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I have an elementary function $p:\mathbb{R}^2\rightarrow\mathbb{R}$ which (locally) represents an image. It's a polynomial, and its the result of the following 2D convolution: $$p=f\star G\star \mathbf{1}_{A}$$ where $G(x,y)$ is a 2D Gaussian, and $\mathbf{1}_A$ 'box' function: $$G(x,y)=e^{-\frac{(x^2+y^2)}{2\sigma^2}}...
I have one image. I want to segment it by simple approach. Could you suggest to me the best method to segment it? I think one of them is gradient or FCM. But I am not sure. Please see my given image file here and segmented image file below
Why do some tables say that Laplace (or Fourier?) inverse of exponential is a time-shifted delta pulse \begin{align} \delta (t) &\overset{\mathcal F}{\Longleftrightarrow} 1\\ \delta (t-t_0) &\overset{\mathcal F}{\Longleftrightarrow} e^{-j2\pi f t_0}\\ 1 &\overset{\mathcal F}{\Longleftrightarrow} \delta (f)\\ e^{j2\pi ...
In Radar signal processing, one has to separate clutter(unwanted signals) from moving target signals. The picture given below is an example of Radar Signal Power Spectrum(taken from the book Radar Systems Analysis and Design Using MATLAB by B.R. Mahafza). Why target return is coming in between 0(DC) and $f_{r}$ where...
(copy from Proper meaning (and origin) of PCM (pulse code modulation)? - Stack Overflow): Apologies first, as StackOverflow is possibly not the right forum for this question; in that case please suggest a move in the comments. But it is motivated by usage in documentation like ALSA project - the C library reference: PC...
I search to display a time-frequency signal with an original discrete temporal signal (sampling step = 0.001sec). I use Python and the library Scipy.signal. I use the function cwt(data, wavelet, widths) to do a continuous wavelet transform, with the complex morlet wavelet (or gabor wavelet). First step: Obtain a scale-...
I have been studying about boundary discontinuity in DFT and why it is not used as transform in image compression. What i understand is that DFT is N periodic and that causes a discontinuity at the boundary. This means that more number of terms are needed to represent the function and the rate of convergence is slow. ...
I perform some tests where I run cepstrum pitch detection algorithm over polyphonic sound. Result shows that there is almost always occurrence of lowest frequency of tones played simultaneously. I think this is because the lowest frequency has the longest period. But I do not know if it is true or why it is so. Can you...
I have a bunch of audio files that have both speech and music parts (think radio). For instance, a given file might consist of the following sections: music --- speech --- music --- speech --- music --- speech --- music My goal is to detect when the file changes between music and speech, and cut away the music parts l...
Basically I want to perform an FFT to find the frequencies of the sound, then stretch the frequencies (for example a stretch by 2 means copy the value of the fourier coefficient X(n) to X(n/2) then use an inverse FFT. Will this work ?
Let's suppose I have a signal $F(t)$ that is periodic, with two periodicities $P1$ and $P2$, with $P1>P2$ Suppose that I know the values of the two periodicities. Using the Fast Fourier transform I can show the two values as peaks in a power spectrum. Now, let's suppose the second periodicity $P2$ (the faster one), h...
How are Radix-8 and Radix-2^3 FFT are related? Are they same? I got some more details in IEEE paper by S.He and M. Torkelson, "Designing pipeline FFT processor for OFDM (de)modulation," in Proc. Int. Symp. Signals, Syst., 1998, pp. 257-262. FFT Radix 2^3 has complexity of radix 8 but has a structure as that of radix-2....
What is the relation between the AM modulation index and the SNR? (I plotted this and it came as the shape of the exponential graph) Also what is the relation between the channel noise power and the SNR? I plotted this and I got the shape as this graph: (ignore the labels of the axis)
Is there a name for a filter that is using a moving average on the frequency domain? I am searching for such a filter to filter out long "pads" (synths) from music. Do any of you know about such software?
I have a problem with Otsu's method. I would like to implement histogram to compute how many ones' and zeros'. For Otsu's method, I have a code from a book and it is running without problem. This is the picture . I've implemented a code to compute the histogram. This is the output of my code. ------------------ 0...
I am using openCV to find a particular object, and am trying to find a way to make it locate the object consistently. The object's location will be determined by taking 30 frames and trying to find the best guess based on where it is found in each of the 30 frames. Both the camera and object are (mostly) stationary (th...
I am investigating autocorrelation of electrical noise as part of an undergraduate experiment (as detailed in http://physlab.lums.edu.pk/images/a/ab/Correlation.pdf). I sampled noise voltages using an 8-bit AtoD from a noise generator (whose description was not provided). I first sampled the noise directly, then I samp...
I am developing a software for fundamental frequency tracking. For this purpose, I have designed a function which calculates autocorrelation over the signal and a second function which, based on autocorrelation result, finds the peak corresponding to the fundamental frequency. However the code I have written is not ver...
As a follow on Estimate the Transfer Function of an Unknown System which os estimating a transfer function of an unknown system using a Wiener filter, How would you put a minimum MSE criteria on how well the estimated filter weights matched the actual transfer function of the system? [Suppose you needed the MSE to be ...
Folks I've written a small utility that extracts some (and only some) header information from Tifs. To test it fully I need some Big Endian Tifs, but I cant find any, and I can't find anything to generate them. Does anyone know of a public repository for tifs with big endian encoding? Or of something that can generate ...
I am very new to DSP so please excuse any brevity... I have to design a 4x4 CMOS camera array and stitch the images together for further processing... I have tried doing that with opencv but the frame rate count drops drastically... So i want to do the Stitching part using some Image Processor Kit which can take multip...
I am learning about FIR filters and I'm confused. I am trying to find out different types of FIR filters. Is direct form and n-tap FIR filter the same? What does transposed FIR filter do?
Most of the resources I found online go the other way. If I have the transfer function $$ H(z) = 1 - \cos(\theta) \, z^{-1} + z^{-2} $$ How do I get the difference equation from it, so that I can apply the transfer function to a set of data?
Consider the following scenario: Say, one machine is sending out a beep signal every 10 seconds in a very noisy environment. I have two sensors which detects these beeps independently. Device A is near to machine compared to B, so B always have a two samples delay compared to A. Since it is noisy environment, devices s...
Say I have 5 minutes of input and output audio. One method I know is to do FFT on windows from input and output. Then divide FFT output into bins and find average energy in various bands for all the windows. After this, take ratio of energy bands from output windows with corresponding energy bands from input windows. A...
Suppose that I am receiving RF signals contaminated with noise through antenna. This signal is digitally sampled. How can I understand the type of noise(Gaussian,Uniform etc). Any algorithm is there to test the type of noise.
I have the following problem: Let $x$, $y$ be finite real valued sequences defined on $0, \ldots, N-1$ and let $g$ be a non negative integer. Define: $$z_n=\sum_{k=0}^n x_{n-k}y_k, \quad \text{also on $0, \ldots, N-1$.}$$ Note that the sum is from $0$ to $n$, not $N-1$. In addition, the DFT of $y$ is known in closed fo...
I'm having some trouble implementing a Kalman filter in MATLAB. I have an Android phone connected sending data from accelerometer for 10 seconds. After i have the data I take out the x-axis vector. What I want after that is to get better readings using the Kalman filter. The output I am looking for is both the accelera...
I'm trying to get a handle on the importance of the error feedback term in noise shaping operation in typical audio dither algorithms. I'm thinking in terms of four signals. The original signal, the output (quantised) signal, the dither signal, and the error signal (the difference between output and original). A simpl...
I am new to DSP, and currently trying to implement a simple polyphase decimation program in matlab. M is decimation factor, x is input (for now it is a square wave 8096 samples), y_polyDec is result from polyphase, output is result from regular filter then decimation. code : FilterBank_LPF = fir1(127, 1/M); y_polyDec =...
I would like to compute how many zeros and ones (e.i. 255) in a binary image. The following code to generate a binary image by using Matlab. I = imread('images.jpg'); level = graythresh(I); BW = im2bw(I, level); imshow(BW) imwrite(BW, 'img.jpg'); This is my C++ code. #include <opencv2\core\core.hpp> #include <opencv...
In Audio Experts book, it's given as Sound pressure meters must be calibrated at factory to a standard acoustic volume. So what doth this standard acoustic volume means?
Why is MSK (minimum shift keying or FSK with a modulation index of 0.5) said to be a spectrally efficient form of FSK?
I understand the process of using a Sobel kernel for edge detection in greyscale images. The input is a greyscale image, and the output is a greyscale image. I'm having trouble, however, figuring out how to apply the Sobel kernel to a color image. Do I find the color gradient for each color channel (R, G, and B) then a...
I wanted to simulate OFDM system in MATLAB. I'm using Fixed WiMAX OFDM parameters in my simulation, where the number of data sub-carriers are 192, number of Pilot sub-carriers are 8, number Null sub-carriers are 56. Null tones can be obtained in MATLAB by generating zero matrix with the required rows and columns. % Nu...
let us consider represented of damping exponential model by prony method,there is source code y=zeros(1,N); for i=1:N y(i)=x(800*i); end d=zeros(1,N/2); for i=1:N/2 d(i)=y(i+N/2); end D=zeros(N/2,N/2); for i=1:N/2 for j=N/2:-1:1 D(i,-j+N/2+1)=y(i+j-1); end end a=pinv(D)*d'; muhat=roots([1,-a'])...
What is the difference between soft thresholding and hard thresholding. Where we use soft and hard thresholding in image for denoising. I understand that in hard thresholding, the coefficients below threshold value are set to zero and the value above the threshold is set to one. Please explain me about soft threshold. ...
I don't really know anything about signal processing. I need to find similairty between two 1-dimensional time series. I read somewhere that spectral coherence could be used to find similarity between two time series. But i am not able to get around how to use it. Can some one tell how i can use it to compute similarit...
I have a large data set that was collected by a 5 GHz oscilloscope, so I basically have an array of voltages with each index being a very small time step apart. I am looking for a specific signal in this data set that would only be present for a very small time frame and so most of the data will not contain this signa...
I can not understand the following sentences, could you please anyone guide me ? "Comparing with the embedding in audio signals, the usable bandwidth is greatly reduced, from 22.05 kHz to 4 kHz. According to Costa’s result, the corresponding capacity will be reduced to l/5 th accordingly." The costa formula is Cw,Cos...
here is my scenario: one speaker is sending a chirp, and a microphone is recording it as well as all following echoes. The function of the chirp is already know. My goal: subtract the original chirp from the recorded sound, and leave only the echoes. I think this questions is hard for me in these ways: To match the or...
I wonder if the following acoustic system is i) considered to be linear time-invariant, and ii) if it is fully described by it's frequency response? A speaker produces waves at the input location of a duct system with a branch and the output is measured after the branch point. The anechoic terminations completely absor...
I have two questions regarding signal processing: 1) In my analysis I generate a signal in the time domain with a frequency of 0.1 Hz. I apply the FFT and then determine the peak frequency of the signal, which should be 0.1 Hz. But I notice that when I simulate my signal for 200s, the peak frequency is always close to ...
I am working to make a music player with automatic genre classification application in Java. On Google searches I mostly found the works of George Tzanetakis who describe an algorithm from Music Genre Automatic Classification. I have big problems with obtaining the beat histogram. http://dspace.library.uvic.ca:8080/bit...
Since my algorithm needs to detect long, coherent edges, the estimated local magnitude P for each pixel is normalized with respect to the magnitudes in it’s local neighborhood W (i.e., a 5x5 window). $P_{normalized} = \dfrac{P - \mu _w}{\sigma_w}$ where $\mu _w$ and $\sigma_w$ denote the average and the standard deviat...
I have below doubts, so confusing! As I don't want to assume from what I read, I am asking for help here! Are Second Order Sections another name for biQuads ? If I have 2 single pole transfer functions, then cascading them equals multiplying the polynomial; such as below? $$H_1 =\frac{b_{10}+b_{11}Z^{-1}}{1-a_{10} Z^{...
I read in a paper that a sound signal can be delayed by D degrees. How can such a thing be possible? Isn't sound signals expressed only in terms of the time axis? Thanks in advance
I have a Output Curve of following characteristics https://i.stack.imgur.com/1fTVl.jpg Following is the data which represents the output curve cited above 0 1228.15406117455 1213.71796132282 1199.44623423626 1185.33715849069 1171.38902630825 1157.60014358826 1143.96882993237 1130.49341866405 1...
I was expecting that when I take (for example, in MATLAB), s2 = fft(s1/norm(s1)), the norm of s2 is not equal one, norm(s2) $\neq$ 1. I cannot figure out why this is! Can anyone shed light on this?
I have to develop an underwater positioning system which has to determine the position of a ROV. Four buoys will be placed on each corner of a swimming pool. Each one of these buoys will be equipped with an hydrophone and a transducer (which emits acoustic signals). Both devices will be connected to a Raspi for perform...
I have a motor, that I want to analyse, using its sound, so I can its 'character' I was using the the simple fft, and figured out, that it wouldn't help, so I tried the order analysis, at least I incremented the speed of the motor and I can say it in spectrogram like here: My question isn't about codes but about theo...
I am working on LMS algorithm,i am not getting exactly how we get desired response of the filter that is compared with the estimated output
Can anybody tell me how to calculate the order analysis / spectrum of a signal? I've been searching in vain. I want to use MATLAB to analyse the signal / sound of a motor. For more information about the order analysis - here is an example Thanks in advance for any help!
I have $m$ complex-valued time-series $z_m(t_n)$, which I am currently analyzing. I started by calculating their PSDs and CSDs. These exhibit an unexpected peak at a given frequency, which is present for all PSDs and CSDs, signaling some kind of coherence between the time-series. Interestingly this peak disappears, whe...
I got an $N$, in my case 512, point FFT of a real-valued signal. Based on some calculation in my application I determine the parameters $k \in [1, N-1]$, the number of oscillations per period, $\phi \in [0, 2\pi[$, phase. I use these parameters and the function $\cos\left(2\pi\cdot \frac{k}{N}\cdot t + \phi\right)$ to ...
I'm curious about the feasibility of designing a noise shaping filter (but might generalise to any recursive filter) with the constraint that the most recent output samples aren't available for several iterations of the filter. The use case I have in mind is reducing the word length of an audio stream sampled at 48kHz,...
How do you determine if you are using analog or digital communications? For example when dealing with PCM and noise how can one notice any difference in the signals when using the two types of communications? Thanks
let us suppose that we have given following model $y(t)=A_1 \sin(\omega_1*t+\phi_1) + A_2 \sin(\omega_2*t+\phi_2) + A_3 \sin(\omega_3*t+\phi_3)+ \ldots +A_p \sin(\omega_p*t+\phi_p)+z(t)$ where $z(t)$ is white noise,we have everything unknown,$p,A_i,\omega_i,\phi_i$,what i want to remove noise using wavelet,i know that...
I encountered the following interesting computational problem in the field of computational statistical mechanics. https://www.dropbox.com/s/8yimd9uy5u0jrdj/dftpr.pdf thank you :)
I read from Wikipedia that signals obey the uncertainty principle. The product of the measured uncertainty in frequency times the measured uncertainty in time is greater than a constant. The localization of both frequency and time is not possible simultaneous. Can two people play a signal game with signal acquisition?...
I want to detect, in real-time, a 1-2 word spoken phrase within continuous speech, speaker independently, in hardware. The smaller the number of gates the better. What is the best way to do this? Are the requirements feasible?
I recently read a research paper that discussed how to correct lightcurve data (flux vs. time) by using the method as explained below: We have considered as outliers in the datasets the individ- ual measurements showing a point-to-point deviation in the two-point dierence function of the light curve greater than 3...
I'm hoping someone could point me in the direction of some signal processing methods to clean up my data. I'm collecting physiological data from human muscle (lower leg; gastrocnemius) during walking/gait. I know the sensor is picking up contractions, but the problem is it's so plagued by motion and skin artefacts I c...
I know that the error probability for antipodal systems is Q (√((2Fs)/No)) and for orthogonal systems it is Q (√(Fs/No)), however, can anybody give their derivation? Because I am finding different ones on the internet I am getting confused. I found pages 74-75 useful but I think their derivations are quite short here:...
For this type of input (top) and output (bottom) which of these equations is required for the SQNR: or
I am new to audio processing. I wrote a simple real-time playback program (when you speak on the mic your voice is immediately played on the output), however it is not working. I am using a 8Hz ADC with 128 frame size, and I output these 128 samples directly to audio PWM. It doesn't play the audio even though there's a...
what are the applications of discrete sine transform . can discrete sine transform be used in place of discrete cosine transform ? please state the reason. can it be used in compression.? regards, phani tej
I'm looking at the problem of integer upsampling (i.e. interpolation). This can be achieved by first increasing the sample rate by factor $L$ (i.e. inserting $L-1$ zeros between the original samples) and then low-pass filtering the signal (see e.g. Oppenheim/Schafer, "Discrete Time Signal Processing", Sect. 4.6.2). Whe...
I wrote a simple lowpass filter in python to run against lena. Now I'd like to add Gaussian noise to the lowpass filtered data and then run an inverse filter against the lowpass and try to get the original back (well, as close to original). I'm new to programming in python and not quite sure how to add noise and write ...
I work mainly in GIS (geographical information systems) but I would like to deepen my knowledge of remote sensing (analyzing imagery acquired remotely from airborne or satellite sensors). I was advised to learn as much about signal analysis as possible. I am assuming that signal processing and analysis refer to the sam...
I had asked a question last night in regards to how to process my data (Noise rejection / feature extraction) but I have a more specific question now that I hope someone can answer. As mentioned in my other post, I have a microphone based sensor to detect muscle activity in the lower leg during walking, however the da...
In the code I am trying to understand, there is an usage of Hann window multiple times (There's overlapping frames, the size of each is 512 and the overlapping factor is 75% so each time we capture 128 samples we build a new frame by shifting the previous one by 128 to the left). For each Frame we apply a Hann window ...
I am working on a signal processing assignment where I need to find out one common time domain signal from 4 observation. The math is like this: \begin{array}{lcl} y_1(t) & = & a_1(t)*x(t) + b_1(t) \\ y_2(t)& = & a_1(t)*x(t) + b_1(t) \\ . & & \\ . & & \\ ...
I'm looking for a dedicated DSP chip that will allow sampling and delayed replay of a signal, given the following parameters: buffer should be integrated configuration of the DSP chip (if needed) should be possible via host CPU (SPI or serial) starting and stopping the sampling/replay should be possible via a dedicate...
I want to calculate Morlet time and frequency resolution. the Morlet wavelet function is define as : $\psi(t)=\frac{1}{\sqrt{\pi f_b}}e^{j2\pi f_c}e^{-t^2/f_b}$ Note: I know the answers, but I don't know how to achieve it on my own.I want someone to explain the time and frequency resolution relation to me please. here ...
As it has proven in some paper and books the time and frequency resolution of Morlet wavelet is : $\Delta t=\frac{f_c\sqrt{f_b}}{2f_i}$ $\Delta f=\frac{1}{2\pi f_c \sqrt{f_b}}$ I don't know how these relations have obtained, I have asked it in detail in here.you can answer that if you want. but lets accept these r...
From my days at college I rememeber that LPC splits a signal into a series of slowly changing filter coefficents and an audio-rate residual error signal, such that the original signal gets reproduced when you run the residual signal as "excitation" through a time-varying filter which is controlled by those coefficients...
So I just read something that two signals are aliasing to each other. I know what a single signal is being aliased - that is when sampling rate is not higher than or equal to the Nyquist rate. But I am just confused of what two signals aliasing to each other mean. Let's say that each signal only carries single frequen...
let us consider following code function [sca_1,sca_2,sca_3,sca_4]=calc_wavelet(y,wname,scales,freq,fs) %y-input signal %wname-wavelet basis name %freq-test frequencies %fs-sampling rate TAB_Sca2Frq = scal2frq(scales,wname,1/fs); [~,idxSca_1] = min(abs(TAB_Sca2Frq-freq(1))); sca_1 = scales(idxSca_1); [mini,idxSca_2] =...
Thanks a lot to whoever downvoted my question for no good reason. I wrote an implementation of the Goertzel algorithm in JavaScript and made a demo that detects DTMF dial-tones. You can find the code here: http://github.com/ravenstine/goertzeljs The demo is below, and you can try it out if you have a web browser that s...
I'm sampling two signals with fundamental frequency of 50 Hz. I want to sample them at the same instant, but there's a delay between the first and the second conversion. I don't keep the previous values of the signals in an array, at most I can store 10/20 samples. I want to correct this delay. I thought of a FIR fract...
What is the bandwidth required to transmit baseband modulated signals like an antipodal and orthogonal signals?
I am attempting to use ICA (FastICA via scikit-learn) on EEG signals from seven electrodes per subject for feature extraction and identity classification – that is to extract signal which is related to a specific cortical area without noise and feed it to e.g. MLP for classification. As I currently understand it, I sho...
I understand that compression works by exploiting certain psycho-acoustical characteristics of our auditory perception, but I'm still confused. I mean ultimately, the waveform must be recorded regardless of what has been done to do it, and I was under the impression that all waveforms were born equal? Does removing a b...
We know that Heisenberg uncertainty Principle states that $$\Delta f \Delta t \geq \frac{1}{4 \pi}.$$ But (in many case for Morlet wavelet) I have seen that they changed the inequality to an equality. Now my question is when are we allowed to change the inequality to an equality: $$\Delta f \Delta t = \frac{1}{4 \pi} $...
I am totally new to image recognition and image analysis and have a project concerning detecting a smooth curve on a uniform background (white/black/any color) and then be able to follow this curve until its end. The curve may cross itself but the path that should be recognized should follow along the "smoothness" of t...
I have some code here that I don't quite understand. It calculates the energy for a given frame the following way: En=log(1+energy(32767*frame)) The energy function is defined as energy = sum(s.*s)/max(size(s)) Where 's' is te given frame. What I don't understand is why the frame needs to be multiplied by 32767, whi...
I am reading a paper in which they filter EEG signal using a notch filter to remove the power supply component of 50 Hz, then a high pass filter at 0.3 Hz to remove artifacts resulting from breathing, and finally a low pass filter with a cut-off frequency of 40 Hz. Is there any difference to simply using a band pass fi...
I know that similar questions have already been asked – I have read the answers, yet I am still not sure if I understand the topic properly at some points. I want to use FFT to calculate PSD estimate. FFT gives me a set of complex numbers that I want to transform into the PSD values. I am aware of the fact that only t...
The problem is here. Now I know the definition of mixed/max/min phase wavelets, whether the roots lie within the unit circle or not. Starting from n = 1, let $$ x_t = ( 5, 6) $$ $$ X(z) = 5 + 6z $$ This would have root -5/6, making it a max phase wavelet (since it's within unit circle). If we switch 5 and 6 around in ...
I'd like to sample band-limited Gaussian white noise non-uniformly. One way to approximate this would be to filter Gaussian white noise with a lower cutoff frequency and non-uniformly pick samples from this. However this approach would limit me to a fixed number of time-offsets and require many samples of noise to achi...
I'm having an assignment of computing narrowband spectrograms using MATLAB. And I totally have no idea about the code. Can someone help me writing the code and explaining what's going on to me. Lots of thanks!! Followings are the requirement: B.1. Load the recorded .wav file using the function “wavread”. Let y be the v...
I have a homework problem that I'm not quite sure how to complete. The problem is as follows: PROBLEM Write the definition of the Fourier coefficients, and show that $$f(t + \frac{1}{2}T) = f(t) \rightarrow c_{2n+1} = 0$$ $$f(t + \frac{1}{2}T) = -f(t) \rightarrow c_{2n} = 0$$ Also show that this is a consequence of th...
i'm trying to get an FFT of a signal my FFT has 1024 samples i added two sin functions one of freq 86,13Hz (2*44100/1024) and one of freq 129,19Hz (3*44100/1024) the WAV looks like that (for 1 period) : i then use this code to inspect the FFT FFTSize = 1024; Y = wavread('2sins.wav'); current_window = zeros(FFTSize,...
I'm trying to plot fft in python. I use pyalsaaudio for capturing audio in PCM (S16_LE) format. I use the ion() and draw() functions in matplotlib to have the fft plotted in real time. This is the program I wrote : import alsaaudio as alsa import numpy as np from matplotlib import pyplot as plot from matplotlib imp...
Let $\textbf{h} \in \mathbb{C}^{N\times 1}$, $\textbf{a} \in \mathbb{C}^{N\times 1}$, $\textbf{b} \in \mathbb{C}^{N\times 1}$ and $\textbf{c} \in \mathbb{C}^{N\times 1}$. The variable $h_i$ is defined as $h_i=\left(a_i+b_i\right)c_i$ $\textbf{a}=[1~2~3~4]^T$, $\textbf{b}=[5~6~7~8]^T$, $\textbf{c}=[9~10~11~12]^T$, Then ...
Last week I've been trying to implement the Order Analysis in MATLAB in vain. I've read a lot of docs about it like this one , but I still can't figure it out. This will be my last shot. Say I have recorded 10sec audio file of a rotating machine, the sampling frequency is 96Khz. In this 10sec interval, the speed of th...
I am trying to smooth a series of data in order to obtain a continuous function that could represent that given data set. It came out that the Savitzky-Golay method could be a good way. Now, I don't know much about smoothing and/or interpolate, but I managed to obtain a good result, at a first glance: The blue points ...
I have an EEG data set downloaded from physionet. I want to play that data into a brain monitoring device like for example BIS monitor or Narcotrend or any other similar brain monitoring device.I use Matlab GUI to visualize the data. Can I send it from a Matlab GUI to any one of the brain monitoring devices with a suit...
In this presentation on slide 15 there is a corollary: Rule of thumb: the dark noise must be larger than 0.5 Corollary: With a N bit digital signal you can deliver no more*) than N+1 bit dynamic range. *) You can if you use loss-less compression They give an example: Example : A 102f camera with 11 bit dynamic ran...