instruction stringlengths 31 27k |
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I don't understand this: the HT works as it should but in an .AC simulation, the phase shows up like this (wrapped):
Why doesn't it look like a constant -90deg phase, like in a differentiator?
[edit]
The thing that also confuses me is that a regular low-pass FIR has the same phase, so does that mean the FIR also add... |
In paper:
A. Vlad, A. Luca, and M. Frunzete, "Computational measurements of the
transient time and of the sampling distance that enables statistical
independence in the logistic map," in Proc. Int. Conf. Computational
Science and its applications, Berlin, 2009, pp. 703-718
it is mentioned that for a time serie... |
I am having difficulty in developing an algorithm for checking if a given system is time invariant or not.
For example, let there be 2 systems with input output relation given as:
$$y(t)=x(t-2)$$ or $$ y(t)=tx(t-1)$$
Now the first system is time invariant whereas the second one is not. But how do I write a function in... |
Consider how the Hanning window is defined:
0.5 - 0.5 * cos(n*2*Pi/(N-1))
By this definition, it has a gain of 0.5, which is simply the average value of the coefficients. By contrast, Flattop windows, as defined, have unity gain, presumably by design.
It would seem appropriate to scale the Hanning window by a factor ... |
I have a signal with half-amplitude $A$ and period $T$ and I want to calculate the average power of the signal.
I think I remember the power of a signal $x(t)$ to be $$\frac{1}{T}\int_{0}^{T}|x(t)|^2dt$$ but how do I apply this to a square wave?
|
From my understanding of Fourier transform, a Fourier transform of a signal in time domain will give the different frequency components of the signal in frequency domain, specified by their contributions(energies) in the original.
The Fourier transform of rectangular function is a sinc function
If I take Fourier transf... |
I have data that looks like this:
Sometimes the data has a higher point in the middle of the shallow slope
I want to find a way to filter the data such that it smooths it and leaves the first rising edge as the peak like so
Is a weighted average a good way to achieve this? or would that not do the trick? Are there b... |
I have a question: Do DCT each output coefficients have to be orthogonal?
Here DCT is DCT-II type. It is used by the jpeg compression algorithm.
|
I want to detect multiple rectangles in a video, each of which features a different content, but running simultaneously. This is typically seen during debates on news channels. For example, in the image below, there are 5 rectangles (each containing a person) that are of interest to me and I want to keep a track of the... |
As we know, a complex analytic signal can be obtained through Fourier transformation of a real value function, and the imaginary part can be reconstructed by Hilbert transform of real part. But I have simulated a real value function using Matlab and performance FFT on it, the imaginary part obtained through FFT is very... |
I need to perform a deformable image registration (DIR) for delineation of the different brain parts on the brain MRI 3D image. I faced with a problem that there are a lot of different approaches for solving this problem while I don't have enough experience to choose the right approach. First off I tried a lot of diffe... |
Suppose I have a DFT matrix $F$ of dimensions $N \times N$ where $N$ is a power of $2$. I convert it into a unitary matrix $\displaystyle U = \frac{F}{\sqrt{N}}$ and compute $S = U^* U^* $.
$S$ seems to have a pattern. $S$ has at most one $1$ per row. Applying $S$ on my $N$-sample signal $x$ essentially provides me ei... |
I'm trying to determine whether or not a system with impulse response
$$
h(t) = \sum_{n=-\infty}^{\infty} \delta(t-2n)
$$
is BIBO stable. I haven't touched this material for a very long time -- could anyone lend a helping hand? I recall needing to show that
$$
\int_{-\infty}^{\infty} |h(t)| \, dt < \infty,
$$
but $h(t)... |
How would you compare the SNR gain from a longer FFT (more samples) to overlapping FFTs? Given the choice, which one would you pick and why?
Update: Suppose you have 2048 samples of a signal. Assuming the sampling rate divided by the fundamental frequency is an integer, compare/constrast the SNR gain/noise variance red... |
Like the title says...
I want to split audio file where a single instrument is played (not so fast)...
I just need to detect the attack phase of the waveform to know where a played note begins. Can someone suggest a way to do this? It should be robust, not to react to isolated peaks...
If there are open source implemen... |
The seminal paper of Cooley and Tukey provides an iterative method for computing the DFT for a sequence of length $N$. Specifically, they mention a method which utilizes the fact that $N$ can be written as $N = r_1 r_2$. The two intermediate results during computations themselves are DFT computations, therefore, their... |
I was thinking, if a realtime system is one in which the processing time is shorter than the sampling period, how does a feedback filter such as a biquad perform the computation without waiting for, say, in the case of the biquad, two additional samples?
|
I'm familiar with the general methodology for checking if a system is linear:
Define output $y_1[n]$ for input $x_1[n]$, define output $y_2[n]$ for input $x_2[n]$
Define $x_3[n] = ax_1[n] + bx_2[n]$
Compute output $y_3[n]$ for input $x_3[n]$ and verify whether $y_3[n]=ay_1[n]+by_2[n]$. If it is, the system is linear.
... |
I am not overly familiar with image processing. I know a little about it from University, but I have not really done any for a while and I have found myself with a problem that requires it.
What I have is frames of a video which may or may not have blocks (usually 16x16) of solid colour (either purple or green) randoml... |
I've written my own FFT but wanted to know if there was perhaps a way to combine the Cooley and Turkey methods. So far I've got N/2 log N real multiplies for transforming a real signal.
My current implementation has N log N multiplies (well a little more because I've not optimised out the case of the complex sinusoid a... |
Weighed overlap add Filterbank structure is often used in speech coding.
(https://www.onsemi.com/site/pdf/Filtering_Hearing_Ai.pdf)
I have basic question with respect to analysis.
what is the difference conceptually between:
a) direct analysis: e.g La=R=N1=256 This includes:
extract R=La samples of frame from speech ... |
I have found this interesting question here:
Which of these two waveforms has a richer spectral content, and why?
Intuitively speaking, I think that the first one is richer, but I cant really justify it! Cheers for any input.
|
I am doing template maching using Normalized Cross-Correlation for measuring similarity between template and image.
However, some images from my dataset contain saturated pixels (overshoot) which should be avoided in the measurement.
How to update the cross-correlation computation to avoid saturated pixels?
|
I have data from an experiment and apply a lowpass filter (digital Butterworth filter, in Matlab).
My question ist what happens if I apply the same filter e. g. 2 times on the data?
What is changing? Is it like I have the double order?
The question refers to the filtfilt function in Matlab:
http://www.mathworks.de/de... |
With help from the dsp guide and DSP-Related, I tried to implement a leaky integrator in octave/matlab. It seems to work in general, but there are a few problems.
So that's what I do:
Calculate the optimal integrator (1/jω, damping below fc Hz) in the frequency domain
Band limit this filter (using custom_filter.m)
App... |
Consider a noise-cancelling headphone:
If I have a noise signal from the outside world, mic it, flip the polarity by multiplying by -1 (with an op-amp or digitally), delay by the appropriate amount, and playback out of the speaker -- the summation of the noise and anti-noise at the ears should approach zero (with bette... |
There are 2 audio signals, X & Y.
X is of 15 second duration. The duration of Y is not given.
Y is a part of signal X (ie a small part of audio X is extracted & made into audio Y)
I have to find the starting & ending location, in seconds, in the signal X from where signal Y is derived.
I was told to NOT use built-in MA... |
I'm not sure that's the right forum to ask this question but I'll do it anyway.
In my project, I've to listen to different machines using a MEMS mic by fixing the microphone directly on to the machine. The problem that I get, is that all of the records that I need, are overcharged (clipped) so useless. My question is ... |
Long term evolution (LTE) systems use single-carrier frequency division multiplex (SC-FDMA) in the uplink. In such a system, there is a DFT block and a sub carrier mapping block before the IFFT block. What is the significance of DFT and sub carrier mapping? And how do these blocks make the system single carrier?
|
in wireless communication,compared to the OFDM system , why does the SC-FDMA (single carrier FDMA) system has a lower PAPR (peak to average power ratio)?
regards,
phani tej
|
I am applying a flat-top window to signal because I am very interested in its actual amplitude in the time-domain. I have read that overlapping will improve results. I am using MATLAB and the following is the code that I am using to compute FFT with flat-top window:
time = 0:0.1:500;
data = 1.8*sin(2*pi*0.2003*time);... |
I know LTE doesn't use the DC subcarrier for OFDM/OFDMA modulation for the downlink, but no author gives a detailed explanation to this stuff.
Can you tell me the reason behind it?
Thank you!
|
I have a problem in my DSP class that I could use some advice with.
An analog signal of length 2.5s is sampled with a Nyquist frequency and the sampled sequence has 5001 values. What is the highest frequency contained in the analog signal?
What I've gathered that the sampling frequency is $5001/2.5 = 2000.4 \text{ Hz}$... |
Ok, so we have an image that is a Fourier inverse of the original picture. We want to get the original picture back. We use Matlab to get that job done. We import the image and then we invert it with the help of ifft(), this gives us a matrix with complex numbers. But to get the original picture we need to do some oper... |
Have you ever seen any articles about using morphological-like technique for denoising color images?
|
in an OFDM system having a radix two FFT (Say N=64) . all of these sub-carriers are not utilized. while some of the carriers are nulls to prevent leakage in to adjacent bands, say a four to five carriers will be pilots. and the remaining will be the data carriers. what is the significance of pilot carriers in OFDM syst... |
I'm learning (trying to learn) how to find a frequency response. I have a discrete time equation and want to understand the process for finding a frequency response.
Ex:
$$f(n) = 2\,\delta(n)-5\,\delta(n-1) $$
Thanks
|
This is an image processing question.
In 2-D Euclidean space a pillbox function, $f(x)$ is given by:
$$f(x,y) = 1\text{ if } x^2 + y^2 < R, 0 \text{ otherwise } $$
$R$ is come constant.
If I try to approximate the top hat when I'm looking at an image matrix of points, a naive algorithm would make some of the image pi... |
I'm trying to do bicubic interpolation on an 8*8 matrix(image) shown below.
A = 10 20 30 40 50 60 70 80
90 100 110 120 130 140 150 160
170 180 190 200 210 220 230 240
250 10 20 30 40 50 60 70
80 90 100 110 120 130 140 150
160 170 180 190 200 210 220 230
240 250 10 20 30 40 ... |
I’ve been trying to estimate the frequency of a signal I obtained at 1000Hz sampling frequency. The dominant frequency is expected to be not more than 4 or 5 Hz. I’ve estimated the PSD using autocorrelation method however for some signal it seems to noisy and it’s difficult to find the dominant frequency. I’ve tested t... |
Inside ADC we have a quantizer, when the sampled signal is passed through it, the signal values get discretized. The number of discrete levels in which the signal is discretized is called the resolution of the quantizer. I read that if we oversample the input signal and then pass it through the quantizer and pass the o... |
I am trying to calculate the disparity between stereo image pairs. I'm using what appears to be the standard approach of correlating between a window of pixels from one image and a sliding window from the other. The horizontal shift that gives the best correlation is the disparity.
However, the images contain large a... |
As seen from the image below, an ideal high pass filter was applied. Hence, stark changes in pixels from lighter to darker or vice versa show up on the filtered image. However, slower changes such as on the cheekbone appear to be this gray color.
What is this gray color? It seems to be quite prevalent in all my graysc... |
I'd Like to ask two questions :
What is the difference between studying Signal processing (both Deterministic and statistical) in Department of Electrical Engineering versus Department of Mathematics and statistics
What's All the mathematical topics (both time/n-domain and f/S f/Z domain ) which is a prerequisites of s... |
Say for example, I have a signal with maximum frequency at 600Hz, and I originally oversampled it at 2000Hz. Afterwards I was only interested in frequency less than 500Hz. I guess I could first filter out frequency >500Hz, then downsample by taking every second sample point; or I could first downsample by taking every ... |
Discrete time systems work on sampled continuous time signals. We need samples only at discrete time to be fed as input to a discrete time system. Does this mean discrete time systems are clocked so that the values only at discrete time are available to the system? How are these discrete time systems implemented physic... |
We try to multiplication operation of a black & white image with RGB image in same image but their class is not same: BW -> logical, RGB -> UINT8.
How can I convert from Logical to UNIT*? Or, can we do images which is in different class multiplication?
|
I know MATLAB has a built in Radon function, but I am working on implementing the radon transform in order to perform filtered back projection. As I started, my idea was multiple nested for loops for the summations at different theta values, this became pretty complex. I am not looking for source code, just suggestio... |
I have one confusion in QAM about in-phase I and quadrature Q part, i.e. do they transmit identical data or not? For e.g. if i have 11001001 as a byte to be transferred, so will 4-QAM separate it like?
I 01 00
Q 10 11
or both I and Q are similar at any instant of time?
Thanks for answers
|
Let assume we have a signal as:
$\color{blue} {x(t)=M e^{(-\beta.t)} e^{j(\sigma .t+\delta )}}$
By applying phase angle operator we have:
$\color{blue} {Phase[x(t)]=Phase[M e^{-\beta.t} e^{j(\sigma .t+\delta )}]=\sigma.t+\delta}$
I have two questions:
1. How can I implement the Phase angle operator, i.e. Phase, in pra... |
I have a sound that is being recorded but I am trying to apply bandpass filtering to the sound so that it filters out anything above and below a selected frequency range. I have tried the tool in matlab to create the filter but for some reason can not get it to work. The code below is the record portion that I have the... |
I am currently pursuing a Master degree of Engineering and I will have to write a thesis about solving a specific Machine Vision application. I have done some CV projects in the past but now I am tinkering about the methods of problem solving in the broad field of object detection. What I noticed is that it looks like ... |
How to constrain the object detection problem?
It would be tempting to ask for precision such as: "It depends, does it have to be viewpoint invariant, is there going to be viewpoint changes, etc.?" Actually, it is exactly what I want to know. I have a specific application in mind and I need a structured way to help me... |
so, first let's do this for the circular correlation case (it might be easier).
let's say we have three real and periodic functions or signals $x(t)$, $y(t)$, $z(t)$ all with period $2 \pi$:
$$ x(t+2\pi) = x(t) \quad \forall t $$
$$ y(t+2\pi) = y(t) \quad \forall t $$
$$ z(t+2\pi) = z(t) \quad \forall t $$
define the c... |
I hear the word warp a lot, but cannot quite get a feel for what it really means
I suppose it's literal meaning would be to change in shape, but what does it mean with respect to DSP?
More specifically, I have a DSP chip manual that says a certain parameter adjusts warp for DACs and another one that adjusts warp for AD... |
On Video recording with IR camera, a file in .ptw is obtained. I want to analyze the file using standard software, like MATLAB. However, video processing in MATLAB requires the file to be converted to some standard formats like .avi or .mp4 or .wmv files. Please provide a way to make similar conversions, with the help ... |
I am measuring wind speed using a number of different anemometers. These anemometers are placed on a structure which exhibits sinusoidal motion. Thus, the signal of the anemometers contains both the wind speed as well as the sine wave to which they were exposed.
I wish to accurately determine the amplitude of the sine ... |
I am currently struggling with a problem when calculating the instantaneous phase of a wavefield using a 1D Hilbert transformation.
The scheme is as follows.
1D Hilbert transformation of wave field $U(x,z)$
$$
q_z(x,z) = H_z[U(x,z)] = \int_R U(x-\xi,z)\frac{d\xi}{\xi}
$$
Calculation of the instantaneous phase $\phi(x,z... |
I am making a 9th order digital bandpass filter with lower and upper corners of 200 kHz and 40 MHz respectively. I am using this filter to filter a 1D time domain signal which is 64k samples long sampled at a frequency of 150MHz.
I have done some digital filtering before in university, so I know what to expect, but i... |
I am looking for a treatment of fundamental DSP theory where only discrete/finite signal models are used. To make a concrete example, every time standard textbooks write a convolution as a continuous integral, I would like to see instead a product between a Toeplitz matrix and a vector. It seems to me that developing s... |
I'm trying to create a digital filter in code(C) but any language is fine. Now I've got an analogue filter that I have represented by an equation in the Laplace domain and I want to try and implement it digitally.
So my filter has this form in the Laplace domain:
$$\frac{as+b}{cs^2+ds}$$
I then use MATLAB's c2d comman... |
The code in this page solves the Least Squares problem for the following dynamic model:
$\dot{y}=ay+bu$
where $a$ and $b$ are constants, $u$ is an input. The code is as follow:
t=[0:0.1:10];
m=length(t);
y=zeros(m,1);
a=-1;b=1;c=1;d=0;u=[100;zeros(m-1,1)]; % u: impulse with magnitude of 100
[ad,bd]=c2d(a,b,0.1);
y=dlsi... |
I am having a comparison of different feature detection methods. So if we compare the SIFT (Scale-invariant feature transform) and Shi-Tomasi method, will their feature locations agree and why? Are they comparable? Thanks in advance.
|
I am trying to add a few samples for a publication on identification of clean signal from the one which has echo on it. For this purpose I need data set that includes playback of an already available signal in a big room/hall and the recording of the signal with a mic in this hall that includes the echo.
Does anybody k... |
I have written the following code which is supposed to put echo over an available sound file. Unfortunately the output is a very noisy result which I don't exactly understand. Can anybody help me with regard to this? Is there any skipped step?
#convolving a room impulse response function with a sound sample both of ste... |
We know that OFDM eliminates the inter-block intereference. But there are still multipath interference in a single OFDM block(symbol). How does LTE handle this intra-block interference induced by multipath components so as to successfully demod. the signal ?
|
I was actually preparing for my semester project and decided to take up Frequency estimation. I ll directly come to the point that I want to know how should I learn MUSIC algorithm. How do I start it? What are the pre-requisites? Do I need to learn any specific mathematics?
Thanks in advance.
|
I have 2 data sets sampled at regular intervals which represent the input and the output of a system. A quick analysis of the plots shows the following:
There is a proportional factor between the input and the output.
There is a delay before a change in the input is observable in the output. This delay depends on the ... |
Hi someone please explain me the translation property of $\textrm{DFT}$. I am not able to understand it neither from Gonzalez nor from internet. I have done extensive study on this but not able to get it. Really Frustrated!!
I don't understand how shifting is done? What is the difference between $f(x-x_0, y-y_0)$ and $... |
I have a signal from a photodiode sensor that has two types of noise. One type of noise is ambient light white noise that gets introduced just from the surrounding environemnt. The other type of noise is due to the motion of the sensor.
Currently i don't seem to have a problem filtering out the white noise when the sen... |
I'm working on some problems that instruct me to find the impulse response of a given system. For example
$$y(n)=4y(n-1)-4y(n-2)+x(n)-x(n-1)$$
The first step I take is to find the homogeneous solution of the equation like so:
$$ \lambda^n-4\lambda^{n-1}+4\lambda^{n-2}=0
\\\lambda^{n-2}(\lambda^2-4\lambda+4)=0
\\\lambda... |
I want to try to improve an image processing technique for finding faces in images. I want to stop the Viola Jones face detector before it finds all faces, and do histogram equalization and other techniques on "candidate" faces. For example, say we have an image with one face. I would like to cut off facial detection b... |
I have seen a lot of posts here and on SO and on various websites, using Matlab, OpenCV, name it, on how to detect rectangles in an image. I have done it myself too, using Matlab. My usual steps were:
Grayscale the image (if not done)
Blur the image (if needed)
Apply an edge detector (e.g. Canny, LoG, etc., tried a lo... |
I really having a hard time to tackle the following question.
It is asked to load a ".mat" file using load file.mat. And the result is two variables: Fs varaible contains the sampling frequency=44100, y is a 441344x1 matrix that contains data points of this audio.
It is asked to amplitude modulate this audio by multi... |
I once learnt that the Huffman Encoding is used to rid of redundant information, thus reducing file size. I am not sure how using a DCT, or FFT on an image can help in reducing file size. It does not work on redundancy, then how does it reduce the total amount of data?
|
From wikipedia:
"Modern general-purpose speech recognition systems are based on Hidden Markov Models. These are statistical models that output a sequence of symbols or quantities. HMMs are used in speech recognition because a speech signal can be viewed as a piecewise stationary signal or a short-time stationary signal... |
Is there any way I can apply the concept of FIR filters in chroma key composition, where I will remove a green background to focus on a subject and replace a different image on the removed part, sort of like how they impose graphics on weather reports? If so, how can I implement it in MATLAB?
|
Please check the link
What effect does a delay in the time domain have in the frequency domain?
Here it is said that if you delay your input signal by D samples, then each complex value in the FFT of the signal is multiplied by the constant exp(−j2πkD/N).
My question is that if i have a DFT vector F(u)={11,-3+2j,-1,-... |
I am trying to take a section of a song (approx 1.5 seconds) 44100 sample rate equals sample size of 65536 which is 2**16.
I want to find the frequency of the piano note that is played in the window of time.
I have implemented the four1 function - real data starting at array pos 1 and I data at 2 ...
four1 uses are re... |
One method of producing a discrete approximation kernel to a Gaussian filter of variance $\sigma^2$ is to assume a cutoff of $5\sigma$. This seems to suggest the following procedure:
Create a uniform grid $\mathbb{x}$ of size $11(=2*5+1)$ between $[-5\sigma,5\sigma]$
Evaluate $\mathcal{N}(\mathbb{x};0,\sigma)$ and ob... |
I am new to dsp and just have a vague understanding of fft process. i know fft is used to convert time to freq domain.... i need to generate a sequence of frequencies such that F=[F1,F2,.....Fn-1] where n=16 and then need to do the ifft of it.... i need these freq sequences to be random... My friend told me that the se... |
I am using an accumulator in the form of
X + [Y-1], X plus the unit delay of the output.
Is there an accumulator that follows this form, but will fall to 0 over time, similar to a differentiator?
|
I'm trying to understand the aim of overlapping frames in the short term fourier transform. Why is it important to do it? I have been told that it's to avoid "problems" at the frontiers of frames. But does anyone have a more precise or mathematical explanation?
Thanks.
|
I came across this root locus diagram in my DSP class. It has a pole at the origin, another pole within the unit circle and a zero just outside the unit circle. I understand that the poles and zeros should be complex conjugates for an all pass filter. So will a pole or a zero at the origin prevent the system from being... |
I have data from a power transformer that has a frequency of 60 hz. Sometimes there are faults in the system and causes a large spike in the reading of the transformer. What I want to do is run the data into a signal processor to detect when these large spikes occur and either remove the spike value or present the ti... |
can anyone please help me with finding Convolution for following problem .....
1) $h(n) = \delta[\sin(1 + |n|)]\quad \text{and}\quad x(n) = \sin(n^2)$
2) $h(n) = \delta[\sin(2\pi n/N)] \quad \text{and}\quad x(n) = n^2$ where $N$ is odd.
|
Suppose we have two signals( focus to audio) recorded with different methods,
or more simply, suppose we have two audio signals recorded with different microphone, and we want to confirm that these signals are of the same audio.
What are the available alogorithms or libraries that make this possible?
|
I calculated the order of the FIR filter to be 31(N). so, the number of coefficients has to be 32 (N+1). so, I have to increase it to 33 to make it an odd number.
Why the number of filter coefficients is required to be an odd number?
|
I'm writing a modulator in matlab to simulate IQ data that I would get from my radio to test my demodulator.
I've read that $I = x(t) cos(\omega t)$ and $Q = x(t) sin(\omega t)$. What is the $\omega$ term? If it is $2 \pi f$ then what is the frequency value?
I have a modulated composite signal that I need to split i... |
I want to transform a studio audio dataset into a smartphone audio dataset (used for a speech recognition application). Would there be any difference in the resulting audio files using one of the following two aproaches?
Replay all audio files and record with smartphone.
Convolute files with smartphone microphone impu... |
I'm trying to do bandpass filtering of a EEG signal samples at 250Hz and benchmarking the following 4 methods of FIR filtering for different filter orders. The length of the signal is 15000 samples.
Time Domain Approach (convolution)
Direct frequency domain approach
Overlap-add
Overlap-save
For follow the procedure ... |
Have many audio files 1 minute long. Some of them are normal. Some of them are noise.
Here is a normal file:
This is a noise:
And this is noise too:
How do auto detect noise using python?
BTW: if you suggest any external console program which can do that it will be cool too.
|
In LTE systems the in the scrambling section, a gold sequence is used. But I've studied that gold sequences are used in spread spectrum communication. This generally increases the bandwidth. So doesn't this mean that scrambling results in an increase of bandwidth.?
regards,
phani tej
|
let us consider following code
>> fs=100;
>> t=0:1/fs:2.93;
>> x=25*sin(2*pi*30*t)+20*cos(2*pi*25*t)+10*rand(size(t));
>> y=23*sin(2*pi*30*t)+21*cos(2*pi*25*t)+11*rand(size(t));
i would like to ask question related to Cross Power Spectral Density function , which in matlab can be easily estimated using function
>> c... |
I've studied that when the bandwidth increases, the number of taps we consider for channel modelling increases(example in WLAN or LTE). why is that happening. why signals with lower bandwidth (example : voice channel) see only a single tap/ a flat fading channel?
regards,
phani tej
|
I have this system:
$$ y[n] = -\frac{1}{2} x[n+2] - y[n+1] $$
I have no idea how to prove if the system is linear or not, because, it depends on future outputs...
Thanks for the help
|
For an FIR filter, with symmetrical tap values $h[N-1-n]=h[n]$,
why is the group delay $\frac{N-1}{2} T$ (where $N$ is the number of taps of the FIR filter and $T$ is the sampling time)?
Why is linear phase so important for the filter response?
|
I am attempting to write a FFSK (differentially encoded MSK) modulator. I have found and implemented a demodulator that works with sample data taken from a radio of a FFSK signal. The demodulator uses a FM Discriminator to transform the IQ data to a FFSK signal that can be decoded. I am using the method described in... |
I compared the difference between seperable and direct filtering for MATLAB's cameraman image. The difference is huge. I would like to understand the reason for this. Code below.
I = imread('cameraman.tif');
% Do seperable 1-D filtering
H = 11;
sigma = 2;
g1 = fspecial('gaussian',[1 H],sigma);
I2 = double(I);
I_seper... |
I am trying to use FFT as a fast way to convolve a image, however one of the downsides of this is that the "kernel" image will repeat when reaching the edges of the sample.
Is there a way to avoid this to make it more like a classic convolution, that cuts off at the edges?
|
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