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This is my first ever question here so the help is really appreciated.
I am performing FFT on a signal. I want to perform windowing, 50% overlapping and averaging to the signal. There is a function scipy.signal.welch to perform this automatically but the output is in power spectral density. I want the output in magnitu... |
I know four common methods for numerical integration of signals such as Midpoint, Trapezoid, Simpson's rule, and FFT integration property. Are there other methods?
|
I am enjoying reading (a used copy of) Statistical Digital Signal Processing and Modeling by Monson Hayes.
A considerable chunk of the fourth chapter is devoted to deterministic all-pole modeling. I think that I understand the value of such deterministic models to designing filters and to synthesizing voiced speech (wh... |
I've seen figures in various books about the the tradeoff between aliasing and blurring when using a gaussian-like filter: the narrower it is, the more it cuts off low frequencies and thus blurs it, and the wider it is the more it leaves high frequencies and aliases. Often this is accompanied by a helpful arrows, whic... |
I need to perform automatic segmentation of dark blue spots on light yellow paper. Here's a very simple example:
In this case, a simple threshold based on hue or brightness works well. But here are some more challenging real-world examples:
Clearly a simple hue/brightness threshold will fail when dealing with shadow... |
Why are digital moving average filters and integrators considered low pass filters in baseband signal processing applications?
|
The microscopy image below is corrupted by an irregular checkerboard-like grid. I attempted running 2D-FFT on the image, but was unable to isolate the frequencies which cause the pattern.
How would I go about removing the interference pattern? The method should preferably generalize to instances where the grid has a di... |
What are the pros and cons of arranging a set of biquad filters in series and parallel? I gather that series should be used in cases where the filters contribute to the same passbands and vice versa as stated here. Is this because of precision issues in fixed point?Is this still true for floating point schemes?
|
I am working on a system that simulates an encoded channel receiver in software. I have a functioning symbol spaced Decision Feedback Equalization (DFE) that works well to track the channel and estimate the output on a per-symbol basis - this is all working as expected.
The incoming and outgoing data stream is highly... |
The np.fft.rfft2 function exploits the Hermitian symmetry of the transform of a real input to more efficiently calculate the transform by omitting the negative frequency terms along the 1 axis (i.e., as opposed to the output of np.fft.fft2, which is ny$\times$nx for input ny$\times$nx, the output is ny$\times$(nx//2 + ... |
Lower stopband edge = 25 Hz
Lower passband edge = 100 Hz
Upper stopband edge = 225 Hz
Upper passband edge = 150 Hz
Stopband attenuation = 18 dB
Passband ripple = 3 dB
Sampling frequency = 500 Hz.
|
I'm relatively new to DSP so excuse my ignorance but I was hoping to have an audio related question I had answered. If we're able to decompose audio into frequencies (e.g. mel spectogram) why is there a whole machine learning field dedicated to 'pitch detection' - wouldn't the pitch just be the mean/mode frequency in a... |
I'm preparing for upcoming exam.
I'm having difficulties with deciding if system is shift invariant.
System is described as
$$y[n] = \sum_{k=-n+1}^\infty x[1 - k]$$
I'm not sure how would output for delayed input look like and what would be delayed output here.
|
I am looking for methodology on how to correct data for losses present in RF signal paths.
The basis for my objective is to convert an discreet time domain signal into the frequency domain with an FFT, fold S-parameter info from a network analyzer into the FFT data, then IFFT the new corrected data back to the time dom... |
For a continuous time signal with spectrum
$$X_c(j\Omega)=1 \quad\text{for} \quad10\pi < |\Omega| < 50 \pi$$
Choose the minimum sampling frequency so that there is no overlap in the spectrum of replicas.
I want to know the formula for calculating minimum sampling frequency using the frequency range given above to avoid... |
I am seeking to compute a kernel radius to use with my gaussian convolution filter, and inspired by https://stackoverflow.com/a/68050503/, I came up with:
$$r=\sqrt{-2\sigma^2\ln\left( \epsilon\sigma\sqrt{2\pi} \right)}$$
This is basically inverting the normalized gaussian, setting $g(x,\sigma)=\epsilon$ and solving fo... |
Given a continuous-time filter with impulse response $h(t)$, is it possible to represent any $h(t)$ in state-space form?
More precisely, given the state-space representation of a single-input single-output LTI system $$\dot x(t) = ax(t) + bu(t)$$ where $x(t)$ represents the system's state and $u(t)$ represents its inpu... |
"In the dechirped SAR signal range spatial frequency is RF frequency scaled by $\frac{4\pi}{c} $. This quantity is has units of radians per unit length and is denoted by the symbol $K_R$. It varies from a minimum of $\frac{4\pi}{c} (F_c - B/2) $to a maximum of $\frac{4\pi}{c} (F_c - B/2) $ "
This is taken from Spotligh... |
So I know that the two methods aim to change the intensity of the pixels. My question is in which way do they differ?
|
I sample a signal that consists of a train of pulses which are amplitude and frequency modulated. I would like to generate a plot that is instantaneous frequency vs time. I compute the Hilbert transform from a down-converted signal to obtain the instantaneous phase and the envelope. Then I use the envelope to gate the ... |
I am trying to retrieve the origin signal in time domain (or path-length domain) from a spectrum obtained by CCD-matrix of interferometer.
I understand how to transform a real signal in the time domain into a complex spectrum in frequency domain. But it's difficult to do the opposite. I only have the real part of th... |
I am aware that some filter implementations such as lattice/ladder and SoS sections are advantageous over high order transversal filter structures in terms of coefficient update convergence in adaptive environments but I don't understand why.
My first thought was that the transient response is altered because the numbe... |
I am using Empirical Mode Decomposition (EMD) algorithm to decompose a vibration signal generated by outer race bearing defect using case western réserve University data.
I am facing a problem: unlike noise-free faulty signal results, noisy signal results (I add noise to this signal with snr=3 or snr=-3) in only the fi... |
On the top graph, we can see a discrete-time signal $x[n]$.
I don't understand how for the signal $x[3-n]$, the impulses with magnitude $1$ still are at the positive indices $n = 1, 2, 3, 4$. Why are they not at negative indices side since we have no positive $n$ this time, but have negative $n$ (see Figure 1.22(b)).... |
I've just purchased a pair of small diaphragm condenser microphones (never used one of these before) and was surprised by the signal shape of a sample recording (the recording is of me speaking some 20 cm from the microphone, with neither a windshield nor a pop filter):
I can't figure why the shape is not only vertica... |
I’m working independently from Ken Steiglitz’s Intro to Signal Processing textbook.
Problem 9 in Chapter 3 asks about the period of a sampled square waveform. I understand that he’s getting to the fact that the “fractional” sampling frequency per wave period has to be accounted for (in the example in the text: 40k samp... |
I am a Ph.D in pure math. In the literature of signal processing, I observe that 16 types of discrete normalized/non-normalized Sine and Cosine transforms are being extensively studied. They have some particular formulas.
What is the origin of their definitions? Why are only these special cases considered and named? (s... |
I have a set of IR LEDS, and two different cameras. The resulting image from one of them is purple-ish, the other is grayscale when blocking all visible light with a thick plastic bag. Both cameras have different sensors. Does anybody have a good technical source to read about this or could provide an explanation?
|
Distributed Acoustic Sensing (DAS)
I have an iDAS (intelligent distributed acoustic sensing) dataset obtain from an undersea optical fibre. iDAS data have a 2D dimensional representation. On the one axis we have the channel axis, i.e. the point on the cable from which we measure the strain rate obtained from the backsc... |
Having all but given up on getting USB audio to work reliably, I'm looking now at taking the stereo 16-bit I2S stream from the Raspberry Pi 4's 40-pin GPIO header.
Unfortunately, that signal stops when not playing anything (some people have made a daemon that constantly plays silence, but I'm not sure that I want to: i... |
A narrow-band beamformer for $0$ degree in the frequency domain is created for $8$ sensors and compared with a usual delay and sum beamformer. The question can also be expressed as : can we have better beamforming solution by "masking the desired spatial frequency band" to our usual beamforming result?
A beamformer cr... |
I wanted to practice Impulse balancing and tried using it to solve this IAR system in one go as opposed to using its LTI attributes. The exercise wants me to find the impulse response of the system:
$$y’’(t) +4y’(t)+3y(t) =x’(t) +2x(t)$$
$$h’’(t)+4h’(t)+3h(t) = \delta’(t)+2\delta(t)$$
I just need to get starting condit... |
Quick question:
Is it correct to define Discrete Fourier Transform like this, if my input signal is real valued:
$$
X[k] = \sum_{n=0}^{N-1} x[n] \cdot e^{-i2 \pi n \frac{k}{N}}
$$
Where $k \in \{0, 1, \dots, \frac{N}{2} -1\}$
This should result in a spectrum where the highest frequency is the Nyquist frequency right? A... |
I have to use the PID algorithm that I describe below, obtained from a PLC in the company where I work, so I want to understand it. This is used for control a valve that opens to let gas pass and reach a certain pressure (SetPoint).
There is a Timer with a preset value of "Tsample", with an output "Tsample.Q" in low le... |
I am reviewing the simulation of a control system. One of the system's requirement diagrams currently contains the following:
N-DOT is the shaft acceleration, DT is the update rate in milliseconds. This is how the above was implemented years ago, based on an earlier revision of the requirement document that is no long... |
I have an OFDM Tx Rx chain with different sampling rates at the Transmitter and Receiver with the same center frequency used for transmission and reception. I understand that different center frequencies at the Tx and Rx cause CFO and SFO but I need to check the case for the same Tx and Rx center frequency with differe... |
Consider a discrete-time system which in response to input sequence x[n] (n integer)
outputs the sequence y[n] such that
$y[n] = 0, n = −1, −2, −3,...$
and $y[n]=2 (y[n − 1] + nx[n]), n = 0, 1, 2,...$. Find out if the system is linear and time-invariant.
How do I check if it is time-invariant?
I know how to check for... |
I have been reading Leon Cohen's book "Time Frequency Analysis" as part of a project for university. On page twelve or equation (1.57) during his derivation of a representation of the average frequency in terms of the time-domain signal he provides the following relation which from my perspective came out of thin air, ... |
I am attempting to find the time derivative of physical system measurements in the Fourier domain. The system can be modeled as an ODE and has a periodic solution.
$$\frac{d}{d t} y{\left(t \right)} = - A \left(y{\left(t \right)} + K \sin{\left(B t \right)}- C \right)$$
Using the analytical solution, I generated test ... |
I am trying to understand the state space equation for a simple first-order LTI system. Suppose I have a system with impulse response
$$h(t) = \frac{1}{\tau}e^{\frac{-t}{\tau}} \ \theta(t)$$
In this system, the state ($x(t)$) and "output" ($y(t)$) are the same (both scalars). $\theta(t)$ is the Heaviside unit step fun... |
Let there be a signal f and its low pass filtered signal be fL. Then what can we say about the spectrum of f/fL ?
To be specific I am obtaining fL simply using gaussian blurring in the spatial domain.
If it helps in giving some intuitive explanation, f can be assumed to have large low frequency components and small hig... |
I am trying to high-pass filter a signal using a Butterworth filter, but I am getting the following results :
As in the picture, there is no signal decay.
python code:
import required modules
import numpy as np
import matplotlib.pyplot as plt
from scipy import signal
import math
# Specifications of Filter
# sampli... |
I have read that people often use a zero-padded DFT with Hann-windowing to get the amplitude+phase information at one frequency (where the Hann window is used to reduce the effect of a small/finite sample size). Does it make sense to use this kind of windowing if I just do a DTFT at the frequency of interest? I want to... |
I have hit a problem which could be probably solved via discrete time derivative filter able to calculate the derivative of two specific harmonics of the input signal (namely frequencies: $f_1 = 100\,\mathrm{Hz}$, $f_2 = 142\,\mathrm{Hz}$) and the rest of the input spectra let unchanged. My question is whether such a s... |
Years ago I was working with a sound editor program called Sony Sound Forge. It had this filter of smooth/enhance which I liked and used a lot, but now I can't find such filter in Adobe Audition or Audacity. Unfortunately that editor is not available on linux and windows version is beyond my budget. So I wonder if:
Is... |
Having a bit of confusion about what the initial process covariance (P) should be. Assume a 1-D tracking problem where I am measuring the distance/position of a static object. Would P not just be process error (R) in this case?
// EDIT:
Based on Tim's provided info, my thinking is that the accuracy of my measurement sy... |
Even in the simple case of doing the FFT of a simple harmonic $\lambda \sin(t)$ I am struggling to obtain the value of $\lambda$ from the FFT data. My code is
T = 100.
N = 300
t = np.linspace(-T, T, N)
y = 2.*np.sin(t)
y_ft = scipy.fft.rfft(y)
frq = scipy.fft.rfftfreq(N, 2*T/N)
plt.plot(frq, np.abs(y_ft))
plt.show()
T... |
To be as succinct as possible, I've written a Java method that implements resampling of a signal using the sinc function interpolation formula. When I test this code with a simple sinusoid, I'm able to take a test signal, upsample it, downsample it back to the original frequency, and it comes out identical. Perfect! Th... |
What would be the effect of dropping LSBs of an ADC on the performance of a DSP system apart from the obvious reduction in dynamic range. For example if there is a 14 bit ADC and only 10 bits are used, will it affect any other area of performance? especially if the ENOB of the ADC in the system was 10 bits to begin wit... |
I am trying to recreate the algorithm used by Shazam to recognize music, as outlined in the paper An Industrial-Strength Audio Search Algorithm. The algorithm performs a specialized peak detection to a spectrogram in order to recognize songs. The developers give the following details about the peak detection.
A time... |
Let be two 2D functions $f(x,y)\geq 0 \ \forall (x,y)\in \mathbb R^2$ and $g(x,y)=\frac{1}{\sqrt{x^2+y^2}}$.
I have, furthermore,
$$\int_{\mathbb R^2} f(t_1,t_2)g(x-t_1,y-t_2)dt_1dt_2=\int_{\mathbb R^2} f(-t_1,-t_2)g(x-t_1,y-t_2)dt_1dt_2,$$
which are two convolution products.
Can I prove that $f(t_1,t_2)=f(-t_1,-t_2), ... |
I have a FIR filter which is described via following transfer function
$$H(z) = h(0) + h(1)\cdot z^{-1} + h(1)\cdot z^{-2} + \ldots + h(23)\cdot z^{-23},$$
where the coefficients have below given values:
$$
\begin{eqnarray}
h(0) &=& 1.192308e-002 \\
h(1) &=& 1.055184e-002 \\
h(2) &=& 9.226209e-003 \\
h(3) &=&... |
I understand that there are two ways to get rid of DC offset: (1) subtracting mean of original signal in time-domain, (2) using high-pass filter. I'm trying both approaches on my data (using Matlab) but I'm not sure if I'm doing it correctly.
The plot in (2.1) is obtained by doing new_signal = orig_signal - mean(orig_... |
Assume I have 2 sensors capable of measuring distance to an object of known distance. If I apply a Kalman filter to these 2 sensors, I would have 2 correction and prediction equations. If I have 2 predicted distance values from this, what are some methods to return a final predicted value with this information? Or, in ... |
I'm trying to implement Farina's method in python to measure impulse response with an exponential sine sweep.
I convolved the output of the system with the inverse filter of the exponential sweep.
Here is my code for the generation of the exponential sweep and the inverse filter:
def logsweep(fmin,fmax,duration,phi0=0)... |
I'm trying to understand how the adaptive phase rotation in iZotope RX 8 works. In the papers I've looked at, the problem is called Peak-to-Average Power Ratio (PAPR) reduction. However, they seem to be for multiplexed signals and cannot be used for audio without greatly altering the sound (e.g. companders).
The proble... |
Let $\mathit{S}$ and $\mathit{T}$ be audio tracks, and suppose that $\mathit{S}$ is the superposition of
An audio signal $x(t)$ of interest, and
A signal that is audibly quieter than, but otherwise identical to, the signal represented by $\mathit{T}$
Isolation of signal $x(t)$ might then be attempted by forming the s... |
I am looking for an image similarity measure that can measure the differences between the structural composition of two images. For example these two images have structural differences between them
So I am looking for an image similarity measure that can identify that structural difference, the structural similarity ... |
Let $n$ be even and consider the non-normalized discrete Sine transform of type 5 which is
$$S=\left(\sin(k+1)(l+1)\frac{\pi}{n+\frac12}\right)_{k,l=0}^{n-1}$$
Let us denote $s_{-,l}$ by the $l^{th}$-column of $S$. It can be considered as a $n$-tuple in $\mathbb{R}^n$.
Q. I am looking for $n$-tuples $v=(v_0,\cdots,... |
Does the equality $H(n_{f}) = \frac{Y(n_{f})}{X(n_{f})}$ hold for discrete signals in the discrete Fourier domain?
|
(Updated)
I am working on a classification project in which I am required to detect a component of a railway switch using time-series data collected from an inductive sensor.
After some signal processing, I have implemented Short Time Fourier Transform (STFT), which yields a feature matrix (51 features in total) and ti... |
My background: until very recently in my studies I was dealing with analog systems and signals and now we are being taught discrete signals.
I am stuck at this question:
Suppose the impulse response of a discrete linear and time invariant
system is $h(n) = u(n)$ Find the output signal if the input signal is
$x(n) = u... |
EDIT: I have debugged the runtime warning, and now I am able to get an output image. However, the output image is still blurry. Increasing the constant value heavily distorts the image. Setting constant within a range of .001-.0001 (as recommended in the text) produces the results shown below. My code and output image ... |
First the simple questions:
Is there an effect on the Nyquist frequency when I apply a moving average filter on the raw data before I downsample?
And what does this do to aliased frequencies?
Background of my question, to help you answer in a way that I can understand it:
I am working with very large amounts of data fr... |
While teaching communication system to my students, I showed different ways to define the diagram block of the system responsible for transforming a bandpass signal, $x(t) \in \mathbb{R}$, into its lowpass equivalent signal, $x_l(t)\in\mathbb{C}$. This module is commonly called a down-converter (some authors ambiguousl... |
May I ask why does the frequency Domain Picture of a signal look like that? Is the signal built with many different frequencies?
|
I have looked at previous similar questions, but I am still not very clear on this. Most real-time or otherwise fft functions suggest adding zeros at the end of the input to make the sample size an integer exponent of 2 and make the resolution better (as the step size of the output is inversely proportional to the numb... |
I want to calculate the data rate in uplink transmission from ground IoT device to satellite.
Is it possible to calculate it with the traditional Shannon equation:
$$R = B \log_2\left(1 + \cfrac{P\cdot g}{N_0}\right) \tag 1$$
where $B$ is bandwidth , $P$ is transmit power , $g$ is channel power gain, $N_0$ is noise pow... |
I'm quite new to signal processing and am trying to find the short time fourier transform of an audio sample and plot it (I believe the best way is as a spectogram?) using Octave (similar to this MATLAB example).
I haven't gotten very far, I've successfully plotted the fft to get more familiar with MATLAB but I'm now s... |
Previously, I answer for this PID controller that I want to understand completly.
This is used for control the opening $U_n$ of a valve let gas pass and reach a certain pressure (SetPoint).
There is current error, and Tsample and 2 Tsample before. Aditionally, there is a variable that store the current change in the er... |
I'm struggling to convert three simple lines of code into a difference equation to calculate the frequency response.
The C-code is as simple (and legacy), as
float temp = 0.0f;
const a = 6;
void filter(float *inOut)
{
temp += *inOut; // add *input* sample to accum
*inOut = temp / 2^a; // scale accum and assi... |
I am trying to find an algorithm to determine the similarity between 2 screenshots of the same screen of a mobile application. I have tried structural similarity index, Peak signal-to-noise ratio and other similar IS algorithms. These algorithms generate a lot of false positives. I have also tried open open cv2's match... |
I just start learning how to calculate Shannon capacity and trying to understand the relationship between power and path loss.
I would like to make sure if I use the equation correctly.
The path loss can be express as
And the received power would be
If I would like to calculate the Shannon capacity , it should be lik... |
I'm trying to solve a problem regarding communication and I'm stuck. The questions asked me to find the maximum frequency deviation are usually written in cos or sin wave. But this question isn't, so I'm baffled and don't know how to start.
|
I would like to limit the noise after the FFT. I have used this strategy:
Divide the signal in segments using a Hann window
Perform abs(FFT(segment))
Ensemble average
Apply the gain factor of the Hann window.
This ends up being the same as doing the square root of the PSD obtained with the Welch method. I am satisfie... |
My question is based on the paper. This paper introduced a loss function by calculating two-loss: the loss between the white noise input and style image and the loss between the white noise input and content image. So my question is, why is the loss function not include the loss between the style image and content imag... |
In common literature, when generating spectrograms, mel-spectrograms, and cochleagrams, the log of the resulting filterbank energies is taken. Why is this done? I notice that my convolutional neural network does not fit very well when I don't take the log. Is this done every time a filterbank or transform is applied... |
After some research, I got a rough idea of how TDOA can be used by a microphone array to multilaterate the location of a sound source. Specifically, I'm not referring to a clustered array of many microphones that only give direction of arrival, instead I'm interested in a spread-out microphone array, similar to the Sho... |
I know mean is constant for a WSS process but I am still confused about the mean for this process. My process was by integrating $X(2t)$ from $0$ to $T$, then substituting $t′=2t$. So the limits become $0$ and $2T$ and $dt=dt′/2$. So putting these substitution, we get same mean for X(2t). Let me know if my approach is ... |
When I study some paper, I have some questions about usase of convolution theorem.
Laplacian = [0 1 0;1 -4 1;0 1 0];
F = padarray(Laplacian,[m-3+1,n-3+1]/2,'pre');
F = padarray(W,[m-3-1,n-3-1]/2,'post');
F : filter
From this, first I want use convolution theorem :
fft2(image).*fft2(F).
But, in the reference, the met... |
I am trying to understand the DSSS (spread spectrum) performance improvments.
Let's consider that we are in BPSK and on a AWGN channel without any interferences or adjacent signals.
Is the theoretical BER for a given Eb/N0 the same using or not DSSS? Does the DSSS improve the sensitivity level of a receiver?
|
Let's say I have two different real signals $x(t)$ and $y(t)$ in the auditory range of humans. What differences in properties of these signals would be minimized for these two signals to sound approximately same to a human ear?
I'm looking to use this to find the closest match $\hat{x}(t)$ to a given sample $y(t)$, but... |
I am very new to DSP, and I'm just trying to figure stuff out here. I saw a video which stated that a filter implemented using an FFT would alter the phase of different frequencies differently. The way I'd assume such a filter works is that it takes the DFT of the input, scales amplitudes based on what frequencies you ... |
I have a real audio signal $x[n]$, and I'd like to apply a frequency shift by $f$ and then envelope it by $\cos(\omega n + \phi)$.
If I wanted to do the frequency shift in the ideal sense, from posts here like this one, I think I'd first construct an analytic signal using the Hilbert transform of $x$,
$$x_\mathrm{a} \t... |
The equation of the double tuned amplifier is shown below in the image, the plot of the spectrum should look similar the the plot in the image below.
|
What is "modified frequency slice wavelet transform"? It was written in an article but It's not been described on the web.
|
What is clock frequency used for? for example, a 64MHz microcontroller is used for sampling with a rate of 256 Hz. Why? I think we can use a simpler one with lower clock frequency?
|
HRV is heart rate variability. As far as I understood, it is the time interval between two consecutive R peaks in the ECG signal. So an HRV dataset may contain two columns, one containing the interval number (1, 2, ...) and the other containing the time of that interval in milliseconds. So why in some articles it is co... |
I am using the matlab online simulator and taking the course "Wireless communications Onramp" where I need to use a Raised Cosine Transmit Filter to filter the data both in the transmitter and the receiver.
Now, the comm.RaisedCosineTransmitFilter has a property FilterSpanInSymbols,so if the FilterSpanInSymbols = 10, t... |
Why do we cascade a median filter and a moving average filter?
I know how each filter performs, but I do not know the difference between these two filters in denoising application and what is the effect of cascading them?
|
I am looking at the output of the FFT and trying to understand the meaning of the real and imaginary parts of the output.
My signals (real in time) are comprised of tonal peaks and the magnitude of the FFT output reflects this. When looking at the real and imaginary parts of the FFT output, they oscillate around zero w... |
I am studying the Wavelet transform and I am considering this example that I took from PyWavelets documentation. The signal in time domain has the following shape:
Till the value of zero on the horizontal axis we have a signal with a constant frequency. So I would expect the scalogram to have something like a constant... |
I have $N$ signals, each of length $\tau$, with $N\ll \tau$, eg. $\tau=10^8$ samples and $N=100$. I want the $r=10$ first components of all pairwise cross-correlation for the $N$ signals.
The naive way to do this, is to for every signal, take the dot product between the signal, and the $r$ shifted versions of all the o... |
Update: the yellow area in the graph below can be ignored, it shows power produced. I'm only interested in the blue line, and how to separate power consumed by the heating system from the rest.
I have data for a signal (blue line) showing overall power consumption. The spikes come from a heating system. The image below... |
How do I prove or disprove:
Smoothing an image with a 3x3 filter using full convolution and then downsampling an image by 2 produces the same result as downsampling an image by 2 and then applying full convolution with a 3x3 smooth filter.
My Thoughts:
It doesn't seem to be true because the information of an image is l... |
I want to compute the following convolution sum:
$$r_h[k] = h[k] * h^{*}[-k]$$
To do that first let $g[k] = h^{*}[-k]$
\begin{equation}
r_h[k] = \sum_n h[n]g[k-n]
\end{equation}
where $g[k-n] = h^{*}[n-k]$ according to my definition. Then the result is:
\begin{equation}
r_h[k] = \sum_n h[n]h^{*}[n-k]
\end{equation}
Thi... |
I'm trying to figure out what length of linear sine sweep I need to cover 100Hz to 9KHz at 1Hz frequency resolution. The max sweep length of my software is 160s, so for this bandwidth it sweeps 55.625Hz/s. Based on the required frequency resolution each FFT will take 1s. Does this mean that I need to split my sweep int... |
I am performing noise cancellation using Wiener filter for a project. The project statement is "to record our voice and take it as desired signal, add a noise, use wiener filter to estimate the previously added noise and finally plot the mean square error and estimated signal". So far, the algorithm looks correct but i... |
I am not sure whether I have chosen correct forum for my question. The question regards the implementation of a digital PID controller. From my point of view it is a kind of digital filter which is a chapter of the DSP.
I have implemented a digital version of the PID controller. The controller is implemented in the vel... |
I am trying to determine the thermal noise for a Frequency modulated continuous wave radar. The radar is the TI IWR6843 chip. Currently, I am using the radar to transmit a chirp with a bandwidth of 4 GHz.
I want to calculate the thermal noise power. For the IWR6843 chip, the noise figure is stated as 12 dB. The only ot... |
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