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I am developing a web app that does real-time processing on audio on a server and streams the processed audio to the client.
The current implementation works using wav format, but obviously using uncompressed audio is terrible for streaming and storage requirements, so I want to do it using mp3 instead.
The server is w... |
I have a LTV (linear and time-varying) system. So, $h(\tau, t)$ is the "instantaneous impulse response" at time $t$ such that if the input signal is $x(t) = \delta(t - t_0)$ (an impulse at time $t_0$), the output signal should be $y(t) = h(t - t_0, t_0) = h(\tau, t_0)$ where $\tau = t - t_0$ is the delay/lag from the i... |
I am using librosa.stft to apply an STFT to an audio sample and display a spectogram of the results when using different sampling windows (n_fft = 1000 and n_fft = 3000). I am curious why the different sampling windows results in different range of values used on the x axis? The original audio sample is 4 seconds long.... |
I need a method to measure image quality via computer programming (like Python). I'm doing this so as to compare quality of multiple images. I have found stuff like NMSE, PSNR and SSIM, but don't fully understand them, and am not sure which one is the best or if I should use multiple.
I was wondering what is the best m... |
If I have the system that could be observed in the next Image:
I want to know the transfer function, where the external force $f$ is the entry and $x_1$ is the output. The direction and positive sense of movement and force is right to left, as the image shows.
Assume that the mass $m_x$ is zero, the initial conditions... |
What is the effect of a passband slope on a QPSK (or any QAM) constellation? Specifically given a linear slope (in magnitude, not dB), and no phase distortion (linear phase or even zero-phase) how would the following constellation appear after being passed through such a channel?
This is a "DSP Puzzle". Please preface... |
I have the following power values in Watts of a signal:
6.5e-8, 4.6e-8, 7.1e-8, 1.01e-7, 9.98e-7, 2.5e-6, 3.96e-6, 2.84e-5, 2e-4, 4.05e-4, 1.7e-6, 8e-7, 8.23e-8, 8.32e-8.
How do I normalize this to 1mW (0 dBm)?
Thanks.
|
I am looking to compare the color depth of multiple images, in order to see which one has the best color depth. I was wondering what is the best way to compute color depth computationally?
I've seen something called bit depth used to measure color depth, however that doesn't seem totally right...
Like between these 2 i... |
Using formula on Audio EQ Cookbook I implemented biquad IIR filters in Excel.
As in my previous question Phase Response Function / Plotting in Excel (IIR Filter), Hilmar helped a lot with the phase formula and Robert Britsow Johnson mentioned about unwrapping the response.
Now I'm able to calculate phase response with ... |
Let's say we have a white noise as input. Total RMS value is -20dBFS.
When I apply a parametric filter at 250Hz with Q = 4 at 3dB boost, what will be the final total RMS of the signal? How can I calculate that?
Is it same with the shelvings?
I have IIR coefficients of filters and an Excel table that plots the transfer ... |
I have verified that for a pure carrier the power is 0 dBm. But when i see a qpsk/psk modulated signal the power reduces. How can i calculate how much the power has been reduced?
Is there any mathematics behind this
|
I am so confused about the high pass and low pass kernel. I saw some posts, and it says the sum of all coefficients is 0 is a high pass and if it is 1 is a low pass. And I think I did understand the concept of nulling out the DC part. However, when I was reading my lecture note, I saw this kernel, and it says to be a h... |
What is the effect of a passband slope on a BPSK constellation? (This is a companion question to the related "DSP Puzzle" for QPSK)
Specifically given a linear slope (in magnitude, not dB), and no phase distortion (linear phase or even zero-phase) how would the following constellation appear after being passed through... |
$$ x \left[ n \right] = 3 \sin \left( 4 \pi n \right) + 2 \cos \left(0.36 \pi n \right), ~~~ 0 \leq n \leq 100 $$
The fact that my signal is composed of both sine and cosine makes it tricky for me.
|
Consider a sinusoid in AWGN:
$$Y = A \cos(\omega t+\phi)+n $$
Assume the frequency and phase are known. To estimate $A$ we can use least squares (which in this case is equivalent to the DFT):
$\hat{A} = (H’H)^{-1}H’Y$
where $H$ is the design matrix constructed from sines and cosines. We know that least squares is unbia... |
When signal processing experts say something is analogue, what do they mean?
If I understand correctly from prior research, they mean to an analogue system, i.e. a system with by-definition uncountable number of modes/states.
|
I have noticed that when you create a crossover using even-order Butterworth filters, you need to use two filters instead of one (Linkwitz-Riley) in order to obtain a flat amplitude response (two low-pass and two high-pass), whereas using odd-order you can only use one (one low-pass and one high-pass). Both even-order ... |
If we call one signal $x$ and the other signal $y$.
I want to find a transfer function that takes $x$ as input and gives $y$ as output. I have recorded data of signal $x$ and $y$.
$$x(t) \overset{H}\rightarrow y(t)$$
$H(\omega)$ is the transfer function I want to find using Python.
|
I am trying to get an FFT to work with Verilog to bin frequency data to be able to tell what note is being played.
I am using an FPGA BASYS3 with a pmodmic3 which samples at $20\,KHz$ and outputs 12-bit data.
I am using this FFT core generator to generate an FFT that I would like to use: https://zipcpu.com/dsp/2018/10/... |
I've read an article about "mains hum" forensics in http://hummingbirdclock.info/about and about Electrical network frequency analysis.
The idea is to be able to measure the small variations of the 50 Hz power line hum.
Is it said that the power line frequency is changing every second.
Given a 96 kHz audio recording si... |
I have a simple signal chain with a differential sinusoid as the input. This passes through the first amp which converts the signal to a single ended signal. The next block is a 4 pole low pass filter with an Fc~100kHz. After this LPF I have my ADC. My ADC is sampling at 500ksps.
(input signal 100kHz differential) ... |
I want to see the frequency response of 1024 order bandpass filter in MATLAB.
The sampling frequency is 16kHz and the pass band frequency is [376Hz 480Hz]. I have implemented the bandpass filter using inbuilt function fir1 as b1 = fir1(1024,[376/8000 480/8000],'bandpass').
To plot the frequency response of the filter, ... |
I'm trying to find the optimal probability of detection/false alarm for the following detection task. Given $N$ signal samples with $d$ dimensions (independent channels) each, assign the samples to one of two distributions:
$$
H_0: \mathcal{N}(0, \sigma^2I^d) \\
H_1: \mathcal{N}(\boldsymbol{\mu}, \sigma^2I^d)
$$
The va... |
I have to design a 1024 order band pass filter in matlab. The pass band frequencies are [376 480] and fs=16kHz.It was written in documentation of MATLAB that "the cutoff frequency is the frequency at which the normalized gain of the filter is –6 dB". How can it be -6dB? It should be -3dB right.
|
The discrete system has poles at points $z_{1,2} = 0.8e^{\pm{i\pi /6}}$ and $z_{3,4}=0.8e^{\pm{i\pi/2}}$, and two-fold zeros at $1$ and $-1$. The task is to determine the impulse response of the system. The wording is weird, but I assume this means the transfer function is $$H(z) = \frac{(z-1)(z+1)}{(z-z_1)(z-z_2)(z-z_... |
Generate a sound wave of saxophone frequency and compare it to the original sound clip and play both to listen to if there is a good match.
I am trying to generate a sound closer to a saxophone using its Fourier Transform computation and picking apart different frequencies, building their respective sinusoids, and play... |
I'm studying Mathias's thesis Algorithms for the Constrained Design of Digital Filters with Arbitrary Magnitude and Phase Response. In section 2.1.2 the complex LS approximation problem is defined by an overdetermined linear system
$$
\mathbf{W}^{1/2}\mathbf{C}^H\mathbf{h}=\mathbf{W}^{1/2}\mathbf{d} \tag{2.8}
$$
where ... |
Let's say a I have a signal
x= [1,2,3,4,5,6,7,8,9,10,11]
y= [0, 2, 4, 6, 8, 10, 8, 6, 4, 2, 0]
How do I normalize this signal so the integral from 0 to 10 (along x-axis) would give me 1? Meaning the total power of the signal will be 1 after normalization.
|
A system is said to be stable when any of its poles are <0.
However I don't get why that is the case. Negative poles mean negative angular frequency, and negative angular frequency is equal to positive angular frequency, so the definition contradicts itself. What am I missing?
|
The chart should have at least 9 distinct color patches for 3×3 CCMs (which are generally recommended). Excellent results can usually be achieved with the inexpensive, widely-available 24-patch X-Rite Colorchecker.
(source)
Is there a particular reason that we need at least 9 patches?
I am asking this because I am wo... |
I have a continuous (in time) non-linear system in the form $\dot{x}=f(x(t)) + Bu(t) + w(t)$ which I would like to track with a UKF. $w(t)$ represent white noise (in particular, the acceleration and angular acceleration component of the dynamic system), $u(t)$ the known input, f(x(t)) the non-linear continuous function... |
I am given the z-transform of a filter $$H(z) = \frac{z^2 - 1}{(z-z_1)(z-z_2)(z-z_3)(z-z_4)}$$
where $z_1^* = z_2$, and $z_3^* = z_4$. I'm interested in the operation of this filter, and what happens when its input is gaussian noise. Now, taking the inverse z-transform of $H(z)$ gives the impulse response. The output o... |
I am working on a project to reduce resonances caused by motors.
The vibrations in the material are measured and sampled by ADCs. The measuring works as intended and the next step is to reduce the measured resonances. My problem is the comparison of different measurements of the same resonances. I have an example:
T... |
I watched this video by Engineer Guy and decided to build the same, but digital. I did it, but I decided to also calculate the sine, to have a complete complex number and be able to calculate the magnitude. So I did and it works fine. Here it is at work: I generated the signal, containing two frequencies, and wrapped i... |
Context: I am solving a 2-dimensonal discrecte Poission equation ($\mathbf{\nabla}^2x = f$) by assembling a system of linear equations $A x = b$ and then solving it using numpy's integrated solver (np.linalg.solve). To assemble this system, I am currently using the well-known 5-point Laplace kernel:
I somehow felt tha... |
Lately, I've been doing some FIR-filter simulations as a part of a programmatic circuit design course. One task was to use different constants $b_i$ for the multiplying stage of the filter and see how the response of the filter changes.
The filter was a 2nd order FIR-filter. First, we used the constants [1,1] for the ... |
If I'm watching a DVD, how many [inverse?] descrete cosine transforms per second is my computer performing?
If I'm understanding this correctly, DVD video uses MPEG 2, which is basically MPEG 1 but with only one possible resolution and frame rate. [Also interlaced, because legacy.]
I'm by no means an expert in this ar... |
This question is related to white noise representation of WSS sequences using AR(2) (autoregressive) model
The function is given as:
$$x(k)=\frac{1}{p_1-p_2}(p_1^{k+1}-p_2^{k+1})w[k]u[k]$$
where $w[k]$ is white noise sequence with variance $\sigma_w^2$, and $p_1<1$ and $p_2<1$.
I have to prove its autocorrelation funct... |
$$ h[n]= \left( \frac{8}{9} \right) ^n u[n-3] $$
And the function is:
$$x[n] =
\begin{cases}
2 & \text{if } 0 \leq n \leq9, \\
0 & \text{ else}.
\end{cases}$$
In order to find the convolution sum $y[n]=x[n]*... |
The SIFT paper and the paper of Anatomy of the SIFT Method do not mention that the input images should be preprocessed (normalized, re-scaled) before feeding images into the standard SIFT algorithm. But the following two SIFT implementations both normalized images to 0-1 beforehand.
The implementation from the author ... |
I have two noisy sinusoidal signals and I am currently investigating solutions to synchronize them using python. Here is an example I made to illustrate my goal, supposing that I generate these two sinusoidals (both with same frequency and amplitude but a phase difference):
time = np.arange(0, 60, 0.005)
amp = 2
F = 0.... |
I try to compute the FFT2 of the white noise image.
I use this image
with Entropy 7.995 and
pixel scatter plot:
based on this python code:
import numpy as np
import matplotlib.pyplot as plt
image = plt.imread("White_noise.png")
image = image - np.mean(image)
ft = np.fft.ifftshift(image)
ft = np.fft.fft2(ft)
ft = np.f... |
Sorry if this is a basic question, but suppose I have a signal that is locally periodic over short time intervals but is not globally of that period. As an example:
import numpy as np
import matplotlib.pyplot as plt
x_limit = 100
res = 50000
random_factor = 0.345
periods_per_slip = 4
x = np.linspace(-x_limit, x_limi... |
I have two different datasets.
Dataset 1 has a frequency of 1000Hz/sec and dataset 2 has a frequency of 50Hz/sec, how can I synchronize the two datasets so that they can be used as training data?
|
In Matt L's answer he states that an ideal phase shifter with a phase shift $\theta$ has a frequency response
$$
H(\omega)=
\begin{cases}
e^{-j\theta},&\omega>0 \\
e^{j\theta},&\omega<0
\end{cases}
$$
But what about DC and Nyquist? If I want to shift the phase of a real-valued signal $x[n]$, I suppose to get another re... |
I'm working on ADAU1772 evo.board with a PDM MEMS microphone.
Microphone's clock is 3.072MHz.
DSP Master Clock is 12.288MHz.
DSP sampling rate is at 192kHz and all DSP modules work with 192kHz.
1st Problem:
It was a clear sound from the microphone, but when I add ~35dB of gain, I started to hear a strange cracks from t... |
In quadrature modulated signals, for instance:
I am unclear how the "discontinuities" in the waveform at the junction of each data point can be physically produced?
Taking a speaker as an example, how is it possible to produce this "discontinuity" in the sound wave by using the speaker cone's mechanism?
|
I have 64 samples of a signal, two sines, one at 0.1 max freq, another at 0.5. Wrapped in gaussian:
In my JS version of FFT I get an array of numbers, but the size is double (128 elements) and the second part is mirrored:
But in Python, numpy's FFT returns an array of 64 complex numbers, and I like it because I can c... |
Is there a relationship between the Shannon entropy of image and the output of the 2D Fourier transform (DFT) of the image?
|
Based on this link
Is it possible to consider a model to introduce FFT2 as a parameter to determine how random the image is? (randomness test like Entropy)
consider Lena's image and its FFT2:
If we do the FFT2 calculations mentioned in the above link, we have:
Compared to the output of FFT2 of the white noise image:
... |
I made a program that can change pitch of a recorded audio clip while maintaining its duration. Basically, it takes the FFT of windowed segments of a recorded signal, processes them using phases to find frequency values, synthesizes the modified FFT values, then runs an IFFT on the windows. The issue is that the modifi... |
I have an array with 2 columns, the time index and the samples, What I first did was to segregate the samples since those are the only values I need but I am lost on what my next step will be to produce the frequency plot.
|
The impulse response of an ideal low-pass filter can be determined by setting $H(\omega)=1$ in the Fourier-representation $$h(n) = \frac{1}{2\pi}\int_{-\omega_c}^{\omega_c} H(\omega)e^{j\omega n}d\omega$$
The solution will be a function of form $\sin(n)/n$. Now, the motivation behind the substitution above is the desir... |
I'm creating a polyphonic audio sample player and I want to let the user play each voice faster or slower using resampling by a rational factor (classical interpolate by L / decimate by M scenario), in real-time (the user could send a command to set the speed of a voice and the system would adapt the resampling procedu... |
I came to the conclusion that synthesizing pure tones in Frequency Domain is much less efficient than synthesizing cosines in Time Domain and then computing the FFT. But I suspect this is the case here only because I am ignoring a better formula.
Namely:
Synthesizing a cosinusoid of given frequency, phase and magnitude... |
The 'binning' process consists in summing the energies (squared magnitude) within groups of adjacent FFT values. This will give you the total energy in a set of disjoint frequency bands.
A more elaborate form consists in using overlapping triangle filter banks - you compute a weighted sum of the energy in a range of FF... |
In dsp it is somewhat common that in filters require a buffered cache of data to fed into an array in order to process data.
Hold that thought.
In SSB demodulation a signal is passed through the IF mixer and heads straight though a filter and mixed again with the carrier VFO to make the AF components.
Back to dsp
Trad... |
I have bunch of gray-scale input images with blob-noise-like parts (however, they are not an actual noise):
Example of one such image:
I need to keep only areas inside red squares and remove the rest as much as possible. Which filter or process is best for this task?
I have tried:
Convolution filters but none of them... |
In the realm of LTI systems (perhaps even in general systems, I am not sure) it is so that if a system is asymptotically stable then it is also BIBO stable. Is that the case when a system is marginally stable too? Meaning that if it is, then it too is BIBO stable.
For an LTI system to be BIBO stable we simply need that... |
I got a question while solving the questions below.
A system is described by $$y[n]=\sum_{m=-\infty}^{n+1} x[m].$$
Classify this system as to time invariance, BIBO stability and invertibility.
As far as I know about checking invertibility of system is to check whether system's input and output is a one-on-one corresp... |
In a VLF (Very Low frequency) communication, I saw that FSK or differential type modulation like DPSK were used.
But is it possible to use spread spectrum communication with DPSK for VLF communication, it will improve transmission reliability? Has spread spectrum communications ever been used in the literature and if s... |
I set up a vanilla BPSK modulation in GNU-radio, but the constellation is showing me a line.
I am using: firdes.root.raise_cosine(1.0,samp_rate,samp_rate/sps,excess_bw,11*sps)
It has been a while since I was last involved with DSP. I know if timing and phase are off then I would expect a line in constellation for BPS... |
I have an image that I've quantized like so:
And I would like to get rid of pixels that aren't much like their neighbors (basically do a low-pass filter). The goal would be to get rid of isolated pixels of a particular value or narrow strips of such pixels.
I've tried doing a Gaussian blur and then re-quantizing, but... |
I am trying to use plt.specgram
But I found something weird in the x-axis setting.
My data is as below..
Date Value
2018-01-01 0.1
2018-01-02 0.4
...
2020-02-27 0.7
2020-02-28 0.3
The length of data is 821. (The dates I wrote are temporary values)
When I plot the spectrogram as below:
Pxx, freqs, bins, im=ax... |
Consider a cascade involving a downsampler (factor $M$) and an upsampler (factor $L$). For the sequence
$$x[n] \rightarrow D \xrightarrow{{v_{1}[n]}} U \rightarrow y[n]$$
where $D$ denotes downsampling and $U$ denotes upsampling, we can write in frequency domain
$$V_{1}(z) = \frac{1}{M}\sum_{k=0}^{M-1}X\left(z^{\frac{... |
I am trying to simulate and focus the point response for an FMCW stripmap SAR system. I have specified my target coordinates as
targets = [100,100,0;400,100,0] and I understand from several resources that the transmitted signal model for the chirp is
$$s_{\text{TX}} = \exp\bigg[j 2\pi \bigg(f_c t + \frac{1}{2} k_r (t -... |
Let's say, I have got a spectrum of a signal with noise using fft:
For a human it's pretty obvious that there is a signal with a frequency of about 51Hz. But what is the mathematical apparatus behind such conclusion and is there a tool for this in python?
Easiest solution in this example would be to just take max(y) o... |
The Solution is given above:
The Question is, how did the $\mathbb{E}{[x(k)f(l)]}$ and $\mathbb{E}{[x(l)f(k)]}$ become zero? is there some rule that correlation between Random Process and Deterministic sequence is zero?
|
Our Professor couldn't explain the clear difference between the Yule Walker equation and the Modified version of it that is used in Stochastic models. Please explain both the equations and why we modified them. Also, its significance, if you can.
The image is from Statistical Signal Processing by Monson Hayes
|
I was analyzing the following code for the MRC scheme from the book MIMO-OFDM Wireless Communications with MATLAB. Here, I am not able to understand based on what logic the sigma value calculated sigma=sqrt(0.5/(10^(SNRdB/10)))
It would be very helpful if anyone could give me any hints on the calculation of sigma from ... |
I have the ROC of a signal $x[n]$ with $z$-transform $X(z)$ as below:
Now I am wondering how I can find $x[0]$ by not calculating inverse z transform based on the roc, I am looking for a simpler, and faster way.
|
Given the following signaling elements,
\begin{align}
g_1(t) &= A \quad \text{for} \quad -T/4 < t \leq 0 \\
g_2(t) &= A \quad \text{for} \quad 0 < t \leq T/4 \\
g_3(t) &= A \quad \text{for} \quad -T/2 < t \leq 0 \\
g_4(t) &= A \quad \text{for} \quad 0 < t \leq T/2 \\
\end{align}
I have deduce the orthonormal basis to... |
Can you tell me how the $P_x(z)=P_x^*(1/z^*)$ is mathematically correct. I can understand the $P_x(e^{jw})=P_x^*(e^{jw})$ as $P_x$ is real value. But why take the Z domain representation in this way ($P_x(z)=P_x^*(1/z^*)$). Why can't I take it like $P_x(z)=P_x^*(z)$ like the Fourier transform representation?
|
I have a 25 MHz IBW real time signal analyser throwing out 25 MHz wide complex data at 50 MHz sampling frequency.
I want the frequency resolution to be 12.5 kHz (width of a vhf channel) so need 2000 (2048 pt FFT).
Since I'm sampling @50 MHz, i have 50000 samples per millisecond.
It's it the right approach to calculate ... |
A book reports that ICA cannot be used if the independent components of the analyzed data are Gaussian (at most one can be Gaussian, but no other). However, in the same book, the following example is reported:
A number of 1024 samples of a two-dimensional normal distribution was generated with mean μ and covariance ma... |
I'm trying to plot some sinusoidal signals in Matlab.
But while frequency is getting higher (closing to fs/2), results are getting distorted.
I guess it's lack of my knowledge but distortion is starting soon. My peak points are getting away from the original amplitude.
What is the reason and how can I correct it?
Code ... |
In these power spectral density plots shown above (taken from the web 1,[2]), the spectra show a roll-off at the highest frequencies, as shown in the figure. For my own example, when I calculate PSD from time domain data taken with a sampling rate of 10 MSa/s using Welche's method, this roll-off appears at half the s... |
For OFDM, we usually perform least-square (LS) channel estimation using a set of training symbols. We usually do LS channel estimation per subcarrier in the frequency domain as follows:
$$
\hat H_{LS}[k] = Y[k]/X[k]
$$
where $\hat H_{LS}[k]$ is the estimated channel (in frequency domain) for subcarrier $k$, $Y[k]$ is t... |
I am using the following code in Matlab, from some source (sadly I cant remember).
The code is used for smoothing signals using a low pas filter.
function y = fftSmooth(resp,srateCorrectedSmoothedWindow)
L = length(resp);
window = zeros(1,L);
window(floor((L-srateCorrectedSmoothedWindow+1)/2)...
:... |
I am working on calculating the SNR for my network model.
Considering AWGN at the receiver with zero means, the SNR can be presented as
$$\text{SNR} = \cfrac{P \cdot|h|^2}{\sigma ^2}$$ where $P$ and $h$ are the transmitted power and the channel coefficients, respectively.
I understand for a zero-mean signal, the varian... |
I have a time series measurement $v(t)$ from a physical nonlinear system and its power spectrum $E(f)$ look like the following
From a theoretical point of view, the solution of the system is modeled as
$$A(t) = |A|e^{i\phi(t)}$$
(of course the amplitude $|A|$ by itself is a function of $t$). This physical measurement ... |
Following the example of the Lorentzian noise power spectral density shown above (ref), I would like to clarify the following:
In the first figure (labeled by (c)), May I please know why the constant levels of the noise PSD increase? Is the change in $\tau$ the only reason for this behavior?
In the second figure be... |
I am trying to create a GNU Radio flowgraph for identifying a predefined preamble in satellite telemetry packets. For instance, the preamble can be "SAT_SAYS_HI". After identifying this sequence of bytes, the entire packet including "SAT_SAYS_HI" is written to a buffer of size equal to the packet size (256 bytes in thi... |
Determine below signal is power or energy and derive its power or
energy accordingly. $$x(t)=\sum_{n=-\infty }^\infty e^{-\lvert
t-n\rvert } \tag 1$$
we have :
$$E_x=\int_{-\infty}^{\infty}(\sum_{n=-\infty }^\infty e^{-\lvert t-n\rvert })^2dt \tag 2$$
$$P_x =\underset{T\to +\infty }{\mathop{\lim }}\,\frac{1}{2T}\i... |
I used the following code to convert the audio to spectrogram, now I try to get the spectrogram back to audio again, but don't know how to write the code.
import os
import librosa
import numpy as np
import matplotlib.pyplot as plt
import cv2
def get_spectrogram(path):
data, fs = librosa.load(path, sr=None, mono=T... |
I have been working on a conference solution based on a smartphone grid. Right now, I am stuck on Android Smartphones. Although I have managed to normalize loudness programmatically when it gets played on different phones then, due to the hardware difference of speakers, it plays with different loudness.
I am keen to ... |
I try to express the relation between:
the delay obtained by maximizing the cross-correlation between two signals $x$ and $y$ noted $\tau_{x,y}$ and
the phase velocity noted $v_p(\omega)$
In the case where:
The signals $x$ and $y$ are spaced by a constant distance $\Delta$
The signal $y$ is a shifted version - at sp... |
I am new to the field of signal processing. So, this might be a very basic question.
Recently, I am trying to build a Heart rate sensor for a cycle. I am mod-ekg (olimex) board with TI's MSP430FG439 microcontroller. From its serial port, the board is sending some kind of pulse signal, that I have to process to find the... |
Through this platform, I want to ask that how can I remove unwanted noise from the signal when you do not have much information regarding the frequency at which they appear? Data is collected from an inductive sensor and sampling frequency is 30000 Hz. There are a lot of electrical noises in the signal, the easily dist... |
Ok so as part of a project for school I have been designing an FIR bandpass filter. The output should have been a signal which looks like a sampled sinusoid but there was a strange alteration on the amplitude. I began to investigate and generated a signal with the same F/Fs. I noticed that signal also looks the same. J... |
in trying to understand the convolution theorem for DTFT, I'm faced with the following problem which I can't get my head around.
First, let me state the convolution theorem for the DTFT as follows:
\begin{equation}
y[n] = (x*h)[n] \iff Y(\omega) = X(\omega)H(\omega), \quad n \in \mathbb{Z}, \omega \in \mathbb{R}
\end{e... |
I have a signal $x_1(t)$ with baseband bandwidth $W_1$ Hz over a carrier frequency $f_1$ and power $P_1$. I also have another signal $x_2(t)$ of baseband bandwidth $W_2$ Hz over a carrier frequency $f_2$ and power $P_2$. The carrier frequency $f_2$ is random, and thus part of $x_2(t)$ spectrum may overlap with that of ... |
What works:
I generate real-valued white gaussian noise with standard deviation $\sigma$
I calculate the FFT of this noise. The Abs of the FFT is a flat noisy trace over frequency, as expected for white noise.
The standard deviation of the resulting FFT returns $\sigma$. -> Expected, good!
Where I am stuck:
I use th... |
Why do we have a negative gain(i.e. not negative as in polarity but gain < 1) after a certain point (1500 Hz). What is the reason and purpose for this. Why is that point the same for both DC and AC coupling of the baseband channel filter - Butterworth?
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I have a process that captures interactions between a source and a sink. Unfortunately, the timestamp on the sample is one day. This means that regardless of the number of interactions (flows of varying volumes) on a given day, they all have the same timestamp.
Practically this means my signal consists of daily sampl... |
A linear frequency modulated chirp signal produces a Spectrogram and Wigner-Ville Distribution that look like this:
My understanding of the fractional Fourier transform is that it corresponds to a rotation in the time-frequency plane - from Wiki:
[FT goes time to freq. IFT goes freq to time.] Fractional Fourier trans... |
For most noise reduction algorithm, the same process is applied to every pixel no matter
the pixel belongs to one of three types of pixels such as homogeneous regions, edges or textures.
Different filtering strength is required for these three kinds of the pixel because human visual system
has different tolerance of no... |
I am working on a project in which I am required to use Interpolation to create equally spaced position values as the position samples are not equally spaced due to varying train speed.
So, the total length of my data is 93636218. After interpolation, the length has reduced to 5506829. This is creating problem for me a... |
I expect to identify the first harmonic (FFT's pitch) by looking for the first batch of Matlab's fft magnitude greater than half the maximum.
I suppose that in human's voice and in violin's sound the first harmonic's FFT magnitude is greater than half the maximum ( $2.3610
\cdot 10^8$ in code).
Then with the max func... |
Let's say I want to increase the sampling rate of my signal, $f_t$, from a sampling frequency $f_s$ to some multiple $Mf_s$. One way to do this is to add zeros between the samples of $f_t$. This increases $f_s$, but also adds unwanted high frequency content and because of this the modified signal has to be filtered wit... |
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