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We have the closed loop transfer function:
$$T(s)=\frac{L(s)}{1+L(s)}$$
So as far as I understand we check along the $j\omega$-axis on the Bode plot whether $L(s)=-1$, cause that's when $T(s)$ has pole on the axis and that leads to instability.
My question is, why don't we need to check the entire right half plane whe... |
I have been trying to understand the weighted overlap-add (WOLA) method. By searching papers and posts in dsp.SE, I found that the window length can be larger and smaller than the FFT size.
1. Window length larger than FFT size.
See the comment in this post. It says that "WOLA allows you to use an N-point DFT with an M... |
This link gives the procedure of implementing the WOLA method. It can be summarized briefly as windowing the extractd data frame, FFT, modify the spectrum, IFFT and overlap-add the windowed IFFT result to the reconstructed data buffer.
Suppose the modification to the spectrum of each data frame in the WOLA is time-inva... |
According to my understanding, IIR filters are somewhat analog as they are based upon analog prototypes such as butterworth,chbeyshev etc. Can we usefreqscommand to display frequency response of digital IIR filters? If we can use this command for IIR filters then what about FIR filters?Can we also use this command for ... |
I am trying to fit data of magnitude and phase to get complex pairs(for FFT), but I think it would be a wrong approach as different pairs of complex numbers can give same amplitude and phase , is my assumption correct? is there some obvious example for this assumption through which I can prove?
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I am getting confused in SINR for the following received signal expression.
$$y(n) = (h+fg)s(n) + w(n) \tag{1}$$
Where $h,f,g$ are zero mean circularly symmetric Complex Gaussian (ZMCSCG) channels. All these channels are independent in nature with variance $\sigma^2_a$. $s(n)$ is transmitted signal and is also ZMCSCG w... |
I have trained three deep learning models, and used majority voting for the final decision. Applying majority voting has led to improving the accuracy of the classifier with all test samples, except with only two samples, where the three models differ in their decision, and thus no majority was obtained. How can I deal... |
Suppose I have few samples (e.g. 20) of a decaying sinusoidal function and I am heavily noise-dominated (SNR ~1, noise is Gaussian or Poissonian). Let's think of e.g.
\begin{equation}
y(t) = A \, \sin(2\pi f+\phi) \, e^{-t/T} + \mathrm{noise}(\sigma)
\end{equation}
with $f \sim 1\,\mathrm{MHz}$ and $T \sim 10\,\mathrm{... |
Uniform sampling can be thought of as multiplication of a function $x(t)$ with a Dirac comb function: $$\text{III}_T(t) = \sum_{k=-\infty}^{\infty}\delta(t-kT)$$
Multiplication of $x(t)$ with $\text{III}_T(t)$ is equivalent to convolution with $\text{III}_{1/T}(f) $ in the Fourier domain, giving rise to copies of $X(f)... |
Hope y'all are doing well. Recently I've picked up a LimeSDR and interfacing it with GNURadio and been having a blast with it. I got my first FM receiever demo working using examples from LimeSDR FM Example (Simple)
My next step was to make it variable, as the Simple FM receiver demo was fixated on one frequency. I fo... |
I see mention that normalized LMS "usually converges faster than LMS", in Diniz "Adaptive Filtering" p.152, can this be made more precise? IE, for which signal distributions does it hold?
I'm in particular interested in overparameterized regime where misadjustment is zero for LMS and NLMS.
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I am working on an open source project s2p which creates digital height model from satellite stereo imagery. The procedure of how s2p works can be roughly summarized with the following steps.
Split a large stereo image pair into smaller tiles (aka windows).
Use SIFT and RANSAC to find feature points and rectify each p... |
I'm working on target extraction with raw FMCW ADC data. I arranged my received data in an (numADCsamples x numAntennas x numChirpsPerFrame) radar cube.
So far I am using range FFT and doppler FFT to generate range doppler map. Next I apply OS-CFAR for target extraction.
Using 3D-FFT on the extracted targets (numTarget... |
Cheers, I have I am given the following signal $$A \cos ( 2 \pi f_o t + \Theta)$$ and $\Theta$ a random variable with pdf of $\frac{1}{2 \pi } ,0 \leq \theta \leq 2 \pi$ and 0 elsewhere and I am asked to find its mean. I know this is a simple exercise, but I cannot grasp why the computation would be $\frac{A}{2 \pi}\in... |
I have performed reconstruction of images in Fresnel transform using a desired algorithm. Now the aim is to find an optimal value of image plane distance at which the reconstruction is accurate. I have formulated the problem as below:
||X_recon - X*e^{ax^2}.e{-bx^2}*e{cx^2}||_2^2 such that ||c||<=1
here X_recon = recon... |
For one clean image, one person can identify every details at first. Then white Gaussian noise is added to the image.
With low noise level, one person can still identify every details even though noise is present.
With a litter higher noise level, small details and low contrast edges will be not be identified.
With muc... |
Cheers, I am demodulating a QPSK signal on GNURadio, the data transmitted is a known HEX file with the "1ACFFC1D" foreword every 124 bytes, to check if the demodulation is a success, it contains ascii code saying "Bonjour :)" indefinitely to spot easily an error.
I also compare the bits in a time sink. The signal is mo... |
I have an analog value coded as an analog PDM stream, which has a toggling frequency $f_{PDM} \approx$ 100 kHz when the value is at midscale.
I am demodulating this value into a 32 bit result with a simple shift-subtract IIR lowpass filter. Precisely, I am taking the difference between input value $x[n]$ (either 1 or 0... |
I am trying to obtain the range, Doppler and angle of arrival estimation with an TDM MIMO FMCW radar.
Actually, I have completed this process. In doing so, however, a few things remained unclear. First of all, I ran the process as an FFT in fast time to estimate the range then another in slow time to estimate the Doppl... |
I'm trying to reduce the noise in this photo. What type of filters/techniques should I use to cancel out the noise?
I've tried using box filter and Gaussian filter to blur out some of the noise, but i had no luck getting satisfying results. Any suggestions?
|
I am given the following problem: I have a filter with impulse response $h(t) = e^{-10t}, t \leq 0$, and autocorrelation function of the input signal, which is WDS and Gaussian with median equal to 0, of: $R_X(\tau) = \frac{\sin 20 \tau}{\tau}$. Amongst other things, I am asked to find the probability that the rate of ... |
In an old paper, accessible only for a fee, I found:
(where the LHS of Eq (5) lack the symbol of derivative). What is the Wiener-Kolmogorov whitening procedure? I would like to study this procedure, can someone explain to me what it is and/or suggest some reading? Thank you in advance.
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The Fourier basis is a common choice for transformations, but a lot of times, it's not the best for a specific application. For instance, wavelet bases give us better spatial / temporal locality than the Fourier basis, and in PCA, a specific basis is selected based on statistical analysis of a dataset.
However, if you ... |
I'm a beginner in audio signal processing, and I'm trying to figure out how to compute the signal-to-noise (SNR) ratio from 2 audio clips.
I have two single-channel audio clips, where one is a recording of room noise(/ambient noise) and another one is a recording of human speech in that room. I find a few equations tha... |
I've designed an IIR Low pass filter in MATLAB with the following parameters.
%IIR Filter Design
SamplingFrequency = 122e6; %Hz
PassbandFreq = 9.5e6; %Hz
StopbandFreq =10.5e6; %Hz
PassRipple = 1; %dB
StopbandRej=38; %dB
FilterIIR=designfilt('lowpassiir','StopbandFrequency',StopbandFreq, 'PassbandFrequency', PassbandFre... |
I ran some code in matlab to show what is hue, saturation and intensity to get an intuition of what they mean.
But the result of the code does not match the explanation i got from here
Hue is a color attribute that describes a pure color (yellow, orange,
or red).
Saturation gives a measure of the degree to which a pur... |
Phase modulation causes instantaneous frequency to vary according to modulating signal.
In the same way does Frequency modulation by direct method causes any phase variation in proportion to modulating signal?
|
I am working on Compressive sensing algorithms for DOA estimation. Can i get any references for writing BPDN code in MATLAB.
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What is meant to be explained in the section between the 2nd paragraph and the folding section in the Sampling sinusoidal functions section shown in this link?
Why is $Nf_0$ written in $f+Nf_0$ instead of frequency in the given equation?
Also, what exactly do I need to understand from the gif on the right?
|
Currently, I'm learning about the UKF and in order to understand it in a good way, I programmed it, however in order to see it working I need a problem to solve which I can't now find.
Can you please recommend a very basic problem in which all the equations and the data are explained and ready to implement in my code?
|
In GNURadio's PSK tutorial they place the costas loop after clock sync and linear equalizer. This seems odd to me.
I'm thinking that it's easier to lock onto a carrier frequency at the higher RF sample rate rather than on the clocked symbols. And similarly it seems to me that it should be easier to to clock recovery on... |
I have a Fourier matrix $F$ with size $N \times N$, such that $y = F \times x$, if I have the vector $x$ contains four identical parts, for example $x = [x_1, x_2,x_3,x_4]’$ and $x_1 = x_2 = x_3 = x_4$ can I simplify the multiplication $y = F \times x$ into smaller matrix to reduce the complexity? . For example use sm... |
When I compute a spectrogram of (say) a piece of music, there is a lot of frequency "smearing." Often we can reasonably expect that the "true" generating process is much sparser in frequency (i.e. maybe 10-20 active frequencies instead of hundreds).
We also know that plain old sine waves become smeared in spectrograms.... |
I'm trying to learn some signal processing and recorded some iq data from the meteor satellite which transmits data with qpsk. I read the file with python and tried applying a rrcos filter on the signal.
sample_rate = 1152000
file = open("file", "rb")
data = []
for i in range(10000):
point = file.read(8)
real,... |
Good afternoon. I have the key to my gate. I received its recording in the form of a PWM pulse train. Instead, I converted the impulses into binary code. Now I should get the hex numbers but it doesn't converge.
I have the same sequence in hexadecimal.
How to translate correctly? I know that you need to read from the l... |
I am studying matched filters and the sense of signal to noise ratio. In a question I was given, stated that:
Show that a matched filter of a signal $s(t)$ maximizes the SNR of the
output on a receiver of AWGN with power spectral density $$S_n(f) =
\frac{N_0}{2}.$$
After they show that that happens when we sample at... |
For regular FIR filters we can feed an impulse signal to get the coefficients, but how about filterbanks?
Let's say I have a uniformly modulated filterbank with hop size H and FFT size M, implemented as weighted overlap-add method [1]. Is there a way to construct an input signal to extract the analysis and synthesis we... |
I am reading wireless research papers and found that mostly the wireless channel is considered to be Zero mean circularly symmetric Complex Gaussian (ZMCSCG) with variance 1 or 2 or 5 or 10 etc.
My query is why generally these values for channel variance are considered.
Any help in this regard will be highly appreciate... |
I have the vector $y = Dx$ where $D$ is a complex matrix with dimension $N \times N$, and $x$ is a complex vector of dimension $N \times 1$.
If the vector $y_2 = [y'_N, y'_{N-1}, y'_{N-2},.... , y'_{1}]$, where $ y'_{N}$ is the conjugate of the element $ y_N$. How can I express $y_2$ in function of $D$ and $x$?
|
I am currently working on a research project based on the paper titled A deep learning approach for anomaly detection based on SAE and LSTM in mechanical equipment. The paper uses data in the acoustic domain and processes it with techniques such as FFT and WPD to extract a total of 11 features per recording channel.
In... |
In my class on wireless communication, I came across the following sentence:
The round trip path-loss can be defined to include aspects such as
attenuation and/or latency etc
Hence I am looking for relation between Pathloss, attenuation and/or latency.
|
For the inverse continuous wavelet transform, I am using a for-loop which runs through each frequency. At each frequency, I convolve the corresponding morlet wavelet with the signal at that frequency, and add it to my result.
This code is based on the inverse transform outlined in "A Really Friendly Guide to Wavelets" ... |
I have seen few papers where generalized order statistics is used to obtain statistics of the instantaneous SNR.
"Performance analysis of monostatic multi-tag backscatter systems with general order tag selection" by Yazan H. Al-Badarneh, Mohamed-Slim Alouini and Costas N. Georghiades.
"Order statistics in wireless co... |
I have a nice and clean noise signal sampled at a fixed rate for several hours. When I calculate the PSD I see it's pure flicker noise. But I don't know much about the process.
I've seen some discussion in articles about wether flicker noise is stationary or not, it doesn't look like a settled issue.
How can I check fo... |
I have a sine waveform that is a result of simulation.
This is always single tone with a constant offset, but with distortion and noise and may have some jitter:
$$s[k] = A\sin\left(2\pi \frac{f}{f_\text{s}} k + \phi\right)+ \alpha+N[k].$$
(since SNR after distortion is better than 40 dB, and jitter probably not domina... |
In fact, I am doing research around SDR and possible dangers. Then I stumbled upon the most well known "replay attack". Via Black hills, I came across the following diagram in gnuradio:
The lower part (so not greyed out) is for transmit signals. Because I want to do this legally and as safely as possible without harmo... |
Looking forward to explaining what this structure means, and the relationship between the two structures.
Paper: An Efficient Spur-Aliasing-Free Spectral Calibration Technique in Time-Interleaved ADCs
|
While developing an application, I noticed strange behaviour while using scipy.io.wavfile to write audio files. The documentation says that floating-point data is written within a range between -1 and 1. However, files written this way seemingly can go past that. This is a MWE to illustrate what I mean:
import numpy as... |
First of all I am totally new to the DSP field and I have no background in it whatsoever, but my work in biology has led me to data that would greatly benefit from DSP. Any answers devoid of specific DSP math notation, verbage or symbols would be greatly appreciated.
My question is: how do I calculate the frequency va... |
I am trying to pass a BPSK signal through a root-raised cosine filter in Octave. To do this, I convolve the signal with the impulse response of an RRC filter, defined as follows (source):
The BPSK signal has a symbol rate of $$\frac{1}{T_s} =10 \text{ Mbps}$$. Would this not result in a massive value for pretty much t... |
I wish to estimate the noise of given trajectory from a measurement (data which is provided).
I consider that the noise is random gaussian noise.
I add here examples of two trajectories, one of the original data, and the other smoothened.
The trajectories are the measured velocity of some particles.
The model which is ... |
My question is with reference to wireless communication.
There are $N$ independent and identically distributed random channels i.e., $Z_i \in \{ Z_1, Z_2,\ldots, Z_N\}$ with PDFs $f_{Z_i}(z)$ ordered in ascending order. Then using order statistics the PDF of $Z_1 = \text{min}_i(Z_i)$ is given by
$$f_{Z_1}(z) = N (1-F_{... |
I am currently using the multiple signal classification (MUSIC) algorithm provided by the Pyroomacoustics library in order to estimate direction of arrival angles and it works pretty well for both narrowband signals and wideband signals.
However, after looking into the source code, I have few questions regarding the im... |
I am looking for an efficient image warping function $$f: (\text{image}\in \mathbb R ^{M\times N\times3}, \text{heatMap} \in \mathbb R ^{M\times N}) \to \text{warpedImage} \in \mathbb R ^{M\times N\times3}$$ that would "inflate" certain pixels in a base image, with the amount of inflation represented in a heat map that... |
Given a complex vector $x[n]$, we can find the magnitudes of the spectrum by computing:
$X_m[k] = |X[k]| = |FFT[x]|$
This involves performing a complex FFT and computing the absolute value of the result. There is consideration to take advantage of a real FFT for hardware simplifications, by computing the magnitude of t... |
Suppose I have a blurry image: a photo convolved with a gaussian blur kernel of unknown sigma. I would want to deconvolve the blurry image using several gaussian kernels (with different sigmas). Is there some metric function that measures "sharpness" of the deconvolved image? I could have a user look at a sequence of i... |
I have to design a notch filter based on an allpass but when I plot the resulting filter in MATLAB it does not fit my specification. I'll provide you with some background.
The filter has the following equation:
$$H(z) = \frac{(1+k_2)}{2}\frac{1+2k_1z^{-1}+z^{-2}}{1+k_1(1+k_2)z^{-1}+k_2z^{-2}}$$
where
$$k_1 = -\cos\omeg... |
How can an SDR, which has a sample rate of about 32MHz record signals with a frequency of 1GHz? Shouldn't the sample rate be at least two times as high as the frequency or am i missing something?
|
How can I add the BAQ (Block Adaptive Quantizer) compression noise to the input SNR? In the end is a quantization noise to be added to a thermal noise. I got the following formulation, but I dont understand the second one. I would have expected the total snr to be 1/(1/snr + 1/sqnr) instead. Can someone explain? thanks... |
Is it possible to reverse engineer a short white noise audio file and imitate it with a function instead?
The file doesn't contain pure white noise, there has been EQ applied to it, also some compression. I would most like to model this in something like MATLAB, the function needs to be simple and should NOT be neural ... |
As far as I understand, the 2d fourier transform is calculated as following:
step 1:
| a b c | -> 1D FFT -> | A B C | (calculate discrete fourier transform seperately for all three rows)
| d e f | -> 1D FFT -> | D E F |
| g h i | -> 1D FFT -> | G H I |
step 2:
| A B C | -> transpose -> | A D G |
| D E F | -> t... |
I have a real matrix $Z$ which is following the form as following:
$Z = \begin{bmatrix}
x_1& 0& 0& 0& 0& 0& 0& 0\\
0& x_2& 0& 0& 0 & 0& 0& -x_4\\
0& 0& x_3& 0& 0& 0& -x_3& 0\\
0& 0& 0& x_4& 0& -x_2& 0&0 \\
0& 0& 0& 0& -x_1& 0& 0& 0\\
0& 0& 0& -x_2& 0& -x_4& 0& 0\\
0& 0&... |
I'm trying to demodulate a qpsk signal with gnu radio. The signal is from the meteor satellite on 137MHz https://www.sigidwiki.com/wiki/Low_Rate_Picture_Transmission_(LRPT).
I recorded the signal with gqrx without any special settings.
This is my flow graph:
After filtering the signal looks like this which doesn't loo... |
I have a signal with a sample rate of 8.9286 MHz and I want to downsample it to 500 kHz.
Since 8.9286 is not an integer multiple of 0.5 I can't simply decimate. Which downsample techniques are recommended to be used?
If I use this solution then I there will be a step to decimate by a factor of 647 $$(89286=2\times3\tim... |
I want to analyze FFT and Harmonics of my signal.
As a proof of concept I tested FFT against standard signal.
I am unable to understand why the frequency output is going to -1. Shouldn't it be +1 at +pi and -pi.
Code :
x = -pi:0.01:pi;
sin_x = sin(x);
fft_x = fft(sin_x);
subplot(2,1,1);
plot(x, sin_x),
grid on;
subpl... |
I've a little question regarding interpolation in GNU Radio:
I'd like to do an interpolation (by a factor of 2) and be able to change the sampling period of the samples interpolated but im having a hard time trying to do it, does someone has any idea on how to do it in GNU Radio ?
As im very knew to this, here is a lit... |
I recently started working on PSD of seismic signals. PSD signal can be expressed in 2 ways. One in ($PSD=g^2/Hz$) and other in $PSD=((meter/sec.^2)^2/Hz)$ and $ASD = \sqrt(PSD)$.
More details have found in the link https://ucum.org/ucum.html#section-Special-Units-on-non-ratio-Scales
I want to find the amplitude of the... |
I'm trying to derive the overall state-space system model for a hybrid system, in order to plot its eigenvalues.
The system is shown as follows:
Which is originally from this paper: Modeling and Sensitivity Study of Consensus
Algorithm-Based Distributed Hierarchical
Control for DC Microgrids.
The blue part is a discre... |
I am studying interpolation filters. I'm under the impression that when talking about interpolating a signal, we filter and then upsample the signal.
BUT
In an article, Interpolation (Digital Filter Design Toolkit, is given a block scheme, where we upsample first and then filter.
The same explanation was given as an a... |
I assume I have the length of such vector $y$ is $N$. In the first time I divide that vector into two columns and then sum them point-wise summation. The second time, I divide the same vector $y$ into four columns and then sum them. Then, I divide $y$ into eight columns and sum them. I need to mathematically find the ... |
In DSSS, the spreading sequence is known only to the receiver and the transmitter. But for example, if we consider a spreading sequence built around a certain signal known only to the receiver and not to the transmitter. On reception, the signal is a key making it possible to decode the spread message with the construc... |
In the 2004 paper Designing good permutations for turbo codes: towards a single model it is mentioned you mention that :
"turbo code permutation of DVB-RCS [5] and DVB-RCT [6] standards, which was devised by the authors of this article was inspired by relations (11)."
I believe this design knowledge was also used for... |
I am analyzing a series of oscillating data of items that gradually become less "floppy"/oscillatory and I have observed that those items that are less floppy/oscillatory tend to have, what has been explained to me, a prominent "DC component" in their FFT profile plot and their FFT profile looks "cleaner" (maybe becaus... |
I am studying a thesis about nonlinearities in the transmit signal and it mentions the following:
The first 2,500 samples, which correspond to the Tmax of each of each
sweep period and are affected by ‘fly-backs’ from the previous sweep,
are set to zero, so that the number of processed samples is =
10,000.
The thesi... |
I am trying to work through an issue with an FFT for audio : I have no spectral leakage while playing back audio, but the moment I change or zero out a value in any of the FFT bins, artifacts are introduced.
I have a basic understanding of what causes spectral leakage while working with an FFT : the rectangular window ... |
I'm looking at spectral data collected through Cavity ring-down spectroscopy. I'm comparing a few samples of similar analytes and expect slightly different peak heights and trends. However, there are peaks and valleys that are present in all samples that are artifacts of the instrument the data was collected on. Is the... |
Single ADC is sampling at 16000 Hz the I and Q outputs of quadrature detector. So each I and Q signal is sampled at 8000 Hz, with phase offset 90 degrees between them. Bandwidth of interest is 3400 Hz. To synchronize both signals in time, I interpolate them with the factor of 2. New sampling rate of I and Q signals bec... |
I am implementing a continuous wavelet transform and its inverse using morlet wavelets. When I compute the inverse, the resulting signal is off by some constant factor (but otherwise correct). Depending on which frequencies of wavelet I use for the transforms, the resulting signal is off by a different constant factor.... |
I'm working on a dataset of spectroscopies and i'm classifying them by transforming the 1D signal into a 2D one by the ssqueezepy library.
For instance, consider to have a 1D signal and the corresponding 2D image, generated by executing the function:
Tx, _ = ssq_cwt(input_1D_signal, wavelet="gmw", nv = 64, scales=0.1)
... |
Can we use"audacity"for teaching undergraduate labs/practicals of digital signal processing? or only MATLAB/Octave should be used?
Weblink of "audacity"
https://www.audacityteam.org/
|
I am currently building a FIR filter based on frequency sweeping data (0 Hz to Fs/2), i.e., transfer curve in frequency with the sampling rate of Fs.
The phase response is shown in the following figure. From the measurement results, I saw that the phase did not end at plus/minus multiple of $\pi$. As I know, a normal ... |
In CDMA signals (such as GNSS) there are two unknown parameters when one receives the signal, residual carrier frequency and an unknown delay. Assuming the carrier frequency residual is known, what is left to find is the delay between the input signal and the gold code modulated into the signal. With no prior informati... |
I am familiar with the continuous Fourier transform yet the more I try to understand FFT, the more I'm confused. If we've got a discrete N sample signal, FFT is a faster way to calculate DFT. If calculating the DFT is independent of frequency bins why do they prop up? For example in the code below that I took from webp... |
I'm actually doing a FM receiver with an antenna linked to an USRP N210 and demodulate it.
I have two channel: one with delay and one without.
I set my delay in seconds (samp_rate*sec) and everything is working fine, I hear both the delayed signal and the "real" time signal. I can set the variable "sec" to 10,20 to ha... |
Suppose I have a set of estimators, $$\{S^1, S^2, S^3, S^4,\ldots,S^n\}$$ that output at each timestep $t$ a measurement representing an estimate of the true signal $y$, however the output of each sensor is delayed by $n$, so $S^n_t$ is an estimate for $y_{t-n}$. Moreover, the accuracy of that estimate increases as $n$... |
I have a locally generated 10 GHz LO signal that should go into the LO port of a mixer. The LO signal strength is -1.7 dBm. The mixer has a required LO-drive of about 13 dBm. Before the LO signal hits the LO port of the mixer it needs to go through a 4-way splitter with a gain of -9.4 dB (insertion loss).
To achieve th... |
Generating time series from a given PSD has been discussed in this and some other forums already (which I've referred some of them below) but almost all of them are mostly descriptive and none of them gives the exact details or code that implements and verifies it. As always the devil is in the details. So I went throu... |
The optimization problem is to find $n \times n$ matrix $C$ such that $\left| \left| x - d C \right| \right|_{2}^{2}$ is minimised where $x$ is $1 \times n$ and $d$ is $1 \times n$. Is this possible also $d$ is not invertible matrix?
|
I've found an interesting exercise which I have been trying to solve for a couple days, without success.
Let $x(t) \in \mathbb{R}$ be a periodic signal with fundamental period $T_0 = \tfrac{1}{4}$, mean value $\overline{x(t)} = 1$ and power $P_x = 10$. Knowing that $x(t)$ isn't affected by an ideal lowpass filter with ... |
Human hearing range is commonly known to be 20 Hz to 20 KHz
But what about speaking range? We know that speech Sounds are generated through our vocal cords. So What is the maximum value of frequency that human speech can have?
|
As part of an exercise, I have been assigned to find an alternative method of executing the additions of a transposed FIR filter. There is a hint stating that the alternative method should be in a form resembling a binary tree.
I figured that the leaves of the tree should be the product of x(n)*h(i), i=0,..,k but I am ... |
In the paper about SIFT algorithm (Distinctive Image Features
from Scale-Invariant Keypoints) it says at the end of section 3.3:
We double the size of the input image using linear interpolation prior to building the first level of the pyramid. [...]. We assume that the original image has a blur of at least σ = 0.5 (th... |
I read an audio file (an instrumental music file that is mono and 16000) with wav.read (read function from scipy.io.wavfile.) This function gives me a data file in int16 format. When I change the type to float and write it again I hear a lot of jitter. Why does that happen?
Here is my script in python 3.6. This is the ... |
I am trying to understand the physical significance of Bit Error Rate (BER) and Outage Probability especially in low power wireless communication such as internet of things (IoT).
Any help in this regard will be highly appreciated.
|
I have 2 signals $s_1(t)$, $s_2(t)$. The magnitude of the autocorrelation functions verifies: $|R_{s_1}(\tau)|<|R_{s_2}(\tau)|$. But the amplitude difference is very small.
When the signals pass in the same channel (Rayleigh for example), I noticed that the relation was always verified. But if I transmitted these signa... |
Let us consider a random wide-sense stationary process $n(t)$, which passes through a filter $h(t, \tau)$. Its autocorrelation function is
$$R_{n^\prime}(t)=\int_{-\infty}^{\infty} \int_{-\infty}^{\infty} R_{n}(t-\tau+\tau^\prime) h(\tau) h(\tau^\prime) d \tau d \tau^\prime$$
For white noise input, $R_{n}(t)=\delta(t)$... |
I have a system which has the following transfer function
$$H(s)=\frac{\beta + s}{s^{2} + 2\alpha s + \beta^{2}}$$
where $\beta = \sqrt{\omega^{2} + \alpha^{2}}$ and $\alpha>0$. The degree of numerator is less than the degree of denominator, moreover the zeros of $s^{2} + 2\alpha s + \beta^{2}$ are:
$$s_1=-\alpha+i\ome... |
According to Andrew S. Tanenbaum (in "Computer Networks", Chap. 2, Section 4 "The Maximum Data Rate of a Channel"), the Nyquist/sampling theorem states that "if an arbitrary signal has been run through a low-pass filter of bandwidth $B$, the filtered signal can be completely reconstructed by making only $2B$ (exact) sa... |
I have Fourier transforms of two images which I wish to add (Basically I have an input Fourier transform which I mask, reconstruct the underlying image using an algorithm, and then try to replace the existing values in the input Fourier Transform in the Fourier transform of the reconstructed image.) They are masked in ... |
I am calculating the convolution of two functions $F(x), G(x)$ in $\mathbb{R}^{n}$, n-dimensional space. I have another function $h(x)$ that is a Gaussian.
What effect does multiplying $F(x)$ by $h(x)$ and concurrently dividing $G(x)$ by $h(x)$ have, on the resulting convolution?
Specifically, I am interested in the qu... |
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