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First-order allpass filter coefficient calculation

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 ...
user70319's user avatar
2 votes
0 answers
9 views

HSI formula inconsistency in Gonzalez & Woods

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 ...
Mehrin's user avatar
  • 33
1 vote
0 answers
13 views

Early-Late TED in SDR

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 ...
Nic's user avatar
  • 11
3 votes
1 answer
69 views

Why is the CT inverse Fourier Transform also an integral?

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 ...
Yulia's user avatar
  • 51
0 votes
0 answers
17 views

How can i do a horizontal compression of my frequency spectrum for pitch modification

I am working on a dsp project inorder to make a male voice more feminin and vice versa and yes I know there are multiple layers for this but i am focusing on pitch now and later I will move to ...
Panik's user avatar
  • 1
0 votes
0 answers
13 views

How to impose power constraint on an encoder's output while training a deep NN?

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 ...
Sudharsan Senthil's user avatar
1 vote
0 answers
16 views

implement gradient descent with sparsification algorithm

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 ...
Simple's user avatar
  • 145
1 vote
1 answer
37 views

Is this FIR filter considered symmetric?

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. ...
Chris_F's user avatar
  • 298
0 votes
1 answer
22 views

What is wrong with this FM synth implementation?

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 ...
TiagoLr's user avatar
  • 101
1 vote
1 answer
47 views

Power Spectral Density and Total Power Calculation for Signal

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 ...
Javad Ibrahimli's user avatar
1 vote
0 answers
40 views

Spectrum and Fourier transform of a sine wave

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 ...
fi-xgamex do's user avatar
3 votes
1 answer
95 views

Audio EQ Cookbook Stability

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 ...
Michael Beckwith's user avatar
1 vote
0 answers
38 views

Help with Implementation of Communication System and Understanding Frequency Response in Python

Professor gave us task to implement communication system as in picture Communication system is made for the transmission of 32 frequency-multiplexed analog signals through a communication channel. ...
Drapic Edin's user avatar
1 vote
1 answer
39 views

Resolution after Oversampling, Filtering-Decimation

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 ...
MrT's user avatar
  • 21
0 votes
1 answer
30 views

What is the denominator of the transfer function when PI-regulator is connected?

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-...
per persson's user avatar
0 votes
1 answer
41 views

Calculate Total Harmonic Distortion (THD) from FFT Plot

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 $N = 40,001$ ...
user70278's user avatar
0 votes
1 answer
39 views

Why is my FM synth getting out of phase?

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 ...
TiagoLr's user avatar
  • 101
1 vote
1 answer
30 views

output of CIC filter is scaled by what frequency?

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 ...
CakeMaster's user avatar
0 votes
0 answers
22 views

EIRP for JWST HGA

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-...
Steve J.'s user avatar
0 votes
0 answers
38 views

Separating 'heart sound' from 'lung sound'

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 ...
hamza nawaz's user avatar
3 votes
2 answers
62 views

Why not overlap save for inverse stft

Since STFT uses overlapped sections of the input signals and compute DFT, it resembles overlap-save method used for block convolution, instead the overlap-add method for block convolution uses non-...
Ras's user avatar
  • 31
0 votes
0 answers
23 views

Z Transform and Difference Equation [closed]

Given that $$H(Z) = 4z+2/(4z^2+2)*(2z-1).$$ Find the difference equation of g(n) such that g(n) and h(n) are two cascading filter such that output is same as input. Moreover find the frequency ...
Rohit Kumar's user avatar
0 votes
0 answers
18 views

Looking for paper about applications of using phase screen to model space laser communication

A phase screen is a concept used in wave optics to model the statistical properties of wavefront aberrations caused by a medium. In the context of light propagation through atmospheric turbulence, for ...
138 Aspen's user avatar
  • 178
12 votes
3 answers
2k views

Why is the time domain low-pass filter the "sinc" shape?

Consider: I'm looking at low-pass filters, and I see that the time domain representation of an "ideal" filter resembles the shape above whereas the frequency domain is a box. I also get the ...
thepman's user avatar
  • 123
1 vote
0 answers
15 views

Confused About Relationship Between Number of Points and Solving Wiener Filter Coefficients in Time Domain vs Fourier Domain

I'm interested in computing M, FIR filter taps / coefficients using a Wiener filter approach with known signal and noise timestreams. I know of two ways to compute the coefficients: one in the Fourier ...
Jenny Smith's user avatar
2 votes
1 answer
64 views

low shelving filter frequency response diagnosis

I am building a second order modified biquad filter based on an implementation by Will Pirkle in "designing audio effect plugins in c++". However when I look at my implementation's ...
Brian Barry's user avatar
1 vote
0 answers
52 views

What can be the pole zero diagram [closed]

Does the pole zero plot for $$H(z)=(1-z^{-1})^3(1+z^{-1})^3$$ have 6 poles at origin and 3 zeroes at 1? Just answer yes or no, not your homework question knowledge
Bitsaa's user avatar
  • 21
6 votes
2 answers
319 views

Modelling MEMS accelerometer noise

I am trying to compare two MEMS accelerometers in terms of their noise for ground motion monitoring applications. I would like to take some "clean" data (recorded by good quality ...
veitha's user avatar
  • 61
0 votes
0 answers
13 views

Computing Channel Power from file containing RMS measurements from tri-axial spectrum analyzer

I have some files from a spectrum analyzer in RMS mode equipped with a tri-axial antenna. I am supposed to compute the channel power evolution for two distinct LTE channels. I have files containing ...
veitha's user avatar
  • 61
4 votes
0 answers
41 views

Discrete Fourier Transform of the Gaussian

Cross-posted from here I encountered the following question in a Digital Image Processing examination: Find the 2D DFT of $\frac{1}{2 \pi \sigma^2} e^{-\frac{(x - x_0)^2 + (y - y_0)^2}{2 \sigma^2}}$ ...
kaddy's user avatar
  • 41
0 votes
0 answers
48 views

padding zeros at the center: where to place the -ve frequencies

There are several thread regarding zero padding at the center and padding at the end. I have a spectrum and want to interpolate it. DFT of real sequence: $\left[X_0, X_1, X_2,\dots,-X_2, -X_1\right]$ ...
jomegaA's user avatar
  • 655
0 votes
0 answers
27 views

Is there any way of calculating the portion of signal lost as we keep on reducing the passband edge frequency of a band-pass filter?

Any kind of references would be appreciable.
Hardik Dey's user avatar
0 votes
0 answers
73 views

Prove the following [closed]

If $$x_1[n]=A$$ where $A$ is a constant, $$x_2[n]=u[n]$$ and $$x_3[n]= \left \{ \begin{array}{l} 1, \mbox{for n=0}\\-1, \mbox{for n=1}\\ 0, \mbox{otherwise} \end{array}\right.$$ Prove that $$x_3[n]\...
Bitsaa's user avatar
  • 21
0 votes
0 answers
30 views

Driving bandpass DSM with tone close to full-scale - stability issue?

I am experimenting with Dr. Schreier's DSM toolbox for MATLAB. I stumbled over a strange phenomenon when having a look at a bandpass DSM in demo-script 2. If I increased the amplitude of the test-tone ...
MisterFilter's user avatar
0 votes
0 answers
14 views

Channel Estimate for OFDM shows distortion for first last order samples

The Channel Gain Estimation was obtained using Least Square method as follows: Hk= mean(Rx_LTS ./ Tx_LTS, 2); k= -26 to 26 Where Rx_LTS is the received long training sequence (52 samples after ...
Walkürie's user avatar
1 vote
1 answer
77 views

Continuous-time convolution of signals with negative amplitudes

While preparing for a mid-term exam, I encountered negative amplitudes for the first time while convolving two signals. I've already solved the problem, but my result and results from others conflict ...
kesetovic's user avatar
0 votes
0 answers
34 views

Calculation of response of continuous-time LTI system [closed]

Consider an LTI system defined by the following impulse response $S_{1}$ $$ h \left( t \right) = u \left(t - 6 \right) - u \left( t \right) \tag{1} \label{1} $$ Assuming $x \left( t \right) = 5 \left[ ...
user avatar
3 votes
3 answers
586 views

Help me understand the concept of IQ signals

I am a computer science student working on a school project involving RTL-SDR receivers, and I have a hard time grasping the concept of IQ data. Basically what I think I have learned from all of the ...
Kuba Lhotský's user avatar
0 votes
0 answers
17 views

Allan variance plot does not match with the true plot

I am trying to simulate an Inertial Measurement Unit (IMU) by generating sample noisy gyroscope and accelerometer data. Currently I am analyzing the simulated gyroscope( gyro) data. Using the noise ...
user146290's user avatar
1 vote
1 answer
54 views

How to specify the cut-off frequence in cascaded biquad filters?

As part of my effort to understand how a high-pass filter can be implemented, I'm reading this section of HandWiki. My question here is: How do the 2 simple formula presented there filter out at a cut-...
DannyNiu's user avatar
  • 125
1 vote
1 answer
53 views

Understanding how to implement high-pass filter with cascaded biquad filters

I'm trying to understand how to implement a high-pass filter using, quote: a cascasded biquad direct form II IIR filter with a cut-off of 8kHz. DSP is a complete new field for me, and the ...
DannyNiu's user avatar
  • 125
0 votes
1 answer
65 views

Energy and power of time-continous signal $x(t)=t$?

Preparing for my exams, I stumbled upon this particular problem. I tried to calculate it using following formulae: $$ E = \int_{-\infty}^{\infty}|t|^2dt $$ Dividing it into two cases because of the ...
kesetovic's user avatar
0 votes
0 answers
22 views

Desired trajectory generation from transfer function in control theory

I am trying to implement the backstepping techniques for quadrotor described in this article : Backstepping control for a quadrotor helicopter: Madani & Benallegue 2006 Since I am new to the ...
aRedDish's user avatar
1 vote
0 answers
33 views

Testing CNN model using sound generated from pyroomacoustics room simulation

I have generated a large set of audio using Pyroomacoustics for room simulation, employing a circular microphone array and a single sound source. I have successfully trained and tested a CRNN (...
KEHINDE ELELU's user avatar
1 vote
1 answer
77 views

Fourier transform magnitude of the sum of two signals

Let $$\mathscr{F}\Big\{x_1(t)+x_2(t)\Big\}=X_1(f)+X_2(f)$$ I think that in general $$\big|X_1(f)+X_2(f)\big|^2\leq\big|X_1(f)\big|^2+\big|X_2(f)\big|^2$$ but I was wondering if $$X_1(f)X_2(f)=0,\qquad\...
giannisl9's user avatar
0 votes
1 answer
37 views

Welch method implementation not smooth

I have been trying to recreate the welch method implemented in scipy.signal by hand but my results don't look close enough in my opinion. How come the scipy.signal.welch method is displaying much ...
Flitschi's user avatar
1 vote
0 answers
18 views

please help me solve the question which proving X[N-k] is the complex conjugate of X[k] [closed]

Show for the DFT that if all x[n] are real, then X[N - k] is the complex conjugate of X[k] for k > 0. (i.e. if X[k] is a + bi, then X[N - k] is a - bi)
Reece's user avatar
  • 11
0 votes
1 answer
40 views

Characteristics of linear systems in the time and frequency domain

Why is the output signal obtained through frequency response different from the other two. ...
Alice Khomich's user avatar
1 vote
0 answers
61 views

Which filter is suitable for reducing noise in feature detection?

I have a feature detection algorithm that is called FAST - Feature Accelerated Segmented Test. It's a very fast algorithm for finding feature points, e.g "corners" inside an image. Here is ...
euraad's user avatar
  • 403
1 vote
0 answers
23 views

RLS Adaptive Filter for estimating signal

I am currently working on a project where i am to estimate a signal x_T using x_1 and x_2 with an RLS filter. I have a problem where i don't quite get the results i am looking for. I think there is a ...
Ramin's user avatar
  • 11

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