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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?
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 ...
I have read some about Karhunen-Loeve Transform (KLT) and its application to the field of seismic data processing. The method as I understand it based on decomposing the data (actually mostly used in image processing) using SVD on its covariance matrix and projecting the data back after manipulating one or some of the ...
I have tried Region based Segmentation - through Seed Growing, Watershed algorithm, and Graphcut algorithm, but none have been successful so far. I am using MATLAB.
If a system is given by a transfer function in the $z$ domain that has all poles and zeros inside the unit circle except for a factor of $z^{-1}$ in the denominator (pole at infinity), can it still be considered minimum phase? If not, how would I create an all pass system to neutralize this pole and obtain a minimum ph...
I am trying to implement Matlab's cross correlation function but I can't seem to replicate the results obtained when using Matlab's built in function "xcorr". Documentation for xcorr can be found HERE. Looking at the Matlab documentation it seems as though the cross correlation is implemented as $$ R_{xy}[m] = \left...
In a debate on the value of 24b/192KHz consumer audio (using a good brick wall filter, minimal jitter), a friend posed the argument: Nyquist never properly dealt with harmonics, which change sound pressures by riding a higher frequency on a lower frequency "carrier", increasing the propagation of the higher freque...
I have 2 vectors, $a$ & $c$, both of length $M$. I know they are related by $a*b=c$. My goal is to recover $b$. Obviously $b = \mbox{deconv} (c,a)$. I am only interested in the first $M$ elements of the convolution (i.e., discarding the last $M-1$ elements). I have two methods of doing this deconvolution in MATLAB and ...
Summary I need to remove artifact that appears strongly on all channels in my EEG data. It's already recorded (from another lab) so I can't use hardware solutions. Also: band overlap and no reference channel. Less Short I have 28 EEG channels with strong common-mode artifact (artifact amplitude is on the order of $10^4...
I'm developing an Artificial Neural Network based Speech Recognition System using MFCCs. Suppose I have 260 input nodes in the ANN, and this number of nodes corresponds to the number of MFCCs that I will use. During feature extraction the number of total coefficients vary with respect to the duration of the sound file....
I understand why there need an low pass filter before down-sampling, the sample frequency need to be at least twice of the max frequency signal, else there will be alias issue. But base on the Wiki page, we need an lowpass filter after up-sampling. Upsampling requires a lowpass filter after increasing the data rate, ...
I'm working on a audio sample recognition project, where first you profile a sound and then run the sound detection function. Currently during the 'profile' stage I'm grabbing copies of the magnitude frequency bin from the sound you want to detect (for a changeable pre-set amount of FFTs). During the detection phase I...
I am currently reading "A parametric texture model based on joint statistics of complex wavelet coefficients", by Portilla and Simoncelli. In this article, they use the "steerable pyramid" to represent the image. This uses a set of filters to decompose the image in a set of something around 16 subbands in the frequency...
I have a C# desktop application. Its purpose is to detect motion changes between consecutive frames. Currently I am using the excellent Emgu wrapper to OpenCV. Now, I know this question may seem ridiculous but it is an extreme example to demonstrate best what I mean by the question. I have a complete white background. ...
I am working on a eye blink detection project in Matlab. By this moment I have been successfully extracted the images of pupil of eye from the real time video captured via a web cam. But I am stuck at the process to differentiate the open eye pupil to the closed eye lid state. Open eye pupil 2.closed eye lid Could ...
I have posted this question "Electrical Engineering", but this seems a more appropiate place. I am trying to model a bireciprocal Cauer filter in LTspice but I don't get the expected results. More precisely, using this formula for the coefficients $\gamma=\frac{re(p_i)−1}{re(p_i)+1}$ where $re(p_i)$ is the realpart of ...
I am quite confused whether the following numerical differences i find are just severe round off errors made by matlab, or something i am doing wrong. The following happened when trying to see what a low pass filter looks like in the spatial domain. By mathematics, the inverse fourier transform of a real symmetric imag...
I've a noisy signal, which is the sound of motor with a constant speed, so the sound "should" be periodic, I know that there is a way to use the autocorrelation function to get the period,I did it, but I can't figure out the period. Any idea how to do that bellow the signal and the result of the autocorrelation : th...
Say I have two signals, a(t) and b(t) where the former is the input and the latter in the output. These signals are both recorded by sampling at every 0.01s. The Fast Fourier Transform was applied to both of these providing me with two large arrays of complex numbers. The FFT of b was divided by the FFT of a to provide...
I have an image taken a night time. I apply _EqualizeHist to it. But the result is a high level of whiteness running through the image. This is the image: I had asked this question in a slightly different way and was given a link to improve the white balancing of the image. This was the original question This is that ...
I have a signal of length 'N' which looks like this [.......p-set...] [SINE-WAVE] [....q-set...] amplitude of points in p-set << amplitude of points in sine-wave Sometimes there are a few sparse points in p-set whose amplitude is comparable to sine-wave points. However, number of such points is much less than the len...
I have a 10-bit ADC with 8.5 bit ENOB. I have 2.5 quantization levels of thermal noise in front of the ADC. Is there a reason to process more than 8 bits from this ADC?
I wish that all those people on that flight were with us, and maybe they still are, as I like to think they are on an island somewhere waiting to us to find them ... however, it does not look good... Recently, a spy plane searching for possible remnants of Malaysian Flight 370 took this photo: However they cannot tel...
I have a problem right now to solve. I have a Gaussian peak, which I need to divide into three components (three Gaussian distribution). For those three component Gaussian distribution, I already know the mean and standard deviation, if I want to do this by using least squares, where should I start with in Matlab? Than...
Bit of a long post, but this is a complex topic: I want to hash of a series of video files so that they can be quickly checked for duplication. I know that any amount of data can be reduced to a hash, but I'd like to do this in a way that is: Format independent, i.e. not a binary hash, but one based on what you see Re...
I have an analog filter with its frequency response curve in dB described by the following expression: $$ N_{dB}=20log_{10}\omega t_1 \sqrt{\frac{1+(\omega t_2)^2}{1+(\omega t_1)^2}} $$ This expression is derived from the series connection of two lowpass filters each associated with the following RC circuit: where, fo...
I'm trying to find the count of horizontal columns in an ordered image (a sprite image that is used to generate an animated image): From this image, I'm trying to generate a meta-data file like so: { "frames": 12, "cols": 6, "width": 102, "height": 89 } Frames would be the total count of "elements" in the image, ...
Is it possible to find time delay of arrival using demodulated signals (of bandwidth ~200hz)? I have a setup in which I receive (multiple receivers, of course) modulated signal from a source. I want to find the angle of arrival of the signal. The signal is amplitude modulated. Due to hardware limitations, cross correla...
I am sending 6.6Vpp square wave across a narrow-band, 40 kHz Transmitter to a narrow-band, 40 kHz Receiver. I want to find a very accurate distance between the two (sub millimeter). My assumption so far is that if I can get very consistent results at a fixed distance, that I can calibrate the speed of sound and an off...
I have many close-range images of a single wind turbine blade (taken from different positions, same angle): I need to separate the blade image from background like this: The problem arises in image alignment application where the blade image moves about large amount while the background does not (parallax effect). Th...
I've started to implement an MFCC algorithm on an embedded DSP, using the CMSIS DSP library. However the results I'm getting don't appear consistent. This is my algorithm so far: Gather 32ms PCM sample from microphone, 512 samples long. 16000hz Apply a Hamming window. Compute the 512 FFT using this. Compute the magni...
I have a question related to wavelet transform: we know that while the Fourier transform is good for a spectral analysis or which frequency components occurred in signal, it will not give information about at which time it happens. That's why the wavelet transform is suitable for the time-frequency analysis. It is als...
I have a set of audio files which contain a word each one, and I'd want to be able to compare a new word (coming from a smartphone mic) to check if it resembles to a specific file and how much it resembles to that audio file. I don't need to know which word it is, or doing a STT system, actually, just checking how simi...
I have a problem with a FFT analysis in MATLAB, which is probably related to my limited understanding of the fundamentals of Fourier analysis. I use MATLAB’s built-in fft analysis function. The sampling frequency is 48000Hz The samples signal is a fairly clean sinus at 3000Hz (se below) I use a flat top windowing....
I am trying to make filters like sepia filter in my app like other photo editing applications , till now I don't have any idea about it , Can any one help me out for it about its algorithms and how to do it ? and Can I implement in opencv using native c++ or we have any of its build in functions ?
I have some code that collects amplitudes from a complex sound like a song. I now want to plot a graph of amplitude vs time, for the graph plotting I use JavaFX. However I get about 9 million amplitude shorts for each channel, so how to I select a few samples to plot (as plotting 9 million points is obviously out of th...
I looked over the other entries regarding peak detection and none seem to answer my question. I'm working with Fourier spectra of digitized audio that can't be measured again. There are no constraints on the nature of the audio. Finding outliers is easy. I'm worried about "fat" peaks whose contours might be made jagged...
Say we have a sampled signal for which the inputs can be out of range. For instance, say a sampled sensor has a range of 0 to 1, and is presented with a time-varying stimulus where a peak occurs with max amplitude 1.2. In the data we would see a saturated or clipped peak. What are some of the approaches to making estim...
I am interested in characterising the time-dependent response of a system to a sinusoidal input. The steady state response of the system is also sinusoidal, however I wonder if there is a standard way to represent the information in a manner that is more detailed than the Frequency Response Function to capture the tran...
I have been studying carrier frequency offset estimation techniques in OFDM systems. I have understood several of them. when using Pilot tones, as stated in paper (1), the process involves two methods namely: acquisition and tracking modes. a large range CFO is estimated in the acquisition mode while in the tracking m...
I'm new to DSP and trying to do some example problems for a class and am confused by this one. Given a data set, and knowing it contains a signal that is a single sinusoid with random content noise, how can I calculate the signal to noise ratio? I also have the sampling rate. I have been able to plot the autocorrelatio...
I am interested in removing oscillations from a signal to capture the lower-frequency variations, similar to the objective of this problem. The oscillations vary in frequency in the time domain, so wavelet shrinkage seems to be a reasonable option, but most of the literature on wavelet shrinkage is applied to denoising...
What could I say about stability of those systems by looking at the scheme? System 1 System 2 $y_r(s)$ is input, $a(s)$, $b(s)$, $p(s)$, $q(s)$ and $r(s)$ are some polynomials and $y$ is output. I calculated tranfer $\frac{y}{y_r}$ and it's the same for both systems. $$\frac{y}{y_r} = \frac{r(s)b(s)}{a(s)p(s)+b(s)q(s...
it is a couple of weeks that I am thinking about the following: I have a tool which is able to simulate the dynamic behavior a steel tower exhibites due to the coupled action of wind and waves. Currently, to damp the oscillations, I am controlling the acceleration of the tower top by means of a standard feedback loop. ...
I have an array x of length 1024 (stored as 16 bits integers, named for example np.int16 in numpy/python), i.e. the size of x is 1024*2 = 2048 bytes. (Remark : x comes from an audio .wav file, stored as 16 bits integers, as it is very common. But it is also very common to interpret it as a float array, with values in $...
I have c# desktop application and I am using emgu framework as a wrapper for OpenCV. I am trying to sharpen an image because the input images are not great. I tired a technique I found via searches that suggest from the original image: I take a copy and blur it. I then take another copy of the original and invert it. ...
let us suppose that we have following signal values,which consists by deterministic components and random noise(white noise) 56.69 75.24 13.77 8.56 -12.88 -65.34 -45.33 -48.78 -22.23 54.12 83.77 11.84 2.31 39.59 -32.09 -88.86 5.45 50.24 -37.39 -35.69 38.62 7.06 -30.01 22.36 60.71 30.96 5.90 -38.91 -58.15 -40.87 -13....
If I want to measure entropy of an image how should I do it? My aim is to ignore things like tree leaves moving in the wind with my C# desktop application. I know matlab has this functionality but I cannot see it within Emgu which is what I am using. I got as far as finding 'Shannon Entropy' is probably the way forward...
when it comes to digital I/Q signal power measurement we know that this power is viewed as $I^2+Q^2$ (as its polar form). This is the power as referred to final analogue signal. Yet the IQ modulator does not convert I/Q to polar form, instead only the final I is output to real world say in the RF modulator. When I mode...
I conducted a research on how to implement a QPSK demodulator. I found out the Quadrature QPSK demodulation technique which seems very straight forward. Here is what i understood from the explanation of the block diagram, and would like to confirm from this platform if my understanding is right. From the block diagram...
If I have the following power spectral density of the quantization error can I tell that quantization noise is approximately white? Can you please provide some justification?