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I have an audio file (u8, 128kbps) of the sounds produced while playing a game app and clips of its three distinct sounds.​ ​ The file consists of sequences of these, separated by silences, indicating next move (0.5 sec), correct move (0.5 sec, immediately followed by next move), and game over (5 sec).​ ​ I want to con...
I'm learning about Z-transforms and transfer functions in DSP. Suppose I have the following transfer function: $H(z) = \frac{1}{z-0.5}$ I want to find the corresponding $h(n)$. If I factor out $z^{-1}$ from the numerator, I'll be creating a new zero and a pole in $z=0$. $H(z) = z^{-1}(\frac{z}{z-0.5})$ Which its in...
I implemented a data transfer system using two tutorials: Guided Tutorial PSK Demodulation and Packet Communications. After that, I decided to change QPSK to 8PSK. My scheme: Constellation Sink: For some reason, the Costas loop turns the constellation, although this should not be. If I put the Order = 7 value in the ...
I found the ofdm flowchart in the Gnu radio examples.(ofdm_loopback) Scheme: The spectrum and the received data of the received signal: After that, I connect the USRP Sink and the USRP Source. The spectrum and the received data of the received signal(USRP): A similar question has already been encountered here, but,...
I have implemented a packet reception and transmission scheme using QPSK. Next I decided to add Polar/LDPC encoding. As a result, nothing is written to the received file. I think the thing is that I'm unpacking the bits incorrectly after demodulation. For convenience, in the diagram, I highlighted packet formation and ...
Assume we need to solve the model: $$ \arg \min_{\boldsymbol{x}} \frac{1}{2} {\left\| \boldsymbol{h} \ast \boldsymbol{x} - \boldsymbol{y} \right\|}_{2}^{2} + \frac{\lambda}{2} {\left\| \boldsymbol{g} \ast \boldsymbol{x} \right\|}_{2}^{2} $$ Where $ \ast $ is the convolution operator. How can we solve this in the freque...
I want to estimate the SNR value at the input of my USRP. To do this, I connect two blocks: MPSK SNR Estimator and MSPK SNR Estimator Probe. Next, I look at the values that I got using the MSPK SNR Estimator block. QT GUI Histogram Sink: Constructed histogram using File Sink. The values I get using MPSK SNR Estimato...
Does anybody have an idea or has tested the transmitters to see what at what frequency they transmit?
How to implement sliding 2D FFT for audio data (shown in the link - R1-chirp)? I have audio data for an array of 1024 microphone which record audio (for 2 seconds) from a sound source. Microphones are arranged in plane (17 rows and 60 columns). I used 2D FFT to estimate frequency-wavenumber spectrum, however the x-shap...
I was playing some very loud pop music through a studio monitor (ADAM A8X), and after about 5 seconds the sound stopped coming out of the cone and only came through the speaker. After shutting it off for a few minutes and trying again, no sound was coming from the cone. I assume I broke something, but don't know where ...
In this question I proved that for a linear linearly time varying system $S$ such that if $t_{2} = t_{1}+t_{0}$ then $h(t,t_{2}) = h(t,t_{1})+h(t,t_{0})\rightarrow$ $h(t,t_{0}) = g(t_{0})h(t) $ where $g(t)$ is a linear function $(At+B)$. But now since we have values for $t_{2},t_{1},t_{0},g(t_{2}),g(t_{1}),g(t_{0})$ we...
Let's say I have a digital integrator with transfer function in following form $$ \frac{Y(z)}{U(z)} = \frac{T}{2}\cdot\frac{z + 1}{z - 1} $$ I have been looking for a mechanism how to compensate the phase delay introduced by the integrator. My first idea how to do that was to use a digital derivator with a filtering po...
For education purposes I decided to try to implement a Cooley-Tukey style FFT algorithm in python, I must say that I am in no way a professional python dev nor a signal processing expert, so ugly code and approximate comprehension of the subject matter are to be somewhat expected. Anyway, I followed the wikipedia artic...
I have been looking at the FFT (in python) of various simple functions. Everything was working as expected, except when I take the Fourier transform of the function: $$ f(t) = e^{-\gamma t}\cos^2\omega_1t $$ where I obtain a peak at $\omega=0$ which I would like to remove. This peak shows up in $g(t)=\cos^2(\omega_1 t)...
I have two MPSK SNR Estimator and MSPK SNR Estimator Probe blocks that insert a tag after a certain number of samples. In my case, tag = "snr" with the value of SNR is inserted every 10000 samples. The question is, with which block or in what other way can I extract the tag value in File Sink(text file) ?
I'll start by mentioning that I have very little experience in signal processing, so my question might be trivial. I'm working with a signal that represents a patient's blood tests over the course of approximately 5-10 years. I'm looking for a method to assess whether this signal has a trend (i.e. improving, worsening,...
About impedance calculations, I have to build the quotient of voltage and current. This is done in the frequency domain after, e.g. Fourier transform to get a spectrum. I'm trying to calculate the SNR of the impedance based on the individual SNRs of the voltage and current. In question 31440, the multiplication of two ...
Signal processing often involves managing and optimizing Signal to Noise Ratio (SNR). To achieve this, one may need to work with noise budgets and ensure that any additional additive noise sources are kept x dB below the current noise floor, with x typically in the range of 10 to 20 dB. These additive noise sources can...
This is a simplified version of the Savitzky-Golay MATLAB functions (sgolayfilt, sgolay). function y = sgolay_simplified(x, order, frameLen) % Compute the Vandermonde matrix S = (-(frameLen-1)/2:(frameLen-1)/2)' .^ (0:order); % Compute QR decomposition [Q,R] = qr(S,0); % Compute the projection matrix B B = Q*Q'; ...
can I know what is the model of a quasi static block fading channel in a MIMO system, in function of angle of departure and arrival?
Why in the attached image for a simple 3 point moving average that has been converted into a TF (z domain) is there a wired dip? It seems that when I change the sampling time, the dip shifts to the right. And why does the black veritable line at the end seem to be equal to /? It looks like the black line is 2 frequency...
For my assignment, I am required to blend multiple square images into a single large image where each square image represents a part of the final image. To accomplish this task, I have successfully completed all necessary steps except for one. I have been asked to provide a coverage count for the final image. However, ...
I would like some help to better understand the Fourier transform of a discrete time signal. My doubts are: The sampling of a signal can be seen as $x_s(t)=x(t) \cdot \sum_{k=-\infty}^{+\infty} \delta(t-kT_s)$ so its Fourier transform is $X_s(f)= f_s \cdot \sum_{k=-\infty}^{+\infty} X(f-k f_s)$ where $f_s$ is the ...
I produced the "elevation vs distance" profile data and their PSD plots, I am trying to get the smartphone estimated profile close to the reference profile. As you can see, the PSD of the smartphone estimated is100 times higher than that of the reference. Judging from profile and PSD plots, what filters could I apply t...
Can I use filtfilt function in realtime filter. filtfilt first filters a piece of data forward and then filters it backward. I want to collect a piece of data first and then output a piece of data after each input. Is this possible? Can you introduce me some methods in filtering, the filter delay is no more than the s...
I have an optical system, which is commonly characterized by its point spread function. Somehow by the method, which resemble slanted edge method, I have end up with discretized line spread function of the system. I have it in array lineSpreadfit so that len(lineSpreadFit)=2*tailFitLength+1, where tailFitLength is some...
DESCRIPTION OF SIGNALS I have several sung melodies under each of nine experimental contexts ("conditions"). The melodies are all of slightly different lengths (ranging more or less from 3s to 12s), and each is encoded as a discretised melodic contour (pitch profile), at fs=1000 Hz. Those contours (rather than the orig...
I have to connect the unfinished edges of the contours in the red color bounding box. I used distance transform, watershed, and morphology, but still, edges are not connecting. import cv2 import glob import numpy as np from PIL import Image import matplotlib.pyplot as plt og_path_test = '' predicted_path = '' IMG_WI...
My aim is to find the optimal cut off frequency for a low pass filter for a normal walking activity. Based on accelerometer data (x y z axis, of a wearable watch), I compute the FFT for each axis and finally the magnitude out of these three axis. When I plot the results in frequency domain, I usually get peaks around 5...
I came across this example the stream processing chapter of a programming book that I'm reading: Streams as signals We began our discussion of streams by describing them as computational analogs of the signals in signal-processing systems. In fact, we can use streams to model signal-processing systems in a very direct...
the sample rate of signal is 1khz,which filter method can i take to realize a 3-5ms delay. I would like to ask if there is a way to pass the signal through a low-pass filter and then adjust it through PI to adjust the error and reduce the filter delay? Or is there another way to reduce the delay?
I am having trouble with figuring out if the following system is time-invariant (with $y[n]$ being the output and $x[n]$ being the input): $$y[n] = 1.02 \cdot (y[n-1] + x[n])$$ I should also add that the initial rest conditions are : $x[n] = 0 $ for $ n < 1 $ and $y[n] = 0 $ for $ n < 1 $ and lastly i managed to rewrit...
I am simulating the MIMO linear detector algorithms including Matched Filter, Zero Forcing and Linear MMSE. I simulated three configures: 4x4 ( 4 tx antennas 4 rx antennas ), 8x8 and 16x16. For Matched Filter and Zero Forcing, 4x4's BER is better than 8x8's, 8x8's BER is better than 16x16's. But for LMMSE, I got the di...
In a multi-antenna system, we often mention diversity gain in fading channels. I am doing simulations with multiple antenna settings at the receiver considering TDLC channel, for example, 1rx, 2rx, 4rx.. The performance improvement I am seeing in the case of more antenna --> should I call it an array gain or diversity ...
I need to implement a real time butterworth low pass filter on some data, I have an already implemented and tested butterworth function, but when filtering data from one specific variable I get an offset. Removing the filter also removes the offset. The following image shows the reference data (green) and the filtered ...
I am confused about signal processing broadband noise like white noise and single-tone noise (sine wave) using FFT. As a starter, I am trying to understand how to normalize white noise properly after FFT. My understanding is that once properly normalized to dBm/Hz, the power spectral density of white noise should not d...
So I have a pretty robust and fast pitch tracker I've been working on based on this thesis. I want to use it for real-time pitch shifting. However, one of the simplest algorithms for formant-preserving pitch shifting (PSOLA) doesn't seem to have a standard easy to find real-time implementation. As such, I bodged the fo...
I'm working on a Fourier-Mellin transform on images to find angle and scale. In most cases when I have scaled and rotated an image, the Fourier-Mellin transform gives me the almost exact angle and scale. When I have a moiré pattern on my original image or some corrupted area of the original image, the Fourier-Mellin tr...
I have a given $h[n] = 0.96^n$ for $n \in \{0,1,...,119\}$ and a test signal $x[n] = \cos(0.05\pi n)$ for $n \in \{0,1,...,N_x -1\}$ where $N_x = 120,000$. Now, I can analytically find the output $y[n]$ using the convolution sum: $$y[n] = \sum_{k=0}^{N_x + N_h - 1} x[k] h[n-k] = \sum_{k=0}^{120119} \cos(0.05\pi k) \cdo...
I am recently trying to use ADC and DAC with low latency and high speed. I wonder what are the ADCs and DACs with the lowest latency over 1Gsps?
I used to read some white papers online, where they mention the below statement: If we have a real vector $\mathbf{y} \in \mathbb{R}^{N \times 1}$ which is an output of $IFFT$ process with size $N \times N$; $\mathbf{y} = \mathbf{F'} \times \mathbf{x}$ where $\mathbf{F}$ is the fast fourier transform matrix and $\mathb...
I am studying OFDM and have some questions to ask. Assuming my WiFi network operating on 5GHz with OFDM channel BW of 20MHz and N = 10 subcarriers: Does the RF spectrum start at 5GHz (frequency shifting by fc=5GHz, single sideband) or centered at 5GHz +- 10MHz (fc = 5.01GHz double sideband)? Why is BW equal to the sam...
I am a newbie in Optimization. I have this Optimization problem, could anyone help how can I analyze it and solve it? \begin{align}\label{Problem_formulation} \mathbb P_1& ~~~~~~\mathop{\max}_{{ \eta, x_0 }} ~~~~{B}\log _2 \left(1+ \frac{{{P_M}\Vert {\textbf g}_d\Vert ^2}{( {\frac{c}{{4\pi {q}}}} )^2 {x_3}^{ - 2}{e^{ -...
I am trying to model the pdf of the peak and rms of the time domain response $y(t)$ in a reverberation chamber. The system is stimulated by pulsed linear chirp $x(t)$ for 200 ns that goes linearly from 4-5 GHz and its Fourier transform is denoted by $X(f)$. The frequency response of the system, $H(f)$ is a random vari...
I'm running a study with continuous stimuli, I send triggers at the beginning and end of a period of recording (the time duration for each period is not fixed). I'm planning to analyse the data with temporal response function but not sure how to format the EEG data to epochs that correspond exactly to the stimuli. Can ...
Exactly which FFT algorithm does Matlab use for the fft function? I though originally it was the Cooley-Tukey Decimation in Time algorithm but the Matlab documentation does not give a reference to exactly which algorithm is used. It does however reference to this page but there isn't any mathematical information that I...
Noticing a strange problem when I upsample an OFDM symbol to a "transmit" sampling rate, pass it through a channel, and then downsample it back to its baseband sampling rate. After doing this, the equalization scheme (H = Y/X) no longer seems to work. I'm trying to follow this method (Sub-carrier Equalization and Pilot...
We're evaluating whether to implement a complex image processing algorithm using Matlab's Sysgen or translate it directly into VHDL. As a start, the algorithm is given to us in Matlab. Traditionally, teams have implemented the algorithms using Matlab's Sysgen. However, it hasn't been the most maintainable or testable c...
I have a question to ask you which has been bugging me as of late. I have s-parameter data with 5001 samples acquired from a vector network analyzer. I am trying to 'gate' this s parameter data, i.e. take IFFT of the data, get a time vector, select a time range I want to keep, and nullify everything else. As per the co...
I am calculating the SNR (signal power to noise power) for a sine wave. I don't have an integer number of periods in the waveform being analyzed, so I am using a flattop window to reduce spectral leakage. If I start with a long sine wave and select a subset of a given length, I find that the calculated SNR vs the start...
I do have differential air pressure data, recorded linearly (in 1/240Pa per step) as signed 16bit integer. Given this is the atmospheric air pressure sampled slightly into audible range (100Hz sample rate), the resulting range is roughly +-32. How would I convert this into audio data for a .wav file? Would a simple sca...
I know this has been covered thousands of times on the internet, but I'm still struggling with it. My end goal is to cancel out the noise in an Inertial Measurement Unit chip readings so I can double-integrate acceleration to get reasonably accurate vertical displacement (of a buoy bobbing on ocean wave swells). I bel...
Suppose you have a set of $n$ sensors with overlapping sensitivities--like the cone cells of an animal retina, which is in fact the sort of system I am trying to model. Given the frequency-response curves for each sensor, how can I calculate a matrix transformation that will a new set of $n-1$ maximally-decorrelated di...
I'm trying to understand "interleaving" from a theoretical point of view. This paper claimed that by sufficient long interleaving, the channel can be treated as memoryless. Does this holds for all memory noise, no matter if it's stationary/non-stationary or ergodic/non-ergodic? Is there mathematical proof for this memo...
I am trying to work through the Cyclostationary Blog to create a cyclic autocorrelation function: I have been given the above equation to determine the cyclic autocorrelation function where tau is the lags for the overlap for the autocorrelation function and alphas are the cyclic frequencies, which are varying from -0...
Decimation by a factor $D$ is the combination of anti-alias filtering and down-sampling (selecting every $D$th sample). The frequency bands where aliasing occurs will be centered of $nf_s/D$ for $n=0, 1, 2 \ldots D-1$ and sampling frequency $f_s$. Conveniently, moving average filters have their nulls at these frequenc...
I have some doubt about this exercise. The Bernoulli random process $X(n)$ with means $p=0.5$ is sent in input to a LTI system with impulse response $h(n)= \cos(\frac{\pi n}{3}) R_3(n+1)$ , where $$R_3(n+1)= \begin{cases} 1 , & \text{if } -1 \leq n \leq 1 \\ \newline 0 , & \text{otherwise} \end{cases}$$ To find t...
I am trying to implement a Phase Locked Loop in a microcontroller STM32. I have a reference signal and feedback signal which has $\phi$ difference wrt reference signal. For the moment, I am not using a real time feedback signal, rather simulating it inside the microcontroller itself, but later extend this to Real time....
I am trying to segment a song into its different parts. In pop music, it's common for a song to have a verse and a chorus. And they repeat. So it should be possible to use deep learning to find the segments in the song, no? Also it should be possible to categorize the similar segments into the same class (aka Bridge, C...
I am trying to decode an unknown signal, which has those really weird spikes/jumps. My assumption is that it is somehow (B)PSK modulated but the signal looks really odd as shown in the screenshot. The screenshot shows the Phase plot of the recording drawn by Inspectrum. I recorded the signal close to the source multip...
I studied a conference paper, High Resolution Capabilities of MIMO Radar and confused about its formula (3). As mentioned in paper that the noise $n_{k,l}(t)$ in the received signal $r_{k,l}(t)$ ($r_{k,l}(t) = as_k(t-\tau_{k,l}(X))+n_{k,l}(t)$) is white Gaussian noise, so for the pdf, I think it should follow the Gaus...
Aliasing is bad, and we want good filters when downsampling. While what qualifies as aliasing is well-defined mathematically, and we can manually design filters with various tradeoffs, how do we actually measure the "amount of aliasing" that a filter admits or rejects? What about measuring aliasing due to directly subs...
When receiving bandwidth (sample rate) is small (<~28 MHz), the signals are received without any distortion. When I choose high bandwidth (for example 40 MHz), sometimes the signals become interrupted for a small amount of time and/or some noise is added. At the same time CPU utilization is about 40-60%. I use differen...
I am trying to find the phase of my dataset between a range of frequencies. I can't post my code or data but generally speaking, the frequencies range from 10 to 100kHz and I need the phase between 20 and 30kHz. The data is read into Python as real numbers in the form of 64-bit floats. I was thinking to first pass the ...
I'm trying to implement 2D cross-correlation to acquire the displacement of these two small images (interrogation windows). The issue is that the result from Cross-Correlation (CC) is different compared to the FFT approach; therefore, the displacement/fitting would be wrong. The ffwtools package in R uses the C library...
I recorded a simple FM signal using my SDR (centered at 100.3e6 hz and sample rate of 2.048e6 per second). Without any transformation, I ran an FFT on the signal and it showed a signal with some width to it at 0 hz (not a line that would indicate a bias but signal centered around 0). I would have assumed the FFT would...
I have a real signal f sampled at 96Hz composed of clusters of harmonics scattered tigthly around evenly central frequencies close to 1Hz and 2Hz (multiples of 0.966Hz) that I'll call features. These are modulated in amplitude and phase by a slower nonstatinary system $C(t)$ that is approximately constant over each fea...
Is there any resource or literature where I can find derivations/proofs for all four linear phase filter types starting from an arbitrary real valued impulse response? I looked at some of the literature such as Introduction to Digital Filters but couldn't find anything concrete.
I have done some pressure measurements of flowing air with transducers that had a frequency response of 5 kHz. The data from these transducer was sampled also with 5 kHz and this is where my problem lies. After some research I gathered that the data from these transducers is probably completely aliased. Is there a way ...
I am writing an SDR application which searches for CW signals at various predefined frequencies. Currently, the radio tunes to the appropriate LO frequency and calculates a number of FFTs to produce an averaged result, and a CFAR process searches for any signals above a given threshold. I am keen to understand the diff...
I was wondering, how can Parsevals theorem (energy in time domain is equal to the energy in frequency domain) hold for sampling. I take three sampels from a sinc-function. This reduces the energy in time domain, but in the frequency domain the energy gets bigger. Thank you in advance.
I have a particle velocity sensor that I'm recording, which has a non-linear frequency and phase response. I have frequency and phase correction curves for it, however, the proprietary acquisition software I'm using can only apply single value sensitivities on channels. Is it possible I could apply the frequency and ph...
I'm looking into different ways to get the Power Spectral Density (PSD) (while toying around with python) of a discrete signal/time series, and I'm struggling to understand why I get (very) different outcomes using different approaches. In particular, when computing the PSD from the FFT of a signal - using the squared ...
I am attempting to load several FITS files using the HARPS-N data set of CCF files. r.HARPN.2018-01-18T14:20:10.645_CCF_A.fits This is not cubed data, it is in the following structure. r.HARPN.2018-01-18T14:20:10.645_CCF_A.fits No. Name Ver Type Cards Dimensions Format 0 PRIMARY 1 PrimaryHDU ...
I have a GNU Radio flowgraph where I send bytes (preamble, header, payload) from an external python program to the flowgraph through a ZMQ socket. The bytes are then modulated in the flowgraph and transmitted through the radio sink. What I would like to do is send an arbitrary bursts of data, after which the transmitte...
As a beginner in DSP, I would appreciate your advice and recommendations on building a PCB with eight PDM microphones arranged in a circular distribution. My objective is to connect the PCB to a computer via USB3 and receive raw audio streams from each microphone in Matlab or any processing software, so that I can appl...
I have approached wavelet transformation from a projection perspective. Specifically, we can show for a certain class of functions (in the continuous setting) that they can be written in terms of a series of scaled and shifted wavelet functions: $$f(x) = \sum_{j, k = 1}^{\infty} \langle f, \psi_{j, k} \rangle \; \psi_{...
I've been looking into both of these quantities and feel like I have a good understanding intuitively of what they each represent but according to Wikipedia both the PSD and the ESD can be computed as the Fourier Transform of the Autocorrelation function of the time-domain signal. $$ PSD = \lim _{T\to \infty }{\frac {1...
I have a measurable $y[n]$ that is the sum of two input channels $x_1[n]$ and $x_2[n]$ according to $$ y[n] = x_1[n] + h_0 x_2[n] + h_1 x_2[n-1] $$ where the scalar weights $h_0$ and $h_1$ are known. In this example, $x_1$ is unfiltered, and the channel effect on $x_2$ is to pass $x_2$ with an echo one sample later wit...
While researching receiver architectures for the demodulating constant envelope signals like a GFSK, I stumbled over the concept of phase domain ADCs, where after $$\boxed{\text{LNA}}{\longrightarrow}\boxed{\text{Mixer}}{\longrightarrow}\boxed{\text{Filter}}{\longrightarrow}\boxed{\text{ADC}}$$ instead of the classical...
I have an FPGA that is limited in clock frequency, but receives data from a higher sample rate ADC by receiving N samples per clock cycle (all N samples are from a single ADC channel). So the FPGA has to operate on N samples per clock cycle instead of 1. Now say I want to implement a generic FIR filter to operate upon ...
Background: I am working on a Unity Project (C#) and implementing a Room Impulse Response synthesis. I do ray tracing to create energy histogram for different bands. With these histograms it is possible to synthesize a room impulse response. My guide is following theoretical Reference: Physically based real-time aurali...
I need to calculate initial states of a second order direct form 1 (DF1) filter that result in zero transient upon reset (initialization). I'm starting with setting filter input (x) equal to filter output (y) (desired steady state condition) which results in 1 equation but 4 unknowns (filter states s1 to s4), hence no ...
When studying neural networks from Neural Networks and Learning Machines, by Simon Haykin, the author highlights the close similarity between of adaptive filtering and neural networks. From a scalar-valued signal, if we put a tapped delay input along with an ADALINE (Adaptive Linear Neuron), do we have an adaptive filt...
I am confused about linear time-varying system. For a time varying system, the output is given by \begin{align} y(t)=\int x(\tau) h_{\tau}(t) d\tau, \end{align} where $ h_{\tau}(t)$ is the output of the system given the input $\delta(t-\tau)$. By manipulating the integration above, we have \begin{align} y(t)=\int x(t-\...
I've been working on solving Poisson problem using CNN model (you can ignore Poisson problem part if you not familiar and jump to image processing/CNN part). More specific, I am solving electric potential problem ($\phi$) by CNN algorithm with particle distribution ($\rho$) as the input in cartesian coordinate, similar...
SVD can be used for image compression: Tim Baumann - Image Compression with Singular Value Decomposition. Balabit Unsupervised - Image compression using singular value decomposition. I think there are many image compression algorithms now. My question is: Is the SVD based image compression popular for actual applica...
I am learning signal processing and would like to know what's the python equivalent of matlab's cwtfilterbank. I looked through pywt but did not find anything. Some background information: I would like to process audio files with a CWT filter bank, where the wavelet filters are logarithmically spaced bandpass filters. ...
I am trying to compress some binary images i.e. images with bit depth equal to 1. For JPEG-XR encoder I used jxrlib but it is not able to compress the image. I am using specified format bmp. Even with some of the online compressors present I am not able to compress my bmp file to jxr. I have even tried with tiff files...
First of all if this isnt the place to post these types of exercise please point out where do i post it and thank you in advance We are asked in this exercise to print out the SNR vs BER graphs for every given mod, these mods being BPSK,QPSK,8QAM,16QAM We are asked to Define the classes and the methods and i belive ive...
First of all, DSP is not my expert area. I didn't take any DSP course and don't know DSP in detail but I have a question regarding Fourier Transform confusing me. Well, suppose that we wanted to take the fourier transform of $\cos(2\pi\cdot 3x)$ and integrated over a finite interval $[-a, a]$, in that case, we would en...
I want to record, modulate and transmit three different voice signals. I will be doing this in MATLAB. Please can you help me figure out how can I do this? Or give me a block diagram that can show what exactly I need to do.
Given input images x and h of same size, greyscale 2D and potentially complex, we seek circular cross_correlate_2d(x, h) done with fft (so no padding) in image processing context. I offer 1000 bounty to an answer that shows it's not exactly as described here. If it is exactly as described, then answers may state as suc...
Suppose we have a system $S$.If I know the impulse response $h(t)$ I can predict the output for any input $x(t)$ if a system is linear and time invariant.However how do we prove that statement?
What does it mean, when my system has controllability matrix that is full rank and identity matrix? Having full rank means that the system is controllable, but does the Identity feature of my controllability matrix say anything specific about my system?
How to replicate scipy.signal.correlate2d(x, h) with arbitrarily sized x and h? ifft2(fft2(x) * conj(fft2(h))) gives bad results. I've read related Q&As but they either do circular cross-correlation, or do convolution which doesn't easily translate.
I am trying to detect anomalies (interferences for example) given signal spectrum data in discrete form (frequency vs power spectral density) in python or matlab. I am trying to do this in 2 scenarios. Scenario-1: The expected normal (without interference) signal spectrum shape is given. In this case I tried to find an...
I have some doubt about the following exercise. Let's consider the signal $X(t)= \operatorname{rect}\left(\frac{t}{2A}\right) $ , where $A$ is a discrete random variable which can assume one value between ${2,4,8}$ with equal probability and $$\operatorname{rect}\left(\frac{t}{2A}\right)= \begin{cases} 1 , & > \text{i...
I have a signal that is $x(t)=e^{-3/t^2}+2\cos(3\pi t^3)$ and I'm trying to find its proper value for fs. But I couldn't find a way on how to do it. I'm working on an interval $0<=t<=2$ and $x(t)$ in this interval is between $-2$ and $3$.
I am struggling with a task I thought would be a good exercise. I am tasked with calculating the bandwidth of the equation below and I'm struggling. I managed to find and understand why $\text{sinc}(t)=1$. However, it's the other function that I can't figure out. $$x(t) = \left[\frac32 + \frac3{10} \sin(2π t) + \sin\l...