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I have audio files recorded from electronic stethoscope and in those files I want to filter out heart sounds and retain just the breathing sounds. How can I do this using just the signal processing techniques? Since I don't have clean data, can't apply machine learning.
|
I am attempting to compute the EIRP for transmissions from the James Webb Space Telescope using the high-gain antenna. I found the following reference...
https://www.ioag.org/Public%20Documents/2019-02-22_LEO26SG_Report_Final.pdf
...from which I obtained the following data (Table D-18):
Center frequency: 25900 MHz
Tra... |
Suppose I have a CIC with a down sampling rate $R$, differential delay $M$, and number of sections $N$. My sample rate is $F_s$.
If I pass a signal through my CIC I will down sample by $R$. If I plot an FFT of the signal that passes through my CIC filter how should I scale the frequency?
My initial thought is that it s... |
I'm trying to create a software FM synth following the formula for frequency modulation:
$$
y(t) = A \cdot f_c(t + I \cdot f_m(t))
$$
Where $A$ is amplitude, $f_c$ the carrier function, $f_m$ is the modulator function, $I$ the modulator index.
I have two generators that read waves from buffers/wavetables, I do the foll... |
I am new to FFT and currently studying it for harmonic analysis. I have a set of data of inrush currents from the results of simulation involving equipment energization. The data has $N = 40,001$ sample points evenly spaced at $t = 50 \mu_s$. I generated the FFT spectrum through a python code below.
import pandas as pd... |
A system is described by the transfer function
$$G_p(s)=\frac{s+2}{(s+1)(s+3)}.$$
A PI-regulator is connected to the system making it a closed loop system.
So the transfer function for the PI-regulator is $$G_r(s)=K\left(1+\frac{1}{T_is}\right).$$
So the transfer function for the system becomes $$G(s)=\frac{G_pG_r}{1+G... |
The ADC I am working on has a resolution of 24 bits and a Signal-to-Noise and Distortion (SINAD) of 108.4 dB. Assuming only quantization noise is present, the Signal-to-Noise Ratio (SNR) of the ADC should be 146.24 dB. However, due to factors such as thermal noise and Total Harmonic Distortion (THD), the ADC cannot ach... |
I'm implementing 6 cascaded biquad peaking filters to make a 6-band EQ. In the system I want to calculate the coefficients on the fly (it's just gain thats changing, center frequency and BW aren't). I am using single precision floating point data, and everything is based off the Audio EQ Cookbook that's posted here: Au... |
Take for example: $x(t)=A\sin(2\pi f_0t)$. The Fourier transform of this signal is $\hat{s(f)}=\frac{A}{2i}(\delta(f-f_0)-\delta(f+f_0))$. If we want to represent the spectrum of the signal we would compute the modulus of the Fourier transform, and it will look like the figure
So the amplitude at $f_0$ and $-f_0$ will... |
I am currently working on analyzing a signal with the autocorrelation function given by:
$$R(\tau) = 100 \cos( 10000\pi\tau) (Λ(2000\tau))^2$$
I need assistance in plotting the power spectral density $G_f(\omega)$ of this signal. Additionally, I am interested in calculating the total power of the signal and specifica... |
I'm trying to get an FM synth to work, the synth uses wavetable oscillators, I am adapting the formula for FM synthesis:
$$
FM(t) = A \sin(\omega t + B \sin(\omega_m t))dt
$$
To use wavetable oscillators as carrier and modulator, so I am doing something like this pseudo-code:
freq = 220;
osc1.set_frequency(freq)
signal... |
My goal is to generate linear phase FIR filters. Currently, I am generating an array of frequency response magnitudes for the positive frequencies and then I mirror those with the complex conjugate and take the inverse FFT of that.
In the image bellow, you can see that the peak of the impulse response is centered at po... |
I am trying to implement the algorithm in this paper for finding a sparse vector x that minimizes the square error $\lVert y − \Phi x\rVert_2^2$ where $\Phi$ need to satisfy the restricted isometry property.
The gradient update rule is
$x\leftarrow\mathcal{H}_s(x+\gamma\Phi^{T}(y-\Phi x))$
where $\mathcal{H}_s(x)$ is t... |
A deep encoder which takes in a 4 bit input gives out an 8-dimensional vector. If I want the power of this 8-D vector constrained to be 1, then I should multiply all the elements by A/B? Where A is sqrt(8) and B is Sum of squares of elements.
|
Intuitively it seems that to get a function back that was integrated, you would take the derivative. Instead, with the Fourier Transform we take an area under a curve of a modified function, and to get the function back we again take the area under a modified resulting curve.
I am trying to more intuitively understand ... |
I am reading about Early Late TED from "Software Defined Radio using Matlab & Simulink and the RTL-SDR". This is from the book:
Now I'm trying to understand the 11.7 formula in a 'for dummies', very simplistic way. This is my thinking and I could be way off. If early-late uses 3 samples, I assume:
x(kTs) is the E ... |
I'm confused about how the book Digital Image Processing by Gonzalez & Woods explains HSI model. It includes a double cone-shaped space for the model (like the figure below) and says every color on the perimeter of the middle circle has intensity 0.5. These include red, green, and blue.
At the same time, it says that ... |
Given the first-order IIR allpass filter depicted below, I would like to calculate the coefficient $a_1$ such that a phase shift of $\frac{\pi}{2}$ occurs at a desired frequency $f_\mathrm{c}$ (the center frequency or break frequency).
Some sources (e.g. Zölzer's DAFX) suggest $a_1 = \frac{\tan\left(\pi\frac{f_\mathrm... |
In my research work regarding wireless communication, I came across many research papers wherein AWGN is assumed to be modelled as "complex Gaussian with zero mean and unit variance".
I understood why we model noise as Gaussian, but I am not getting why we model noise with zero mean and unit variance only.
Also, what e... |
I am wanting to use the discrete cosine transform to relate the autocovariance function of a process to its periodogram. Following Chris Chatfield's book (Time Series Analysis, p129), I am wanting to use the transformation in the form:
\begin{equation}
I_{xx}\big[\omega_p\big] = \frac{1}{\pi}\left(c_0+2\sum_{k=1}^{N-1... |
My question assumes that I want to transmit some data using QPSK using and SDR.
The SDR will be fed with IQ data and this will transmitted at a sample rate Fs and be up converted to a carrier frequency Fc.
The data needed to be transmitted will gets mapped onto QPSK symbols which are transmitted at a symbol rate lower ... |
I have made recordings of a single source (musical instrument) with multiple microphones and in the interest of limiting the size of the final project, I would like to find a way to get the impulse response between a ref microphone and the others.
I have tried doing this in python by simply taking the FFT of both signa... |
The task is to implement communication system for transmission of 32 multiplexed signals through communication channel represented by copper cable whose frequency characteristic is given by:
$$H(f) = e^{-d\gamma(f)}$$
$$\gamma(f) = j 2\pi f \sqrt{LC \left (1+ \frac{(1-j)k}{L\sqrt{2\pi f}}\right)}$$
where: $d$=1200m, $L... |
As ilustrated in controlsystemsacademy shown relation between z-domain and s-damain poles by this image.
with contour for natural frequency and damping ratio given by these equations.
However there are some special case which is real-negative pole in z-domain can not represent with one pole in s-domain also mention m... |
Hi I am a newbie to signal processing and I am trying to better understand how inverse DFT works under the hood. Consider this signal and its DFT:
(Source)
For the sake of this post, let's assume that I have no idea what $x(t)$ looks like. I need to reconstruct $x(t)$ from its DFT $X(k)$.
It is known that a signal $x(... |
I have multiplexed 32 signals into 1 signal with python.
Now I want to plot that signal in time domain and to plot it's amplitude spectrum.
Professor gave us his plots so we can look up to it.
These are his plots:
These are my plots:
As you can see plots in time domain are very similar. They can't be identical bec... |
I have been trying to receive different types of signals using a USRP X310 but all I can receive is noise. I tried different types of antennas and designed many different kinds of blocks but I get nothing but noise. In my recent attempts I have been trying to transmit and receive at the time so that I can make sure the... |
How to avoid the edge effects that may occur when applying the Hilbert transform (by mirroring) for image or image series signal?
Code is appreciated (in python) please.
|
As I started researching FMCW radar signals, I discovered a popular reference code for FMCW simulation in Github. One of the most important things that I still don't understand is the parameters Number of Chirps: Nd=128. The code shows that the instantaneous frequency of transmitted FMCW signal still increases after th... |
I was given a .mp3 audio file needed to be deciphered however I don't know how.
I found the spectrogram which outlines the words "youtube", so I suspect the final answer might have a youtube link.
the audio file name was "19559q", could that mean something?
Kindly help.
https://gemoo.com/tools/upload-video/share/589650... |
In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it capitalized ? It doesn't seem to be the name of a person.
|
I was reading the 3gpp specification to understand the CDL channel models. (See section 7.7.1 and tables within).
It says that for a cluster $C$,it has angles of arrival/departure along the azimuth $(\theta)$ and zenith $(\phi)$ with respect to the transmitter/receiver. ie, $(\theta_{D,Tx}, \phi_{D,Tx}, \theta_{D,Rx}, ... |
I have sound pressure data in the form of Pascal,Pascal A-weighted, dB, and dBA. I want to compute a spectrogram in MATLAB.And I want to show color values as sound pressure level (dBA).But when I am computing the spectrogram the values are higher even higher than the maximum possible values. Can anyone please let me kn... |
I have a radio that stays on a channel for 500 milliseconds to listen to the packets and increment the detection_counter if packet is found( by correlating the preamble which is about 48us)
For some reason, radio goes to OFF state (multiple times) during the 500ms period (to deal with other higher priority tasks) and a... |
I have an audio file where people are talking, and Gaussian noise and high-frequency signals have been added to the file. My goal is to remove the noise from this file using Python. For this assignment, I am not allowed to use ML, so that really limits my options. Does anyone have any tips on the best ways to go about ... |
If I use a open source simulator (OAI 5GNR ULSIM/DLSIM) for realizing a wireless uplink transmission channel and transmits a LDPC coded data. My question is how to use it to compute the energy consumption (total power i.e. dynamic and static times execution time) on a device. I had implemented LDPC encoder c language.
|
In Scipy's implementation of STFT, an argument allows to scale the result as a spectrum or as a psd :
if scaling == 'psd':
scale = 1.0 / (fs * (win*win).sum())
elif scaling == 'spectrum':
scale = 1.0 / win.sum()**2
However, I haven't found a mathematical derivation of these two normalizations. I suspect t... |
Whenever the bandwidth of a medium linking two devices is greater than
the bandwidth needs of the devices, the link can be shared.
Multiplexing is the set of techniques ...
Forouzan, Data Communications, p. 156
I always knew that bandwidth is always of a signal. Composite signals (made of many simple sine signals) ar... |
I have many datasets like this:
to which I am trying to fit lines:
$$
y = a + bx
$$
However as you can see there are some outliers in each dataset. The number varies from data set to dataset.
I am looking for a simple but good way to filter out these outliers.
So far I am doing this:
Calculate a and b from all points... |
I am working on some code for estimating the position and orientation of a camera.
From an image taken by the camera I require that the user select lower left interior corner: green, the lower right interior corner: blue and the upper left interior corner: red:
Knowing the number of columns and rows I then calculate t... |
I'm learning about poles and zeros of transfer functions in my signals class and we have just covered the effects of poles outside the unit circle on the pole-zero plot. Intuitively, it makes sense that a pole outside the circle would cause the system to become unstable. However, I am a bit confused about the effect of... |
Amateur radio receivers often have several types of audio filters, and a common one is simply called "Noise Reduction." Compared to notch filters, impulse filters, and adjustable bandpass filters, where the operation is fairly obvious, I have little idea how "Noise Reduction" works.
Some googling implied maybe it was s... |
I am currently wondering about the frequency spectrum when performing upsampling using linear interpolation. As far as I know, linear interpolation corresponds to a $\text{sinc}^2$ function. If I apply a low-pass filter up to the Nyquist frequency before multiplying this filtered signal with the spectrum of linear inte... |
I am trying to intuitively understand the mystery of why the DFT of a complex vector with real values produces apparent symmetry in the magnitude plot (see the second plot here for example). In DFT $X[k]$ of a $N$-dimensional complex vector with real entries $x[t]$, the relation $|X[k]|=|X[N-k]|$ holds for each frequen... |
I am currently working on digital filters that can predict my input signal(assume that input signal is bandlimited). In other words, I want my filter to have a flat magnitude response in bandwidth of interest (let's say, $0$ to $\pi/4$), as well as a constant and negative group delay in bandwidth of interest. Since I h... |
Which other non-trained pitch detection algorithm is similar to PYIN in accuracy?
PYIN worked great for me, but all the code is GPL licenced or unavailable, so I am looking for an alternative?
Trained algorithms are out of the question, because I think gathering good samples for training is more difficult.
|
I have the the following transfer function (TF):
$$
G(s) = \frac{b_2 s^2 + b_1 s + b_0}{a_3 s^3 + a_2 s^2 + a_1 s + a_0}
$$
I want to program the numerical solution of the underlying system of ODEs myself and compare it to MATLAB's lsim.
This TF should be equivalent to the system of ODEs:
Then, I solve it using some d... |
I want to simulate the IEEE 802.11ax waveform and include the IQ impairment and its mitigation.
What is a reasonable/expected value of the Amplitude and Phase Imbalance coefficients?
Please specify the units, linear or dB and degree or radians.
|
A baseband signal is just a signal that has not been used to modify a carrier signal. Now, I have noticed that all baseband signals are low-pass too. Why is that?
I can understand why digital baseband signals are always low-pass: because the frequency 0 is always one of the constituent frequencies of a digital signal.
... |
In Rudolf Kalman's seminal paper, the problem of filtering is defined as below (p. 2):
We are given signal $x_1(t)$ and noise $x_2(t)$. Only the sum $y(t) =
> x_1(t) + x_2(t)$ can be observed. Suppose we have observed and know
exactly the values of $y(t_0), \dots, y(t)$. What can we infer from
this knowledge in regard... |
I'm looking for the simplest possible method for estimating the process noise covariance $\mathbf{Q}$ and measurement noise covariance $\mathbf{R}$ for a Kalman Filter. This is assuming a time-invariant system, that is the state transition matrix $\mathbf{A}$, control matrix $\mathbf{B}$ and observation matrix $\mathb... |
A CDMA system is capable of operating with three different processing gains $G_p$: 16, 32, and 64. It is known that when $G_p=32$, the number of bits per slot is 64.
Can we assume that the product of $G_p$ and the number of bits per slot is constant? I've completely lost my track with CDMA, and I have no idea how to ap... |
This feels like a real basic question but I'm struggling to find a clear answer.
I have PSK signal that subjected to a fixed phase shift noise, introduced between the transmitter and receiver - so the received signal has a constant phase shift from the expected signal without noise.
The question is how to compensate fo... |
I am exploring the Blahut-Arimoto algorithm (or any numerical method) for channel capacity calculations, particularly for continuous channels. I understand its application in discrete scenarios, but I am unsure about its adaptation to continuous channels such as AWGN channels. Could someone provide insights or a step-b... |
Consider a desired (pink) signal as well as its observation in heavy, non-stationary interference (green, this is the desired signal plus interference). As seen in the plot, the interference can also display quasi-periodic behavior.
Initial filtering removed frequency bands where the interference dominated. The post-... |
I have a discrete-time complex exponential
$a[n]*(\cos(\omega*n/N) + j \sin(\omega*n/N)) + b[n]$
where $a$ is constant and $b$: $\{0,1\}$
So it is basically an offset-modulated sequence of bits and I would like to detect when one of the sinusoids peak, because it is the right time to read out a new value. An easy way t... |
I have written a code that simulates a target at 400 m and velocity of 100 m/s.
I calculated tau_t the time delay due the the range and the velocity, then I created a received signal from the transmitted signal with samples corresponding to the delay, as you can see.
then I produced the doppler range map by match filte... |
I have a nonlinear (blackbox) system which gets a singular frequency (f) signal as input:
input = A*sin(2*pi*f) + B
And I get an output signal, which has oscillations + unknown trend that looks like this:
My question is what would be the best way to extract the phase between the two signals? Are there specific method... |
Someone said to me that Bipolar AMI is not low-pass; it does not include the 0Hz in its frequency domain. I doubt that that is the case.
However, at present I am unable to do Fourier transformations. Does anyone have a diagram of the frequency domain of any Bipolar AMI signal. I will use it to verify the claim mentione... |
The conventional signal power calculation can be:
$$
P(n) = \frac{1}{N}\sum_{i=0}^{N-1}x^2(n-i)
$$
$x(n)$ is the signal. However, I have seen another method:
$$
P(n) = \lambda P(n-1) + (1-\lambda) x^2(n)
$$
Here, $\lambda$ is the forgetting factor. So what is the difference between these two methods? Which one is widel... |
Good morning, i had a question regarding the Nyquist rate :
Let's say i have a signal $x(t)$ with Nyquist Rate $w_0$. I need to find the Nyquist rate of the following two signals :
$$(x * z)(t) \text{ where } z(t) = \sin(\omega_0 t/3)$$
$$(x * z)(t) \text{ where } z(t) = \cos(\omega_0 t)$$
If I understand, it means $x(... |
i have a 192kHz sampled signal that has been measured using an ultrasound MEMS microphone and i want to visualize the spectral envelope. I have found here and here ideas on how to do it (mainly "peak picking in the power spectrum followed by parabolic interpolation"). My data looks like this:
This is not a good envelo... |
I have formed my OFDM symbol his length is 88
(Nsubcarrier=72,Ncp=16) with Fs=1080000 Hz
we call this symbol as Signal_from_baseband
I want to modulate the OFDM symbol with Fc carrier=2.1 GHz and plot the frequency spectrum of the modulated signal with x-axis as the frequency axis in Matlab.
Here is my tentative:
Ts=1/... |
I have two signals of the same wave $f(t)=\sin(2\pi\cdot 70\cdot10^6 t ), t \in [0,1e-3] $, which I created with Matlab synthetically.
My first signal has 1e6 samples and the second has 1e5 samples.
syn_signal = sin(2*pi*70e6*linspace(0,1e-3,length(1e6)));
syn_signal = sin(2*pi*70e6*linspace(0,1e-3,length(1e5)));
W... |
I'm trying to identify the relevant specifications/calculations for building a system (at least theoretically) to accurately triangulate the position of a 915MHz CW transmitter to within 1cm. I have an array of three yagi antennae to receive the signal which I would read in with an appropriate ADC/SDR. This whole setup... |
I want to detect every buttons in every application windows.
For example, in explorer window, I want to detect every Tags, folders, files and Buttons like "page back", "page forward", "Min", "Max" and "Close". how to solve it?
For it, I used tesseract OCR but It cannot detect them at once. How to improve of accuracy of... |
I am developing an algorithm that automates the processing of audio files for the assembly of datasets of machine learning models.
Among the steps to be followed for the processing of the files is to apply filters to remove noise, reverb, etc..
For now I am only going to focus on my main concern, the reverberation. Sea... |
Good afternoon,
I am having a question regarding the following function :
Let $g(x)$ be :
$$\begin{cases}\pi + \frac{\pi}{2}x \text{ if } -2 <x < 0\\
\pi - \frac{\pi}{2}x \text{ if } 0 <x < 2\\
0 \text{ otherwise}
\end{cases}$$
From this I understand I am doing the convolution of two box functions that have are bet... |
I'm confused about image gradient computation mechanism in Python. I try to find gradient magnitude and orientation of a pixel. Here is my example image:
I = [[255, 255, 255],
[0,0,0],
[0,0,0]]
Now I compute vertical and horizontal gradients of the central pixel and gradient magnitude-orientation as follows... |
I have observed that different citations/codes use signs in different (reversed) columns/rows for many kernels. Here is an example for Prewitt (but same for Sobel, Scharr, etc); in some cases is written such:
prewitt_v = np.array([[-1, 0, 1],
[-1, 0, 1],
[-1, 0, 1]])
prewitt... |
I am currently working on a MATLAB project where I aim to replicate the functionality of certain Phased Array System Toolbox functions (phased.ULA() and phased.PhaseShiftBeamformer()) using pure MATLAB code. I've made progress in replacing phased.ULA() as shown below:
clear all;
% Provided data
retVal = 1.0e+03 * [
... |
Description of my scenario
My main computer is connected through network as a JACK client to an embedded board working as master with a loopback connection via I2S to another board that acts as a filter. So far I've been able to play some audio using applications like mpv and get the filtered audio back into my compute... |
I don't know where I went wrong, this might be a dumb question. I am solving for the z-transform of $$ y(n) = x(n+2)u(n) $$.
To solve, I used the z-transform equation,
$$ Y(z) = \sum_{n=-\infty}^\infty y(n)z^{-n} $$
Now,
$$ Y(z) = \sum_{n=-\infty}^\infty x(n+2) u(n) z^{-n} $$
The unit step function changes the lower li... |
I'm quite new to digital signal processing and currently learning the ropes. I'm fascinated by the concept of emulating analog circuits with DSP and I'm eager to understand how one might emulate an analog oscillator digitally. What are the basic principles and steps involved in this process? Are there specific algorith... |
I've read up a little here on how to deal with multiplicative noise vs additive noise, with the most simple approach to take the logarithm of the signal to remove the noise. I'm wondering how this translates to FFT / spectra. For my case, I'm dealing with time series, so the noise has a known form in frequency space bu... |
I have an audio file consisted of a sole 440 Hz sinusoidal wave, which can be heard here.
After some manipulation, I've added a noise to it, resulting in a second audio file, avilable here.
The noise is pretty noticeable with the human ear (especially when using earphones), so I wanted to visually evidence this noise a... |
I was going throught ORB paper. It states that it adds an efficiently computed orientation to FAST features. The description explains how it calculates
Moment of patch $$m_{pq}=\sum_{x,y}x^py^pI(x,y)$$
centroid $$C=\left(\frac{M_{10}}{M_{00}}\frac{M_{01}}{M_{00}}\right)$$ and
orientation $$\theta=atan2(m_{01},m_{10})$... |
I have an assignment on a time series that contains the x positions of a point in space. This series comes with another one of the same length, where every element goes from 0.0 to 1.0 and represents the accuracy of the measurement with the same index: 1 is perfect accuracy, while 0 is missing/completely garbage data.
... |
I'm working on an app to process Distributed Acoustic Sensing (DAS) data from oil wellbores. This data comes in a massive H5 format matrix. The columns represent channels or sensing positions in the fiber, the rows are the time series of the sensing recorded at a certain rate, and the values indicate a phase shift at t... |
I have simulated a FMCW radar for a single stationary target, now the next step in my project is to simulate multiple moving targets. For this I had thought of using doppler shift to show a difference in received frequecy, but for some reason after calculating the effective frequency the values of frequency for differe... |
I have read a number of posts that seem to ask this question but have not been able to find a definitive answer. Here goes...
I understand that BPSK with NO pulse shaping would produce a signal in the time domain something like this:
But, for wireless communication, this takes up a lot of bandwidth. Therefore the da... |
I use the FDAtool package to work with digital filters. I have always worked with coefficients in float format. I noticed that FDAtool always makes the b coefficients as integers. It also always adds gain before each filter section.
Here is an example of coefficients generated by FDAtool for a 4th order low pass filter... |
I have a complex signal and I'm trying to use Octave to plot a spectrogram. I found the specgram function, and it's close to what I want, but I can't figure out how to get it to display negative frequencies. Can it be done?
|
I want to normalize my FFT signal. From this page it says that we can normalize it by dividing the FFT result by the lenght of the signal in time domain.
On the other hand, my supervisor told me that to normalize it, I need to divide the FFT by the sampling frequency. Both methods don't deliver the same results, as you... |
I'm trying to "reverse engineering" a sample-based synth from the 90's
(an Ensoniq SQ1, which uses a OTIS chip for the samples; if you're interested you can find the manual of both on the web) but i think i lack the theory to complete this step.
What i know and have:
I have the original sampled "instruments", and a tab... |
I am reaching out because I’ll soon have to tackle signal processing questions that were so far handled by colleagues of mine. I come from the macroscopic link budget computation world in a satellite communications context, so I am used to summing powers directly, applying impairment budgets and so on to compute C/N an... |
In the book Modern Digital and Analog Communication Systems by B.P. Lathi and Z. Ding, on page 21, the authors wrote: "A necessary condition for the energy to be finite is that the signal amplitude $\to 0$ as $|t| \to \infty$."
The energy of signal $g(t)$ is defined as
$$ E_g = \int_{-\infty}^{\infty} |g(t)|^2 dt. $$
H... |
I want to analyze a signal after up-conversion in discrete time and frequency, for example:
Let's assume a continuous up-converted signal is:
$$e^{j2\pi ft} \cdot e^{j2\pi f_c t} = e^{j 2 \pi (f+fc) t}$$
What is the discrete-time/frequency version of the above signal? Is it
$$e^{j 2 \pi (k+kc) n /N }$$ ????
Have you ev... |
Recently I met an optimazition problem, i.e., finding an optimal quantizer which maps floating point LLRs to integers of range $-11,\ldots +11$, where the quantizer boundaries have been calculated out.
For example, let the quantizer boundaries and corresponding LLRs be:
quantizer upper bound
float LLR
-0.25
70.... |
The input sequence is spherical harmonic function of certain degree
x =
7.0807 + 0i
-5.4512 + 4.2999i
1.9123 - 6.3227i
1.2376 + 6.1254i
-3.0466 - 5.2350i
3.7713 + 4.8249i
-3.7158 - 5.2318i
2.8246 + 6.1738i
-1.0723 - 6.9629i
-1.0723 + 6.9629i
2.8246 - 6.1738i
-3.7158 + 5.2318i
3.77... |
I am trying to understand the zero-padding in frequency domain and discovered that original sequence in interpolated data only when (N/M) = integer where N and M are the lengths of output and input.
clear all;
close all;
M = 9;
N = 18;
m_p = (M - 1) / 2;
x1 = [2 4 5 2 1 3 1 7 9];
y1 = fft(x1);
y2 = (N/M) * [y1(1... |
Suppose I have the following signal:
I can compute its power magnitude spectrum:
PSD = np.abs(np.fft.rfft(signal))
Now I would like to pick a specific amplitude range, say between $10^3$ and $10^0$, and find a signal whose power magnitude spectrum matches PSD in that amplitude range and stays around the amplitude ra... |
Problem is to find 1 second moving average of a signal having sample time as Ts.
To solve this, it is suggested to pass the signal through a Butterworth low pass filter having the following specifications:
Cutoff frequency = $\cfrac{1}{2\pi\times0.5} \texttt{Hz}$
Can you please suggest how one second moving average is ... |
It is known that the discrete-time Fourier transform (DTFT) of a complex exponential
$$
x[n] = e^{j\omega_0 n}
$$
is
$$
X(e^{j\omega}) = 2\pi \sum_{k = -\infty}^{\infty} \delta(\omega - \omega_0 + 2\pi k)
$$
but what about the phase $\angle{X(e^{j\omega})}$?
|
my goal is to generate scalogram from .wav file and save as jpg or png format,i used following code :
import cv2
from scipy import signal
import numpy as np
import pywt
import matplotlib.pyplot as plt
import librosa
y,sr =librosa.load("genres_original/classical/classical.00000.wav")
print(sr)
width =np.arange(1,31)
... |
The task is to calculate power of signal below:
$$ x(t)=\sum^{\infty }_{k=-\infty }(t-k)[u(t-k)-u(t-k-1)]$$
The answer is 0.33 but I don't know how to calculate this even though I applied the formula, then I don't know what should I do next:
$$ \lim_{T\rightarrow \infty}\frac 1 T \int^{T/2}_{-T/2}\left(\sum^{\infty}_{k... |
I'm trying to figure out the implementation of IIR filters with integer coefficients, but I've run into a problem that I can't figure out.
There is an excellent post on this topic, and I took it as the basis for my implementation. To explain my whole problem, I will try to be consistent.
I needed to make two filters: a... |
I am very new to signal processing and I have been doing a lot of reading on it but I don't seem to understand everything. I'm hoping you can help my confusion a little.
I am simulating a lidar signal that has a lot of noise and the end goal is to get the power of the sinusoidal component. My algorithm for getting the ... |
For continuous phase modulation (CPM), the circular phase trajectory is expressed as follows:
$$\phi(t) = 2\pi h\int_0^t\sum_k \xi_k g(\tau - kT_s)d\tau + \Phi_0\tag{1.1}$$
Where:
$h$ is the modulation index, a constant
$\xi_k$ is the symbol time at time $kT_s$
$g(t)$ is a frequency pulse shape with duration $LT_s$
F... |
I'm trying to implement a waveform such the frequency will look like this -
$f(t)=f_0+\left(f_1-f_0\right) e^{\frac{\left(t \pm \delta_t / 2\right)^2}{2 \sigma^2}}$
To do so, I calculate it like this -
def Gaussian_frequencies( t , f_0 , f_1 , dt , sigma ):
return f_0 + (f_1 - f_0)*np.exp(-np.power(t+dt/2,2)/(2*(s... |
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