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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...
I'm struggling with a signal analysis electrical engineering college assignment. We need to a derive differential equation for a low-pass filter, high-pass filter and a band pass filter (made by connecting the output of a low pass filter to the input of a high pass filter). Here are diagrams for reference: Low pass fil...
I have just bought a new video capture card whose output format is YUV 4:2:2 (as my old one) and surprisingly when I capture images with both cards I get different results. Differences are very small (have to zoom on edges if you want to see something) but I would like to know where do they come from? I was wondering i...
For a certain experiment of which I have data, I want to calculate the autocorrelation of the measurements. For this I used the expression from Wikipedia, which reads Here, $n$ is the dataset's size, $0 < k < n$, $\sigma$ is the standard deviation and $\mu$ is the mean. Now, I've read the descriptions on the page, bu...
I have the the noisy curve defined by numpy 2D array: As you can see, it has the first flat segment, then rise, peak and decay phases. I need to find the starting point of the rise phase, marked here by the red dot. What algorythm can I use for that? I need it to be as noise-tolerant as possible.
I was thinking sometime back about how to explain the Continuous Wavelet Transform ELI5. So this is what I came across. The correlation of two exact signals is 1. So if I have an input signal $f(x)$ made up of an array of frequencies, how can I find out what frequencies exist at what points? Well, slide a signal $m(y)$...
I am having a problem understanding how the author gets the onset envelope of an audio song in this paper. The entire section in question has been quoted below: First the input sound is resampled to 8 kHz, then we calculate the short-term Fourier transform (STFT) magnitude (spectrogram) using 32 ms windows and 4 m...
How can I convert a time signal (audio) into the frequency domain, multiply frequency amplitudes by a vector of scalars, and convert it back into the time domain, without messing up timing information (ringing?)? Some context: I am calibrating ultrasonic speakers. I have a frequency roll-off curve of the attenuation dB...
i am trying to reconstruct time series from SSA ,because according to this link http://en.wikipedia.org/wiki/Singular_spectrum_analysis there is procedure 4th step: Diagonal averaging. i have with started following code clear all; B=xlsread('data_generations1','A1','g8:g301'); n=length(B); l =input('Give the ...
For my project, I needed to detect the primary oscillation frequency in a signal data. As I googled around, it turns out that DCT is the best tool to find this (http://www.mech.pku.edu.cn/robot/teacher/wangjiandong/WangHuangLu_ImprovedDCT_CEP_2013.pdf) My question is: why would you base the analysis on DCT, and not DF...
I am trying to understand some basic concepts with IFFT as represented here OFDM Modem page 50-53, and 108-114. Some basics that are represented - The base sampling rate is 8khz. The symbol period is 32ms, which would calculate to 250 total samples. The fft size is 64. The bin separations are 32hz. The bandwidth used i...
I have a time series (of 23 timestamps) of which I take the Fourier transform. Now the fourier transform has 23 imaginary values and each has an amplitude and a phase. When I get the phase angle, it given me arctan value of the angle. Now how can I determine if two fourier coefficients are in phase or out of phase.
If I have a time series (for eg. for 23 timestamps) and if I plot it and see that it is bimodal, that means it might be having high value of second fourier coefficient (with frequency = 2). But when I do the fourier transform I find that the value of second fourier coefficient is very low. Now, is the following reasoni...
Connected with this topic , I would like to have some clarifications about the Fourier Transform and the Python tools that make it. Having my data set whose plot is showed in the above link, my array of data is stored in rate. Then, if I check the result from the Fourier transform: yfft = np.fft.rfft(rate) y_smooth = n...
I have two sound clips $C_1,C_2$ which are pretty similar. I'd like to measure how perceptually similar they are, i.e., how similar a human would perceive the two to be. Is there a way to use a lossy compression algorithm (e.g., a MP3, AAC, or Ogg Vorbis encoder) to compare the two clips? It occurs to me that that au...
I have been reading about eigen values and eigen vectors but I haven't been able to find any decent explanation relating their application in Image Processing / Computer Vision. For example, How can those be used for face detection and eye detection? Can somebody please through some light on this?
let say we have some data,some signal written in this form $x(t)=s(t)+\epsilon(t)$ where $s(t)$ is pure signal and $\epsilon(t)$ is white noise with mean zero and some variance ,question is that if we have enough sample data,should periodogram always estimate frequencies correctly?let say we have following sin...
I am reading Direct Synthesis versus Wavetable Synthesis. The quote below explains code I have read but I don't get why it's a division by the integer multiple, when multiples increase in frequency. Doesn't that mean we need more samples (higher sample rate mapping to more samples for same accuracy) to deal with the hi...
let us consider following graph of singular values i want to make some kind of clustering of these data,namely to seperate main components from non main components,let say signal components from noise components,i would like to do like this(any software matlab,etc is great,better matlab)let us start like this ,f...
My problem should probably be built up from the beginning, so lets start there. I performed a certain experiment 25 times. Every time, the experiment consists of 5000 measurements, and each measurement returns either 1 ('yes') or 0 ('no'). I know that each time I measure 1, the probability that this measurement is corr...
I read that multiplication is convolution in frequency domain. I also understand that convolution is just polynomial multiplication. Can somebody explain what are the advantages of doing convolution in frequency domain? Also, i would like to know how to multiply two signals in time domain and what is the significance o...
I have a discrete signal of random length (let's say between 500 and 800 in time). This signal has an important information which I want to collect and to put it together in a feature vector to train a classifier. The signal also comes in random phase and in random scale (it depends on the distance to the sensor). So t...
How can you filter out a person's voice from a group of people talking? We have a sample of each person's voice from the group, and the sample of the entire group talking at once. Both samples are uploaded into matlab for analysis. Is there a way to single out any one person's voice?
I am trying to do the complex trace analysis of a seismic trace in Matlab where I need to calculate the instantaneous frequency of a discrete signal. I have used the formula according to Barnes. I am running an algorithm where I need to iteratively run complex trace analysis on the residual trace obtained after subtrac...
This would most likely be the opposite of this question ( Mathematically inclined Signal and Systems/Signal Processing book? ) I figured I'd ask here if there are any good books that while, necessarily, focus on the math are more oriented to what these concepts can DO instead of the mathematics behind it, so I guess mo...
I found a really intuitive explanation of Discrete Fourier Transform here. To summarize the author's approach briefly : What if any signal could be made from circular motion (repeating cycles) ? This is the central theme of Fourier Transform Every circular path needs a size, speed, and starting angle (amplitude/freq...
I am trying to develop some code to find a transient signal in a data set I have. Before I get there however I am trying to write code using simulated data. I first am creating a signal with a specified frequency and then adding white noise to it. I then iterate through a loop where I create a reference signal at an i...
I was trying to understand the Otsu thresholding algorithm in image processing. For that purpose I found a useful link. I got the algorithm flow, but a fundamental doubt arises. Why 'minimum within class variance' ${\sigma_W}^2$ (notation given in the link) gives the optimum threshold? Please can someone clear my doubt...
Is it a realistic mapping to divide the frequency bins in 3 parts, high/mid/low frequencies? How can these bins then be manipulated to control the level of these bands in the time domain? Is it sufficient to adjust these bin segments and do an iFFT to time domain?
From discussions all over the internet, I understand that in a 512 sample point fft, only the first 256 contain relevant information provided that my signal is real. The other half contains complex conjugates of the first half. I base my question on this discussion at stack exchange. I'm trying to plot the fft of a wav...
I'm trying to measure the angle between two needle positions in an analog meter (gauge). such as: After the image processing of several images, I get the processed image below(a image show different needle positions: Here, the black dot points were the feature points of needle area, so I use the line fitting algorith...
I know that the gradient function is supported by MATLAB, but the local gradient is not, so how do you calculate it? Local gradient is another texture descriptor, which is defined as: $$I_g=\sum_{z=-r}^{r} | \nabla I(x+\tau) |^2 $$
Suppose that a signal is decomposed by using discrete wavelet transform (DWT) and transmitted. Is it possible for receiver to find which type of wavelet is applied in transmitter side ? I mean, if the signal is decomposed by Haar or DB and then reconstructed. Is it have any way to know the type of wavelet (like haar) t...
I have a conventional phaser setup with allpass filters and feedback. I can calculate the peaks frequencies of the phaser when the allpass delays are set to 1 (then they work as simple one sample delays). Adding two allpass delays will give an additional peak, and the peak frequencies for peaks i = 1 to N is found by: ...
I've got following problem: assume you've this continous system 1st order: $$ H(s) = \frac{1}{1+s} $$ s = tf('s'); sys = 1/(1+s); I'm going to discretize this system via impulse invariant transformation and plot the results. sysD = c2d(sys,1e-3,'impulse'); impulse(sys); hold all; impulse(sysD); So this seems to work...
I'm currently studying sound design, however over the past couple of years i started creating audio effects/instruments. I have also been programming in C++ for quite some years. I noticed my shortcomings though; i can design compressors, envelopes and simple stuff like additive synthesis quite well along with basic s...
I'm currently doing my final year Degree in computing studies. The project I'm working on is a video of a webcam or computer webcam to have 2 pre-defined ROI (Region of Interest) on top and bottom of the video. It is to detect a person when he does a simple push up whether he reaches the area of both top and bottom RO...
I am measuring the phase shift between two pseudorandom signals using correlation ifft(A.conjugate()*B) and picking out the maximum). That works very well, but takes a lot of time. I roughly know where the peaks are, and don't need the correlation for other shifts. Is there an adaptive cross-correllation algorithm that...
I working on writing a program to get the BPM rate and the Beat Times in audio songs but I am having trouble trying to come up with reliable testing strategies for my design.What do I benchmark my BPM rates and Beat Times against? Should I just use other softwares or is there ways of checking if my BPM rates and beat t...
guys! I have carried out some experiments using matlab warp tools from here. But the results are not good, so I decide to read its original paper Principal Warps: Thin-Plate Splines and the Decomposition of Deformations from here to find reasons. However, I am stuck to part II..... It says the energy function $\mathbf{...
I'm currently working on a piece of DSP/Synthesis software and I have run into the issue of generating band limited waveforms. I have been doing some research to try and find a good solution but I haven't had much luck. I have found some things about a BLIT to generate complex waveforms but I have not found a good exam...
float xx = acceleration.x; float yy = acceleration.y; float zz = acceleration.z; float dot = (px * xx) + (py * yy) + (pz * zz); float a = ABS(sqrt(px * px + py * py + pz * pz)); float b = ABS(sqrt(xx * xx + yy * yy + zz * zz)); dot /= (a * b); I run into the equation above on github, but I don't know what kind of DS...
Can somebody please explain how Photoshop vibrance function can use in opencv c++ ? so that I may reproduce the same effects in my application. Here is the vibrance documentation
In Matlab we use below function to read an image : var = imread('IMAGE‌‌ ‌PATH'); But, how to do same thing in Scilab ?
I am using OpenCV C++ for making the blending mode like in photoshop, I want to make overlay mode in it. I search its alternate in OpenCV in which I found this blending way, but its not the overlay as I want to use the overlay method in it. Overlay method formula from this documentation (Target > 0.5) * (1 - (1-2*(Targ...
Could someone please provide a method for inverting random fixed point numbers. I have as an input both positive and negative floating point numbers that should be converted into fixed point and then be inverted. An example could be 1/0.2453 where floating-point 0.2453 is converted into a Q18 fixed point format and the...
I have three signals (below) each having the same standard deviation, however, are clearly very different temporally. Is there some such metric that could be calculated for each of these signals to give an indication of how they vary over time, such as a measure of continuousness or intermittency if such a metric exis...
I have recently started learning DSP and the book that I am reading challenges the readers to find a: continuous time-domain electrical signal that only has a finite number of amplitude values I obviously could not think of an example and hence I am here. Please suggest an example of continuous electrical signal t...
I'm pretty new to Image Processing and am using MatLab. I have an problem were I take an image of light passing trough an set of optical fibers (bundled together) and one were I pass laser energy trough the same bundle of fibers to burn the image of these fibers onto paper. I then try to match these two. Up to now I j...
I plotted a sine wave(blue) and a 90 degrees phase shifted cosine wave (red) expecting them to overlap each other. x = 0:0.001:1; %ploting sine y = sin(2*pi*x); plot(x, y); hold('on'); %ploting cosine z = cos(2*pi*x-90); plot(x, z); The two graphs do not overlap. Whats wrong with my assumption ?