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I want to develop a procedure to remove some inpainted text from images, i.e. I have to estimate the inpainted pixels. I will do that with first an imputation step and then applying SVD (and perhaps later further improvements). The imputations step just have to give an initial guess of the inpainted pixels. I thought o...
I'm trying to use a cardinal sine as a lowpass filter for a cosine signal with a fundamental of 1k. In frequency domain, what really happens is that I'm multiplying two impulses (centered at -1k and 1k) with a rectangular pulse of width equals 2 (convolution is multiplication in frecuency domain). The result would be ...
I want to transmit data like radio stations do. Like radio stations broadcast music and etc. using frequency, so I am also curious to know about it that actually how it works. I also want to make such a device that will broadcast my data or voice in my apartment and if anybody will on that frequency they should hear my...
If I have an image that is severely corrupted by Poisson noise, and I want to fit a function to the image, is it "better" to attempt to denoise the signal first before fitting, or should I move straight to the fitting stage? In the example below, a 2D Gaussian function has been corrupted by Poisson noise. Should I fit ...
I am currently working on a application which requires an inverse Fourier transform (IFFT) and am using a function library provided by Texas Instruments, in the datasheet they give an example on how to use one of their IFFT functions (shown below). My question is: whilst I understand the need to perform bit reversal ei...
I am new to signal processing and am currently doing research into the subject. I am learning about Orthogonal Frequency Division Multiplexing and have a question. Most definitions of OFDM mention its modulation scheme can be adapted to suit channel conditions, an example being changing to a more robust scheme if heavy...
When considering the uncertainty of a signal which is determined by multiple inputs propagation of uncertainty states that for measurement $$ y = f(x), x=\{x_1, x_2,..., x_N\} $$ uncertainty in $y$ is $$u_y=\sum_i \frac{\partial f}{\partial x_i} u_{x_i} $$ My question is how do we get to this rule. I know that it can b...
I am working on some waveform generation software and I am trying to figure out what the best type of filter would be to use a an Anti Aliasing filter for my algorithm. I am generating my waves in a "Raw" mathematical way, meaning that i am creating a ramp for a saw wave and my square wave consists of pure 1's and 0's....
I am very new to DSP, so excuse my ignorance. Using Audacity, I generated a sine wave of: frequency 440Hz amplitude: 0.8 duration: 1 minute Now when i plot the spectrum for this sine wave, I get an image like this: This is really surprising for me. I was expecting a single surge at 440 Hz for the pure tone. But ...
A very naive question... A signal with bandwidth of 20 MHz is modulated using a carrier of 1 GHz. What will be the sampling frequency? I was asked this question during an interview. I answered 2 GHz but the interviewer said it will be 40 MHz. Can anyone explain it? Thanks.
During the multi-camera tracking, there is a situation where one person moves from one camera's coverage area to another. In that case that person has to be re-identified(as the one from previous camera's previous view) in order to keep tracking the particular object. What is the easiest way to do this? Color matching?...
I wish to low pass filter an image with a Gaussian kernel. However, I also wish to select a filter that has the property of producing images that have have no greater than N cycles/pixel. Let's say I wish N to be close to, or exactly .025. By what method is $\sigma$ selected in order to obtain such an image?
I am a student of computer science (not the university but technical school). I am interested in the world of DSP, and I saw that it is closely connected with mathematics. I noticed for example that a lot is based upon the Fourier I'm slowly trying to learn. I also want to buy the book "Understanding Digital Signal Pr...
I have multiples records of a motor sound, I would like to synchronize them first and then calculate their spectrogram. My question is how can I use the autocorrelation or the crosscorrelation to synchronize them? I want to make sure that the only difference I see in the spectrogram is related to the frequency, not t...
How do you derive the theoretical symbol error rate as a function of $E_\mathrm{b}/N_0$ for 4QAM? I know that the result should be $Q\left(\sqrt{2E_\mathrm{b}/N_0}\right)$ but I am ĺooking for the derivation. Also, what are the symbol error rates vs $E_\mathrm{b}/N_0$ for 16QAM and 32QAM?
What options exist for discrete audio compression that is good quality, moderate compression (e.g. 3+:1), and low complexity (e.g. hundreds of processing cycles to encode each sample without the use of floating point), especially ones that are not patent-encumbered? The specific application is ambient sound - vocal and...
I am developing an image processing algorithm / plugin for Fiji (ImageJ) that processes scanned microscope slide images (20x magnification) and extracts features from the slides. The slides contain histological tumor sections that are stained immunohistochemically with an antibody that identifies nuclei positive with t...
write an m-file that will take the"cameraman" and produce a image quality or difference matrix for spectral selection method.The image properties are a sfollowings: For spectral selection: image1:DC coefficiant only Image 2:first A/2 coefficiant including DC,where is the number of coefficiants prior to the EOB symbol. ...
Iam having 1 min video.I need to cut the video into 5 segments depending on changing background.Please suggest me good algorithm which can be easily implemented in matlab.Please provide me helpful links.
Inspired by the talk "You and Your Research" by Richard Hamming, I started wondering what are the important research topics on signal processing nowadays. Certainly an extensive list of topics would not be easy to find. I do not know the trending topics, for example, on Circuits and Systems, but I could imagine it for ...
Is baseline wander in ecg is low frequecny noise in ecg and power line interference is high frequency noise in ecg? Is both the artifacts present in EMG signal?
I have a functioning DTMF program. It takes a larger signal and breaks it up into many small components, then analyzes each segment with an FFT to determine if the magnitude of the response for a particular "bin", corresponding to a specific frequency, is present, thereby indicating that the number which is respresente...
I am new to signal processing, so sorry if the question is superbanal. I have been spending more days than I would admit on this and can't find the error. I have two triangle waves that are emitted from loudspeakers and M sensor in a linear uniform array. I need to estimate the delays of the single wave in the single s...
I'm currently collating images to try and develop a HAAR classifier using the OpenCV traincascade functionality. The objects I want to detect are small and numerous (bees), and I'm curious as to whether it would be better to train the image with lots of small cutouts of the bees as positive images, or to use bigger pos...
I was looking at the filters, and the ones for 1st derivatives, gradient http://en.wikipedia.org/wiki/Savitzky%E2%80%93Golay_filter#Tables_of_selected_convolution_coefficients by testing the 9-points one: D=[-4;-3;-2;-1;0;1;2;3;4]/60; y=sin((0:0.2:6)'); yd = conv(D,y); plot([repmat(0,128,1) y yd(5:end-4)]) we see tha...
I am trying to apply phase lock loop to calculate phase shift due to doppler effect and finally calculate the doppler frequency($\Delta f$). I am assuming that the received wave is of the form, $r(t) =\cos(2\pi f0 + \phi) $ .The receiver in my setup is at rest and the source is moving at velocity of about 1m/s. But I a...
I'm currently working on implementing an efficient method to segment an organ on 3D CT images. I used an algorithm (SLIC) to generate superpixels on each slice of my CT 3D matrix. The process is similar to the k-means algorithm although we do not need to visit the entire image for each pixel. Then I build a graph base...
I'm trying to fuse different images of the same scene which each image have a different focus. I've been told that first I have to perform an image registration to the images. Right now, the result from the registration has errors, which can be seen by "aura" around the objects in the images and shrinking the image edg...
Can someone explain me how are the different frequency bands useful for detecting bearing faults using resonance demodulation?
I would like to create a Band Limited Table Lookup Oscillator that functions much like the sin() function in C++ works where i would inout a phase value and receive a sample value. for example Sample s = saw(x); or Sample s = square(x);. To fill the table I would first generate an oversampled table by a factor of "X" ...
i have a legacy processed audio wav file. Another UN-processed copy of the source is processed by my algorithm. From both processsing blocks i get processed wav files as output. Now i want to do null tests, because i read (will soon quote the reference) null tests are suitable for objective loudness comparision. So mer...
Suppose that we have the following model $y(t) = A_1\sin(\omega_1 t+\phi_1) + A_2\sin(\omega_2 t+\phi_2) + ... + A_p\sin(\omega_p t+\phi_p) + z(t)$ My question is not related to how to determine the number of components, but I am interested in the maximum. For example, let us assume that we have sampled this model and ...
We have measured the human blood pressure with the sphygmomanometer connected to relative senzor of the pressure and osciloscope. We measured the offset of the sensor as $U_{OFF}=1.01\ V$ and the sensitivity $k=0.02425\ \frac{V}{mmHg}$. By measuring the pressure on the human's arm, we got this image: How to find syst...
I have a doubt regarding multi track/multi layer recording of audio signals. in recording studio each instrument or chorus,playback will use a independent layer/track..so after recording a all tracks ..how do they mix all of those into a single track ..any mixing scheme?..or it is the simple time division multiplexing?...
In ofdm systems after applying IDFT we take the last N/x samples and place them before these N samples. This is called cyclic prefix(CP).So why to use a cyclic prefix? Why is the CP known to combat the inter symbol interference.Why can't we use some junk data or zeros over there? Can't this junk data combat the ISI ?....
Just wondering what the opening operator is in image processing? What does it do? Just deletes bits of the image that you don't want?
How do I calculate the maximum number of harmonics a complex wave can have based on the sample rate? Say I Have a 20hz fundamental and a sampling rate of 44100 hz. My instinct is to take 44100/2 = 22050(nyquist) and the take the fundamental and multiply by 2 while the frequency is less than nyquist and the count how ma...
I had an exam one of these days and one of the questions was: "Knowing that an auto-correlation estimator of a sinal x[n] could be defined by: $$ R_{xx}[k] = \sum_{n=-\infty}^{+\infty}h[n].x[n+k] , k=0,1,...,n-1 $$ and that the non-periodic convolution: $$ y[n]=x[n]*h[n] = \sum_{k=-\infty}^{+\infty}h[k].x[n-k] $$ show ...
I'm using Filter Banks (Narrow Band Pass) created using the functions in Matlab. However, I wanted to know if there are fundamental limitations like that of FFT which also affect the Filter Banks. Is there any co-relation between the two? Basically, I don't know if they share the same problem of time and frequency bei...
Actually I am using DSP kit TMS320C6711 through which I want to remove 50Hz component out of the signal. I am unable to interface Code Composer studio (v2) with matlab. Therefore, I have designed FIR notch filter using Code composer Studio in which coefficients are taken from fdatool of matlab. Kaiser window has been u...
I am trying to calculate the mfcc of an audio sample of 5sec duration. How can I frame this audio sample. Is there any criterion for selecting the frame size and number of frames? What all change will be there if the audio file is 30sec long?
I have created a fairly simple TDOA system that uses ultrasonic signals emitted from two speakers to geolocate (relative to the speakers) mobile phones. The two signals are separated by frequency. The system has the following constraints: The signals must be inaudible. To that end we stick to frequencies above 17 kH...
I am looking for a great introduction to FFTs for my job. I studied math, but never heard of FFT until I got this job. There is not a lot of documentation at my place of work and the articles I have found online have been somewhat confusing, even the wikipedia entry. So I am looking for a recomendation for a higher lev...
Is there any difference in energy content of baseband and bandpass signals? When we see their representation in frequency domain, it looks like a bandpass signal has double the energy content compared to basband. But my friend says there's no difference. Someone please clarify.
I am trying to work out this problem: Assume that we are transmitting data from a dictionary of 15 symbols. Let the probability of sending one symbol be the same as that of sending any other symbol. Calculate how often we need to fetch a symbol to transmit information at a rate of 5 bits/s. How can this be found?
How exactly is beat division applied to a signal? Does the rhythm need to be detected or can just the onset be manipulated? For rhythm detection is it better to look for patterns in the frequency domain or time domain for accuracy?
Assuming that a computer is interfaced with a device that has 210 characters and that each character is transmitted using binary words what is the minimum number of bits needed to represent each character? To find the minimum is there any specific formula ?
I am reading "Small Unmanned Aircraft: Theory and Practice" (page 7). We have the following Transfer Function (TF): $\phi(s)=\frac{a_{\phi 2}}{s(s+a_{\phi 1})}(\delta_a(s)+\frac{1}{a_{\phi 2}}d_{\phi 2}) $ where $\delta_a$ is the output signal to the actuator and $d_{\phi 2}(s)$ is an unwanted disturbance. Initially,...
What's the best way to incorporate polynomials or sinusoidal equations into a linear model to predict a future amplitude in this sort of data? You can see that if those polynomials continued to go on, they wouldnt be anywhere close to fitting the final location. Im trying to use the greeen shaded area to predict the no...
What is the name of the following bit encoding generator matrices?
In figure 11.23 of Monzingo's Introduction to Adaptive Arrays, the author uses a sinewave in the frequency domain to model the amplitude ripple in the passband of a channel of a receiver. What does he mean by array bandwidth? How does one create a time domain filter (FIR or IIR) with this custom/arbitrary frequency r...
Here are two images. What kind of filter/processing is done here? It looks like an emboss but the bumps are smooth. 1) Original : 2) Processed :
I want to clear my thinking on how to approach problems of the type characterized below. I guess what we want to do is select a sample rate of a sufficient size such that this signal can be fully captured therein but also small enough so as not to waste all the space from 0 to, 156? (77*2). I've seen in some example...
first of all, sorry for this basic question in signal processing, I'm pretty new in this field and have been stuck in a situation a little beyond my expertise. The idea is to derive the instantaneous wavenumber direction as an estimation of the direction normal of a 2D field by using a directional Hilbert transform in ...
Can the Fast Walsh-Hadamard Transform be used to find a Walsh-Hadamard frequency spectrum which some how relates to a Fourier frequency spectrum? Is the Fast Walsh-Hadamard Transform more accurate than the Fast Fourier Transform when analyzing signals with very low bit resolution? Below are the different Walsh-Hadamard...
In the literature, I've found that DFS and DFT are one and the same. If they are one and the same why to use two different names for them? If there is really a difference what is it and what is the significance of discrete Fourier series?
I'm trying to code the HPS (harmonic product spectrum) algorithm and the problem I'm facing is that using my guitar, the fundemental frequency (Here it's 82Hz) just isn't there, so is the 5th Harmonic at 410Hz (Actually it's the lowest local point). In my algorithm, I did the step of finding the max of all (Mul(harmoni...
This is my very first signal processing exercise so I'm very new to this space. That being said the goal is to do isolate voice recognition in identifying someone saying yes and no.As the title suggests I have a 1x26 vector that looks as follows: Columns 1 through 10 (P)160.2504 66.7557 -35.9298 144.1598 -11....
Anyone can explain me about how measure side lobe roll-off rate of kaiser window from Spectrum in matlab where side lobe roll-off rate is the asymptotic decay rate, in decibels per octave of frequency. How adjust Frequency axis if length of FFT = "500" by padding "473" zeros with length of window= "27" to frequency in...
I'm new to OFDM modulation. During OFDM modulation in the frequency domain, are the negative frequencies modulated with a conjugate version of the data of the positive frequencies, or do the negative and positive frequencies carry completely different data?
I am working through previous exam questions for a class I'm currently taking in signal theory, and one problem, although it looks easy, stumps me. PROBLEM Two real $N$-periodic time seires $f_n$ and $g_n$ are given, where $n = 0, 1, 2, ..., N-1$. In addition, the complex time series $h_n$ is given as $$h_n = f_n + i...
Is is possible to use a digital filter (eg. Butterworth) for smoothing sensor data that is sampled at a non-constant rate? The time between two consecutive samples is between 12 to 40 ms.
Are there any computer vision libraries in PHP, or any server-side programming languages? If not, maybe we can make a list of some -already implemented- methods (Viola Jones face detector, landmark detector etc.) to help someone who wants to implement such a library?
I have just read that the Gibbs phenomenon can cause artifacts in the use of Hilbert transforms for audio processing applications, but the reference contained no more specific information. Can anyone expand on this remark? When does the Gibbs phenomenon cause problems for the Hilbert transform, and how does one solve...
I found many readings on Non rigid registration, also saw some videos but didn't find a clear explanation anywhere. What I understood is the following Rigid Registration: It is the alignment of two point clouds without changing the shape (ie only by rotation and translation). ICP algorithm is used. The closest point in...
Im trying to generate a sweep sine wave that increases its frequency till an upper limit and then starts decreasing its frequency to the lower limit at the same rate. Im trying to simulate in matlab. The frequency of the wave will change based on the following formula: f = fmax-a*abs(T - t); a in the formula equals (f...
I am trying to segment the nucleus of the cell in an image so I can extract its boundary. The nucleus and the background of the image are very similar so I am finding it difficult to do this using the something like erosion. If anyone could suggest a way to remove the background and keep the nucleus that would be very ...
I'm trying to figure out what a good DSP unit would be to try and learn how to do some embedded programming on? I would like something relatively inexpensive but that still has a decent amount of "kick" to it performance wise. I was looking at the TI ez-DSP but I don't know what to think of it. I would like it to be pr...
I have the following block diagram I know that it's solution is: $$\dfrac{10 K_p}{Js^2+(B+2K_pK_v)s+10K_p} $$ But when I try to derive the solution with reduction rules I get different result. Assumptions: $5$ and $K_vs$ are in parallel: $(5+K_vs)$ $K_p \cdot \dfrac{2}{(Js+B)s}$ result: $$\dfrac{K_p \cdot \dfrac{2}{(...
I'm implementing a dictionary learning algorithm for doing sparse coding on images and I have a problem how to learn a dictionary corresponding to several iamges. Let’s say X is the data matrix. If I apply dictionary learning only to one image then X is equal to the image. But what if I want to learn the dictionary for...
Assume an arbitrary (discrete) signal that is periodic and known over a whole period. I need a way to select a characteristic point along the signal such that I can always retrieve it even when the signal is phase shifted. Candidates could be zero crossings or extrema or inflections... provided they exist and are uniqu...
Multiplications of time domain vectors of two signals is equivalent to convolutions in frequency domain and also use circular convolutions for when FIR filter design using window method in frequency. if working Discrete Fractional Fourier transform (DFRFT) i.e. on bins between time and frequency domain then which i...
Laplace&Z transform is just frequency analysis of a system with a multiplication of decaying exponential. We analyse frequency with varying Laplace Exponential, which can be seen in the formula below for laplace transfrom. From my simple aspect I can't make the theoretical connection, how inspecting frequency with ...
I am trying to make filters for days but now I just want to know the experience of others rather than go on to wrong way and finally decided to put my query over here with full detail rather then short question of every day . What is the best way of making image filters like Instagram or aviary or many other photo edit...
I'm experimenting with linear predictive coding as a low-complexity audio compression method, and I'm observing something very odd. I'm compressing a 20 MB mono 16-bit PCM song in different block sizes using 2-sample LPC (increasing the number of samples appears to benefit little compression-wise). The optimal block si...
I intend to whiten the spectrum of a time series using a point-wise normalization in the frequency domain. In my first test, it looks pretty good. figure; n = 2e3; t = 1:n; m = n/2+1; f = linspace(0,1,m); x = rand(1,n) - 0.5; y = fft(x); subplot(2,2,1); plot(t,x,'k'); axis tight; xlabel('t/s'); title('original signal ...
I don't have a thorough background in Signal processing and require some information for an application pertaining to computer science. Minimum Entropy Blind Signal Deconvolution with Non Minimum Phase FIR Filters paper proposes to use entropy for blind system identification. My questions are the following based on ter...
Ok, so I recently started playing around with an SDR in an effort to challenge and educate myself. So, let's say the sampling rate (Fs) is 3MHz and of those we grab 3M samples. Now we are left with 3M samples consisting each of an IQ pair. Complex data. My understanding is that I would perform an FFT using the entire s...
I have already calculated the cdf values of each image now i have to map the reference image values to target . I cannot use histogram equalization or interpolation I simply have to map them . function S = histM(t, target) if size(target,2)>1 %an image, replace the image with its cdf target = getImageCDF(target...
I try to obtain fundamental frequency from cepstrum. I decide to use peak picking algorithm, which pick maximum value of the Cepstrum (c) with sampling rate fs. function [f0] = spPitchCepstrum(c, fs) % search for maximum between 2ms (=500Hz) and 20ms (=50Hz) ms2=floor(fs*0.002); % 2ms ms20=floor(fs*0.02); % 20ms [...
I have a discrete signal $x[n]$ of length say $1024$, such that : $$-1 \leq x[n] \leq 1, \qquad \forall n$$ and let $y[k]$ be its DFT. Is there an upper bound for $|y[k]|$ ? (i.e. for the DFT of a bounded signal)
I am getting a high frequency at 0Hz. How can I remove it? I am not supposed to get it. My code is W = 3.64*2*pi; Fs = 50; % Sampling frequency in hz T = 1/Fs; % Sample time L = 1000; % Length of signal t = (0:L-1)*T; % Time vector v = 113*(1...
It is know that low frequency signals travel longer distance that of high frequency signal. If so, our speech signal is low frequency signal, why it is not travelling further?
I'm trying to make an oscillator using IIR filter and I'm having problems tuning it so that it's stable on the platform I'm using (ARM Cortex M4F and CMSIS DSPLib's arm_biquad_cascade_df1_f32). Note that this is my first time doing this and that I don't know much about control theory, so I could be missing something ve...
The other days I read a useful article (on BER v.s. Eb/N0 curve of ofdm) whose link is as bellow. While watching the plot inside, a question was arisen in my mind. I wonder if the BER v.s. Eb/N0 curve of BPSK alone (without using OFDM) and one using OFDM could be the same in both AWGN and fading (flat and selective) ch...
I'm wondering if there is a way that I can reconstruct an image, given the output vector obtained from the convolution with a certain filter. For example I convolve my image with a filter from MR8 filter bank, so I got a feature vector v. Which is : v = img ** filter so can I define a function that takes v and filter ...
When the following code is written,the resulting magnitude spectrum shows magnitude values represented by two peaks is different even though they have the same amplitude. why is that? a fast reply would be much appreciated Fs=250; %sampling frquency of 250Hz i=0:1/250:4; %1000 data points x1=sin(2*pi*7*i); %create...
We are trying to implement Alamouti STBC on underlying ofdm signal. We were thinking as to where to write alamouti coder block. I thought we could write it after the cyclic prefixer block as we get a symbol after the IFFT and cyclic prefixer. we could take two such symbols,send both of them at one time instant and then...
I try to detect and count circles in image (for example smarties) I use HSL color space. But I am not able to distiguish colors od the same color if they are in touch. I try to erode and dilate picture. But result is the same, I have only some blobs(connected components) od the same color. Do you have some general alg...
Can any one help me in how to extract a table from the whole document image and segment it? I have done segmentation for the document image which has only text by using the projection profile. So please help me how to extract the table from the whole document image. I want only table to be extracted from the document. ...
my_signal = 8192*cos(2*pi*6*(0:7)/8) + 1i* 8192*sin(2*pi*6*(0:7)/8); my_signal_fft = fft(my_signal); plot(abs(my_signal_fft)); This is matlab code. What do you notice about the spectrum? Is it symmetrical? Obviously not, because I think it has an imaginary part. Would you enlighten me for the analysis of this signal...
When we have FFT of Square wave of a pure real signal with high frequency(close to f(max), less than half of sampling frequency) we see only 2 symmetric high peaks but when we decrease the frequency we see the two peaks and peaks of harmonics and they are getting smaller and smaller, why? Also the highest peaks are get...
Good day everyone. So, I run in to trouble - How to add white noise to signal samples? For now I'm working with MSK signal - quadrature modulator and quadrature demodulator + Viterbi processor. My signal samples at TX and RX side represents Q and I signals of modulator(real and complex part of the signal). Model works...
I am quite new to MATLAB and I am currently trying to do some Fourier synthesis. In order to do the Fourier synthesis I need the phase information of the harmonics as a fraction of the period of the fundamental frequency. I am a bit lost on how to do this. Also I am trying to do a Fourier synthesis using the equation $...
My goal is to apply the Remez exchange algorithm in R to design an FIR band pass filter for a specific frequency window. The question is not supposed to be referring to R of course but about how to choose the parameters correctly and why the result looks like as it does. The toy signal is an additive combinations of fo...
I try to find about the delay between two audio files using Cross Correlation in Java. I've already done this algorithm so far that i get a idea about how many samples is the delay. FFT x1 -> Zero Padding to length: x1.length() + x2.length() FFT x2 -> Zero Padding to length: x1.length() + x2.length() X3 = FFT{x1} mul...
I'm working on image processing that involves facial features. Where can I get a generic 3D model of a human face? (any format is fine as long as I can manipulate the data using C or Matlab)
Consider 2 undirected weighted graph as shown in figure. Consider first graph as $G_1$ and second graph as $G_2$. Graph $G_1$ consist of vertices $V=\{F_1,F_2,F_3,F_4,F_5\}$ and graph $G_2$ consist of vertices $V=\{N_1,N_2,N_3,N_4,N_5\}$. According to logic of my project we have to match vertices of 2 graphs as shown ...
Why are Peak filters called "Peak" and why are Shelving filters called "Shelving". I am not a native English speaker. So excuse me if it is so dumb question.
I want to do a convolution from the bottom right and not as usual from the top left. I think conv2 of Matlab only does from the top left. How can I do a convolution in Matlab from the bottom right? Thank you very much for the answers.