverse engineering of the... |
Let $x[n] = \cos(\frac{\pi}{8}n)$ and $X(k) = \text{DFT}_{16} \{ x[n]\}$
In order to find $X(k)$ I write out the DFT sum:
$$X(k) = \sum_{n=0}^{N-1} x[n] e^{-j2\pi kn/N} = \sum_{n=0}^{15} \cos(\frac{\pi}{8}n) e^{-j2\pi kn/16} $$
I could expand the $\cos(\cdot)$ as two complex exponentials inside the sum but that would m... |
I'm reading a paper - Total Variation Superiorized Conjugate Gradient Method for Image Reconstruction on total variation regularization and conjugate gradients. In page $3$, the authors define the total variation norm of a (vectorized) image as the $l_1$ norm applied to a matrix-vector product. More precisely, they def... |
I want to plot a specific frequency (e.g., 1 kHz or 100 Hz) of an audio file (sampling rate 44.1 kHz). I have done fft and spectrogram analysis on the file, but I am now interested in plotting only one frequency (as a sine wave) of the signal versus time. Here is also an example signal I created with only two frequenci... |
I have a large signal (coverage file) as below and my goal is to
detect the change points in the coverages
identify change points that have gradual increase and decrease in coverages (point C1 and C8)
identify points where there is a substantial increase in coverages (C6 counts, while the dip between C3-C4 does not co... |
I am using complex demodulation to estimate slow amplitude
and phase changes over time of a single sinusoid. The time
series consists of continuous blocks with large gaps (~30%
of block lengths) between them. The amplitude results are fine,
but although the estimated phase evolution is smooth within blocks,
there are s... |
Can someone explain the difference between 5G Cellular modems used in our Mobile phones to that of modems used in the wifi router?
TIA
|
My question is related to Solving regularized least squares problem using black-box computation of $\mathbf{A}\mathbf{x}$ and $\mathbf{A}^T\mathbf{x}$.
In case, the problem is formulated as:
\begin{align}
\min_{\mathbf{x}} \frac{1}{2} \| \mathbf{y} - \mathbf{h} \ast \mathbf{x}\|^2 + \lambda \phi(\mathbf{x})
\end{align... |
I am working on the "System identification : theory for the user" by Lennart Ljung (freely available here) and it is one of these books which contains exercises but no answers... My exercise is the 2G2 p47.
The context is as follows :
s is a discrete scalar real quasi-stationary signal, a quasi-stationary signal being ... |
Conventionally, the definition of the system function for a IIR digital system is:
$$H(z)=\frac{b_{0}+b_{1}z^{-1}+b_{2}z^{-2}+\cdots}{1-a_{1}z^{-1}-a_{2}z^{-2}-\cdots}$$
where coefficients are the ones from the difference equation:
$$y[n]=a_{1}y[n-1]+b_{0}x[n]+b_{1}x[n-1]+b_{2}x[n-2]+\cdots$$
In both equations + and - ... |
Supposing that there are $N$ particles distributed inside a periodic cubic box of volume $V=L^3$, I want to divide the cube into a regular mesh and calculate the following summation at each grid point $r_i$,
$$z_i = \sum\limits_{j=0}^N K\left(\left\| r_i-r_j \right\|\right)y_j$$
where $r_j$ is the position vector of th... |
I'm having trouble with this exercise.
The signal $x(t)= \cos(2\pi f_0 t)$ is sent in input to a non-linear system with input-output relationship equal to: $g(x)= 0$ if $x<0$ and $g(x)=3x$ if $ x >= 0$.
Find $y(t)$ and its Fourier transform.
The output of this non-linear system is $y(t)$ and can be written as $$ y(t)=... |
It is probably very well known that dBFS cannot be directly converted to dB SPL without knowing other information such as the technical specifications of the microphone.
The questions are:
What is missing for the conversion?
What is the formula?
What I now have in hand:
A dBFS value: -12 dBFS (just for example);
Mic... |
I am starting to learn MIMO transceivers and I would like to compute the virtual array of my configuration. To do so, I am following these set of equations 2 and 3, to achieve (3) I was trying to use Matlab conv and conv2 but the results that I am getting are not the one I would expect as if I do the direct implementat... |
I find myself confused how two properties of the ambiguity function relate to one another.
The properties in question are (presented here from wikipedia, but similar formulas can be found in books by Levanon, Mahafza, etc...)
The maximum value of the ambiguity function is in point $(0, 0)$
$$ \tag{1}
|\chi(\tau, f)|^2... |
I have an analog signal as shown below.
As you can see it has a DC component, a 1MHz component and some high order component.
The period of the signal is 8.0199667µs.
I want to see a good accurate frequencies up to 10MHz.
I have captured a signal 78.97518µs long.
Its edges are not absolutely equal so my fft will g... |
I was studying Rayleigh channels from Wireless Communications, Second Edition by Andreas F. Molisch. The book states that by adding different in-phase and quadrature components multipath components the distribution of amplitudes of in phase and quadrature should be Gaussian due to central limit theorem, but in-phase co... |
I'm trying to simulate a Fourier transform spectrometer in Python. I started basically with a simple single frequency (f=1e10 Hz) sine signal coming into the spectrometer. I obtained the following plot of the signal against time:
What the spectrometer is measuring is only the intensity of this signal - that is, the fi... |
In Solving inverse problem using black box implementation of the kernel the solution depends on solving the equations of the form:
$${\left( {H}^{T} H + \lambda {G}^{T} G \right)} x = y$$
Where $H$ and $G$ represent convolution operation of the $h$ and $g$ kernels.
Given we have a a black box implementation of the conv... |
I have response time (RT) data (values between 0 and 3) to a memory task. Participants made a response at about every 9.5sec with several breaks of 25sec. I want to analyse the whole timeseries of (RT), which yield at about 30 datapoints.
Due to the unequal distances between samples, I took the Lomb-Scargle periodogram... |
I wish to convolve/cross-correlate two images
and
but, only horizontally, yielding 1D output.
So, first output point would be sum(x * h), second sum(x * h_shift1), where h_shift1 is h horizontally shifted by 1 sample, in Python np.roll(h, axis=1). Basically, pass images through each other horizontally.
I know that 2... |
I'm trying to implement the approach for processing audio like it is done in this paper: Block-Level Audio Features for Music Genre Classification
Can someone explain me how the part marked in yellow works?
I'm new to signal processing and I already gathered information on what a frequency bin(intervals between frequen... |
I was parsing the forum when I saw this post surging out of the depths of this forum like an old Kraken. The problem is quite simple.
You have a continuous time state space model :
$$
\begin{split}
\dot{X} &= AX+BU\\
Y &= CX
\end{split}
\tag{1}\label{eq1}
$$
The state equation giving in Laplace :
$$(sI-A)X(s) = BU(s) ... |
Suppose we have a linear linearly time varying system $S$ such that the output depends on the input + the time at which the system becomes excited so $S(t,t_{0}) \rightarrow g(t_{0})S(t)$ and it is true that if $t_{2} = t_{1}+t_{0}\rightarrow h(t,t_{2}) = h(t,t_{1})+h(t,t_{0}) \rightarrow g(t_{2})h(t) = g(t_{1})h(t)+g(... |
I would like to understand how does DFT-s-OFDM reduces PAPR compared to CP-OFDM. I am not able to locate any mathematical explanation for this. If anybody can provide an explanation it will be great.
Thanks.
|
I am trying to reproduce Fig. 2.17 from the book 'Optimum Array Processing' by Van Trees.
The figure is a polar plot of the beam pattern $B_{\theta}(\theta)$ given by
$$B_{\theta}(\theta) = \frac{1}{N}\frac{\sin\left(\frac{N}{2} \frac{2\pi}{\lambda}\, \cos(\theta) \,d\right)}{\sin\left(\frac{1}{2} \frac{2\pi}{\lambda}... |
My lecturer has decided that in order to pass our signal analysis class, we have to give a presentation on the blackboard in front of the entire class, so its kinda critical that i get this right. (Issue is i'm pretty bad at this class). I managed to convolute the subsystems, but i would appreciate some confirmation th... |
I am trying to decipher some notes on a Time-interleaved Analog-to-Digital Converter with 4 sub-ADCs (the author is unreachable at the moment), but there are few things obscure to me so I was hoping to get some hints from the community. Quotations " " will be taken verbatim from the document.
The author starts off well... |
I am making a digital sound mixer and get this wav file (200kb, Wetransfer)
I looked through the waveform using Audacity. It uses 20% of its maximum amplitude and I see no sudden changes in the waveform.
Yet it repeatedly clicks.
It is composed (not maxed) sine waves, first 1, later 4. But the waveform looks like I avo... |
There are many related questions posted about estimating sine wave parameters, and this one is closest: Measuring amplitude of a pure sine wave of known frequency close to the noise floor, but none consider the specific case of providing the optimum strategy when signal is near the noise with unknown frequency in the... |
I'm trying implement SIFT(Scale-invariant feature transform) myself. I've read some documents and stuck at Sub-Pixel Refinement - left without knowing how to actually implement it in code. I do understand its basic idea of estimating the pixel in continuous space using Taylor series of quadratic but how do we go differ... |
I Have a picture of Dominoes on a table, which is noisy.
I am trying to do the following:
removing the background.
counting how many dominoes pieces are there.
identifying what each piece represent.
I am using Matlab here.
I have successfully isolated the dominoes from the background using thresholding then dilatatio... |
I have an audio file (u8, 128kbps) of the sounds produced while playing a game app and clips of its three distinct sounds.
The file consists of sequences of these, separated by silences, indicating next move (0.5 sec), correct move (0.5 sec, immediately followed by next move), and game over (5 sec).
I want to con... |
I'm learning about Z-transforms and transfer functions in DSP. Suppose I have the following transfer function:
$H(z) = \frac{1}{z-0.5}$
I want to find the corresponding $h(n)$. If I factor out $z^{-1}$ from the numerator, I'll be creating a new zero and a pole in $z=0$.
$H(z) = z^{-1}(\frac{z}{z-0.5})$
Which its in... |
I implemented a data transfer system using two tutorials: Guided Tutorial PSK Demodulation and Packet Communications. After that, I decided to change QPSK to 8PSK.
My scheme:
Constellation Sink:
For some reason, the Costas loop turns the constellation, although this should not be. If I put the Order = 7 value in the ... |
I found the ofdm flowchart in the Gnu radio examples.(ofdm_loopback)
Scheme:
The spectrum and the received data of the received signal:
After that, I connect the USRP Sink and the USRP Source.
The spectrum and the received data of the received signal(USRP):
A similar question has already been encountered here, but,... |
I have implemented a packet reception and transmission scheme using QPSK. Next I decided to add Polar/LDPC encoding. As a result, nothing is written to the received file.
I think the thing is that I'm unpacking the bits incorrectly after demodulation.
For convenience, in the diagram, I highlighted packet formation and ... |
Assume we need to solve the model:
$$ \arg \min_{\boldsymbol{x}} \frac{1}{2} {\left\| \boldsymbol{h} \ast \boldsymbol{x} - \boldsymbol{y} \right\|}_{2}^{2} + \frac{\lambda}{2} {\left\| \boldsymbol{g} \ast \boldsymbol{x} \right\|}_{2}^{2} $$
Where $ \ast $ is the convolution operator.
How can we solve this in the freque... |
I want to estimate the SNR value at the input of my USRP. To do this, I connect two blocks: MPSK SNR Estimator and MSPK SNR Estimator Probe.
Next, I look at the values that I got using the MSPK SNR Estimator block.
QT GUI Histogram Sink:
Constructed histogram using File Sink.
The values I get using MPSK SNR Estimato... |
Does anybody have an idea or has tested the transmitters to see what at what frequency they transmit?
|
How to implement sliding 2D FFT for audio data (shown in the link - R1-chirp)?
I have audio data for an array of 1024 microphone which record audio (for 2 seconds) from a sound source. Microphones are arranged in plane (17 rows and 60 columns). I used 2D FFT to estimate frequency-wavenumber spectrum, however the x-shap... |
I was playing some very loud pop music through a studio monitor (ADAM A8X), and after about 5 seconds the sound stopped coming out of the cone and only came through the speaker. After shutting it off for a few minutes and trying again, no sound was coming from the cone. I assume I broke something, but don't know where ... |
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