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I need to analyze some vibration data from accelerometers (g-Hz) but I don't quite understand the meaning of each component that the registered signal contains. Basically it is Magnitude and Phase. I'm able to get the Real/Imaginary parts from those two as well.
Today I plotted Real vs Imaginary parts (Nyquist plot/dia... |
I am making a software model of an existing digital synthesiser and I have made audio recordings of single notes with particular parameter settings. I made a program in C++ which sends a MIDI sysex message to change the parameter in question, then triggers a MIDI noteon and noteoff message. The sound is recorded to a w... |
I have two power plots. I am trying to normalize it for comparing them . So , do we normalize it after converting to db scale? or do we normalize the power as such?
|
Is there a formula to calculate the frequency of a phantom fundamental based on the harmonics, and would this formula be applied directly to the energy bins of an FFT?
|
Firstly I should state that my maths knowledge is limited.
I'm currently designing an acoustics modelling application which uses a rectilinear FDTD grid for modelling pressure variations - but I've become stuck upon trying to implement a fibrous layer boundary. From Kowalcyzk 2008, the fibrous boundary model is define... |
I have got a txt file which contains voltage and time data. When I did fourier transform using fft function in matlab I got complex numbers. I understand that the real part is the magnitude and the imiginary part is phase. How can I determine whether there is any aliasing by looking at that complex numbers?
Another qu... |
whats is the effect of pading zeroes to a sequence in FFT?
for eg: x[n]=[2 3 4 5] corresponds to X[K]=[14 -2+2i -2 -2-2i]
while x[n]=[2 3 4 5 0 0 0 0] corresponds to X[K]=[14 0.58-9.65i -2+2i -3.414+1.65i -2 3.414-1.65i -2-2i 0.58+9.65i]
so what is the relation between X[0] , X[1] and X[2] in the new sequence?
please h... |
I have a signal with 262144 samples and "sampled" with 200Msps.
I want to use the cwt (Continuous Wavelet Transform), for making a time-frequency scalogram.
Also, I worked in some scripts in python and Matlab, but it's very hard for the machine for make the process and empty the memory of the computer with 262144 sampl... |
I'm working on improving the modem communication speed that can be achieved over a 10km cable. The current modem achieves a speed of 9600 bps via hardware (phase locked loop) and I'm looking for ways to at least double this, and do it all in firmware. Here are some specs for the project.
Cable characteristics
Length: ... |
I know there are lots of papers proposing algorithms for the problem of reconstructing a signal from modulus of its Fourier Transform (so-called Phase Retrieval Problem).
Also, recently it is studied for the case of sparse signals.
I want to know whether there is any paper or survey working on "reconstruction of signal... |
This is probably off-topic since it isn't really a question, but I thought that this GIF of the fourier series of a square wave was too cool not to share.
|
I hope this isn't a dublicate . I checked but maybe didn't see everything.
Here goes :
I've been scratching my head over some EQ related questions. I read a lot of papers but my limited math and DSP knowledge prevents me from fully understanding them, so I thought I'd ask the experts.
I hghave a good library (http://... |
I'm reading an article about multi-scale representation of image, and it is said that, convolving the image with a Gaussian kernel at different $\sigma$, then different scale representation is created.
This really confuses me, because I think the image is just blurred by Gaussian with different $\sigma$, each blurred v... |
I'm new to signal processing and I'm learning on my own. Therefore I might ask questions not worthwhile your time.
At the moment I'm trying to figure out the concept of a lowpass filter.
Using matlab I created a complex signal x:
x = rand(1,1000)+rand(1,1000) .* 1i
And i created a filter Num with the fdatool, the spec... |
i have my output as vector of 60000 complex number representing 10 sequence in frequency domain , each of length 6000. i need to plot the phase and amplitude of intermodulation distortion from this output. can anyone suggest me how it can be done. i tried with magnitude and angle unwrap in matlab selecting particular b... |
I recently read that with compressive Sensing it possible to sample the signal at a rate lesser than that suggested by Nyquist Criterion. However, I am still not getting how this is possible. Can someone suggest an intuitive way of understanding this concept of sampling at lower rates? And if this is possible, do moder... |
I'm wondering if someone can clear the following up for me as I'm having a difficult time understanding the concept.
I'm looking to implement and learn about the processes and uses using short-time Fourier transforms. I have had experience using DFT's and wanted to expand my knowledge.
From research that I have read ... |
First of all I apologize for the long question. I am working with a 3-axis accelerometer and a 3-axis gyroscope and I need to calculate the correlation between the different signals (acceleration in x,y,z and the quaternion x,y,z,w). In the end, I need to get 21 correlations, one for each pair. I was thinking of using ... |
I am stuck on a research problem and do not know what kind of methods to use. I hope people in this forum can give me some good ideas, I've always had good brainstorming sessions here.
My problem is that I have a laser and a stack of thin films made up of three different materials. If I shoot the laser at these thin f... |
I want to do sample rate conversion by subsequently upsampling with factor I=5, and then downsampling with factor D=9.
I have designed a nyquist sample rate conversion filter h() of length M, with matlab's filterbuilder tool for interpolation factor I=5, and decimation factor D=9. Since D > I, I have chosen the normali... |
I've found good literature for offline proccesses, and I've gotten good results. However, when it comes to online applications, I've found nothing good. Applying any filter without a linear phase response will give me too much distortion once I get to displacement. However, any filter I've used with linear phase respon... |
I understand that sampling rate should be twice the signal bandwidth for successful reconstruction. But what I don't get is how this translates in the time domain.
For example, I have a sine wave pulse with frequency 100MHz and duration of say 50ns. This signal has a bandwidth of 20MHz (1/50ns = 20MHz). Theoretically, ... |
I'm studing about Kalman Filter and Particle Filter in multiple target tracking in computer vision (tracking pedestrians). Reading sientific papers I'm colliding with a lot of sentences like:
The assumptions of linearity and Gaussianity of the Kalman Filter
cannote manage complex scenario.
Now.. I'd like to find so... |
We call the conversion from a continuous time signal $f(t)$ to a discrete time signal $f_s[k]$ "sampling". Is there a name for the reverse operation, i.e. creating a continuous time signal from a discrete time signal?
|
I'm looking for your opinions.
I have obtained a Speech Signal and computed a STFT on this signal. I have split the signal into blocks of (100 x 256) and computed the STFT based on different Windowing functions. But, they all bring back similar results (outputted on the same block) I know this might make sense, but, I... |
I would like to improve my knowledge about Fourier Transform and related subjects.
What are the recommended resources in order to self study about it?
|
To start with, I'm pretty new to DSP, so forgive me failing to understand any basic concepts.
I'm working with an iOS app that depends on the input from the analog audio jack whilst monitoring the signal through output (headphones, etc). Unfortunately, iOS devices suffer from crosstalk given a high enough volume due to... |
I am trying to form an approximation of the wavelet transform from a partially sampled image.
Reconstruction in the 1D case is easy. We have $w = h x$, with $w$ as the wavelet coefficients, $h$ as the inverse of the Haar matrix and $x$ our signal. $w$ and $x$ are $n \times 1$ and $h$ is $n \times n$. Where we don't sa... |
I would like to recognize the boundaries of a hexagonal tiling in a photograph, like in image below:
It seems to me, that a standard approach at a square grid is to first detect corners (e.g. canny) and then extract the longest lines via a Hough transform or something similar.
This does not look as optimal solution ... |
I am trying to create a program that can count the number of nuclei in such an image:
What I've already done is the following, step by step:
Apply an Alternating Sequential Filter (closing and opening the image with gradually bigger structuring elements)
Apply a distance transform
Apply watershed segmentation using t... |
Matlab says
x = hilbert(xr) returns a complex helical sequence, sometimes called the analytic signal, from a real data sequence.
How did the name complex helical come about? How is it related to analytic signal?
|
I've seen some other questions on here about zero padding (like this one) but I'm still a little confused about my situation.
I'm attempting to zero pad my input time series data, so I get an interpolated frequency spectrum. I had initially tried this with NumPy's FFT package, and I checked my algorithm on generated d... |
I'm wondering if anyone knows of any open source solutions to note recognition. I'm planning on building a program for online sheet music, which recognizes your notes, and tries to tell you what you can improve, and how.
However, for piano for example, I don't even know if it's possible to have software that detects va... |
I am referring to this report on pitch estimation.
The author does envelope extraction using Hilbert transform before applying AMDF.
What is the advantage of doing that?
Is speech a narrowband or wideband signal? I read somewhere that hilbert envelope extraction works better for narrowband signals.
When I extracted e... |
I asked a question earlier but I didn't get any answer for it. So now I am simplifying it: what are Cross-Spectral Density (CSD) and Power-Spectral Sensity (PSD)? What is their application? How can I get them in MATLAB?
$$S_{kl}(\omega)=\lim_{T\to\infty}\frac{1}{T}E\{Y_k^*(\omega)Y_l(\omega)\} $$
$$S_{kk}(\omega)=\l... |
I'm learning feature-detectors from this lecture notes, and I don't quite understand the Normalized Laplacian of Gaussian filtered image.
Here is the original image:
this is the output presented in the lecture notes, filtered by Normalized Laplacian of Gaussian with $\sigma=2.502$:
and here is mine, using scipy.ndima... |
I believe that I read, for digital modulation with multiple levels, that the AGC makes use of the fact that the symbol sequence, before noise etc., can be modeled as a uniform distribution.
$$\text{Source $\rightarrow$source-coding $\rightarrow$FEC$\rightarrow$ interleaving $\rightarrow$ bit-scrambling $\rightarrow$ mo... |
I am using complex morlet wavelet to estimate frequencies and damping ratios in a 1-D signal. I have done it so far, and the results are accurate enough. Now my question is how can I modify or optimize my method to get better result. like mixing it with other method such as Neural networks or anything else. I know this... |
Say I am trying to measure a exponentially decaying signal ($S_y$), that constitutes a portion $\gamma$ (about 5% so $\gamma=\frac{1}{20}$) of total signal measured($S=S_y+S_b$ and $S_b=S_0 \beta e^{\frac{-t}{C_b}}$ and $S_y=S_0 \gamma e^{\frac{-t}{C_y}}$). Where $\gamma+\beta=1$ and $C_Y$ is about $\frac{1}{4}$ of $C_... |
I am trying to speed up some of my code and for physical reasons I need to do an FFT shift to get the FFT data in the standard DFT fourier representation. In Matlab this is performed in the frequency domain, with a data swap.
I had believed that I found a numerical trick, by performing the shift in the time domain, be... |
A bit of noob question, but can't find a definite answer anywhere:
What is the difference between Gray-scale resolution of an image and the Image contrast.
Are they different terminology for the same thing?
|
I have Digital Image processing (DIP) in my academics and I am looking for some reliable and daily updating sources for learning DIP Filters for Enhancements,color conversions, Filtering purposes through MATLAB , please suggest me some websites or tutorials which can help me. Thank you.
|
Background:
I am a newbie in DSP. I am implementing a simple Kalman Filter that estimates the heading direction of a robot. The robot is equipped with a compass and a gyroscope.
My Understanding:
I am thinking about representing my state as a 2D vector $(x, \dot{x})$, where $x$ is the current heading direction and $\d... |
This is primarily a question about how to name my filter. Suppose that I have a photon detector, and that I want to measure the rate at which photons arrive, with a simple gated counter (because this is the only hardware I have). The photon rate is just the counter value divided by the gate interval.
It seems to me t... |
I am using Q1.15 fixed-point values to represent a 12-bit signed integer value. I was wondering if it matters if I use the 11 MSB or LSB of the fractional part to represent my 11 magnitude bits? Only thing I can think of is using LSB might in some cases reduces the chance of overflow errors and maybe allow me to use l... |
As I understand aliasing is because you sample a high frequency with a low sample rate, and the sample points don't have enough information about the high frequency component. Here is an example of aliasing:
w = 5.0;(*signal frequency*)
dt = 0.66125;(*sampling rate*)
f[x_] := Sin[w x]
ls = Table[f[x], {x, dt, 200 dt, ... |
I'm trying to get the average frequency response of an entire piece of audio by splitting it into segments, doing an FFT on each one, and averaging the magnitudes together. Its working great until I get to the last segment, which nearly always is shorter than my FFT size, so I pad it with zeros until its the right size... |
suppose $x_n$ is an $N$-point sequence with periodic auto-correlation
$$
c_n=\sum_m x_m x_{(n+m) mod N}
$$
How to find $N$-point sequence $y_n$ with periodic auto-correlation $d_n = c_n - A$, where $A$ is a constant?
|
I'm developing an application which one of its capabilities is peak detection.
I expect to handle three types of the similar data (Acceleration vs Frequency) but measured in different laboratories and with different sampling and noise. The data is not equally spaced in frequency.
Here is the plot of each of them:
De... |
I'm trying to implement SIFT algorithm.
After I found the min/max points in DOG images I need to find the real min/max subpixels by taylor exapnsion:
$$
\begin{equation}
D(\mathbf{x}) = D + \frac{\partial D^T}{\partial \mathbf{x}}\mathbf{x} + \frac{1}{2}\mathbf{x}^T\frac{\partial^2D}{\partial \mathbf{x}^2}\mathbf{x}
\e... |
First of all excuse me for asking a seemingly vague question. But i am going nowhere in the problem i am trying to solve.
I am working on a powerline communication system where i need to upconvert in 28-56Mhz range and later downconvert it to 0-28Mhz so that it works in the coupler range.So i need to design a band pas... |
I am trying to analysis a system by Fourier Transform. my system has 3 dominant modes. (1,0.05) (0.51,0.01),(0.46,0.01) the pairs are defined as (freq,damping_ratio)
I've got those result with eigenvalue analysis.the system contains 3 signal.so these frequency[1,0.51,0.46] must be find in these three signals too. so I'... |
I have some basic understanding of signals and convolution. As far as I know it shows the similarities of two signals. Could I get some explanation in plain English of:
what are the linear and circular convolution
why they are important
practical situation where they are used
|
I'd like to have your overview if what I'm doing is OK.
I need to plot the BER vs Eb/No. So the noise power which I should generate should be related to Eb/No.
bits are encoded into 1/2 rate encoder.
the output of the encoder is converted to +1 and -1 symbols
symbols are oversampled by OS ratio.
the oversampled sym... |
I just studied about SURF and I'm going for its implementation, but I still didn't understand why we use descriptors.
I understand what keypoints are and their purpose, but when we extract the keypoints than why do we need to use descriptors ? What is their importance and role in recognition?
|
Sorry if this question is too simplistic, but I am new to image processing and after successfully writing two programs, a convolution program and a low pass filtering program, I noticed that for approximately the same output, the low pass filtering method takes 4 times the time!
Lets assume that the image is a square o... |
I'm trying to use Berchin's FDLS methods (for those that don't know what it is, it's a way to design arbitrary magnitude and phase of an IIR Eq)
I tried different things but neither gave me what i asked:
I tried to design an allpass filter with arbitrary phase response (nothing fancy just a linear phase and a short de... |
So the Gaussian function is an eigenfunction of the Fourier transform because it transforms into itself, right?
But this isn't true for the sampled Gaussian in the DFT because the tails of the function are truncated, right?
Wikipedia describes a discrete Gaussian kernel here and here (solid lines), which is different f... |
I've got a high-resolution image segmentation, which I'm downsampling (and non-linearly deforming) to fit on an ROI image. I typically blur the high-res image before downsampling with a Gaussian-fwhm related to the downsampling factor. The result is high-res and low-res images with the same mean.
I'd like now for my hi... |
Consider x(n) as input, y(n) as output & h(n) as filter coefficient. The filter is of 16 taps(N=16) from 0 to 15. The decimation factor of the filter is 4.
When I apply x(n) to this 16 tap decimation filter with decimation factor 4, what is the output that i should get?
What is the general relation(give formula) betwee... |
I'm finally making some headway in understanding data whitening, but I have a question now relating what I think of as "data-based" whitening to the situation where there are time-varying signals.
So in the "data-based" world, suppose I have a mean-centered column matrix $X$ with covariance $\Sigma = \mathbb{E}[XX^\top... |
I have two sets of MFCC coeficients which I get with fftExtract utility from two wav audio files.
How can I compare this sets (matrix 271x13 and 551x13) if they are similar ?
Probably I should use EMD distance ... But Im sligthly lost in all this math and I dont
know how to interpret its results .
How can I make fro... |
The DFT of a real signal is Hermite-symmetric, so you can roughly halve the computation time/memory by not bothering to calculate half the values of the spectrum (and complex conjugating the existing values and copying them to the second half if needed). So the rfft operation takes N samples and outputs N/2 spectrum b... |
I have a signal and from that I want to extract a specific frequency range, e.g. 10Hz to 15Hz. Now I heard that I should use Fast Fourier Transformation.
I'm going to do that in Matlab, but I have no idea where to start. Is there any good example how I can do this by passing the original signal and the frequency band ... |
There is an example of inverse filtering in my textbook which I don't fully understand.
Given the two filters v(t) = (1,-0.5) and w(t) = (-0.5,1). The inverse of these filters are given by:
for v: 1:(1,-0.5) = (1, 0.5, 0.25, . . .)
for w: 1:(-0.5,1) = (-2, -4, -8, . . .)
However, there is no mathematical illustratio... |
What is meant by Least square plus Gaussian method of estimating the parameters of an unknown system? I am aware of least squares in general but some works refer it as least square+gaussian method. What is the role of Gaussian here? Is it for some kind of assumption of the distribution of the residual errors? please ... |
I am averaging several FFT results together in an attempt to get the average frequency spectrum of a track of audio. I'm attempting to use Parseval's Relation to get the RMS of the audio from the resulting average spectrum, but i've noticed the RMS i get is wrong, usually only by a couple dB. Why does this happen? Does... |
I'm new at Fourier Transforms. I'm using AForge.Math FFT/DFT algorithm (C#). Said that I have this situation:
My signal:
$$
1.4 + sin(2\pi t) + 0.4sin(5·2\pi t)
$$
Sampling: 100 samples each second.
Here is the spectrum. The DFT finds the frequencies of my signal: 1Hz and 5Hz (and the constant component: 0Hz).
Here... |
Suppose that I have two signals $x[n] = \left\{2,4,1\right\}$ and $p[n] = \left\{5,1,8\right\}$ and I want to multiply them.
How do you do that?
How different is it from convolving two signals?
I understand that multiplication in one domain is equal to convolution in other domain. How do you choose as to what to use ... |
I'm studying the IIR filter design that is described in the book: Algorithms for the constrained design of digital filters with arbitrary phase and magnitude responses.
You can get the code at page 171 (at least the main function), and here is an example of a filter design :
M=4;
N=4;
tau=5;
om=pi*[linspace(0,0.2,20),... |
I am multiplying two sine waves with the same frequency (f), but might have a phase difference 0 <-> 90 degrees. The product is a two frequency sinusoidal wave with f1 = f-f = DC and f2 = f+f = 2f. I now want to filter out the 2f component. I am currently sampling at Fs = 32 x f. I get a 40th order equiripple FIR filte... |
I understand that digital audio has both a sample rate (time "discretization") and a bit depth ("amplitude" discretization). Also, it can have multiple channels (stereo would be 2 channels).
But things that I can't find readily in the internet:
What is the range that is discretized in the bit depth? My experience is i... |
i would like to add even harmonics from the data of FFT of a signal.
How do i do that ? do i have to search for higher magnitude frequencies then add the $2f$ then $4f$ etc ?
I understand how to do it from a sin wave for example but not from a noisier signal.
|
I need to scan an image in OpenCV and obtain the lines. For this I use the canny edge detector and then the propabilistic Hough transform, with which I actually get the lines.
The problem now is that I get many short lines instead of a few continuous ones. The image I am analyzing has the size 1024x768 and I am wonder... |
I extracted features from an image using SIFT and SURF. Then I applied few transforms on the image and again calculated the features. While matching the features, I noticed that there are some erroneous matches. For example, if using image of a face, some points on face are matched with points not on face in the transf... |
I saw the following sentence from Feature Extraction & Image Processing, 2nd Edition
$$
x = a_0 + a_x\cos(\theta)+b_x\sin(\theta)
$$ $$
y = b_0 + a_y\cos(\theta)+b_y\sin(\theta) $$
This equation corresponds to the polar representation of an ellipse.
I don't know this meaning. If you need extra information, I'll add ... |
There are several kinds of detectors and descriptors, like SIFT, SURF, FAST.
I wonder are they all eligible for real-time applications? Which is the best or better?
And furthermore, is Harris-Laplacian dectector still useful when we already have the above three? Is it better than them?
|
Does anyone know about different adaptive filtering implementations (LMS, RLS ...) in 2D or even 3D ? I have sequences of 2D images and 3D volumes with repeating patterns but small differences. I was thinking of using one as my reference input and extract differences between the pair (A simple subtraction doesn't work ... |
In scale-adapted Harris detector, the scale adapted second moment matrix is defined by:
$$\mu(x, \sigma_I, \sigma_D) = \sigma_D^2\ g(\sigma_I) *\left[ \begin{array}{cc}
L_x^2(x, \sigma_D) &L_xL_y(x, \sigma_D) \\
L_xL_y(x, \sigma_D) &L_y^2(x, \sigma_D)
\end{array} \right]$$,
while $\sigma_I$ is the integration scale, an... |
I am working with phoneme clustering. I have been using $k$-means clustering have done $k$-means on 2 different data: spectrogram vector and MFCCs. I would like to know how I can evaluate the result of clustering quality? I mean I need to know standard ways to validate the clustering. It would great if someone explains... |
I have two sets of data of given Field of view, one of them only covers a subset of the FOV of the other. I therefore want to upsample the one with the larger FOV to combine it with the other one.
So the question is: In the context of X-ray computed tomography, where the x-rays projections are backprojected to obtain a... |
i'm trying to implement an arbitrary IIR filter design. I'm decomposing all the algorithm into smaller pieces that i can test.
I currently am on the part where it designs a FIR filter from an arbitrary response.
Matlab's main function is here :
function h = lslevin (N ,om, D, W)
% Complex Least Squares FIR filter desi... |
I'm developping a people tracking with nearest neighbour data association technique and kalman filter for smoothing/predicting.
The tracker works quite good but sometimes it mismatch objects if they cross them-self.. (I think it is a common problem for a tracker).
I really believe that this tracker could be strongly im... |
I am trying to make program in matlab that counts the number of times the accelerometer was lifted up and down (a rep counter for exercise). Basically my phone is getting accelerometer data on all three axes and then i import this data into matlab. I have tried to filter the data using a simple filter a = [1 -0.9] b=0.... |
I need to create a diagonal matrix containing the Fourier coefficients of the Gaussian wavelet function, but I'm unsure of what to do.
Currently, I'm using function ConstructHaarWaveletTransformationMatrix to generate the Haar wavelet matrix and am taking the rows at dyadic scales (2, 4, 8, and 16) as the transform:
M ... |
If I have a 2D scalar image, it's trivial to display it with MATLAB or matplotlib type tools with the pixel values mapped via a color map, and there is any amount of literature (one example, another one) describing the perils of those tools' default "rainbow" colour-maps and the advantages of various other schemes (col... |
I'm trying to figure out how to convert any phase response to the corresponding group delay response.
Is there a way to do that, and vice-versa if possible?
|
could you tell me step by step the algorithm to design an arbitrary phase response IIR allpass filter design ?
I know Matlab uses a "brute force" algorithm to find an arbitrary group delay response but how do they do that?
|
Are there any good published approach for designing parametric EQs as FIR filters. The standard techniques discussed in DSP textbooks are frequency-domain-sampling or using Parks-McClellan, which this involves passband, stopband, transition band, etc. This is nice for rejecting a certain portion of the frequency spec... |
I have successfully extracts all Pitch Markers from waveforms, I used autocorrelation to find my pitch periods, The markers can be observed in the figure below:
I want change my signal (Pitch or Speed) using PSOLA, I need find the synthesized Marks, for example if I choose change my Pitch one octave UP(0.5), for it I ... |
First of all thanks for your patience: it is the first time for me posting a question in this forum. I am not a DSP expert, but I should get by if you give me in depth explanation.
This is an example of my raw data
and a link to the actual dataset (time=t0, acceleration=acc).
My problem is the following: I am measur... |
I come from Fluid Mechanics domain so pardon me for asking a naive and fundamental question.
I am trying to perform high pass filtering on a set of positive real numbers (~Energy) using FFT. However, the result I obtain has a lot of negative values. I tried to quantify how many of them are negative and it seems $\sim5... |
I'm quite new to machine learning/data mining and I'm struggling to find the correct path for my problem and would appreciate some guidance or criticism of my proposed solution i.e. is there a better/simpler algorithm for the problem?
The Problem
I have a number of features that describe a particular type (label) of wa... |
First of all I'm new to DSP so excuse my simplified words & I'm sorry if it's a duplicated question.
Now, I'm trying to filter a signal but as you all know, the filter step response overshoot the filtered signal at the beginning.
Here it's the signal. It begins with zeros then the signal comes up
Is there any accurate... |
Let $x[n]$ and $y[n]$ be two orthogonal baseband signals, with bandwidth B. Check if the following signals are orthogonal:
$u[n] = x[n]*\cos(2*\pi*f_c*n)$
$v[n] = y[n]*\cos(2*\pi*f_c*n)$
I'm not sure what the problem asks to be honest.
1) What exactly does he mean by bandwidth?? I only know the definition of a bandwi... |
Suppose we have filter $h[n]$, length $N$. If I use it to do convolution in the time domain, then there will be $N$ multiplications each time during the sliding. (Ignore border conditions for now). However if say $h[n]$ had zeros every other sample, then there would be half that number of multiplications, so ostensibly... |
My signal oscillates back and forth on the interval [-1, 1] at varying frequencies, with some added higher frequency noise components (example data shown below). I want to to estimate the phase of the slow oscillation at any given time. This means I would like to filter out some of the high frequency noise, and then ca... |
I am trying to write a connected component labelling program, I have done the first part to create a label based on the neighbours. For simplicity I am using a 4-connectivity. I am using ANSI C and to create the equivalence table I am using two linked lists with structures defined like this:
struct T2_node
{
size_t... |
I am trying to implement audio dynamic range compression in JavaScript (not using web audio API).
There is a lot of articles for sound technicians, and some high level documentation, but I couldn't find any helpful reference for actually implementing digital dynamic range compression.
From what I understand, there is a... |
I'm new to this multiple view geometry issues and I'm a little bit lost, and some concept are not clear at all.
I have a calibrated camera ( = I have distortion coefficents, field of view, and with those I can find what is called intrinsic matrix).
With this camera I'd like to make an application that can estimate the ... |
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