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I have a Problem with the Paper "Topological Structural Analysis of Digitized Binary Images by Border Following" by Suzuki and Abe.
In the Chapter "Appendix I: The Formal Description of Algorithm 1" I think I found an issue:
Consider this input image:
We start by finding the first not zero pixel and decide that it is ... |
I'm using a fairly standard setup after the ofdm flowgraph examples provided by gnuradio.
When I look at the absolute value of the received complex symbols in the first sync word (and every other symbol in the header), it looks something like this:
The location of this peak varies slightly, but is typically around sub... |
I am trying to investigate the performance of OFDM system practically, I am using sampling rate $1 GSps$ giving bandwidth $B = 0.5 GHz$. What I find is when setting the number of subcarriers $N = 64 $ or $N = 128$; the performance is OK. However, when increasing it into $N = 1024$; the performance becomes worse.
Why do... |
If it is given that PSD of a random process is bandlimited to frequency $f_B$, then can we claim that any sample path of the random process is also bandlimited to $f_B$?
Intuitively, I always thought of PSD as spectrum of the random signal in some sense but mathematically, the relationship between PSD of the random pro... |
Sound waves can cause vibration of the particles/objects that are scattering/reflecting/emitting the light. Since vibration is spatial displacement, it causes a phase shift by affecting the value of "$x$" from the light (electromagnetic wave) phase formula: $\phi = k x - \omega t$.
Is it true that the louder the sound... |
I have been working on a text-dependent speaker verification project. I have used mel-frequency cepstrum coefficients(MFCC) for the project. As far as I know, MFCC coefficients depend on the vocal characteristics of the speaker, not what he/she says. What features can be extracted from speech that rely both on the spea... |
I am familiar with the combination of the FIR filter with Hilbert transform, leading to the analytic signal:
$$y_a(t)=y(t)+jH(y(t))=x(t)*h(t)+jH(x(t)*h(t))=x(t)*\left(h(t)+jH\left(h(t)\right)\right)$$
which means that we can define a new complex filter $$g(t)=h(t)+jH\left(h(t)\right)$$ that will represent both filterin... |
I have a digital notch filter (band-stop) implemented in C. It works very well. The issue is that varying the filter coefficients in order to track a noise signal that varies in frequency results in a transient response that can be more noisy than the incoming signal. We refer to this as "shot" noise and it appears thi... |
I've recently asked a similar question aiming at the same problem on
Math.SE, for which I've set a bounty.
Problem.
I have a radio calibration transmitter with known fixed location $\langle t_x, t_y, t_z\rangle$ embedded in a medium whose index of refraction varies as a known function of depth. That function is of th... |
The article here says that
This 20-micropascal reference was selected because it was the quietest
sound pressure level that a group of normal hearing test subjects
could detect.
which makes me think it should be the smallest sound that a normal human ear should be able to detect.
However, if we look at the db HL scal... |
I have to extract a particular frequency range from an audio signal. For that I am using an FIR filter of order 50 and it is written in MATLAB as below:
[input fs nbits]=wavread('female_speech.wav');
input=input(:,1)';
len=length(input);
order=50;
b1=fir1(order,[0.1 0.2]);
input_buffer=zeros(1,len);%%variable used t... |
In the case of CRS collision in LTE, is the serving cell and the neighbouring cell always PRB aligned? For example, if they are PRB aligned, port0 will always collide with port0, otherwise port0 will possibly collide with port1. The details of CRS collision is specified in which specifications?
|
So I am doing some experiments on comparing MMSE and ZF performance. Here is the description of my problem setup:
Consider a pulse response $p_{tilde}$ which is equal to:
[7.41283693728174e-06
-6.84231291007695e-05
-0.0443418886383774
-0.0151503353629903
-0.0272769624198551
-0.0177892552560615
-0.0203350692581077
-0.01... |
In presence of a process that produces a 1D signal that is composed by an initial "transient" regime followed by a decay that we look forward to use for the estimation of a model's parameters using an exponential function.
signal(t)
What's the standard way in the signal processing community to mark an approximate "begi... |
A transfer function can be classified as strictly proper, proper or improper depending on its relative degree, i.e. the difference between the degree of the polynomial in the denominator and the degree of the polynomial in the numerator. If a transfer is improper, it is said that the system it represents isn't causal. ... |
How would the LMS equalizer dimensions change for the MISO case?
LMS adaptive filters are typically described for equalizing a single input signal, $x(t)$. Can the LMS algorithm be modified in the MISO case to perform diversity combining when the transmitter has $N=1$ antennas, the receiver has $M>1$ antennas?
For the ... |
There are many times when the textbooks used by all university students like me make an introduction to the notion of sampling of a continuous-time signal with an impulse train as shown below.
Why do we use this approach? Impulse train is something theoretical and of course it cannot be achieved in real life due to it... |
I wrote some code to demonstrate the Fourier slice theorem and it's relation to the Radon transform. However the sampled FFT from the 2D FFT and the 1D FFT of the projection at the same angle don't match too closely.
The issue might come from two places:
Maybe the issue is undersampling and I should use zero padding fo... |
so I'm reading an academic paper about some state of art algorithm of VO using deep learning, and there are some "known" issues of classical computer vision approaches that the authors bring up which I less known with. I will be glad for some clarification and reading references.
The paper is: TartanVO
I bring here a f... |
I'm trying to remove the DC components from a signal in a static situation (there is no movement), so I'm using a high pass filter with a cut-off frequency of 0.1 Hz (since the DC components happen at 0 Hz). Is this true, and if not how to choose the correct cut-off frequency?
|
I know STFT is generally applied to non-stationary signals but I tried to apply it to a stationary signal to get a working knowledge.
I created a stationary signal composed of three frequencies as below:
x = 3*cos(2*pi*30*t + phi1) + 2*cos(2*pi*45*t - phi2) + 1*cos(2*pi*70*t + phi3);
I then performed STFT (Short-Time... |
my signal consists of data points that represent the (vibration signal) energy measured during process cycles of a machine. Due to the hydraulic power units emptying and filling again, some periodicity is introduced into the yielding signal of energy measurements.
I am searching for a way to remove this periodicity fro... |
Assume the following problem: A deterministic signal $X$ whose magnitude is known to satisfy $0 \leq \Vert X \Vert_2 < \Delta$ for some known constant $\Delta$ is transmitted through a Gaussian Channel with known standard deviation $\sigma$.
It is known that the ideal observer performance on the detection problem (dete... |
The transfer function
$$H_{LPF}=\frac{b_0}{1+a_1z^{-1}}$$
becomes a multiplier-less low-pass filter if $b_0=2^{-M}$ and $a_1=2^{-M}-1$. According to this paper, the cut-off frequency is then given by $f_c=\frac{f_s}{2^M - 1}$. I tried to verify that in a python script, but I get different results. In my case, the cut-o... |
I'm testing implementing a convolution by using an FFT way. Reference Link: Replicate MATLAB's `conv2()` in Frequency Domain
as you see, he uses fft2 & ifft2 and it works perfectly. But I want to use traditional FFT & IFFT for each col, row
My forward direction is ok. However, the inverse direction has problems that ma... |
I have understood convolution and it seems almost same as correlation mathematical expression wise. But in terms of signal processing,both are different operations.
The difference between convolution and cross-correlation from a signal-analysis point of view
I have read above answer and i some how understood through it... |
I am working on an application which consists of cross-roads and roads that connects them. In my design, I am using graph signal processing to estimate the importance of the roads which means edge weights in the Adjacency Matrix. I have the graph signal which consists of number of cars at every minute. I also know the ... |
I'm trying to implement the Butterworth filter as a high pass filter inside a function, where every time stip a measurement will be passed (inside this function I do have buffers for storing the data).
However, I'm using the Matlab built-in function butter to calculate the coefficient. The problem is the filtered signa... |
Python vs Matlab? Which one better for image processing?
The above link is for a question that was 2 years ago and was related to image processing
But what about signal processing these days?Python vs Matlab? Which one is better for signal processing especially audio signal processing?
|
I am trying to convert -13 dBm/ Hz into corresponding dBW/100 Hz? Can you please point me in the right direction.
Do I need to convert dBm to dbW by adding -30 first?
|
I am trying to compute the sampled Fourier Transform of a Continuous Time LTI system output. $x(t)$ is the input of LTI and $h(t)$ is the impulse res. $y(t)$ is the output. we know that
$$ y(t) = \int_0^\infty x(t_1)h(t-t_1) dt_1 $$
after sampling,
$$ y(nT_s) = \int_0^\infty x(t_1)h(nT_s-t_1) dt_1 $$
$$ Y(j\omega)=\s... |
SPL will be calculated as described below:
collect $N$ samples
do FFT for the $N$ samples collected: the signal is now in the frequency domain.
get magnitude of the FFT
get RMS value of the FFT
apply a time-weight filter to the RMS value
apply a frequency-weight filter to the RMS value
compute the SPL using the RMS va... |
Using SDR#, I saved the baseband I+jQ signal centered about a commercial FM station to a *.wav file. I want to write python code to demodulate this signal. I know that SDR# can read in that file and demodulate it, but I want to do it on my own. And no, this is not for a class ...
I assume that the I and Q signals co... |
Existing work
concentrated mostly on the problem of “matrix completion”
where one aims to recover a low-rank matrix from randomly
selected matrix elements. Their result covers this situation as a
special case. I got many confusion of this paper.
Let's begin with the notation I already get confused. $T$ is a linear spac... |
The slide below, borrowed from CMU, shows the typical interpretation of the "filterbank interpretation" of signal reconstruction from a Short Time Fourier Transform. As far as I can tell, the hop-size here is 1, i.e., there is no frame decimation to represent a realistic frame hop.
In that context, it makes sense: yo... |
I am trying to learn about frequency domain and I am using audacity software.
I selected a small YouTube video (link below):
https://www.youtube.com/watch?v=rGHrKkieqCY&ab_channel=PANKAJSAO
I converted it to .wav format using the link below:
https://youtubeto.org/en/youtube-wav.html
When I imported this .wav file into ... |
I want to analyze the auto-correlation of a received power signal that I captured. Unfortunately, I cannot publish the data but I found the same problem arises for a random walk, that's why I used the random walk in the code snippet below.
The problem is that the progression scales with the number of input samples, i.e... |
Cross-correlation is usually used to estimate e.g. lags between signals that are detected by different sensors at different times. In my case, the situation is slightly different in that I measure signals with sensors at different grid points and I want to recover geometric information about the grid. To be more specif... |
I perform real experimental tests for optical communications using DCO-OFDM. The transmitted signal is uploaded into an AWG (arbitrary waveform generator), then bias (2.5 V) is added before adopting the signal into the source light which includes a lens. The received signal is also focused by another lens and passed th... |
Given a harmonic signal of a certain frequency and amplitude, what is the optimal way of "turning on" this signal from zero to full amplitude within a given time window under the constraint to minimize bandwidth (lowest possible frequency cutoff)?
A mathematical formulation or sketch of the solution would be much appre... |
Given an arbitrary configuration of carriers having bandwidths that belong to the 3GPP standard (5, 10, 15, 20, 25, 30, 35, 40, 50, ..., 100 MHz), is there a method/rule-of-thumb to estimate the distance between successive maxima of the magnitude of the resulting time domain OFDM signal?
This is what I have tried so fa... |
Let's say I have an electronic system that's taking a measurement. It provides a simple bipolar excitation current to a resistive load (bipolar square wave so as to cancel out thermal emf), puts it through an analog front end and some anti-aliasing, into an ADC.
What I'm thinking is if during calibration we can just ta... |
Suppose we have two OFDM based systems A & B.
System A has $N$ subcarriers and bandwidth $W$, and the number of samples in the CP is 10% of $N$.
System B has $2N$ subcarriers, $2W$ bandwidth and the number of samples in the CP is 5% of $2N$.
Which system has higher capacity?
|
I've noticed that Stack Overflow uses lh3.googleusercontent.com to store and serve the profile images of each user. One of the features this website has is that you can easily request a resized version of the image by using the sz=XXpx url parameter in the image url. Google will resize the image at minimal loss of qual... |
I have collected data from an oscillating system where angle values change over time and I am interested in identifying frequencies in my oscillations.
Given that my sampling frequency is $f_s = 1$ GHz (i.e sampling period of 1 nanosecond), I have provided $f_s$ to the software I am using in order to obtain the DFT bec... |
Hello all hope everyone is doing fine.
I have a vision related problem where I want to calibrate my camera using a 3d object such as rubik's cube with known dimensions. I have a cube with 60mm length, width and height. I want to find the projection matrix "P". Which gives me the relation between 2D and 3D coordinates i... |
I have designed an algorithm for classification and I need to add a suitability or compatibility test for accepting a signal as an input. Basically, not all signals are suitable for processing by my algorithm and I need to find out a way to avoid processing signals which do not apply to my conditions.
My conditions: I ... |
I have a box with four microphones on its sides: one on top, one on front and two on left and right sides.
Is it possible to calculate a vector from the center of the box to the source of sound in real time?
|
I'm validating a C++ oQPSK receiver. The 5Mbps oQPSK signal is generated by a satellite emulator. The receiver structure goes as follows:
SDR --> AGC --> Costa's Loop --> Delay --> Polyphase Symbol Sync (Maximimum Likelihood) --> Constellation demap
I have tested the Costas and the polyphase sync with BPSK and QPSK and... |
I have measured data of a vibration with chirp input were I suspect unwanted shock events occurred.
I therefore thought about computing the spectrogram of my time histories to check the frequency content.
I am however not fully sure about what to look for. My idea would be to look for a straight line at a given step in... |
I have a situation where a linear array of sensors monitoring near field sources has had one of the sensors bumped so that the sensor position is off by a smallish amount and is not in its original geometric position. The following image is intended to aid in understanding where $S_e$ represent the sensors with $S_{e4}... |
I'm trying to understand the derivation of a Wiener filter for a vector-valued signal, found in the paper "Kalman filtering in non-Gaussian model errors: a new perspective," IEEE Signal Processing Magazine, May 2022. This is the section headed "Smoothing Wiener filter" on p. 108.
The author starts out with a definition... |
I noticed a mixed usage of AoA (Angle Of Arrival) and DoA (Direction Of Arrival) and was quite confused about the exact distinction between AoA and DoA with respect to wireless signals. Reference to any document for further understanding will be very helpful. I wish to clarify the below queries as part of this page and... |
I need to find the phase difference between two sinusoidal signals with the same frequency (23.5MHz) but not in same phase.
Test case:
Generate two sinusoidal signals of length 8 with different phases (signal 1 has 0degree phase and signal 2 has 90 degree phase) from a Xilinx DDS compiler IP core.
The two signals are ... |
If $x[n]$ is length 4 sequence, with DFT $T.F\{x[n]\} = X[k]$ and $y[n]$ is a length 8 sequence obtained from $x[n]$ by padding with 4 zeros:
$$y = [x[n],0,0,0,0]$$
For $k$ even, we have $X[k] = Y[k]$
What is the DFT of $y[n]$ in terms of $X[k]$ if $k$ is odd?
|
I'm using numpy.fft in python to compute Fast Fourier Transforms. In particular, I'm using rfft as I have a real signal and don't need negative frequencies.
My question is: my signal has 184320 points currently and encompasses multiple periods. Should I be truncating the signal after a certain number of periods for bes... |
I am trying to calculate the power spectral density step by step using numpy FFT and comparing it to the PSD calculated by pylab PSD. Difference b/w plots of both methods can be visualized from in figures below.
#*************************PSD using pylab PSD**************************
import math
from rtlsdr import RtlSd... |
I'm making a phonographic filter or simulator to make one of my songs to sound as if it was recorded on a phonographic cylinder.
Two important things about phonographic recoding is that sound gets more treble when it's louder and also that the recording of the wave is made top to bottom so the negative part of the wave... |
In the DFT we sample one period of the spectrum in the frequency domain. What is the difference between having an odd or an even number of samples?
We know that DFT is just a sampled version of the DTFT.
My thoughts are that if we use an odd number of samples of the DTFT in our DFT, the frequency content will go from... |
I'm trying to match the amplitudes of a signal before performing DFT and after. So, let's consider a 64-point sine signal with amplitude of $1$:
The DFT of such a signal will give us the amplitude (corresponding to the original signal frequency) equal to $32$, which comes from the DFT formula:
$$X[k]=\sum_{n=0}^{N-1} ... |
I am trying to implement the following line of MATLAB code:
[pxx, f] = pwelch(data,[],[],4096,F_S)
in Python. Knowing that MATLAB using a Hamming window type that uses 8 segment with 50% overlap, my attempt was
freq, PSD = signal.welch(data, F_S, window='hamming', nperseg=None, noverlap=None, nfft=4096)
However, I rec... |
Say that you have a reservoir of fixed capacity, $V_{max} [m^3]$ and some level of fill $V [m^3]$.
Liquid is flowing in a stochastic, but observable rate:
$$x(t) [m^3\cdot s^{-1}]$$
You are given a valve that can drain the pool at any rate:
$$y(t) [m^3\cdot s^{-1}]$$
I want the pool to never overflow or underflow. At t... |
I'm trying to find the phase of a cosine function with the FFT but I don't understand the results from the plots. I did two separate examples for two phases $\phi$: $\phi_1 = 0$ and $\phi_2 = \pi/3$. They end up with nearly the same results but with different values for the phase. I would appreciate any help.
$\phi = 0... |
I found this free DSP book by Rutgers University professor Sophocles J. Orfanidis. The book is called Introduction to Signal Processing and the link contains many different implementations of common DSP functions in both MATLAB and C.
My intention is to eventually implement this Butterworth BPF in C for an embedded pla... |
Consider a wireless communication system having $t$ transmitting antennas and $r$ receiving antennas. Then, the received signal is given by
$y = \mathbb{H}x+n \tag{1}$
where $\mathbb{H}$ is a $r \times t$ complex matrix, $x$ is a transmitted vector such that $x \in \mathcal{C}^t$, $y$ is a received vector such that $y ... |
I need to perform the following integral ($i^2 = -1$) (with a fixed value of $m$):
$
\int_{\phi=0}^{\phi=2\pi} \psi(\phi) e^{-im \phi} \hspace{0.7mm} d\phi
$
for all values of $m$ in $\{-N_{\theta}, -N_{\theta}+1, ..., 0, 1, ..., N_{\theta}-1, N_{\theta} \}$, that would be $2N_{\theta} + 1$ naive calculations of the a... |
In this DCT video isn't $X^c$ the input image and what does the superscript $c$ signify?
Why do they call $X^c$ "DCT coefficients " although it's the input image?
|
I'm fairly new to signal processing and I'm building a beat game using frequency, RMS and hopefully onset detection alongside numerous other functions.
However the problem is I don't want to use an FFT - multiple reasons for this, for example I'd want to write my own which I imagine would take far too long and also I l... |
I'm doing some feature extraction on audio signals.
$M$ being a mel filterbank matrix, and $S$ being the spectrogram (extracted from the Short Time Fourier Transform of my audio signal), we can compute:
The Log Mel Spectrogram: $X_P = \log(M \times|S|)$
The Log Mel Power Spectrogram: $X_{PS} = \log(M \times|S|^2)$
Qu... |
I've been trying to confirm the process for the Cooley-Tukey approach for FFTs. Currently I have a function that generates random input data for a matrix with $n_1$ rows and $n_2$ columns.
The result of my Cooley-Tukey approach is compared with a 1D FFT using the $\text{Scipy}$ package. My current approach is as follo... |
I am working with data from a TeraFlash THz Spectrometer from Toptica, which gives me the pulse in the time domain and the spectrum in the frequency domain
I have been experimenting with the FFT to get to the spectrum myself and I see some discrepancies compared to the spectrum from the software. The manufacturer told ... |
I want to convert the in situ reflectance into satellite reflectance using SRF (Spectral response function). I have downloaded the SRF file but I find that the bands wavelengths are different than in the Sentinel 2 handbook.
For example, in the handbook, the wavelength of band 1 is $443\,\mathrm{nm}$, while in the SRF,... |
I need to implement a basic 6th order Butterworth lowpass filter on a DSP that sampled data at 50kHz. I calculate the coefficients using matlab for a filter with the following step response:
$$\begin{align}
y[n] {} = & \frac{1}{a_0}(b_0 x[n] + b_1 x[n-1] + \cdots + b_P x[n-P] \\
& {} - a_1 y[n-1] - a_2 y[n-2] - ... |
Newbie struggling to understand why a larger chunk of signal bandwidth is needed to carry more information. For example, why do you need more bandwidth to carry video as opposed to voice or CW? Apparently missing some fundamental concept here. Have a sneaking suspicion this will lead to information theory. Thanks in a... |
I'm trying to obtain the envelope of an audio signal by using the Hilbert method. My code generates the analytical signal in the same way as scipy.signal Hilbert() function does (I basically copied the scipy source code in troubleshooting this issue). As can be seen from the pictures below the envelope is not obtained ... |
I am currently dabbeling with programming a real-time audio software and I already got a simple pipeline where notes can be played and released and each note has its own envelope so that the amplitude of each note signal begins and ends at 0 smoothly.
However, I am currently running into the problem that if 2 notes ove... |
I have done a base-band modeling of a 5G NR OFDM (CP-OFDM/ DFT-s-OFDM) transmission system. And I have added a DC leakage as an RF impairment and transmitted the signal through a PA (Power Amplifier).
However, at the receiver side when I try to do DC leakage correction using MATLAB's dsp.DCBlocker, some of my RB alloca... |
I am new to signal processing. Currently, I am reading a paper regarding musical expression. The paper calculates brightness, and I want to reproduce their way of computing it. I was wondering if there's any way to calculate harmonic series from audio data, because I want to calculate the Brightness.
This is from a pap... |
The KL divergence between two real-valued Gaussian distributions with means $\mu_1$ and $\mu_2$ and common variance $\sigma^2$ is well known to be:
$$
D_{\text{KL}}\left(\mathcal{N}(\mu_1, \sigma^2) \parallel \mathcal{N}(\mu_2, \sigma^2)\right) = \frac{\lVert \mu_2 - \mu_1 \rVert_2^2}{2\sigma^2}.
$$
Assume now two circ... |
I don't understand the unit of the pulse shown below.
Hz by definition is 1/sec.
If it's in time domain then BW is in seconds.
If it's in frequency domain its Hz.
What is the logical meaning of the combination between the two?
1/hz/sec?
|
I want to implement a FIR highpass filter for acoustic signals.
I generate the FIR using Python's SciPy code:
import numpy as np
import scipy
# Parameters
nfft = 512 # FFT length
cutoff = 100 # Cutoff in Hz
fs = 16000 # Sampling rate in Hz
# Frequency related sizes
kbins = int(nfft//2 + 1)
# FIR filter via firwin
nu... |
Laplace domain is also known as "s domain".
Is there any difference between "s domain" and "frequency domain"? Can we use both terms interchangeably?
If we want to convert a time domain signal to frequency domain, can we use Laplace transform?
|
Suppose $x$ is sampled from standard 0-centered Gaussian in d-dimensions, and I apply the following iteration.
$$w \leftarrow w-\alpha x \langle w, x \rangle \tag 1$$
What is largest $\alpha$ such that this iteration converges almost surely for any starting $w$? Couple of textbooks I checked solve for $\alpha$ such tha... |
I'm new here and unfortunately I have really no knowledge at all about digital signal processing so my apologies if this is rather a stupid question or a duplicate. I have some audio signal which consists of beep tones. I need to figure out how many beep tones it consists of.
I have the following signal:
But if you lo... |
As per my current understanding, the ADC datasheets for modern high speed ADCs give the noise level measurements in terms of Noise Spectral Density (NSD) in unit of dBFS/Hz. Now to convert that value to the noise floor first the dBFS/Hz value is converted to dBm/Hz by using the value of full scale power in dBm for the ... |
I did a basic google search but, I wasn't really satisfied with the output info.
I would love to know what exactly we can do with the extracted wavelet.
What kind of information extracted wavelet can contain ?
|
I would like to sample an analog 5kHz signal with a 50MHz sample rate. I want to do this with python so I can see if this sort of sample rate is enough for my needs. But I couldn't wrap my head around the problem.
I am so used to Simulink where I can have a continuous signal which I can move to the workspace and sample... |
There is a dataset of $N$ elements, each represented by $K$ bits. Now due to hardware limitations to reduce memory for storage, they have to be reprocessed into $K'$ bits each, and $K'<K$. What is the best coding strategy?
|
How do I determine the wavelength of a noise signal like the one below? I find it easy to understand for sine waves, but it gets tricky for me when the signal is more complex like a noise signal.
If I understand correctly a period is defined as the time it takes for the signal to cycle from one amplitude back to the sa... |
Is it possible to implement realtime DSP algorithms on PC using a NI PC acquisition card like NI PCI-4461?
In A New FXLMS Algorithm with Off-line and On-line Secondary-Path Modeling Scheme for Active Noise Control of Power Transformers, they run a real time ANC algorithm using a PC.
But my thinking is: PC is not real t... |
I need to perform soft Viterbi decoding on a Manchester-encoded BPSK/QPSK signals. To that end, I need a floating point Manchester decoder. Most of the implementations I have seen operate on hard bits. There are of course floating point implementations that monitor "branch weight" like this one. This class of decoders ... |
I am learning signal processing. For a hobby project I wanted to transmit and record a simple ASK signal via my soundcard, to see if I can write code to recover the sent data. After playback and recording, the recorded signal isn't even close to the original. I thought that the problem was my code, but I could not fix ... |
I am trying to generate a baseband (BB) system model with LO leakage impairments and correction.
The channel bandwidth $BW = 20$ MHz, Subcarrier Spacing $= 15$ kHz, oversampling rate = $4$ (just oversampling the generated signal in time domain). Time duration = $10$ ms. Sampling Rate $F_s$= 122.88 MHz
Generate Signal i... |
I teach a Digital Signal Processing Lab to Electrical Engineering undergraduate level students. We have most Labs on MATLAB and some Labs on a dsp kit tms320c6713. I am planning to incorporate topics and contents related to Machine Learning, and I am wondering if it's ok? Anyone else doing the same (teaching Machine Le... |
how to avoid negative frequencies that can be obtained from instantaneous frequency estimation using Hilbert transform?
Here is what I am doing:
compute analytic signal, X = hilbert(x);
from analytic signal, unwrap the instantaneous phase
calculate instantaneous frequency from derivation (np.diff) of instantaneous pha... |
I am currently implementing an acoustic communication system in Matlab. For synchronization purposes, every transmission contains an upchirp preamble. The receiver uses cross-correlation to find the preamble in a captured audio file and decodes the transmitted data afterwards. This works well in case the recording actu... |
I have a question regarding the concept of carrier power, output power and power relative to carrier (dBc). Suppose I have a carrier frequency of 4Hz and signal bandwidth = 4 Hz. The signal span is from [2Hz 6Hz].
My indexing starts at 0 for simplicity.
I have 7 samples with sampling frequency 7 Hz such that:
x(n)=[1+... |
I have two 2D arrays: $A$ is the original matrix that contains only 0s and 1s, and $B$ is the convolved matrix. I know the size of the convolution kernel $K$. Generally, it follows $B = A*K + S$, where $S$ is the additional noise with Gaussian distribution.
I want to estimate the kernel based on $A$ and $B$, how can I ... |
I am about to start a new engineering job in the fields of signal processing and machine learning. I have a bachelor's degree in electrical engineering, but for the last two years I have been doing a master's in theoretical physics, so I have been away from the field of signal processing for a while.
In by bachelor's I... |
I want to invert a $6 \times 6$ complex matrix on the ARM Cortex M4F processor. I have the C code to invert a real matrix using the CMSIS library. Has anyone written a similar C code for complex matrices? Can the CMSIS library be used to do complex matrix multiplications?
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