instruction
stringlengths
31
27k
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.
Let's say we have the following matrix: 0 0 0 0 0 0 0 1 0 0 0 0 1 1 0 0 0 0 1 1 0 0 0 0 0 0 0 0 0 0 0 0 0 1 1 1 0 0 0 0 1 1 0 0 0 0 0 0 I want to calculate the size of every connected region of 1s and also get the corresponding matrix elements of each region. The size should be computed as (width + heigth) / 2. In the...
I want to use Robust PCA for image inpainting, i.e. to estimate anomalous (bad) pixels. I assume that the bad pixels are known (mask). I know how to calculate the Robust PCA using Alternating Direction Method of Multipliers but if somebody has already an implementation in Matlab I would not say no to sharing it. ;) But...
I would like to know whether this can be done and if then how would this feat be acomplished in Matlab given a filter calculated with Remez? h = remez(...); If this is not possible then I would appreciate an explanation whether this can be done approximately (approximations of the zeros and poles) or whether this is si...
Required to find symbol error rate vs $E_b/N_0$ for 4QAM, 16QAM & 32QAM. Thought that SER & BER are the same but did my research to find that BER is $1/\log_2(M)$ of SER. Could you please confirm this? Also found SER for: 4QAM to be: $\displaystyle \text{erfc}\sqrt{\dfrac{E_b}{2N_0}}$ and that of 16QAM to be: $\disp...
I will be using one central node/latop to play a tone and will have a second node/laptop to record and read the sound and be able to say this node is on the left or right. So trying to figure out the best way to distinguish between the two tones. I want to have a tone set for left and one set for the right. I want the ...
I have designed few shelving and peak filter. And I have calculated the a, b coeficients for several gains for these filters. Now I would like to create some kind of characteristic in Matlab so I could present my design of these filters? Something like this for shelving filter
I saw in some codes people always taper the ends of a signal using a cosine taper. Is that necessary and why?
What does End Effect in signal processing basically means? I would like to see some reference book or papers if you suggest me. Thanks
I need to implement people counting(head count) into one of the product. I am able to detect faces and count them but only when the persons are actually looking into the camera, if they are not looking into it they will not be detected. So, what I want is to an idea about detecting human rather than detecting only face...
I apply an FIR filter passing the band around 50/1000 and attenuating the other three frequency components 30/1000, 70/1000 and 110/1000. Both filters are generated using the Remez algorithm. The first filter is of length 300 and the second is of length 100. Why is the beginning of the filtered signal attenuated and w...
I'm new to matlab sound processing and I am trying to create a 2D array for the sound code below. I have to make the sound have 400 samples(i.e 10ms) at 20hz, but having trouble getting the code to work. Any help would be greatly appreciated. The steps are: (1) Construct a two-dimensional array (n x 2 array, i.e.,...
I need to identify certain features of the audio signal recorded from microphone in stethoscope. These sounds are only samples i've found, but the final signal will be probably a bit noisier (maybe not, i don't know yet). Below are examples of three signals. Normal one and one of the person sick with atrial spatial def...
TV is L1 norm of gradient of an image. So we've to find gradient of the image (which is still matrix, right?). Then take the sum of absolute values of the gradient matrix (So now it must be a scalar, right?). Can I simply use imgradient function in matlab for first step? It gives two values magnitude and direction, so...
I have a play function that is called twice, each plays a channel (a column of the two dimensional array) in both speakers. What I am trying to do is to play both channel at the same time. How do I silence a channel and set the the elements in the corresponding column to 0. I also need an output array that im trying ...
Problem: The signal $cos(2\pi14100t)$ is sampled at $F_s = 400 Hz$. It is then upsampled with a factor 3 and then reconstructed ideally with a new frequency $F = 500 Hz$. I now want to find the new signal that is created. Incomplete solution: With the frequency before sampling $\pm 141000 Hz$. After sampling the normal...
I have 2 sounds that I created and trying to make them beeps. Maybe 4 or 5 repeated beeps but have a pause in between each beep but make it still part of the signal. I think I have it so that the sound is a single frequency but just need make it a repeated with pauses and still part of the same signal. Any help would b...
Could some one tell reasons to use the MATLAB imreconstruct function when processing images? I already have studied the topic and what the function actually does to the image. However, I cannot figure what we would use it for. What output should we expect from calling doing this function?
I have a .pgm format 480 * 640 gray scale image named 'columns.pgm'. Using PCA (principal component analysis), by preserving 40 principal components I compressed original image with 1218 KB to 301 KB using a Matlab function which I wrote. Now about the entropy. The entropy of the original file was 6.93 and of the compr...
I was going through some of the basics of fourier series and fourier transform. And I came across one topic "Fourier transform of periodic signal". But I am not able to understnad why we need to go for fourier transform because we already have fourier series to take care of periodic signal. Any help is appreciated.
I normalized the spectrum of a time series and windowed the spectrum, then something strange happened. The code below can run in MATLAB or Octave. 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 ...
assume that I have a set of image and segmented image of a object in image (training set). Now, I want to measure features of object in image based on the training data. But I don't know how many features in image. And which is best feature for image segmentation? Gradient feature or entropy feature or ....Could you ex...
I am trying to store the signal I have created in a single variable and graph the signal in two subplots in one single graph, one for left channel, one for right channel. Fs = 44100; N = 400 t = (1:N)*(1/Fs); freq = 500; sound_array = zeros(N,2); %starts N x 2 matrix sound_array(:,1) = sin(2*pi*freq*t); sound_array(:,...
I'm trying to write a tempo detection algorithm for analysis of audio samples. I'm roughly following the approach described in Scheirer, the first stages of which give me some very nice beat onset plots, which I'm happy are correct. The next stages involve the actual tempo detection process, which go something like thi...
What fading model should I use for simulating rain loss in my network simulations? I am using NS-3 for this and I'm quite lost on how to go about this. My simulation is just simple packet streaming and I want to test the effects of various interference on throughput.
I've been looking at 2 different methods to acquire a known prn code in a CDMA signal. The first trys Xor prn code chip to I sign data. Then average the result over a time period to determine the data bit. I have not had reliable success with this method. The second which I am going to try performs a real multiply of a...
I was wondering what would be the statistics of the output of the filtering of an extremely wideband signal (e.g. 10 Gbps random data stream) through a narrowband low-pass filter (e.g. 10 MHz LP). I have the feeling that the statistics of the output would be somehow Gaussian due to the central limit theorem, but I cann...
I am using FFTW3 to transform the autocorrelation function of a particular function. I expect the transform to be real since the autocorrelation function is symmetric with respect to negative and positive lags and its Fourier transform give the power spectrum. However, I get nonzero complex part and negative real parts...
I'm applying at each 2048 samples the pitch synchronous algorithm to pitch scale some audio files, everything seems to occur well, but I've noticed some glitches, artifacts, clicks between every boundary frame! I'm testing off-line in large áudio files(I'm processing frame-by-frame), what I tried to do was keep the old...
When computing wavelet transforms with PyWavelets (Discrete Wavelet Transform in Python), e.g. import pywt cA, cD = pywt.dwt([1, 2, 3, 4], 'db1') >> print 'cA', cA cA [ 2.12132034 4.94974747] >> print 'cD', cD cD [-0.70710678 -0.70710678] how can I specify the scale parameter of the wavelet? In Matlab for example ...
I want to find the SER and BER of the 4QAM,16QAM,32QAM. I know that the formulas for SER and BER are different but I cant find them in any text book. The formulas for the BER that I found: 4QAM = Q(sqrt(2 ebno)) 16QAM = 3/4 Q(sqrt(4/5 ebno)) 32QAM = (8-sqrt2)/10 Q(sqrt(15/31 ebno) can anyone confirm and give me the fo...
By which value can the constellation of a 32QAM be normalized? for example 16QAM is normalized by squareroot of 10.
My aim is to determine the power spectrum of a signal of finite duration using Its autocorrelation function Performing FFT directly on the time-domain data The function is defined as follows in the finite time interval from 0 to 10: $$ \cos(10t) + \sin(20t) $$ I am using the library FFTW 3.3.4 in C source code and ...
Why do we need different modulation techniques? Is it because we need to convert from analogue to digital data sometimes? Also what are typical applications where modulation schemes are used apart from wireless scenarios?
I was wondering which is the best language for image processing? I know MATLAB has a good library and user community for it. I havent checked it out, its there for python as well. What is the most efficient language for this?
I am looking for methods to enhance noisy images, where: some pixels in the image are very noise, some other pixels do not contain so much noise. My first thought is to build an adaptive Gaussian filter. This means that the Gaussian kernel will depend on the (estimated) noise status of the current pixel (large radi...
I have been working on Digital signal processing since Feb-2013. I have good command over it. Now I want to expand work in field of IMAGE and Video processing. Please suggest me best practical guide books, blogs, online links and other supporting matters related to IMAGE and Video processing .
I've got a FIR filter, which I developed in IC. Now, when I got my results, I ended up with two signals: Theoretical signal calculated through MATLAB as a response to my input. Real response given by the hardware. What is the best method of calculating the error of the real response with respect to the theroetical on...
When transforming a noisy signal via Fast Fourier Transform from time to frequency domain there is a "Processing gain" of the FFT which increases as number of bins increases. I.e. the more bins I have the more the noise floor in the freqeuency domain is reduced. 1. Actually I don't understand completely, from where thi...
If $\omega$ is the frequency of a continuous time sinusoidal signal, $-\infty < \omega < \infty $. But when it comes to discrete time sinusoidal signals, the range becomes limited : $-\pi \le \omega_{discrete} \le \pi $. Does that mean discrete sinusoidal signals cannot represent all frequencies possible ? What is the ...
From this derivation to get the differential of a band pass filter: Id like to find out how we set the upper frequency cut-off and the lower frequency cut-off. $$i(t)=\frac{v_i(t)-v_o(t)}{R}=C\frac{dv_o(t)}{dt}$$ So we get $$v_i(t)=v_o(t)+RC\frac{dv_o(t)}{dt}\tag{1}$$ For the high-pass filter we have $$i(t)=\frac{v_o(...
I want to convolve two discrete functions $f$ and $g$ using convolution stride size $a$ to get the result as $s_{a, i}$: $$s_{i,a} = \sum_i g_k f_{ai-k}$$ I know that simple convolution with $a=1$ can be calculated via FFT. I can do it even now, but I have to drop out all the points except $ai$. I think this way is not...
I am trying to follow a paper where they say they apply the Kalman filter, but don't give the forumulation for the Kalman filter! Moreover I have looked at two references Wikipeadia and Durbin and Koopman 2001, but they seem to give slightly different formulations, I would like to know which is either is correct! The s...
From the textbook Signals and Systems by Oppenheimer ... If we define $h_k[n]$ to be the impulse response to $\delta[n-k]$, then for the system to be time-invariant, we must have $$h_k[n] = h_0[n-k]$$ I don't understand how the last equation follows. By definition of time-invariance, if the input is shifted by $k$ un...
A median filter is a non-linear and lossy process, so it doesn't have a closed form frequency response as would a FIR filter (say a box filter of the same length) in an LTI system. But how closely can something similar to a frequency response of a median filter be approximated? How would this scale with the length o...
I am analyzing some time series of sea surface water temperature and am using wavelet analysis to do so. I am computing the continuous wavelet transform and then removing some specific frequencies and then converting the series back from the frequency to the time domain. I am following the methods of torrence and compo...
I have read some papers about distance measures like Euclidean, Manhattan or Chi-Square for matching gradient based image descriptors like those computed from the SIFT Algorithm (128-D vectors). Most of them state, that one or the other measure is more suitable for that task and that this decision depends on the noise ...
This might be a terminology question but I am not sure. Basically, what is the difference between conventional beamformers, and adaptive beamformers? I thought that all beamformers were inherently adaptive to some criteria, like minimization of distortion or variance, or some other spatial filtering criteria etc. So w...
I'm using matlab/octave and would like some help trying to improve the speed of my code. I've tried to vectorize the code but I run out of memory so I went back to for loops. What I'm trying to do is have a function that will allow me to shift an array cell by cell. The reason I'm doing this is to have exact precisi...
I use Nearest Neighbour function of Scikit learn module in python for training and compare an image set (https://docs.google.com/file/d/0B6zJ3_hrkW40U182SWNWX3o4dE0/edit?usp=sharing&pli=1) By deafult NN function use euclidean distance, but in these way it mixes all feature without giving a weight. I use feature like so...