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I am creating an Android app which records sound for t seconds at a sampling rate of 44.1kHz with a buffer size of 8192 (16 bit mono samples). I need to plot a graph of amplitude against frequency, and so first I need to run an FFT over the buffer each time it is full (which happens multiple times every second, more so... |
I am detecting roadline pattern in an image. I have applied delaunay triangulation.
I get 160 triangles in that. Now I need to find the area for each triangle. Can you kindly help me. Only after this I should proceed with my project. Please help me...
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I am trying to use Matlab to compute the spectrogram of a signal but I can't seem to figure out how Matlab computes the spectrogram.Consider the case where I have a signal x(t) with M samples and I want to compute it's spectrogram using hann windows of length N and using an overlap of K samples.
Because there are M sam... |
Calculating the coherence (sometimes called magnitude squared coherence) between two signals indicates the presence or lack thereof of a linear transformation between the two signals. Is there an analogue for signals which are related by a non-linear transformation? Let's assume that the expected non-linear transform... |
I am student of network security, and very unfamiliar with the domain of signal processing. Do pardon my naivety at asking very basic stuff.
My work deals with detecting malicious activity in network logs -say, differentiating between botnet traffic and benign Internet traffic. I came across this interesting article ti... |
I am working on bearing faults.I did the high pass filter and then calculated the envelope of the signal in the time domain. I then found its fourier. The fft spectrum is still not clear. My fundamental frequencies are not visible clearly. Do i need to filter it further?
|
I'm trying to write a simple script that should plot the spectrum in scilab, to test it I use a sinus function with 440hz so that I get my dirac in this position , my problem is that it doesn't work and I don't understand why?
here's the code :
Fs = 8000;
f = 440;
t= 0:1/Fs:1;
y = sin(2*%pi*f*t);
nf = 1024; // numbe... |
I am recording a 17khz signal at a sample freq of 44.1khz. I want to perform cross-correlation between the received and transmitted signals for calculating TDOA. But when I do xcorr, the results are inconsistent. Is it because my sample freq is too low for my signal freq? If so, how can I overcome this? Will upsampling... |
I referred the below papers to understand the working of Phase Encoding,but the explanation is not proper.Just a few steps have been given in brief.
ijcsi.org
www.ijarcce.com
Can someone please explain the concept of how Audio Steganography using Phase Encoding works?
|
I have a situation where a sampled signal is passed through a DAC and re-sampled by a switched capacitor circuit.
I can't see a model for this in the literature.
The standard model of a DAC is the ZOH. But when it gets re-sampled by the switched cap circuit it seems to me that ZOH effect of the DAC disappears because o... |
I am trying to get a frequency response vs dB SPL plot for recorded sounds. I have a microphone with a flat response, a sensitivity of 4mV/Pa, and 1 Pa = 94dB.
To get the dB SPL, I have the following formula:
$dB SPL = 94 + 20 \cdot \log_{10}(AI/0.004)$
where AI in the input voltage from the microphone, and assuming no... |
I have an on going input signal, most of the time with no activity.
Once in a while there is an activity (oscillations in different freq).
I need to take good 205 samples out of the signal.
what i mean by good?
really oscillating signal for 205 samples (not just crossing threshold).
here is an image to clarify:
bub... |
I have I and Q samples of a time domain signal, sampled at a rate T, both of length L.
I create the complex envelope, C = sampleI+j*sampleQ.
Then take the FFT of C based on Matlab's FFT help algorithm:
Fs=1/T;
NFFT = 2^nextpow2(L)
Y = fft(C,NFFT)/L
fvar = (0:NFFT/2)*(Fs/NFFT)
plot(fvar,2*abs(Y))
The frequency content ... |
We know that when we window a signal, we increase the main-lobe width. Let 'main-lobe-width' here be the null-to-null bandwidth of the main lobe. Let us further more say that the main-lobe width of a square window is '1'.
What I would like to know, is given a particular symmetric window, is there a closed form solution... |
first of all, I am not a technician, but I would really appreciate your helping with one (I hope simple or trivial) issue. I recorded a signal (skin conductance level - SCL, uS unit) with my device - 16 bit, range of the measured SCL is 1-20 uS.
In a recording software I changed digital upper range - "digital maximum"... |
I have a small input signal of 60Hz sine wave from signal generator, which is corrupted with 50Hz mains supply frequency. I want to measure the amplitude of the 60Hz signal using FFT because it is very small to see in the oscilloscope.
The Matlab FFT code:
y = data;
Fs = 2048;
[r, L] = size(y);
NFFT = 2^nextpow2(L)... |
I want to create a project that will recognized the chords that i played in guitar.
I recorded a raw sound to be compare to the one that the user will input.
Example
I have a recorded sound chordA then I will ask the user to play a sound in the guitar and check if the signal is similar to my sound chordA..
Getting the ... |
The $\frac{d_1}{d_2}<0.6$ gives a large number of false matches and it assumes the nearest neighbour is a correct match.
Anybody knows any good method to reduce number of false matches or filter them ?
Can we use k-NN matches instead of $\frac{d_1}{d_2}<0.6$ ratio for matching the SIFT features and then choose the bes... |
How can I generate an NxM DCT matrix? In Matlab dctmtx function generates only square matrix (NxN).
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I tried the following code for modulation of two analog waves. Can anyone point to how to modulate and demodulate for an analog and a digital wave? I tried by replacing the second signal (x2) with a digital pulse but I am not able to recover the analog wave after demodulation.
T = 1;
Fs = 44100;
t = 0:1/Fs:T;
Fc = 1700... |
I am reading some paper, and I am having some trouble with some Fourier transform,
Suppose that $F(\omega)$ is the Fourier transform of $f(x)$, i.e. where
$$F(\omega)=\int_{-\infty}^{\infty}f(x)e^{-i\omega x}\,dx.$$
What is $$\frac{1}{2\pi}\int_{-\infty}^{\infty}\frac{F(\omega-a)}{i\omega}e^{i\omega x}\,d\omega ?$$
I w... |
I have project in which I'll work on audio files using Matlab, I'll be listing to different machine and should get thei rotational speeds, which is kind of periodic, my question how should I analyse this should I work with the spectrum and its magnitude using the fft or should I use the PSD , which is the result of p... |
There is a recursive way of finding threshold in the image using histogram that are biomodal, and is as follows:
An automated procedure for bimodal histograms :
Select an initial estimate of T
Compute the means of the two regions determined by T
Set the new T as the average of the two means
Repeat steps 2,3 until th... |
let's assume I have two time series: one with a delta peak at position $x$ and another time series with a minimum phase wavelet that starts at time $y$ with a maximum peak at position $z$.
Now, when I convolve both time series, is the beginning of my minimum-phase wavelet at position $x$ or is the maximum amplitude of ... |
kernel A = [ 0 -1 0 -1 4 -1 0 -1 0 ]
and B = [-1 0 -1 0 4 0 -1 0 -1 ]
i have found the frequency spectrums with freqz2 function of Matlab.
A's spectrum has the highest magnitudes in the corners (1,1),(-1,-1),(1,-1),(-1,1)
and B's in (1,0),(0,1),(-1,0),(0,-1).
I know that the center of the frequency spectrum has th... |
I applied Fourier transform on a sound (PCM generated digitally with sin function) and frequency detection is accurate. However, if I play the sound and record it, and then apply a Fourier transform to detect frequencies, my results are nowhere close to original results. The window size I have chosen for Fourier transf... |
After reading this question: PSD (Power spectral density) explanation I am still a little confused as to what extra information the PSD gives us over simply taking the magnitude of the fourier transform.
So my understanding is that the PSD is equal to:
$$|X(\omega)|^2$$
but what does that tell me that simply
$$|X(\omeg... |
Lets say we have a picture with 6bits/pixel and we use discrete cosine transform with 8x8 blocks of pixels and each coefficient is 8 bits. We can use 2 8x8 masks, a zonal mask A and a zonal bit allocation mask B.
A = [1 1 1 1 1 0 0 0 1 1 1 1 0 0 0 0 1 1 1 0 0 0 0 0 1 1 0 0 0 0 0 0 1 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 ........ |
How do you measure the SNR of a sinewave in the frequency domain? (Assuming no filtering.)
Suppose: $$x(n) = s(n) + n(n)$$
Create the signal (nCycles of a sinewave) in MATLAB...:
sampleRate = 1024;
f0 = sampleRate/8;
nCycles = 11;
time = 0:1/sampleRate:nCycles/f0;
signal = sin(2*pi*f0*time);
Create the noise and sc... |
This is an extention to my other question. My sampling frequency is only about 2x my tone frequency. I want to estimate time delay of arrival using this signal. I am generating the tone signal so I know its properties. Can I use compressed sensing to reconstruct my tone wave received by the sensors and get a good time ... |
I need to generated a high frequency sinusoidal signal for modulation in MATLAB. But it turns out to be something weird. This is the simple code snippet I used...
t = 0:0.001:100;
A = 1;
s = A*sin(2*pi*1e9.*t); %1 GHz signal
plot(t, s);
title('Modulating Signal');
xlabel('Time');
ylabel('Amplitude');
Instead of the s... |
I am trying to learn signal processing and digital pre-distortion.
http://www.altera.com/end-markets/wireless/advanced-dsp/predistortion/wir-digital-predistortion.html
I am currently trying to auto correlate the input signal and from this I make a matrix. I then get the coefficients for this matrix and use them with a ... |
I have two cameras, and I'm up to perform some matching stuff on videos captured by these two cameras since I'm a CV guy.
The problem is that two cameras have very different performance in terms of color and brightness. The same object in the same illumination appears different in two cameras.
Is there any academic wor... |
I have a signal that contains two components. How can I separate it two its two components in matlab?
Please note that the location and shape of components are different in each experiment and we do a lot of experiments. It should automatically detect the two components based on the frequency contents. I have added thi... |
How can I do point by point multiplication of the quantized DCT coefficients with the quantization matrix before computing the inverse DCT for each block? I want to modify the reconstruction algorithm to approximately undo the quantization step shown here:
for i = 1:Blocks
for j=1:Blocks
irange = (i-1)*8+1:... |
I'm coding a phase vocoder in C for time-stretching audio signals. The only piece that is currently not working is the phase propagation.
I take overlapping frames from an input signal, apply a window function, and perform FFT.
Then i calculate the phase propagation (the one I'm stuck with)
Finally I perform iFFT and ... |
This is the logistic distribution of single random variable (taken from Wikipedia).
x = random variable
mu = mean of all random variables
s = variance.
Now, I want to do a Bivariate logistic distribution (having two random variables x1 and x2). My dataset is going to be image pixel values!
When I find the covariance o... |
I want to linearize a system to this form
$$\begin{bmatrix}
\Delta\dot{x}_1\\
\Delta\dot{x}_2\\
\Delta\dot{x}_3
\end{bmatrix} = A\begin{bmatrix}\Delta x_1\\ \Delta x_2\\ \Delta x_3\end{bmatrix}+B\begin{bmatrix} \Delta u\end{bmatrix}$$
$$\begin{bmatrix}
\Delta y
\end{bmatrix} = C\begin{bmatrix}\Delta x... |
I got this simple Discrete time signal question on which i have to perform shifting and multiplication.But I have some Doubt regarding the answer this book shows. Please see the uploaded image.
|
I have no background in signal processing but I would like to know what is the actual state of the art for vocal elimination/reduction. Obviously, I am not talking about the infamous "stereo elimination", but other, more sophisticated techniques that not everybody is aware of.
I understand that for many projects there ... |
I have a noisy signal file in Matlab and I have to denoise the polluted signal using a discrete Fourier transform.
I'm asked to perform the fourier transform, then take its absolute value. Then study/examine the absolute values to then implement a low-pass filter for the actual sound (and corresponding high-pass filter... |
I have a project of recording human's sound via Android's SoundRecoder API, and then displaying the extracted note on The screen. It is specific for Android platform (can't use Jmusic or Jfugue API as javaSE environment is not supported).
Now suppose I get to record a sound using this API . How can I then deploy FFT to... |
I have this simple $x(t)$ graph, and I have some confusion regarding time shifting.
What is the graph of $x(1-t)$ ?
|
Can you explain to someone with no mathematical background why is deconvolution used in telescopes, CT scans and seismographs? How do you explain that the same technique can be used for different things. The distortions in telescopes, CT scans and seismographs are all different from each other.
|
I have this wav file that I recorded using my laptop, It contains the sound of a G chord of a guitar. Since the guitar have 6 strings the sound should contain different frequencies I want to determine the top 10 most dominant frequency in the signal.
[X, Fs] = wavread('chordG.wav');
freq = fft(X);
plot(abs(freq)):
I d... |
I am trying to plot short time energy of an utterance with percent overlap as an input variable.
As I referred to the short time energy equation it can be represented as square summation of $x(n)*w(n-m)$ where $w(n)$ seems to be window length.
But I couldn't figure out how to write it as percent overlap.
|
I have to come up with an upsampling algorithm for a project and cannot use Matlab's built in upsample function. I came up with the following function but it just runs forever when I use a 13 second, 16-bit song sampled at 44.1kHz. Please let me know how I can improve the efficiency of this algorithm:
x = desired_... |
I need to identify two systems. Those are their Unit Step responses:
I don't know, what order of the system that is. It could be second order system with zero in 0. Lecturer told me it was third order system, but then told me to solve it as second order system, so I'm a bit confused. More over we don't have materials... |
I have two audio signal that I want to compare using Matlab, my problem is that I can hear the difference between both loud and clear, but when use the function pwelch to compare their PSD ,I don't see much difference between them,any idea how can I compare them with a different methode ! thanks in advance !
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I am basing my question on this post here, because I would like additional details on it, as I have not had any success in re-creating it.
I would like to simply learn, in no uncertain terms, how to do upsampling via the FFT. There seem to be a lot of details regarding if the signal is even or odd, where exactly to ze... |
I'm obtaining MFCC by using the function wave2mfcc from the Speech and Audio Processing library (SAP) from here http://mirlab.org/jang/matlab/toolbox/
The steps follows the procedure:
Take the Fourier transform of (a windowed excerpt of) a signal.
Map the powers of the spectrum obtained above onto the mel scale, using... |
Calculate the entropy of Turkish alphabet and compare with the entropy of English alphabet with using Matlab?
How can ı do this question anybody help please?..
|
Currently doing some signal analysis in python for a major project in my physics degree which is due really soon. I need some help!
Say I have two signals, f(t) and g(t) which are recorded over the same amount of time.
f(t) and g(t) both feature different amounts of noise.
f(t) and g(t) are recorded at different sampli... |
I have one quite simple question: how does the amplitude of the signal scale when we do FFT where FFT window length is shorter then duration of the signal? Does this means that if FFT size is 12 time shorter that also amplitude of the signal will scale by 1/12? if so what is the mathematical explanation.
Thx in adavanc... |
I've seen the use of a Causal wavelet mentioned a few times but have never seen any examples of them in use with Matlab. Is it possible to to perform a Wavelet Decomposition using a causal wavelet i.e. no future data. If so an example of how to do this would be greatly appreciated.
Thank you.
|
I'm using the MATLAB algotirhm Matching Pursuit to obtain the Gabor atoms forming a signal.
But now I want to graph them in a "Heisenberg boxes" graph.
Does someone how if is there a function to graph them like this?:
Or maybe I should write a function for graph. In that case, I know the time and frequency which are ... |
Say we have a signal S, and we composite it with a 0.1-volume 30-sample-delayed version of itself, so:
T(k) = S(k) + 0.1*S(k-30)
How would one rearrange this equation to make S the subject?
i.e given the resultant signal T, and the fact that we know the parameters of the echo (0.1 and 30), how to reproduce the origina... |
Say I have a transfer function, for example:
$$H(z)=\frac{1}{1+0.1z^{-30}}$$
How can I compute the impulse response?
(This is just an example, the important thing is that it is in closed symbolic form!)
If I understand correctly, if I could just rewrite $H(z)$ as some polynomial in $z^-1$ (by using a Taylor expansion),... |
I am on a project to write a function in Java to perform GMSK modulation. I have a good understanding of the GMSK modulation block diagram to an extent, but I have a challenge on what it means to delay the Q-phase input by 90°.
Do I simply multiply my Q-phase bit stream by a negative sine carrier or what? Please, what... |
I want to implement the 'filter' function in matlab but I just can't seem to replicate the results I get when using the matlab function.
My understanding of the matlab function is that it takes 3 arguments: 'b' = array of numerator coefficients , 'a' = array of denominator coefficients and 'array' = the array to be fil... |
I know this is maybe a very basic question but I am doing this as a hobby and I can't find a solution to this problem.
Basically I am trying to remove some noise from data I am reading from an accelerometer. This is what I want to achieve (Taken from Total Variation Denoising (An MM algorithm)):
I read in Picking the ... |
I'm doing MIT's OCW course with Allan Oppenheim - "Signals and Systems".
I'm given 2 continuous signals$$x(t)=\cos[\omega_x(t+r_x)+\theta_x]\tag1$$ and $$y(t)=\cos[\omega_y(t+r_y)+\theta_y]\tag2$$
Assuming their phases are the same, the notes tell me that the signals are identical if $$\omega_x=\omega_y$$which is quite... |
We are working with speckle noise reduction in ultrasound images by using various despeckling filtering techniques. We compare the performance of each filtering technique by using Quality metrics such as mean square error,signal to noise ratio,peak signal to noise ratio,Edge preservation index,Speckle index.
We know th... |
I wonder what generalization I can make about what happens when I introduce some error in my sampler spacing.
For example, If I sample once every second I would want my sampler to take a signal at 1s,2s,3s,4s.
Unfortunately, my real world sampler looks like more like 1.01s, 1.99s, 2.99s, 4.01s.
I want to make a statem... |
Quoted from this
As a first order approximation it takes ten times the energy to double the perceived loudness
So is there a 2nd order approximation. If so then how much change in RMS value doubles perceived loudness? Is it governed by any system of equations?
Thanks
|
I apologize if this is not the correct forum, but I thought I'd ask.
I have an array of signal samples and I want to detect (retroactively, once I have measured the time/magnitude values) if there are any spikes / deviations during a sampled real signal
How would I do this intelligently?
I've been computing the mean o... |
The goal is to preprocess an image in a way that outside the circle isn't in the region-of-interest so set its value to 0 (black). That way the followed step of segmentation will be easier to analyze all the blobs that exists inside that circle.
My instance isn't this one in particular, but this example is somewhat clo... |
The following integral (perhaps fourier tranform of $\sin (t \omega)$ ) is not convergent:
$\int_{-\infty }^{\infty } e^{-i t \omega } \sin (t \omega ) \, dt$
As, $\sin (t \omega)$ is NOT an Energy Signal (but a Power Signal), then how come we get successful in finding the fourier transform of $\sin (t \omega)$ ?
|
i use this for complex fft.
Output expected
$fft[3].real= 32$ (peak at 3rd bin)
$fft[61].real= 32$ ((peak at negative frequency pair of 3rd bin))
All other values negligibly small
The input is
$ y.real = \sin (2\pi*3i/64)$ where $i = 0 \to 63$
$ y.imaginary = \sin (0*3i/64)$ where $i = 0 \to 63$ (all zero)
The outpu... |
I am caught up in a following question:
I have a mixed (causal + anti-causal) input to a system whose output is causal, with system having transfer function with pole at z=-1 and zero at z=1. What is the ROC for the system (if any) ?
Thanks a bunch.
|
I'm investigating whether it's practical to use a Goertzel filter to decipher a binary message phase-shift encoded into a single frequency.
Here is my result:
I would like the bottom line to reflect the discontinuities visible in the top line.
Here is the code generating the second and third lines:
samps_per_sec = SAM... |
I need a small help for my assignment, which is to solve the 8-point FFT without using Matlab or C.
Preliminary information:
8-point sine wave: [0, 0.7071, 1, 0.7071, 0, -0.7071, -1, -0.7071]
8-point complex exponential: [1, 0.7071-j0.7071, -j, -0.7071-j0.7071]
So here's the attempts I did:
1.) Bit reversing the 8-poin... |
I'm interested in sending digital data over ultrasound.
My initial thought is that it should be first converted to BMC:
And then phase shift encoded (PSK):
I initially thought to use a Goertzel filter to decode it, but this runs into trouble: Extracting phase using Goertzel filter in Python
What would be a practical ... |
It is widely known that matching a FIR filter of fixed length to a band model is an unconstrained QP-problem. The MATLAB function firls() implements a solution to this problem. Basically, one symbolically evaluates integrals of the (Z-domain) transfer function over the unit circle, getting astonishingly simple expressi... |
I am trying to understand how this works, specifically, what the DTFT of each step looks like in each step of the chain (for understanding). I am not looking for an answer like because the input signal has a frequency of $\pi/3$, it will be reconstructed correctly.
I plotted the DTFT of each step in MATLAB and simply ... |
I am currently trying to implement a GMSK demodulator, using Java. I am actualy trying to use the model of this block diagram. However, I really do not know what goes on within the mode selector and the differentiator. Does anyone have an understanding of this?, or is there a better method i can employ which may or may... |
I am reading tutorials about windowing and have read following sentence:
For an integer number of cycles, all smoothing windows yield the same peak amplitude reading and have excellent amplitude accuracy. Side lobes do not appear because the spectrum of the smoothing window approaches zero at Δf intervals on either si... |
I have read about spectral leakages in other posts here. From what i understand, it occurs because you dont have an integer number of time periods in your sampled data. By default a rectangular window (=no window) is used and it corresponds to a sinc function in the frequency domain. If we have an integer number of tim... |
I've been trying to solve a Discrete LTI System problem where the overall impulse response of the following System, whose input and output sequences are, respectively, $x(n)$ and $y(n)$, has to be found:
This is what I've tried:
Calling the output sequence that comes out of the adder $e(n)$, I know that $y(n) = e(n) \... |
How can I find time difference between two same audio signals?
Please do not answer cross-correlation or GCC. I am looking up for a new solution that will help me. Thank you.
|
I've been told that they are influenced by noise considerably more than that of a first order edge detector such as a Gaussian. But why?
|
I have a spectrogram built by breaking a signal into chunks and taking the FFT of these chunks. An example spectrogram is:
In this spectrogram I am looking for a curve (aka pitch contour) that:
Starts at zero
Increases to an unknown value
Stays at that unknown value for at least one minute
Decreases back to zero
It'... |
I'm looking at communicating digital data through the air as soundwaves.
To illustrate the problem, I generate a 10-cycle 2kHz wave, play it out through my MacBook's speakers and pick it up on my iPod Touch:
The echoes are about half the amplitude of the probing signal.
HotPaw pointed out to me in a comment on a recen... |
I am getting error while implementing a simple idea for noise detection in an image. Let I denote noisy image and for each pixel of I is represented by I(x,y). A Sliding window of size 3X3 centered at I(x, y) is defined. At first we estimate that the center pixel of window is noise or not. To differ the noise... |
I am try to grasp the idea of frequency domain for images. I think get the basics, but now I've stuck with a question that I can't find appropriate answer anywhere.
How are frequency domain and complex numbers in relationship?
I've read this article from DSPguide and I understand that after applying DFT to an image in ... |
I got some problem in the equalization techniques. I am new to communication simulations but I have done some data analysis labs in my college. I read some where on Internet that in a research paper that
zero forcing is Optimization technique but on Wikipedia it says it is Equalization technique?
I am confused in equ... |
I'm at a loss trying to answer the following question:
The frequency response $H(j\omega)$ of a length 4 FIR filter with real impulse response has the following specific values: $H(j\cdot0) = 2$, $H(j\cdot \pi/2 ) = 7 − 3j$, and $H(j\cdot\pi) = 0$. Determine the impulse response function $h(n)$.
I can't think of anythi... |
I'm trying to get my dsp sea legs a bit, and am trying to complete a problem that asks for the mean delay and expected SNR boost for a given difference equation: y[n] = (x[n] + ... + x[n-N+1])/N
I'd love some general guidance on how to learn more about how to address this. Specifically,
I'd like to plot the filter fun... |
i came across many terms describing loudness ! I understood perceived loudness which is what the user perceived and is related to the user. What is specific loudness and perceptual loudness?
AES Convention Paper 7180 defines specific loudness as
a measure of perceptual intensity across frequency and time and is measu... |
Are the following frequency domain signals valid DTFT's (Discrete Time Fourier Transforms)? Answer yes or no, and justify your answer.
a) $X(\omega) = \omega + \pi$
b) $X(\omega) = i + \pi$
c) $X(\omega) = \sin(10\omega)$
d) $X(\omega) = \sin(\omega/10)$
I'm just very unsure as to how to go about this question (what m... |
I want to know the difference between the 2D continuous and discrete Gabor filter when it comes to Matlab implementation. I found many files related to 2D Gabor filter on Mathworks File Exchange but I am unable to find any file related to 2D continuous Gabor wavelet. I want to use continuous Gabor filter to enhance blo... |
I am researching in medical image processing area. I want to segment knee cartilage into femoral cartilage and tibial cartilage. But two part have same intensity and only different position(femoral cartilage is above and tibial cartilage is below). I am finding the techniques or method to segment them individualy.Could... |
guys. I am really lacking in knowledge of linear algebra. I am reading slides of Harris corner. But I am really confused about one of them. I know that I can find corners by two large eigenvalues but I can not consider it as an ellipse....can anybody give me any suggestions?
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Say you have a bandpass filter and you place it on 40 discretely sampled points from a sinusoid with a period of 40.
Why is the bandpass filter (green curve on the image) only "in phase" with the sinusoid if you use a window length (impulse response length) for the filter of 1/4th the period of the sinusoid (here 40/4=... |
I don't know how to determine line rate and frame rate of the following system:
Height of monitor: 40cm
Screen Aspect Ratio: 16:9
Pixel Aspect Ratio: 4:3
Progressive scan
Viewing distance from screen: 0.6m
Active line period: 15μs
Horizontal Blanking: 5μs
Vertical blan... |
I am stuck on a problem about finding the complete solution of a Discrete Linear System by using the parametrization $y_{c}(n) = y_{h}(n) + y_{p}(n)$ where $y_{h}(n)$ and $y_{p}(n)$ are, respectively, the homogenous and particular solutions.
The System, in this case, is given by the following difference equation:
$y(n)... |
I am seeking a clarification of the concept of Incoherence within the MC framework.
Specifically,
1) the literature mentions the application of a "strong incoherence" given a set of assumptions. Intuitively, how is the strong incoherence different than "incoherence".
2) Does the Restricted Isometry Property have any... |
I understand that you need to sample at a frequency greater than or equal to the Nyquist frequency to reconstruct a signal/aliasing etc. I read though that a "more conservative" approach is to sample at 2x-5x the Nyquist frequency (Rampil '98, on EEG processing). Assuming signal bandwidth doesn't extend beyond $F_N/2$ ... |
I'm working with AEC of Speex. The algorithm is based on the MDF adaptive filter + an adaptive learning rate. I'm using it like a ANC and it works very well.
Does anybody have some material, as block scheme, data flow diagram of the AEC of Speex.
I have read the documentation but it is not very useful for the dsp conce... |
I am trying to improve an edge following algorithm developed by some students who did a project at my work. The algorithm is supposed to make a robot follow a line with the use of a camera. Their approach was to detect edges with Canny's algorithm and then extract lines from this image using the Hough transform. Three ... |
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