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Assuming all “real” number, consider the detection problem of $x$ given $y_1$ and $y_2$ below
$y_1= x + n_1$
$y_2= n_1 + n_2$
where $x$ is either $+A$ or $-A$ with equal probability, and $n_1$ and $n_2$ are i.i.d Gaussian random variables with zero mean and variance of $\sigma^2$.
Is observation of $y_2$ helpful for d... |
I'm trying to implement a 3rd order Type 1 Chebyshev digital lowpass filter in C++. I've been having a lot of problems and I'm not sure why. I can plot the transfer function in MATLAB and see that it works, but when I do the bilinear transform manually the frequency response isn't as expected.
$\epsilon=\sqrt{10^{r_\ma... |
I have a 3 stage biquad filter, so I have 15 coefficients, that filter was created for a 48kHz signal,
AFAIK, filter coefficients are calculated in a normalized frequency matter, can they be scaled of calculated for another frequency sample? ie. 96kHz? getting the same freq response?
currently using Matlab for this
|
$\DeclareMathOperator{\sinc}{sinc}$ Given 3 signals
$x_1(t) = \sinc(t) , x_2(t) = 10\cos(2\pi f_2t)$ and $x_3(t) = 5\cos(2\pi f_1t)$,
where $f_2 = 100\text{ kHz}$ and $f_1 = 1\text{ kHz}$.
I must create the general form of the simple amplitude modulation AM ($S(t)= A_c\left[1+k_am(t)\right]\cos(2\pi f_ct)$) , the DSB-S... |
What are the downsides of using an LMS (or least squares) equalizer with FSK signals? What are the tradeoffs on doing equalization pre-demoduation vs post-demodulation?
Most literature I can find discusses equalization on linear modulations. I have found very little that covers equalization on non-linear modulation tec... |
Transducers have a certain DC offset and it can drift relatively quickly. After a high-pass filter is used (say 0.1 Hz) will there be any noise in the frequency bands of interest (eg. 30Hz-200Hz) because of this drift? How to estimate the amplitude or voltage density of this noise? This is in the context of online proc... |
When I am reading Uriel Frisch's book "Turbulence", on page 55 he claimed that the Wiener-Khinchin formula for an incompressible isotropic random field, such as the velocity field of incompressible fluid based on the Navier-Stokes equation, is in the form of
$$E(k)=4\pi k^2 E_{3D}(k)=\frac{1}{\pi}\int_0^\infty k\rho \G... |
Consider a discrete-time sequence, $y[n]$, defined as:
$$y[n] = \frac 12 x[n] + \frac 12 (-1)^n\: x[n]$$
where $x[n]$ is another discrete-time sequence.
The DTFT of $y[n]$ defined as $Y(e^{j\omega}) = \sum_{n=-\infty}^{\infty} y[n] e^{-j\omega n}$ can be easily written in terms of the DTFT of $x[n]$ as:
$$Y(e^{j\omega}... |
I have a sampled signal, which has constant segments(I pad the signal if some of the BLE packets were lost). I want to apply a digital filter to the signal, say Butterworth and I don't want it to affect the constant segments. Any advice how to achieve it? An example of such filtering I see in EDF browser, but I it's ha... |
I'm working on porting Opus IMDCT/FFT to a fixed point 16-bit DSP with a 48-bit accumulator. I'm trying to do that with 32-bit fixed point numbers to keep the highest possible quality. I'm basing my implementation on Xiph's reference implementation which can be compiled to use fixed points.
Looking at the macros used t... |
If we have a sinusoidal signal at 50 kHz and we sample it by an ADC with a sampling rate of 7 kHz. What would be the output of the ADC?
Since the sampling rate is way less than the Nyquist rate (100KHz), we definitely have aliasing. When I write down the frequency domain signals, I will end up with this sequence:
$$y(n... |
Im simulating a system in code, the real system uses a 3Mhz sample rate on PDM, trying to simulate the feedback response, in the feedback path there is a system, this system has a impulse response of ~512k coeficients, currently processing the feedback in a time domain basis, meaning, once a sample is received, then a ... |
I am processing an EEG signal acquired by showing 10 image of each of 40 classes. I want to label the signals converted to epochs to their corresponding class labels.
The event_dict contains data
[('initial event', 65280), ('initial event', 65533), ('airliner', 65281), ('watch', 65281), ('folding chair', 65281), ('radi... |
I understand that a sine wave with an effective value of 1 Pa corresponds to a Sound Pressure Level (SPL) of 94 dB. However, when I perform a Fourier Transform on such a sine wave with amplitude of $\sqrt{2}$, the FFT result shows a peak of $\frac{\sqrt{2}}{2}$, which corresponds to a SPL of 91 dB. I'm having difficult... |
I was intending to use the following convolution property in my Matlab code.
$$y[-n]=x[-n]*h[n]=x[n]*h[-n]$$
However, I am noticing that the results seems not the expected ones, for example let have
x = [ 1 0 0 0 0 0 1 0 0 0 0 0 0 0 0 0 0 0 0 1]
h =... |
I am implementing a timing synchronization loop for a QPSK demodulator. I have chosen to use a Gardner timing error detector, which has often been mentioned. I am about to close my loop and I am connecting the output of the detector to the loop.
A question comes to my mind. The output of the detector follows an S-curve... |
Consider the signal of e.g. a photodiode or a spectrometer, where a dark measurement has been taken to account for baseline noise. The signal we want to measure is quite weak, so we need a decently long integration time to almost saturate the device, lets say 1 second for the sake of simplicity. This gives us a reading... |
What would be the best way to plot or graph a mixed signal that contains other signals that can be isolated?
|
I have a system and I know its impulse response $h$, this impulse response was recorded at $f_s = 48000 \texttt{Hz}$, now I can use Matlab to get the system output to an arbitrary signal ($x$) by convolving it with $h$, right?
So now I want to use an input signal that its at a higher frequency [$3.072 \texttt{MHz}$ (f... |
I recently took a dive into DSP and I was wondering what the appropriate scaling factor for a wide-band differentiator is.
Let's derive the impulse response of such a differentiator.
$$\begin{aligned}
h_{wb}(n)&=\frac{1}{2\pi}\int_{-\omega_c}^{\omega_c}H_{\text{ideal}}(j\omega)e^{j\omega n}d\omega\\
&=\frac{1}{2\pi}\in... |
[I'm very new to this website and to DSP in general as well! :)]
Hello everyone!
I have a question on the derivation of the circular convolution.
As far as I understand, we can define a circular convolution between two finite length signals g and h in the following way:
[source of all images: Convolution theorem - Wiki... |
I am dealing with the following problem:
I have $M$ signals ($x_1, x_2, \cdots, x_M$), each of length $N$, which are supposed to be similar, up to a certain delay between them and noise. I am searching for a reference signal $s$ that minimizes the mean squared error (MSE) across all of them, considering delays. In othe... |
I'm working on an audio visualizer using Blender, and I'm wanting to separate input audio files into their respective frequencies to allow my models to react based on the amplitude at different frequencies. Is there a Python library out there that can (somewhat simply) separate an audio file into different frequencies?... |
I have looked at previous questions on this topic but I am still unsatisfied. I want to predict how accurately I can estimate the phase of a tone as a function of signal SNR, tone frequency, and the signal length (assuming AWGN). My method for estimation is to just do the FFT, pick the tone frequency and calculate the ... |
I'm looking for a fast but robust algorithm to detect whether a camera raw image has clipped/saturated highlights and, if so, compute the level at which saturation occurs. This saturation/clipping level is called the white level. For the purposes of my question, a camera raw image can be assumed to be a two dimensional... |
I am trying to write a simulation code to estimate AOA's from channel state informations(CSI) using MUSIC algorithm. I am using OFDM with 64 subcarriers. Is there any material on how to estimate AOAs directly from CSIs using MUSIC?
I have linear antenna array. The number of antenna is a variable that I can play around ... |
I have noise measurements for a circuit, in units of noise power (measured in Watts against a known load), per frequency bin.
For simulation purposes, I wish to synthesize noise with the same spectral shape.
I believe I can model my noise as Gaussian, but it isn't white.
How can I synthesize such noise?
My guess is tha... |
Problem/Question
After taking the convolution of two discrete sine waves of the same frequency and length the result appears to be a sinc function but I was expecting to get a sine wave since this should be the same as multiplying two impulses in the frequency domain right? There is a similar question posted here -> Wh... |
I try to use a denoising algorithm to process an audio in streaming mode.
What I've done is spliting the audio into 0.5s long segments and use denoising algorithm respectively.Then use np.cat or np.hstack to merge them together.But in this way,the resulting audio will have a lot of spikes.And the resulting spectrum wil... |
I have a small issue.
I'm going to do FFT2 onto a matrix $X$, and the only library I have is FFTPack. FFTPack is the oldest FFT library, but it's one of the fastest and 100% portable too.
The problem with FFTPack is that it has no FFT2 function. Only FFTPack version 5.1 has FFT2 functions, but those functions are writt... |
I'm working on an audio visualizer using Blender, and I'm wanting to separate input audio files into their respective frequencies to allow my models to react based on the amplitude at different frequencies. Is there a Python library out there that can (somewhat simply) separate an audio file into different frequencies?... |
(note: I read the site policy and meta questions about whether such a question is ok. I will be specific)
I am looking for books that specifically cover using convolutional neural networks (CNN) for audio processing for signal detection, including the preprocessing steps for the audio and the network architecture. I n... |
I am trying to simulate the reflection of a sound ray, that goes from a sound source, bounces off a wall, and is received by a microphone.
The wall has a an absorption coefficient, and a specular reflection coefficient, both of which vary by frequency. Thus, the sound reflected by the wall specularly can be characteri... |
I am in the process of compressing a set of chromatograms to develop a preprocessing pipeline, that will end with DTW for peak alignment. During this process I have downsampled a signal from 0.4 seconds to 2 second intervals, and am looking for an appropriate similarity metric to summarise the difference between the or... |
To analyse data in the frequency domain I often use functions like stft() in Matlab. Inspired by this video and this article I have compared the results of spectrums generated by stft() and a Filterbank approach. I am quite pleased with the result of the 2nd method. However I wonder if there are other ways or methods t... |
I'm doing FFT2 with FFTPack 5.1 and it's working great. But it does not give the same result as Matlab's FFT2 command.
Turning Time domain -> Frequency domain -> Time Domain give the same result as Matlab's
ifft2(fft2(X))
How to reproduce the example.
/*
===============================================================... |
I am looking for a formal proof of the result:
The Hilbert Transform of a real signal $x(t)$ is generally a complex signal. Can this be proven and if so how? Thank you.
|
I am trying to understand the concept of out-of-band (OOB) which occur with OFDM, I have tried to read about it, but I couldn’t understand how I can illustrate that effect with MATLB.
This is the matlab code I am using:
clc; clear;
% Parameters
fft_size = 64; % FFT size
cp_length = 16; % Length of cyclic pre... |
Say I have a sinusoidal signal of 49.93 Hz, which is sampled at 16 kHz. A period of the input signal would then contain 16000/49.93 = 320.448 samples. Naturally, the number of samples has to be an integer, so it is rounded to 320 samples. Say the RMS value of the continuous input wave is 100 V (volts). When I calculate... |
Recently I meet the following question: Let $X\in {\mu_1,\mu_2,\cdots,\mu_m}$, $Y \in {R_1,R_2,\cdots,R_n}$, where $\mu_i, i = 1,2,\cdots,m$ is means of $m$ gauss probablity density function, $Y$ is the output of a discrete memoryless channel (DMC), probabilities $p_{ij} = P(Y=R_j|X=\mu_i)$ denote the crossover probabi... |
I notice that when I connect two audio clips, there is a jump at the junction.
How do I use interpolation to connect these two pieces of audio?
Thanks for the answers, I've got an answer from my other question：
How to merge audio better?
Some related information:
1.Noise is the sound of aircraft propellers 2.I want to... |
I have EEG signal data from a single channel for 150 similar events (epochs), each spanning 11s which might be accessed from here.
I ran the following code block in MATLAB to remove line noise at 60 Hz using a Butterworth order-4 filter and then, plotted PSD for increasing time durations from t=100 ms to t=3000 ms (alw... |
I have some binary data that mostly comes under the form of a rectangular signal. In order to "idealize" my signal I would like to fit a rectangular wave that best represent my signal (Here I mean, in lack of a better idea, minimizing the sum of squared errors). So I tried that and it miserably failed.
Then I thought I... |
I have a grayscale image consisting of MxN 1 byte pixels.
The image shows K vertical lines. Each vertical line has the same thickness.
Due to a systematic error each line is imaged in a different horizontal position than the actual position. The horizontal shift differs between lines.
Consider a scenario where there ar... |
We currently have a radio that is transmitting at the following RF params:
Modulation: OQPSK
Data rate: 500Kbps
Uplink coding: BCH
I am interested in understanding what is the associated uplink bandwidth. From my understanding, the formula is R = 2B * log2(M) where R is data rate in bps, B is bandwidth in Hz, and M i... |
background:
Trying to filter out all other signal frequencies except -> signal whose frequency can be in multiple's of 16Hz {16,32,48.......160].
so started experimenting with a single tone [no noise] at 16 Hz and i get weird output spectrum.
Input: single tone waveform at 16Hz.
Filter: multi-bandpass filter design u... |
When solving
$$A_{m-1}(k) = \dfrac{A_m(k)-A_m(m)*A_m(m-k)}{1-A_m(m)^2}\,,$$
the $A_m(m)$ here is $A_3(3)=-1$, which results in infinity if put in above equation.
How to solve it?
|
I have a 3D point $(x_w,y_w,z_w)$ with normals $(X_n,Y_n,Z_n)$, with the intrinsic matrix I can project it to image space $(x_i,y_i)$. I want to calculate a homography matrix such that I know how the 2D feature around $x_i,y_i$ looks like when it is located at the center of the image, as well as correcting any perspect... |
I am working on 50 samples obtained from time domain signal by FFT. It appears at certain frequencies which are -- according to our experimentalist -- due to some issues with their oscilloscope. What is the best way to remove these?
They seem to appear at fixed frequencies only but they are not present in all samples t... |
When we need to analyse band pass signals we use complex envelope and preenvelope representations, wherein a signal is given an equivalent low pass representation.
My issue is why it is done, though I agree that using a preenvelope or complex envelope remove the carrier amplitude and shift the signal around zero freque... |
A differentiator with frequency response $j2 \pi f$ is unstable because as frequency increases its response becomes out of bound.
But from a pole zero point of view a differentiator just have zeros and no pole, so I believe it is a FIR filter, but then it should be stable see this
I am confused how one can talk about t... |
Does anyone know how to detect and quantify a digitally time stretched audio signal when made back to tape?
|
I intuitively understand what the DFT is doing and what the equation means. We are essentially adding up all the points of the input after projection onto the unit circle and then taking an average (dividing by 1/N). I note that this division of 1/N is often done in the IDFT (inverse).
What I don't understand is why wh... |
I have a rudimentary understanding of Convolution, the Convolution Theorem and why the output z(t) of an LTI system can be found using the convolution of input signal x(t) and the impulse response h(t).
In an old post, I came across a statement "...assume you want to pass signal x(n) through filter y(n). The output of ... |
I am a PhD in mathematics. Recently, we made an attempt to compute the eigenvalues of non-normalized discrete sine and cosine transforms. Surprisingly, the issue regarding three particular types, DCT-I, DCT-V, and DST-VIII, is rather complicated. Anyway, my question pertains to finding possible applications.
I mean, wh... |
What is the probability distribution of phase of fourier transform of white noise and colored noise (1/f noise in particular) ? Is there a standardised definition?
Often, the colored noise is derived from filtered white noise (whose magnitude pdf can be guassian or uniform or one of the other distributions). While thi... |
I have a signal which is a mixture of 2 close-frequency sinusoids, something like 1320 Hz and 1325 Hz, with an amplitude envelope which is typical for a musical instrument (ADSR).
$$s(t) = a_1(t) \sin(b_1 t + c_1) + a_2(t) \sin(b_2 t + c_2) + \text{noise}(t)$$
with the frequencies $b1$ and $b2$ very close from each oth... |
Passband : 0 to 0.5π
Stopband : 0.7π to 2π
Is there a way to reduce this to a 0 to π scale?
If I had to guess, we should be able to simply divide the edge frequencies by 2 and use the time scaling property of z-Transform. Can anyone confirm if this is a valid technique?
I am not sure if this is even a valid question bu... |
I was asked this question in an interview couple of weeks ago. I think I answered it correctly, but I think I might missing some points. Here is the problem, following by my answer.
Part 1)
In Figure 1, for one of the TX beam, we have two of the strongest RX beams, $V_1$, $V_2$, obtain frequency-flat complex channel es... |
Hi guys I am trying to find a peak in waveform 1-d array data until I can find peaks in Python using scipy.signal find_peaks reference for that https://docs.scipy.org/doc/scipy/reference/generated/scipy.signal.find_peaks.html
but right now I want to implement the same thing with Arduino do you know how I can do it
|
Almost all of the searches online have returned results related to image-processing, but I am looking for a more rudimentary example with some physical intuition (eg; a high-pass filter only lets the high frequency components pass, thus filtering out the low frequency components).
The specific filter that I found and a... |
I have been working on a MATLAB code to perform convolution and calculate the power of the resulting signal. However, I have encountered an issue with the power calculation in my code.
I am convolving two signals, signal1 and signal2, and I know the power of signal1 (input_power) and the desired output power (output_po... |
Recently, I came across a paper that focuses on signal processing using lock-in amplification.
the algorithm involves applying the Fast Fourier Transform (FFT) to a time-domain biosignal.
A band-pass filter is then applied in the frequency domain, specifically targeting the range of 0.7-1.5 Hz. This calculation is limi... |
I'm trying to correctly implement the real-time variable speed playback of my buffer,
to achieve the pitch shifting.
I don't need to stretch to preserve the original length.
Buffer is calculated with 48000 sampling rate.
The code I have so far works well until some high-frequency audio is stored.
Then when pitching up ... |
I have a time-domain magnetic field, $\mathbf{b}(t)$, with components $b_x(t)$ and $b_y(t)$ where $x$ and $y$ denote orthogonal coordinate directions (i.e north and east, respectively). At each time, I can compute the direction (or bearing) that the magnetic field vector points:
$$\gamma(t) = \textrm{atan2}\left (\frac... |
$$ \frac{^2()}{^2} + 2\frac{()}{}+ 1 = 2̈()$$
a. Write Matlab source code evaluate the impulse response of the system.
b. Write Matlab source code to evaluate and plot magnitude and phase the spectrum of
the impulse response of the system
|
I am currently choosing a window function to analyze signals in frequency domain. While I understand the reasoning behind using a window function and what to look for concerning main lobe and side lobe forms, I have stumbled across the fact that the frequency domain representation of a window function that Matlab gener... |
Relatively new to the DSP side and wanted some help to implement this approach using Python (with NumPy, SciPy libraries) or via MATLAB.
Background of the problem:
I'm running a linear dynamic loads model using ANSYS (commercial finite element software) and the run time is on the order of 4-6 days. The input is an acce... |
I've posed the similiar question
I appreciate that the former replier tells me the process is about Hilber transform,but I'm still confused with the positive frequency.
The paper process a cardiac cycles, using a lock-in amplifaction. Chose a region mean pixel value as the time domain signal. After that, it operates ff... |
I have a camera looking down on a flat horizontal surface, Z=0.
The position of the camera with respect to the surface: $[XC,YC,ZC]$ is unknown.
Also the orientation of the camera with respect to the surface is unknown. Ideally the camera should look perfectly vertically on the surface: $\vec{dir}=[0,0,-1]$. However, u... |
I have a PWM signal going into an ADC.
After the ADC the signal is being filtered using an IIR single pole filter.
Performance is exactly what I want with one exception. The purpose of the filter is to average a voltage at the ADC that represents a current. For high duty cycles getting an estimation of the average curr... |
My understanding is that the relationship between complex signals and real signals is such that for some complex signal $f(t) = s(t) \cdot e^{j2\pi f_c}$, the actual broadcast signal sent through the physical channel is $\Re{\{f(t)\}}$, such that for a given complex signal magnitude $r(t)$ and phase $\theta(t)$ of $s(t... |
Question
Is there an existing algorithm that does cross correlation with two images, including a weight image?
In template matching, when should I prefer the sum of absolute errors vs cross correlation?
Background
I have a high-resolution image of a world game map, my base image. My game displays an axis-aligned mini-... |
After I use deep learning algorithm to enhance the speech, the speech will still have a weak background noise.The background noise of this audio has little effect on the calculation of SNR, but it does affect the hearing.I want to use traditional algorithms to eliminate the background noise.The frequency spectrum of au... |
Suppose there are two functions $f(x)$ and $g(x)$. Each function is nonzero only in the specified interval, $[m_f, M_f]$ and $[m_g, M_g]$. What I want to calculate is the cross-correlation between two functions:
$$
(f\star g)(t) = \int_{-\infty}^{\infty} f^\ast(x)g(x+t)dx
$$
which can be obtained by applying convoluti... |
I need to derive code - or preferably see code of already derived - for Winograd's prime power DFT of size 27.
Found the original (1978) paper, but lack the background in number theory etc to be able to follow the process.
Also failed to Google for ready made solution.
No, it isn't a study assignment...
|
I am pretty new to algrebra so if u can use simple terms and explain a bit more it would be helpfull. Ive been trying to use imaginary numbers to get the total square error of a graph of a non linear equation. i figure theres a way to calculate a near perfect equation just using x and i without squareing x or anything ... |
I am working on a project involving an accelerometer that outputs a pdm signal, which needs to be decoded to an acceleration. The device itself outputs a clock and data line, we just have to sample data on the rising edge of our clock. We are able to collect the data without issues with our chosen. Just so we are clear... |
I have a complex sequence $z(t)$ in time which I know to be a Gaussian process. I read that the complex Gaussian process is not only characterized by the covariance, but also the pseudo-covariance matrix. However, I do not have expression for either of this. What I know is that the power spectral density (PSD) of $z(t)... |
I am running a WT using pywt.wavedec on a gait signal (accelerometer and gyroscope data), I'm setting level to 5 and using a db4 Wavelet. I am wondering why we only get the approximation coefficient cAn (cA5 in my case) returned from the pywt function. How do I access the rest of the approximation coefficients?
Would a... |
I am trying to understand something in remote sensing but lack a background in science and technology so it is difficult. Hopefully this is a reasonable question to ask.
In sensing there are different definitions of resolution. One of these is Ground Sample Distance (GSD), measuring the size of the pixel subtended on t... |
I am working on my thesis where I'm employing Butterworth filtering on images using the function butterworth from scikit-image's skimage.filters module. The code/formula I'm referring to is available here
For the sake of academic rigor, I need to specify the filtering frequencies in terms of "cycles per pixel" (c/p), w... |
If it helps, I want to do perform the below computation in python.
Two time-series - 30-min cadence (Nyquist of 24 cycles / day) lasting 27 days, followed by 10-min cadence (Nyquist of 72 cycles / day) lasting another 27 days. There may or may not be a gap in between these two time-series (the 27 day observation may al... |
When I apply a bandpass filter in python to a pure sinusoidal signal, it seems that after filtering I get different frequency components across the spectrum (see orange line below). It looks like leakage, however plotting of the FFT should be "clean", i.e. the frequency of the sinusoid lands on a single bin. Why does t... |
Suppose I have real-valued continuous-time sequences $x_1(t), x_2(t), x_3(t)$ with Fourier transforms $X_1(\omega), X_2(\omega), X_3(\omega)$. I know that $X_3(\omega)=H(\omega)( aX_1(\omega)+bX_2(\omega))$, where $H(\omega), a,b$ are complex, and my ultimate goal is to find $\overline{x_3^2(t)}$, i.e. the average valu... |
I'm trying to evaluate the performance of PLL in compensating the phase noise in the absence and presence of AWGN noise, and the counterintuitive result I consistently get is that PLL performs better in noisy channels. Is this expected? and Why?
The code with which I get this result is a little bit complicated because ... |
I am watching this video and at 7:12 he calculates the frequency deviation $\Delta f = f_{c}f_{m}A_{m} $ where the $c$ underscript is for "carrier" and the $m$ underscript is for "message".
But if you analyze it dimensionally the units dont match $\Delta f$ should be in Hz but $f_{c}$ and $f_{m}$ are in Hz and $A_{m}$ ... |
My understanding is that a Gabor filter gives us information about when in the input signal does the frequency of interest (equal to the frequency of the sinusoid in the Gabor Filter) occur, thus tackling the inability of the Fourier Transform (FT) to give temporal information.
However, a few videos I saw and articles ... |
My understanding is that the standard deviation of the Gaussian window in a Gabor filter dictates the temporal resolution.
Wouldn't it always be better then to make the window smaller, thus achieving a high temporal resolution of exactly when in the input signal the frequency of interest (equal to the frequency of the ... |
I'm currently exploring BFSK and I've stumbled upon a result that I didn't expect. I wonder whether there is an intuitive explanation for it.
The intuition I've had about BFSK so far is that it can be thought of as a sum of two on-off keyed carriers. That appears to be what I see. For my graphs I've used a symbol rate ... |
I have a one-third octave spectrum with 23 frequencies, and I want to use it as a frequency response to filter another spectrum, however that spectrum is specified linear with 2049 points.
I know I can't simply linearly interpolate because a) the OTO spectra are not evenly spaced in their frequency bins and b) the weig... |
When convolving with undefined functions, do we just treat them like variables? (eg. How do I handle $u \left( k \right)$ and $u \left( n - k \right)$ since they are just step functions with no equation listed?)
I treated them like variables and got this answer but it doesn't look right. Am I supposed to put them in th... |
I understand that the Matlab function FFT gives the amplitude and phases. Is there a way to find out or approximate the errors on those values?
|
A paper I am reading (Linear and Nonlinear Encoding Properties of an Identified Mechanoreceptor on the Fly Wing Measured with Mechanical Noise Stimuli) defines power level for a white-noise signal as the area under the autocorrelation function of the signal.
What exactly is power level? I couldn't find a proper definit... |
I've been studying the modulation schemes used in NB-IoT communication, with a particular focus on energy efficiency. I've noticed that QPSK (Quadrature Phase Shift Keying) is used in multi-tone transmission, while π/4-QPSK is often used in single-tone transmission (https://scdn.rohde-schwarz.com/ur/pws/dl_downloads/dl... |
I am trying to find the correlation between the signals $u(t)$ and $\sin(t)[u(t)-u(t-2)]$
The correlation function $C(t) = \int^{\infty}_{-\infty} u(\tau+t)\sin(t)(u(t)-u(t-2))d\tau$
This is my progress so far:
$u(t+\tau) = 1\rightarrow \tau>-t , 0\rightarrow \text{elsewhere}$
So for $t\ge 0$ the graph looks like this:... |
When applying a low-pass filter to a constantly changing signal, there will be a lag between the actual value and the filtered value. For example, if f(x) = x, the regular low-pass filter output will always be less than f(x).
I remember coming across a filter that also tracked the rate of change, thus eliminating the l... |
I am currently working on my master thesis and have so far generated a broadband ESS to obtain frequency response and impulse response.
To generate the ESS as an excitation signal I use the following formula:
()=sin(21(-1)).
See: Calculating the inverse filter for the (exponential) sine sweep Method
For the broadband s... |
I have a random bit string which is being generated by an arbitrary waveform generator (AWG), being amplified by an RF amplifier to drive an optical phase modulator for binary phase shift keying modulation modulation.
The RF amplifier has a saturation power, and I want to know if my signal is going to cause the amplifi... |
I have two discrete-time LTI systems whose transfer functions satisfy $h_1[n] * h_2[n]= \delta[n]$. We also know that system 1 is causal and stable. Does $h_1[n] * h_2[n]= \delta[n]$ tell us anything about $h_2$?
Intuitively, I would say that $h_2[n]$ is stable, but I don't have any idea how to reason about it.
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