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The "Fundamentals of Speech Recognition" text by Rabiner, mentions three approaches for Speech (Endpoints) detection - Explicit, Implicit, and Hybrid approaches. I am studying the block diagram for the explicit approach.
Steps involved in explicit approach are Speech signal -> feature extraction -> Speech Detection -> ... |
I'm trying to implement the code in this tutorial in MATLAB using MATLAB's ifft2() function instead of the algorithm used in the shader (looks to be some type of Radix-2 algorithm). It is used to compute the surface of an ocean using compute shaders in Unreal Engine.
I've confirmed I'm computing FFTGrid.DZ correctly, b... |
We have an adrv9002 that has built in DAC and ADC that takes and gives IQ samples. I did a test to verify functionality but don't understand what I am seeing.
Take a RF tone from a signal generator and feed it to the receiver.
Take this received IQ, and just send the I data to DAC port for output while making the Qs z... |
The idea behind this question is for misalignment fault we say vibration signals across the misalignment are out of phase.
If the signals are just sinusoidal signals with one frequency calculating phase difference can be calculated via multiple ways like cross correlation,fft. But what if the signals has multiple frequ... |
I am working on a personal project that maps bass notes to colors in real-time. However, I'm encountering some issues with oscillations in my frequency bins.
I visualized my frequency bins to illustrate the issue:
30 Hz tone
45 Hz tone
75 Hz tone
Here is an outline of my signal processing chain:
Capture 2205 samples... |
Text from "Fundamentals of Speech Recognition (Rabiner)".
A complex pole of the LPC model spectrum can be expressed as $Z_i =
> r_i \cdot e^{j \omega_i} $, where $r_i$ and $\omega_i$ are the radius
and frequency of the complex pole, respectively.
The bandwidth of the complex pole is related to the radius by,
$ B_i = -... |
I'm looking at two different books written by Oppenheim. In Discrete-Time Signal Processing (source 1) he defines the DFS to be:
where, $W_N=e^{-j(2\pi/N)kn}$ , while in Signals and Systems (source 2) he defines the DTFS to be:
Can someone explain why there is a difference? It appears to me that the only difference i... |
I can't find an axis diagram for the movesense sensor, nor does it say on the spec sheet on the website - Is the magnetometer in the NED or ENU reference frame? On SO there is an answer that doesn't differentiate between the accelerometer, gyro and magnetometer.
Or if we approach it as a black box, how do I tell what a... |
I am trying to do a kind of frequency compression on say 20 most energetic frequencies in a particular slice of a STFT. If w[i] is the frequency peak for a slice of STFT at a particular time, then I want to move w[i] to w[i/n] where n is arbitrary. To make sure invertability is still maintained I'm using a 2 sided STFT... |
I should sketch the signal below:
I think I should use delta dirac function sampling theorem but I don't know how in this special case
I know that $x(t)δ(t-t_{0})=x(t_{0})δ(t-t_{0})$ but I don't have any idea about $x(t)δ(at-t_{0})$
|
I'm doing quite a bit of the samething as I wrote in this post here :
Shifting frequency peaks such that w[i] moves to w[i/n] using STFT and IFFT
I'll quote from it here to present the issue I now face which is different from the one in the post :
I am trying to do a kind of frequency compression on say 20 most energet... |
I am trying to use the Kalman filter for my task:
During the time, I receive data from different sensors. The state of the model may change over time according to the Const Velocity model, or the Const Acceleration model. The data is a true value with Gaussian noise. The main problem is that the Gaussian noise of the s... |
I have a replicated a Direct sequence spread spectrum link (schematics taken from ARRL SS handbook) and would like to measure performance characteristics such as process gain.
I have 2 questions.
I am using LSFR using 74164 for PN code generation which is clocked at 2MHz. Is it reasonable to say that the bit rate of P... |
I am doing an undergrad assignment and one of the exercises is as follows:
Given the following signal $x(t)$:
and another signal $y(t)$ which was transformed from $x(t)$:
Describe $y(t)$ in function of $x(t)$ using signal transformations (compression/expansion, time offset, etc).
My strategy:
I know there are at leas... |
I'm interested in a side learning project in decoding the images recorded as sounds in Voyager's golden discs (https://voyager.jpl.nasa.gov/golden-record/). The instructions to be able to decode these images were represented on the record's cover as you can see in the following explanation of the diagram:
Specifically... |
We currently test bearings on a third-party test machine, which uses a velocimeter and reports through a software front end that outputs a low mid and high band spectrum result, and RMS, Peak and Crest values. This machine also outputs the audio to a speaker, allowing us to listen for structure borne noise which may fa... |
A $17$-level $\left\{ -4,-3.5,-3,\ldots,0, 0.5, 1,\ldots, 3.5, 4 \right\}$ digital signal coming at $125 \, \textrm{MHz}$ is passed through an FIR filter ($10$ taps) operating at $1250 \, \textrm{MHz}$ having coefficients as $ 0.25^{k}$ where $k = 1$ to $10$.
|
In the field of signal processing, when a spectral peak is fitted with a Gaussian, is there a name for the quantity obtained by multiplying amplitude with frequency and dividing by the standard deviation (sigma) of the Gaussian?
It's meant to be a measure of oscillation strength.
I think I saw that quantity named somet... |
I have some questions about the M&M implementation of GNU Radio that uses complex samples.
I started reading the implementation for real values:
while (oo < noutput_items && ii < ni) {
// produce output sample
out[oo] = d_interp.interpolate(&in[ii], d_mu);
mm_val = slice(d_last_sample) * out[oo] - slice... |
I have been working on restoring some old stereographic images. This means I have a Left and Right image of effectively the same thing with different elements of wear on the image for each side. I had the idea that this might give some boosts in terms of using image comparison for denoising - but given the Left and Rig... |
I want to apply the Friis equation of free space path loss on OFDM signal. Do I multiply signal by $\frac{c}{4\pi f_cd}$ before IFFT or after?
|
Could anyone lead me on the right path on where I could get more information in regards to a narrowband Gaussian signal can be modeled as:
$$
X(t)\cos(2\pi f t)+Y(t)\sin(2\pi f t)
$$
where $X(t)$ and $Y(t)$ are the in phase and quadrature phase noise amplitudes.
|
I am trying to make a feedback howling detection program for an audio DSP chip
I made a very crude one by counting zero crossings, doing something like this:
for (n = 1; n < NUM_OF_SAMPLES; n++){
if( (sample_array_int32[n-1] < 0 && sample_array_int32[n] >= 0) || (sample_array_int32[n-1] >= 0 && sample_array_int32[n]... |
I have data captured by a wireless sensor that is noisy. It randomly jumps in value frequently, and I want to know what this signal will look like without these jumps. I am looking for an elegant signal processing technique to do this, if one exists. Below is the time-series signal:
I initially thought to do a Fourier... |
This paper https://ieeexplore.ieee.org/document/10051908 tells that we can implement polyphase fft even when data length is less than filter coefficients length.
I am confused how can we implement the polyphase fft when i have limited data length and filter coefficients greater than data length as we need to achieve be... |
The normalized factor for windowing, is it $(\sum w[n])^2$ or $\sum w^2[n]$?
I try to reproduce the pwelch function, if I set the normalized factor to $(\sum w[n])^2$, I get the same result as MATLAB built-in pwelch function but, if I set the normalized factor to the other one which is introduced in the papers, the res... |
Following this link, we can see that the magnitude of the $n^{th}$ harmonic of a sawtooth wave is equal to:
$$
V_n = \frac{Vpp}{(n*\pi)}
$$
Converting that Matlab code (the code at the end of the page, not the first one) to Python, I got this:
from scipy import signal
from scipy.fft import fft, fftfreq
import numpy as ... |
I loaded a 256x256 greyscale image as an array, A.
Then I Singular Value Decomposed it:
$$
A = U \Sigma V^T
$$
Next I set all singular values except the 64 first to 0.
Then I multiplied together the SVD matrices again.
The result was a pretty good low rank approximation to the original image.
Should I have subtracted t... |
I need your help. Unfortunately I don't have background in signal processing, but have a problem in this field. I'm helping a friend with an interactive art project. It has parts that move based on rotation and periodic manner on predefined path subject to tolerances in the mechanics (and noise in the hall sensor to me... |
Sometimes I see papers that average the log of PSDs (i.e., in dB space), instead of the PSDs themselves. So they are doing a geometric averages. I found a paper that shows that geometric mean has some advantages.
When one does Welch method of PSD estimation, one use arithmetic mean. Would it make any sense to do a geom... |
For a dynamic model I am looking at the appropriate input to analyse the performance under the influence of floor vibrations.
The norm is an ISO VC-C floor spectrum as displayed in the image. However, I am stuck in the derivation of going from a rms velocity to a (displacement) noise signal in the time domain. Any help... |
I am working in a problem where for many stored signals (hundreds of thousands), I need to compute the slew-limited version of each signal. The slew-limited version will start with the same initial value as the original signal and try to follow the original signal, but it's rate of change is limited. An inefficient alg... |
I have had good success using the Distance Transform and Chamfer Matching to locate a template in an edge image. Is there a recommended technique to then find the actual edges that match the template?
For example, in the image below, the template is shown in blue at the location found by chamfer matching. The image edg... |
I am getting position and velocity measurements out of a GPS sensor and I want to filter these data, so I can have a better, less noisy, estimation of the true measurements. I thought of doing this using a Wiener Filter.
Doing a little research about how Wiener filters work, I found that we have to calculate the Wiener... |
I am using GNSS to see the position of the satellite that is sending the gps messages. When using GNSS, the decoder only gives me the coordinates for the receiver at the given time. What I want is to see the original gps messages with time and position of the transmitter (satellite) instead of the receivers position at... |
I want to extract the text from this image. I'm a newbie in opencv. I've tried various opencv codes across various questions, but none is working for me.
How can I extract text from this? Or maybe remove grid lines & that circular bulge in the middle so that image is straightened & then I can extract the text.
Any help... |
The Dynamic time warping is applied for time normalization. As shown in the diagram, two different signals with $Tx$ and $Ty$ time instants, are time-normalized to have $T$ time instants. $\phi$ is the warping function.
A path P is represented as having increments $p_i$, and $q_i$ along their respective axis. These inc... |
The signal quantization consists in assigning to the different values of the sampled signal a value among the range of the $2^p$ values allowed by the $p$ bits used.
As the different values of the sampled signal can be represented by coordinates (sample value and a sampling time), I have a question:
How are the samplin... |
I have to combine three images that are inside one single image into one output image in Python. This has to be done using Gamma corrections, Histogram
equalization, filters convolutions (gradient, median, bilateral, sharpening…
etc), Fourier transform and manipulations in frequency domain.
I know it sounds confusing, ... |
I have a buffer of shape (n_samples, n_channels) in which new samples are coming in at regular intervals and old samples are discarded. In this example, the buffer array is called data and the array of new samples coming is called new_data.
This buffer and all the new samples are filtered with IIR causal filters, but d... |
I was wondering about the influence of phase discontinuities on a power spectrum density calculated with f.e. Welch's method. I know you shouldn't have any phase discontinuities in your signal, but I'm actually not so sure why.
For my own interpretation I made a signal in Matlab:
fs=500;
t=0:1/fs:10;
x= sin(2*pi*100*t)... |
I am working on preprocessing fiber photometry data. The data collection involves inserting fibers into an animal, where the reflection of specific wavelengths of light is captured. We collect two types of signals: an experimental signal (which contains both the true signal and noise) and a control signal (which contai... |
I've developed an UKF for a system, whose dynamics change slowly over time. The state & measurement equations are quadratic and linear equations fitted to experimental data in the following form:
$$ \begin{aligned}
x &= p_1m^2 + p_2m + p_3 \\
y &= q_1x + q2
\end{aligned} \tag 1$$
I then converted my state equat... |
My goal is to produce a new random noise image from two already existing noise images. For that I take the absolute values of one image and the angles (phases) from another image in Fourier Space and combine them to create a new image. I notice, that the back-transformed new image shows a more Gaussian shaped distribut... |
I have to apply the Gram Schmidt algorithm on discrete time series to make them orthogonal and normalized.
This implies that I compute dot products, but it is of course not applicable for very long time series. I assume that this is something done frequently in signal processing but I failed to find any useful help wit... |
Seeking assistance with an optimized and well-documented implementation of the Parks-McClellan algorithm, I explored beyond the original FORTRAN code provided in A Computer Program for Designing Optimum FIR Linear Phase Digital Filter.
Instead, I discovered other available options such as an open-source C implementatio... |
I'm looking for information about how to filter a square wave. I don't want to filter it out, but extract it from the rest of the signal. Most filtering I've encountered usually only care about a limited segment of a spectrum (low-pass, high-pass, band-pass, band-stop). I have not seen anything about preserving the ... |
Suppose I have the following sum, where all the input variables are positive real numbers while the output is complex-valued data
$$a e^{i \left( x + y \right)} + a e^{i \left( x - y \right)} + a b e^{i \left( x + z \right)} \tag{1} $$
Here I'm assuming $x$ is running from zero upwards while all the other variables st... |
Consider the problem of transmitting just two SRRC pulses through an noiseless Channel. The two transmit bits are {+/-1}. In plots below, the individual transmit pulses are shown in black, while the actual transmit sequence is shown in blue. Now consider transmitting just two pulses.
Case 1 : {+1 , +1 }
The total ener... |
I have a BIDMC PPG signals. I'm trying to add accelerometer data as a noise to PPG signals and trying to denoise it using deep generative models like GANs, VAEs, Diffusion Models. So, what can be the best way to add noise so that the it wouldn't distort the signal so much that we lose all the original information?
|
I have implemented a modified linear prediction coding function like this:
def modified_lpc_process(s: NDArray, p: int, fs: int) -> NDArray:
a = lpc_coefficients(s, p)
a_hat = modified_pole_coeffs(a, fs=fs)
# a_hat = a # debug line
s_hat = numpy.zeros(len(s))
for n, sn in enumerate(s):
as_accum = 0
for k in ra... |
I needed help in plotting this in a paper, I have 2 signals:
x(t) = 4 Tri ((t + T) / 2T)
x(t) = sin((2 pi t)/ T).rect((t - 3T) / 2T)
I have added my solutions to these signals in the attached image, do they look somewhat correct, I'm just doubtful with the operations I performed !!
Thanks for any kind of infos on this ... |
Given the autocorrelation function ACF(n)=cos((2/30)n) for n= -5,-4,…0,1,…, 5 use the simplified Lim & Malik iterative algorithm to extend the ACF to 256 points. Note that the simplified version of the algorithm only indicates that you should correct the ACF and the inverse ACF. When implementing the algorithm use app... |
Does TDOA based localization give estimated local coordinates with both positive and negative range error.
Let me explain
When i obtain localized coordinates using TDOA of moving object, sometimes the estimated coordinates lies in front of the actual position (positive range error) and sometimes it falls behind the act... |
I have two signals, both recorded at the same sampling frequency, however one has a phase delay. I am currently playing around trying to figure out how to calculate the time delay between these two signals using the cross correlation method. I want to calculate the time delay between these two signals using windows, as... |
I have an ADC working with a sample rate of 960 Msps, my signal being located at the 3rd Nyquist region (I am undersampling). Input signal bandwidth is centered at 1200 MHz with a bandwidth of 120 MHz. Once sampled with 12-bit the same ADC uses digital down-conversion and decimates by 8. Number of bits increases from 1... |
Given the power spectral density of noise, if the values at different sampling points of the same noise are added, what is the power spectral density at this time? Is there any correlation between the noise at different sampling points?
|
I'm trying to apply a DWT with 3 composition levels and the following question arose when calculating the composition matrix.
The step I'm trying to follow is:
The DWT coefficientes are obtained from filtering operations and are divided in approximation ($cA$) and detail coefficients ($cD$).
Three Decomposition Level o... |
I'm unable to figure out the cause behind my noisy bandpass filtered signal, despite my adaptation of the paper's parameters, in which they specified it to be a 4th order Chebyshev type I bandpass filter for the bandwidth of [6 18]Hz. The filter is applied in both forward and reverse directions, which made me opt for P... |
I have a collection of raw recordings. I also have a text file for each recording with the starts and ends of subparts of interest from the original recording. I want to train a model to classify parts of interest in future sound files.
My intuition says that I should aim to train a BERT-like encoder to classify parts ... |
I have a rotating magnet in front of an iron-core solenoid. When the magnet rotates, it would reduce speed, and then it will stop spinning and just sway back and forth. I want to know when it stops spinning from the signal. When it spins the signal looks like attenuating sine wave, and then the signal distort. Sorry I ... |
I am working on a project where I have two grayscale images of different sizes, named part1 and part2. Each image has some pixels that are interpolated or superposed from the other image. My goal is to combine these two images into a single one without any overlapping. The resulting image should look good and real, mea... |
My answer is that the signal can be splitted into 2 regions, and cancel out the negative and positive areas under the integral, based on the definition
Needed some validation on this analysis
Thanks..
Update -1 (based on comments)
|
While reading about 1st order low pass discrete implementations I found out, the following 2 equations achieve similar results, but apparently not identical, but I thought they had the same origin. One way is to derive from a filter transfer function and end up with something in the form:
$$y[n] = -a_1y[n-1] + b_0x[n] ... |
RINEX files are containers for preprocessed data in GNSS receivers. I want to know what exactly the 'carrier phase' means. As far as I can tell, the carrier phase is simply the phase (that is, a value between 0 and 2pi) of the carrier signal emitted by the satellite. However, carrier phase measurements in RINEX files s... |
I am converting 8-bit LPCM audio to 16-bit in software and, in order to cover the full 0–65535 destination range with my 0–255 source range, I multiply the samples by 65535/255, which is exactly 257.
However I have studied some existing software and, so far, I have noticed that they actually multiply by 256 (or shift 8... |
Suppose you begin with the following model of a discrete impulse response:
$$h[n] = \begin{cases} \displaystyle\sum_{k=1}^K \alpha_k p_k^n & n \ge 0 \\ 0 & n < 0 \end{cases}$$
i.e. a linear superposition of geometric series, where $\alpha_k$'s and $p_k$'s are free parameters.
Am I correct in understanding that
This is... |
Background - Create context
I am trying to synthesise some impulse responses to simulate an active noise control system. I have already formulated the problem in the frequency domain so now I want to move to the time-domain formulation.
The transfer functions I have used in the frequency domain are those of point/monop... |
I have a pulse signal with frequency f(t) And i want to generate from it a sine signal with frequency f(t) (f(t) doesn't vary a lot in time)
Is there a method to do it directly with a function generator ?
If no, How can i acheive this?
|
I'm trying to apply delta filter to remove pre-fault conditions from my signal. It is formed by a vector (7680x1), the signal was generated at 60Hz and has 32 samples per cycle lasting 4 seconds.
The problem is that I was unsuccessful with my attempts. I believe I didn't understand correctly, I would like better clarif... |
I'm writing a MATLAB code to decode LDPC codes using the BP algorithm. I'm trying this with a (7,4) code. I tried the message-passing algorithm between variable and check nodes. L_r is the check to variable update information and it is supposed to be a 7x4 matrix. Instead, it is now a 7x7 matrix. L_q is variable to che... |
My solution is $y(t) = \cos(\omega t) + \frac{1}{2}$ giving it a shift of $\frac{1}{2}$
But I'm a bit skeptic about it, should there be any coefficient of cos to this modification?
|
Based on the recommendation I would like to ask you for your opinion regarding my question I have asked on different forum.
|
I've tried to optimize the LoRa setup using Arduino IDE libraries to ensure accurate or acceptable RSSI measurements and for distance estimation between multiple nodes and a master in an outdoor environment using those WiFi LoRa 32 V3 modules that operate within the frequency range of 433MHz ~ 510MHz with SX1262.
How d... |
For context, I'm basing my code on the information provided in the paper "QRS Detection Based on Multiscale Mathematical Morphology for Wearable ECG Devices in Body Area Networks" by Fei Zhang and Yong Lian which can be found here. I'm endeavouring to the best of a novice's ability to translate the algorithm into a Pyt... |
I've seen IQ data typically combined within a single 32 bit word when sending streams of data between different components. (i.e. I & Q are both signed fixed point integers with 16 bit word length and 15 bit fraction length.) Is there a standard for this, or is this just common practice to format it this way?
|
Following a past question, I'd like to extrapolate a periodic signal using AR Burg, but when doing so, it seems that I need to sample "enough" for that to work. For example, if I use the original question example, but my signal "ends" at t=151, the AR Burg prediction isn't capturing the expected extrapolation, and dec... |
I apologize first for some questions that might be dumb. Unfortunately, I am only a bachelor student (writing my bachelor thesis in physics) and have almost zero image processing experience. And I cannot receive much help offline in uni.
I summarized my questions first, then wrote the details with questions in each pha... |
Unlike LTE, DC subcarrier is used in 5G-NR. DC subcarrier may suffer from local oscillator leakage. Local oscillator leakage gives rise to DC offset.
Are there any DC offset compensation algorithms for 5G NR?
Please note that the following question does not have any satisfactory answer.
What type of LO lekage Correctio... |
In wireless communication, when the transmitted signal is $x$ and the received signal is $y$.
Please tell me why the formula below holds true.
\begin{align}
y(t) &= \left( e^{-\jmath 2\pi f_c\frac{d_1}{c}} + e^{-\jmath 2\pi f_c\frac{d_2}{c}} \right) x(t)\\
&= \left( e^{-\jmath 2\pi \frac{d_1}{\lambda}} + e^{-\jmath ... |
The signal is a waveform generated by an electromagnet placed on a sample of Q235 steel with 6mm thickness. An AE sensor is also placed on the sample to detect the waveform. The excitation frequency is 30hz and the excitation voltage is 40v. The number of samples is 2M. In the image there are 3 envelopes, I want to get... |
In the frequency domain representation of a digital phase modulated signal, why there are prominent other frequency components which are multiples of the sampling frequency, other than the main carrier frequency ?
For example here I am using a sample frequency of 6000 Hz and 8 bits to quantize one sample. Then there ar... |
FIR filters are said to have linear phase when the filter coefficients are symmetric or antisymmetric with respect to time instant n=0. I am trying to design a fractional FIR filter of group delay=1.7 using inbuilt function in Matlab,then the impulse response obtained is as follows
[-0.0579 0.3401 0.7936 -0.0757]
Sinc... |
The ESPRIT algorithm described in wiki or any other literature suggest the usage of autocorrelation matrix instead of output matrix while performing SVD as one of the steps towards estimating DOA. It appears that the output matrix (Y) can be used, mathematically speaking (wiki also shows the steps), but there must be s... |
"The digitised speech signal $s(n)$ is put through a low order digital system (usually first order FIR filter) to spectrally flatten the signal and make it less susceptible to finite precision effects later in the signal processing"
This is from Fundamentals of Speech Recognition (Rabiner) material.
Gathered three reas... |
I have a simple BPSK system using RRC TX & RX filters with 10% excess bandwidth with a receiver that uses a symbol synchronizer of the common type consisting of a polyphase filter bank (performing simultaneous matched filtering and resampling), a PI loop, and a timing error detector (TED). (i.e. pretty much what gnurad... |
My question is tightly related to my different question. Let's say I have an inverter fed three phase induction motor drive where in the braking phase (when the motor operates in a generator mode) the generated electrical power is dissipated in the braking resistor controlled via simple chopper. Below is a record of th... |
So, I tried out what i am learning in digital communication using GNU radio.
The plan is to modulate two baseband signals(sine and cosine) 1kHz with no modulation protocol(don't know any(yet)) to a carrier of 9kHz. I added gaussian noise of 10 amplitude to the passband. Which is followed by BPF, demodulation and LPF. T... |
I'm trying to do a simple modulation and demodulation of a digital/binary source. I'm multiplying it by a 1khz cosine to center it at 1khz instead of 0. Then, to demodulate, I multiply again with the same cosine to have the original signal back. Why instead of having a 0 frequency centered, I have a 2khz centered frequ... |
I am trying to generate 5MHz signal. I am using NI USRP with matlab. I have extended the existing MATLAB QPSK Tx Rx to 16QAM, Now i am struggling with advacing the testbed to send and receive 5MHz signal. At Tx side i have Bits generator, 16QAM Modulator, Raised cosine transmit filter with rolloff factor 0.5 and finall... |
So I have a very basic on which I couldn't find much details anywhere.
I'll start with an example:
I have to transmit 100Mbps of BPSK data. Roll-off factor is 0.25.
This data will be processed at baseband.
For the receiver recovery algorithms, lets say I need a minimum 2.5 samples per symbol.
At TX: I will first pulse ... |
For a sine wave, $A \sin(\omega_0 t)$, with an amplitude $A$, after FFT and calculating the magnitude spectrum we can get a single peak with a height also equal to $A$. This requires proper scaling.
Let us say we have an exponentially decaying sine wave $x(t)$, then, the "recovered" peak intensity in the frequency doma... |
If we have a rectangular pulse function, we know that after a Fourier transform we obtain a sinc:
We know that the left part (negative frequencies) has no physical meaning and it's just specular. It's important from the mathematical point of view but not from a physical point of view.
Anyway, if we had multiplied the ... |
# defining parameters for the signal
A1 = 5 # Amplitude of the first cosine wave
f1 = 2000 # Frequency of the first cosine wave in Hz
A2 = 2 # Amplitude of the second cosine wave
f2 = 1000 # Frequency of the second cosine wave in Hz
fs = 8000 # Sampling frequency in Hz
N = 128 # Number of sam... |
My professor said that: in order to avoid interferences/harmonics caused by the nature of the square wave g(t) (sinc will have these secondary lobes), it's better to change the basic pulse g(t) to something like a arc of sine or something like that instead of having a square wave. I couldn't find anything on web. So I... |
In 1D signals f(t) (e.g. a sound signal - independent variable t-->time) when we want to use informations from the Fourier transform F(ω) in order to reconstruct the signal, using the magnitude |F(ω)| yields better reults than using <F(ω). In 2D signals f(x,y) (e.g. an image - independent variables x,y-->space coordina... |
I am going through Fundamentals of Speech Recognition (Rabiner). I stumble upon the concept of Two Level Dynamic Programming .
Can you suggest me any online resources to study the same?
|
I just finished a costas loop for QPSK signal,I transmit the baseband signal and sent it into the costas loop, which worked well, but when I filter the baseband signal with root raised cosine filter during modulation, the original costas loop did not work properly. I probably know why it did not work properly, but I co... |
Given a continuous time signal $f(t)$, we can sample it signal by multiplying with a Dirac comb (impulse train)
$$\bar{f}(t) = \sum_{n=-\infty}^{\infty} f(nT) \delta(t-nT) \tag{1}$$
where each impulse is spaced by $T$ (Lathi. pp 319).
I have a concern about this claim. When sampling a signal, we are interested in sto... |
I require some advice on achieving a better R^2 value using the cross correlation method. Is there any sort of way to improve an R^2 value to obtain better results? I have been looking at applying some sort of filter to every window before I calculate the R^2 value for that particular window, however, I am unsure if th... |
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