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I don't if I am posting my question in the correct place. I have tried hard to solved it but I am unsuccessful.
I have an structuring element shown below:
1 1 1
1 1 1
1 1 1
My image is:
11 11
11 11
11 11
11 11
1111111
11 11
11 11
11 11
11 11
11 11
I have to apply erosion using the above image and s... |
I want to eliminate spurious peaks of Hilbert transform for finding glottal closure in the linear prediction (LP) residual. I have the following 4 steps:
Down-sample.
Hilbert Transform.
Identify Peaks in Hilbert Transform.
Consider this hypothesis that time gap between two successive glottal closure instance is not li... |
I am trying to get the position data from an IMU 6 Degrees of Freedom (MPU-6150) board using an Arduino Uno. I am trying Jeff Rowberg’s example MPU6050_DMP6.ino that may be found here.
This is the first time I am working with an IMU. I have no idea whether it's possible to get the position data using the using quaterni... |
i am working on simulation of an article "A Real-time ECG Data Compression Algorithm for a Digital Holter System". This algorithm has five steps "Difference=> ECG Signal=>Classification=> DCT=> Window Filter => Huffman" . it described window filter as "In order to implement a high compression ratio, the DCT window size... |
The cyan plot is a spectrum of 50 Hz, and the magenta one is a 50.1 Hz sine wave (having amplitude 0.7). Both are sampled at 1024 samples/s. I performed a 1024 point FFT to get this spectrum.
Why is only the 50Hz spectrum a single value? Why does the 50.1 Hz sine consist of other frequencies apart from 50.1 Hz; where... |
I would like to create an algorithm that, given an iddata, estimates different armax models with different na, nb, nc, nk, and finally show the compare plot.
I know how to use the armax estimation function, but how can I store the model in a variable in a such that I can compare them later?
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I'm testing an OFDM system implemented in fixed point. The data format is Q11. My system work's fine but I need to test it under some channel for evaluation of the design before field testing. The system will be tested on Rician fading channel but for now I'm working on AWGN testing first. I asked a similar question he... |
I was looking for a more efficient way of finding the magnitude and phase of a signal at a certain frequency without performing an FFT because it produces more information than I need and I came across this comment on the Matlab message board:
http://www.mathworks.com/matlabcentral/newsreader/view_original/250421.
If I... |
After so much search on google i find out few things hope so that correct for detecting Left ventricles of hearth and draw the Region of Interest(ROI) on MR Modality .
Can you give me please more detiails idea about "Morphology Algorithm" for tumore detection in heart of LV
|
If i add the two tones then the result if two peaks are added together will be greater then the the sample max (2^15).
For the life of me I couldn't think of the right words to google. Think i have google block.
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I am collecting a 15Hz sine wave at 1500Hz. Therefore, every 100 points of data collected should contain a single complete sine wave. If I sample for 10 seconds, I will have 15000 points of data. If I perform an FFT on this data I will have very high frequency resolution. However, If I wanted to divide these 15000 po... |
It could seem an easy question and without any doubts it is but I'm trying to calculate the variance of white Gaussian noise without any result.
The power spectral density (PSD) of additive white Gaussian noise (AWGN) is $\frac{N_0}{2}$ while the autocorrelation is $\frac{N_0}{2}\delta(\tau)$, so variance is infinite?
|
How to detection of the boundary of the left ventricle from MR images, the crucial step is to determine the region of interest (ROI) or the center point (CP) of the left ventricle.
I just want to show this marker over the Left ventricle of heart of MR Dicom Image
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I was supposed to find state space representation and its matrices of this system:
and I have no idea, how to do this. We were told not to transfer the system to time domain, but I can only do state space representation from time domain schemas.
When I tried to solve this, I got matrices
$$A = \left( \begin{array}{cc... |
I experiment with the coefficient quantization in an IIR filter. I change some values like the passband ripple and also how many bits i want for the quantization method. In the first picture i have 1db passband ripple , 'round'and 7-bits as the quantization method:
In the second picture i have 1.3db as the passband ri... |
I want to implement an IIR filter. The filter is going to implement by cascading second order sections. Below there are 3 pictures that implement this filter using:
Denominator - Numerator using Tapped Delay Line Structure 2:
Numerator -Denominator using Time Delay and Accumulate Structure 3:
Denominator - Nominator ... |
I want to detect the regions of voice in a song. Currently, I am using the fact that the vocals in the song are recorded as center panned. So, I am removing all center panned audio from the song. However, this removes other instruments such as bass also. So, it sometimes gives false indication of voice being present.
I... |
I have noticed that the spectrogram for a audio signal is often calculated in different ways when the result from the DFT is transformed into real numbers; some tend to compute $|k|$ or $|k|^2$ while others use the formula $Re(k)^2 + Im(k)^2\ \ $ for each DFT bin $k$. What is the theoretic difference between these and ... |
The optimal binary detector for asymmetric, equiprobable PAM pulse normally have structure as shown below,
But I cannot understand what is the multiplier and the integrator using for.
Why the detector need to care the signal difference and multiply with the received signal?
Why can't I directly put the received sig... |
suppose that we have a wave which has some peaks in some indexes like 2,12,25,... and otherwise zero.
I want to extract the time lag or lags between peaks of a signal. could anyone help me ?
|
Does anyone knows can i find a FFT of 1536 length input. Its a specification given in 3gpp Lte and we need a transform of 1536 input size which is neither a power of any number i would say. I just need a theoretical idea.
|
Long story short, I have a sinusoidal wave with DC offset going in with a given frequency of say 100 Hz and $\phi = 0$ (which I'm guessing is really $\pi/2$ since it is a sine wave not cosine). I get an output of the same frequency and, using FFT, I would like to find the phase between these two. What I did is I found ... |
@Jim Clay: No if the filter varies slowly, then the clicks go away. So I guess the filter needs to vary slowly to avoid audible clicking sounds.
I created a test in matlab where I'm comparing time-domain filtering with frequency-domain filtering. The two operations produce the same output, but I get clicks in the audio... |
I am trying to understand and visualize the concept of a covariance matrix. Suppose I have a matrix:
$A = \begin{pmatrix} 2 & 3 & 4\\ 5 & 5 & 6 \end{pmatrix}$
how do I calculate its covariance matrix, and what is its practical meaning?
(All I was able to understand by now is that on the diagonal of the covariance matr... |
I have a noisy signal with jumps to high value and back to normal for limited time bins (40 for example) which I want to filter to have a smooth signal with no jumps. Here is the graph, the blue one.
I tried moving average filter (n=20), the green one. Also tried window based (kaiser) LPF with cut-off freq=0.01 (I ch... |
I have already asked this question 2 times, but still i am confused because this is not available anywhere on the internet and explained properly. I found some information in this document and i am going to apply their method , but confused to understand. here is the document Altera Application Note 480: 1536-Point FFT... |
I have been asking around if the way I was extracting phase shift (lag) was correct, and I ran into some trouble.
So in general, given 2 arrays of data of the same length, representing:
the input sinusoid and
the response, also a sinusoid.
So not knowing the functions that produced the input (but knowing the frequen... |
Hi guys, I've been reading some papers on - how to remove ghosting artefacts from the Fourier Slice theorem applied to a 3D discrete image volume. The papers mention that in order to remove ghosting artefacts "employ a better(and typically wider) interpolation filter".
Currently, in Matlab I fft a 3D image space volume... |
I heard that the padding with zeros is mainly for efficiency and speed, other than that, is there a downside to use fft(signal) instead of fft(signal, N) where N is a power of 2 assuming signal length is not a power of 2.
Thank you
|
I have to implement a second order filter in DF1 or DF2. I should avoid housekeeping operations by suitable addressing strategies.
I don't understand what "housekeeping operations" means. Anyone could help me or propose a source i can find more information?
EDIT :
I use DF2 for the second-order filter:
As pichenettes... |
I am trying to convert from a floating point number to a Q-notation signed 16-bit fixed point number (e.g. Q15). How can this be done in C or Java? I searched online but could not find any available code.
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I'm learning DSP slowly and trying to wrap my head around some terminology:
Question 1: Suppose I have the following filter difference equation:
$$y[n] = 2 x[n] + 4 x[n-2] + 6 x[n-3] + 8 x[n-4]$$
There are 4 coefficients on the right-hand side. Are the "number of taps" also 4? Is the "filter order" also 4?
Question 2... |
I have 2 sin waves:
Fs = 1500;
N = 100;
t = (0:N-1)'/Fs;
sinwave = sin(2*pi*15*t);
sinwaveshifted = sin(2*pi*15*t + 1.3);
If I get the magnitude at 15Hz using an fft (need to use an FFT that is a power of 2 for reasons beyond my control):
fftsin = fft(sinwave,128);
fftsinshift = fft(sinwaveshifted,128);
% 2 correspo... |
What is the difference between an impulse and a step response diagram?
|
Does anyone have any experience designing wideband IIR Hilbert transform filters with audio applications in mind? I am using the filter for single side band modulation audio effects discussed in this paper :-http://www.mikrocontroller.net/attachment/33905/Audio_Hilbert_WAR19.pdf.
I tried to implement the design techniq... |
I have been working on a simple low pass filter for < 100 Hz metering in my application. But so far, I am struggling with the theory behind it all. It's cool that I got it working, but I'd really enjoy it if I knew how/why it is working.
I found the following code:
void getLPCoefficientsButterworth2Pole(const int sampl... |
I am trying to bandpass filter a signal using a butterworth filter, but I am getting the following result (click to enlarge):
The original signal is the blue and the filtered one is the green.
Why does the first part of the filtered signal (from 0 to 50) is changing so much from the original? Can you help me understa... |
I have two plots, each having frequency as x-axis and Gain as y-axis. By taking one data set as a reference I have to calculate similarity between them.
The graphs have same values on the x-axis and have same range on the x axis
Can 2D Correlation or Co-variance can do a decent job ? or
should I opt for Fréchet distan... |
I have a discrete signal. I want to do sinusoidal regression to estimate the parameters such as amplitude, phase, frequency, etc.
It is important to note I only have the signal values and have no time when these values were recorded.
How could I possibly do the sine regression?
Thanks.
|
INPUT
x[0] = (0.00 + j 0.00)
x[1] = (1.00 + j 0.00)
x[2] = (2.00 + j 0.00)
x[3] = (3.00 + j 0.00)
x[4] = (4.00 + j 0.00)
x[5] = (5.00 + j 0.00)
x[6] = (6.00 + j 0.00)
x[7] = (7.00 + j 0.00)
x[8] = (8.00 + j 0.00)
x[9] = (9.00 + j 0.00)
x[10] = (0.00 + j 0.00)
x[11] = (0.00 + j 0.00)
x[12] = (0.00 + j 0.00)
x[13] = (0.0... |
I have an image like the one above, and what I need to find are the red circled areas. They are caused by how the nerve layer is in the eye, and I have no idea what method to use to extract them. (I am looking for a brighter background from the nerve layer)
I have found all the blood vessels, so I can ignore those are... |
The periodic square wave,
$$x(t) = 1, |t| < T_1 \\
x(t) = 0, T_1 < |t| < T/2$$
What I wonder is ..
(1) If $T_1$ is constant and $T \to \infty$, is x(t) a constant function?
(2) If $T_1$ is constant and $T \to 0$, is this non-sense?
(3) If $T$ is constant and $T_1 \to 2/T$, is x(t) a constant function?
(4) If $T$ is con... |
I have been trying to implement an algorithm for successfully counting cars in an image. I have tried implementing A Method for Vehicle Count in the Presence of
Multiple-Vehicle Occlusions in Traffic Images
It estimates the background from a set of various images. I have looked into various other techniques for this pu... |
I need to transform this function:
$$
x(t) = 4\sin(20\pi t ) - 5\cos(24\pi t ) + 3\sin(120\pi t )
$$
into a sequence $x(n)$ given that the sampling frequency should be 50 Hz. So that means the sampling interval $T = \frac{1}{50} = 0.02\ s\ $. So would the sequence be:
$$
x[n] = 4\sin(20\pi0.02n ) - 5\cos(24\pi 0.02n )... |
I am working on the implementation of a location recognition pipeline as detailed in the section 5.4 of the paper Leveraging 3D City Models for Rotation Invariant Place-of-Interest Recognition:
Vocabulary tree + Inverted File System + SIFT detector/descriptors (discarding features not lying on a building using geometri... |
I have a bunch of short axis MRI images of the heart(Left Ventricle), in dicom format.
I want to do a Automatic segmentation of various structures in these images.
Does anyone know of a program I could use? (a C# would be fine also)
I basically just need a program which allows me to select a couple of
points on the ima... |
I am currently trying to count cars using OpenCV 2.4.4 with a HOG descriptor. Because a model for car detection is not available in OpenCV, I am creating my own model using SVMLight and a dataset of cars from the INRIA car dataset here (positive and negative samples).
I created the model using the positive and negative... |
What is the definition of Generalized Cross Correlation?
I need it for estimating the delay (time offset) between two audio signals. The time offset between the two signals might be as large as 500ms.
|
I have seen that in OFDM, the sampling rate is much lesser than twice the bandwidth of signal. why is it so? How is sampling rate calculated in OFDM? Does it depend on the modulation type used inside like QAM or BPSK. Does it relate to FFT length by any chance.
|
I am trying to read OFDM synchronization paper by Schmidl and Cox. The math is slightly overwhelming.
What is the formula that they use in implementation that distinguishes between coarse estimation and fine estimation?
Secondly once fine frequency offset is done, how do you determine that you need to do coarse frequ... |
I'm looking for an algorithm which can accurately estimate the time-offset/delay between two audio signals in real-time. One audio signal is a loudspeaker signal and the other signal is the microphone signal.
Audio is processed in blocks of 10ms (sample frequency 16kHz).
There is light office noise embedded in the si... |
I want to convert the two's complement, 16-bit output from an ADC into a generic fixed-point format, $\rm Qm.n$, where $\rm m$ is the number of integer bits and $\rm n$ is the number of fraction bits.
Any hints/procedure to do this job would be appreciated.
|
I use the below matlab function to design the unquantised coefficient set:
[b,a]=ellip (4,2,60,2*[0.2,0.3]);
and then i use the matlab finction "tf2sos" to convert the direct form coefficients to a set of a factorized second-order sections:
[sos,g]=tf2sos(b,a);
My result is:
The normalized version of the differenc... |
Windowing is a popular method of reducing the spectral sidelobes of OFDM. A popular window usually used for this purpose in the raised cosine (RC) window, because of its tapered and smooth edges. I understand the concept of windowing and why it reduces the sidelobes, however what I don't understand is this: "windowing ... |
Anybody knows what is the best pdf for periodic signal ?
I have signal with $480$ samples $\left(16000*0.03=480\right)$ but I could not find the exact pdf for that. sometimes it appear like 2 mixed Gaussian.
|
I am required extremely fast to fill the gap and learn Fourier Transform with application in Computer Vision.
It was easy to find mathematical aspects of Fourier Transform, but I am more interested in application in Computer Vision for a newbie with a lot of examples.
If you familiar with a textbook or have any other... |
so there is this paper I'm reading and trying to understand fully: Towards Practical Identification of HF RFID Devices
http://dl.acm.org/citation.cfm?id=2240276.2240278
I don't want to link the PDF here, but maybe you can catch it from ACM or somewhere else.
Now what we need to know: They record a signal for $5 \mu s$ ... |
Currently I'm working on a project to record ELF (Extremely Low Frequency) signals from a receiver. The highest frequency of interest is about 30 Hz and the system should have a flat frequency response. I'm looking for a simple solution using an 8 bit ADC and I wonder how the resolution can be improved using oversampl... |
I use matlab function tf2sos to convert digital filter transfer function data to second-order sections form:
[sos,g] = tf2sos(b,a)
Taken from the help menu of the matlab:
G is a scalar which accounts for the overall gain of the system
My question is: Because G is the overall gain if i draw the block diagram is goin... |
I am a long time software engineer but have practically zero experience with signal/audio processing. I am interested in learning about signal processing via a use-case we have for one of our audio components. NOTE: this is just a learning exercise for me... it is not a priority that the end result be useful.
We have a... |
Sometimes the terms 'Fourier domain', 'complex frequency domain', 'Frequency domain' and 's domain' are used interchangeably.
Take those answers here for example: http://answers.yahoo.com/question/index?qid=20090924230321AAxvnJg
Can you really use them interchangeably all the time, without being technically wrong? So, ... |
I know overlap save and overlap add are used for long data sequence filtering. Are there any other similar or better techniques like these?
|
I am computing the fourier series of the following function between $[-0.5, 0.5]$
$$\displaystyle f(t) = \frac{1}{2} - |t|$$
According to the definition of Fourier Series the coefficients are given by
$$\displaystyle \hat f_n = \frac{-(-1)^n}{2\pi ^2 n^2} + \frac{1}{2\pi ^2 n^2}$$
That's easy to calculate. So far so go... |
I am performing a system identification by striking an experimental model with an instrumented impact hammer and measuring the strain response.
The timebase for the recorded signals is 5 microseconds and the hammer pulse width is typically around 200 microseconds. The data are recorded for 20 milliseconds prior to the ... |
I am confusing when I find the lpc roots magnitude, f1=0.9313, f2=0.9308, f3=0.91 and f4=0.95 but when I look at LPC envelope the f1 is higher than f4 in term of magnitude in dB.
Could anybody explain it ?
I have wrote following MATLAB code :
frame=wave((i-1)*frame_length+1:(i*frame_length));
a=lpc(frame,30);
r=roots(a... |
I want to implement Wavelet Transform from the scratch, that mean breaking the wavelet transform into its equations to implement in any Programming language. Matlab Comes with built-in functions to implement Wavelet Transform but It is really hard to understand which processes are exactly involved in the implementation... |
I asked myself how to compute dBFS (dB full scale) from a value of sample between 1 and -1?
and in general?
|
I use discrete Fourier transform for digital image processing purposes , but I don't understand basic concept behind it. For example :
What information exists in frequency domain?
What is difference between spatial domain and frequency domain?
|
The question I have is related to image quality evaluation. Suppose you are supposed to perform OCR (Optical Character Recognition) on a set of document images. Some images, however, are very blurring, and for these images blind deconvolution is needed. However, there are good images that do not need deconvolution. The... |
My question is regarding the input size of a signal which is not a power of 2 and we have to take the fft of it. Some solutions say that suppose if we want to take the fft of 1800 we should zero pad it till the length of 2048 to make it power of 2 and then apply the radix 2 algorithm. But there are other solutions as ... |
I'm trying to implement a multipath ricean channel model. I had found several resources but I'm kind of lost with too much information. What I have from now is the following step:
1- Generate the signal without noise/channel
2- convolve with the channel response
3- add white gaussian noise
My problem is how to genera... |
Given a sound clip (whose source is an FM radio signal) I'd like to measure how different from background noise (the "in-between channels" noise) the sound clip is, so that I can compare the "non-noisiness" (and thus the quality) of two radio signals. Is there a good way to do this?
|
I want to up-sample my input signal by a factor of $2$. I saw zero padding followed by low pass filter method being used in few cases. But still I need some help in this.
Say I have $10$ input samples and my interpolating factor is $2$, then if I am doing interpolation using above method, will I get $20$ samples at t... |
I am trying to extract only urban aeas from satellite imagery Using Matlab. I have applied some edge detectors like canny and also used morphological functions and I am able to extract all the boundaries ,but not able to seperate them from various urban aeas like roads, trees and waterbodies.
Please help i can share th... |
I have read in my textbook and i don't understand this part, that the Direct Form II implementation of an IIR filter is a bit tricky in memory addressing.There is need of clever memory addressing. Memory is the place that we store partial summations(states of the filter). Anyone that understands the problem and explai... |
Can anyone tell me how does bit reversal order technique works for FFT when you are trying to break down the FFT to small sizes. Like I want to only for the when the index is odd because that involve some reverse carry propagation, but when the index is even you just add N/2 to it to get your reverse index r. An exampl... |
I'm trying to understand this code, however I'm unable to deduce the meaning of reds and trans. I would be glad if you can help with it:
import sys
import Image
# pixel constants in hex
trans="\x00\x00\x00\x00"
# reds most to least, the last is trans and used as a noise factor
reds=("\xf1\x34\x00\xff" , "\xfa\x00\x2f\... |
I have a wav file which is gained by playing 2 different wav files with 2 speakers and recorded with mic. If I want to get one of the original signals using the resulting wav file and the other original wav file, is it possible to do this? I checked online and only got blind signal separation algorithm. Thank you in ad... |
Three methods of estimating maximum signal level at a node:
Worst Case Min-Max (L1-Norm)
Mean Square (L2-Norm)
Frequency Domain (Linfinity-Norm)
In matlab i found the code for L1, L2
Hfull=[h(r,:),conj(fliplr(h(r,2:end-1)))];
L1(r)=sum(abs(ifft(Hfull)));
L2(r)=sqrt(mean(Hfull.*conj(Hfull)));
For Linf is corr... |
Question Details:
There are a variety of image processing algorithms useful for detecting reference marks (contour tracing, FAST, MSER, Haar Feature-based Cascade Classifier, or even just Harris Edge detector, etc).
There are a variety of simple geometric shapes one might pick for use as reference marks (circle, square... |
This is just an idea. How can we model the kalman filter to get the state representation in continuous space when the observations to the system are actually from the discrete space. The discrete observation, for example, can be sequence of digits (0-9) where the digits themselves do not have numerical meaning but the ... |
Here's my problem. The input signals $x$ and $y$ will be having the time value aligned with each other. However, the data are not evenly sampled. I would like to calculate CPSD of both signals.
The solution comes to my mind is as follow
$R_{xy}$ = cross correlation of $x$ and $y$ (I'm not sure how to do it with uneven... |
I am currently working on the FFT function in MATLAB. In this function, i require a 1024 number of data Time Domain plot and produce a 400 number of data Frequency domain plot.
After applying the FFT function, i obtain a result of a graph going up to 6000Hz.
However, the Frequency band produced is too big and I only wa... |
I'm trying to understand how DFT works exactly. However, when experimenting around, I compared both Matlab generated FFT result with a dummy approach result and I get similar result.
However, the Imaginary part of both results are negated. The code below is my implementation. The implementation of dummy approach is bas... |
I have a raw infrared image with 16384(2^14) intensity levels.The raw image must be compressed into one with 256(2^8) intensity levels. In order to retain the original details( small blocks) , the compressed image must be enhanced. Which the best method I should select?Someone advice that "histogram matching" method sh... |
I want to know how I can determine whether a series of data is periodic or not.
I want to use Fourier transform/series. My data looks either aperiodic
[111100001111000110010101010000101]
or periodic
[11001100110011001100]
and I need to decide which it is automatically. What types of analyses or calculations can ... |
I am trying to implement speech recognition using backpropagation algorithm, and I have been following this paper.
I have followed it all the way, except that it tells me to zero-pad the coefficients when there is an empty slot, because the MFCC coefficients of two samples are not the same.
Now my question is, do I ra... |
I use this snippet of python code to transform data to Fourier phase and magnitude and then retrieving original data.
from random import randint as RI
import numpy.fft as FFT
import math
w = 4
h = 4
random_range = 255
vals = [[] for i in range(h)]
for i in range(h):
for j in range(w):
vals[i].append(RI(0... |
I am implementing phase correlation algorithm to determine shift between two images.
It generally works, but I am not sure how to interpret the resulting shift.
Pseudocode:
ft1 = FT(input1)
ft2 = FT(input2)
for each pixel p
product = ft1[p] * conj(ft2[p])
cross_power[p] = product / abs(product)
cross_correlation ... |
I've used Kalman filters for various things in the past, but I'm now interested in using one to track position, speed and acceleration in the context of tracking position for smartphone apps. It strikes me that this should be a text book example of a simple linear Kalman filter, but I can't seem to find any online lin... |
I have quite a straight-forward question. What I aim for is the generation of a certain set of random numbers with a normal distribution (mu = 0, sigma = 1).
Now, the best way to approach the above mentioned gaussian bell is to generate quite a large number of samples.
Unfortunately, in my case, I can only generate a r... |
In my filter specs, the passband gain is given as 6.02db. How do I achieve this?
Usually all texts give the passband gain as 0db (as well as passband ripples if any).
|
I want to port the following FFT code on a microcontoller having a very little floating point capability. I want to perform FFT of the output from 12 bit ADC. Since my values are all 12 bit numbers it is wasteful of storage and processing power (I do not need very accurate FFT output) if I do every thing in double as t... |
I am an image processing engineer. I my work I often find that there are many aspects for my code to be improved. Who can recommend a book which focuses on optimization methods for image processing? Any advice will be appreciated!
|
I am confused about this Q&A : Interpolation by factor of 2
If my input signal
$$
x[n]= x0,x1,x2,x3
$$
then according to the threads explaination my $v[n]$ will be
$$
v[n]=x0,0,x1,0,x2,0,x3,0
$$
$x[n] \rightarrow$ zeropad $\rightarrow v[n]$
$v[n]\rightarrow$ LPF $\rightarrow y[m]$
My $y[m]$ is the output of LPF whic... |
I'm trying to work out what the correct solution is for a DCT of a 4x4 checkerboard matrix (let's call it A) and for a matrix of ones (let's call it B). So A is:
0 1 0 1
1 0 1 0
0 1 0 1
1 0 1 0
And B is:
1 1 1 1
1 1 1 1
1 1 1 1
1 1 1 1
With Octav... |
I am trying to thin foreground pixels of a finger print until they are one pixel wide without using the built in bwmorph and thin operations.
Can anyone explain to me how to write them in matlab without actually using the built in function. Thanks.
|
I have a bunch of 8x8 images which have been vectorized and formed into a 64xN matrix, X. I'd like to take the 2D-FFT of each of these images without reshaping each column of X into an 8x8 image and using fft2.
Any suggestions? I've tried playing around with properties of kronecker products and vec() operations but h... |
Thanks to everyone who posted comments/answers to my query yesterday (Implementing a Kalman filter for position, velocity, acceleration ). I've been looking at what was recommended, and in particular at both (a) the wikipedia example on one dimensional position and velocity and also another website that considers a si... |
Could anybody tell me a method to calculate the "4 Hz modulation energy" for a speech signal?
Thanks.
EDIT
I get some details : 4 Hz modulation energy: Speech has a characteristic energy modulation peak around the 4 Hz syllabic rate [3]. We use a portion of the MFCC algorithm [4] to convert the audio signal into 40 per... |
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