instruction stringlengths 31 27k |
|---|
Is there a relation between Symbol Error Rate (SER) and the channel capacity?
What is {Modulation Order x (1-SER) / symbol duration [bps]} called?
|
I want to find roll-off (unit dB/decade) of highpass filter $H(z) = 1 - a \cdot z^{-1}$.
I derived magnitude response to be
$1+a^2 - 2a\cdot \cos(\omega)$
To find out roll-off factor from above equation, how should I proceed?
I thought of doing
$1+a^2 - 2a\cdot \cos(10\omega) \over {1+a^2 - 2a\cdot \cos(\omega)}$
But ... |
When splitting higher order recursive filters into more convenient 2nd order sections (biquads), in matlab, we usually use zp2sos to convert. In document, there's a couple of lines , which I am quoting below,
[sos,g] = zp2sos(z,p,k,''order') specifies the order of the rows in sos, where 'order' is
'down', to order the... |
Is it OK to calibrate a camera using effects and filters such as increasing contrast and brightness etc. to improve the detection of corners of a chessboard?
Even if the used mode for detecting the chessboard is just plain gray scale?
|
In GNU Radio Companion, I have created a very simple flow graph where I generate a 1 kHz, 2.0 volts peak-to-peak sine wave, and feed it into an FFT sink and a time domain sink.
The reference dB in the FFT sink is set to 0, and the reference scale is set to 2.0.
As such, I would expect to see 0 dB on the FFT display, b... |
I am planning to do a self project on speech recognition for a vocabulary detection system using arduino. I know CMUsphinx or bitvoicer is probably the way to go. But I would like to code the project from the ground up as the purpose of the project is learning.
I have been reading different papers like
http://revista... |
I have not studied signal processing at all, so please forgive any ignorance in the following discussion.
I have some noisy position measurements that I've been trying to smooth. I've been attempting Kalman filtering but getting either hf noise or lf oscillation depending on the constants I put in. What should I do?
Th... |
I've seen many questions on this forum regarding pitch detection for musical instruments (commonly guitar), and spent a while reading through the answers to create a basic implementation of auto-correlation to make an Android guitar tuner.
This is the algorithm I'm using (implemented in Java on an Android phone):
1) Re... |
I want to do a comparison of power spectral density of BFSK, 4FSK and 8FSK by using matlab baseband module "fskmod".Following is the code:
M = 2;
freqsep = 32;
nsamp = 4;
Fs = 64;
x = randint(1000,1,M); % Random signal
y = fskmod(x,M,freqsep,nsamp,Fs); % Modulate.
ly = length(y);
% Create an FFT plot.
freq = [-Fs/2 : F... |
I'm reading Szeliski's Computer Vision:Algorithms and Applications. On page 212-213, corner detector is discussed. I can understand upto equation 4.8. From then on, I don't understand how come eigenvectors of A specified the direction of slowest and fastest changes, and why does the shorter eigenvector specified the di... |
I have got a set of pressure data from a boiler which has been sampled at 45Hz and I need to extract a frequency component of 250Hz from the signal. Using a standard FFT method doesn't display anything apart from frequency components between 0 and 10Hz.
I have read similar questions posted on the forum with no success... |
I am trying to plot the magnitude of H(z), I got it to factore, and would like to sketch its plot of magnitude. But I am have trouble evaluating the function.
$$
H(z)= \frac{(1-2z^{-1})(1+0.5z^{-1})(1+0.9z^{-1})}{(1-z^{-1})(1-0.7jz^{-1})(1+0.7jz^{-1})}
$$
I know we replace z with $e^{jw}$ and try to evaluate it at var... |
I came across this basic problem in DSP which is not my area owf expertise but I consider this problem to be a common problem which others might be addressing.
I have multiple time series data and I need to find a time series data that is a representative of all those data. One method that I am applying is averaging t... |
Given a 128 Hz signal, in MATLAB I entered the following command
[B,A] = butter(2,[1/64,63/64],'bandpass'); %construct second order all pass filter
Filtered_Data = filter(B,A, Data) %filter data using butterworth filter
And got
This is far from what I expected since the filter should have passed all frequencies. an ... |
I have a sample of graphs (more than 10000...).
that look like in the image below:
I am searching for an unsupervised learning algorithms that can help me to detect anomalous observations.
Here what I suggest for beginning: for every observation I have a collection of points $(x,y)$. With this collection, I find Fouri... |
My advisor wants me to take the area of visible light communication as my MS project, but I don't know whether there are any research value in this area, it seems most of the tenichnical difficulties are solved in this field. Any one has similar experience in this field? thank you !
|
Given a z transform with one pole can you have a 2 sided Region of convergence or does 1 pole limit it to being only left or right sided? I know when you have two poles the 2 sided scenario is when a "ring" forms between the two poles in the z-space but when you have one pole does the case when the entire z space is in... |
Hilbert transform is used to transfer signal into analytical signal. But why after the transform, the envelop and instant frequency (if the signal is monogenic) can be calculated. What is happened in the transform? What is the relationship between these two signals?
BTW, according to wiki page analytic signal is
In m... |
I'm currently taking a course on signals and systems, and even though we are much ahead of frequency response, I still feel like I am missing an intuitive grasp of what the frequency response of a signal is. Specifically, when I look at the various CTFT pairs, it's difficult for me to intuitively understand why the pai... |
I have a 50 Hz sine wave sampled at 1600 Samples/second.I'm computing the 16 point DFT.So my fundamental frequency is 100 Hz.
I then decimated the 1600 samples to 400 samples.I then computed the 8 point DFT using this.Now my fundamental component is 50 Hz.The second harmonic is 100 Hz.
I plotted the above two scenarios... |
I am trying to understand Haar transform of simple vector of numbers.
Some references I found say, that first level transform of $ X = (4,6,8,10,13,9,3,3)$ will be $\sqrt{2}(5,9,11,3, -1,-1,2,0)$. Others say, it will be $(5,9,11,3,-1,-1,2,0)$.
Which one is true? Or are they both correct in some way and am I missing som... |
I am trying to calculate the FFT of a signal stored in a WAV file using SciPy. I found a solution here, but it seems like we need to perform this step before the FFT:
b=[(ele/2**8.)*2-1 for ele in a] # this is 8-bit track, b is now normalized on [-1,1)
where the signal samples are stored in array a. Can someone explai... |
If I perform a number of camera calibration routines at a fixed resolution, with various sizes of calibration patterns (like checkerboards), at various distances and get various values for focal lengths, could I just average them out? Would this just give a more "robust" focal length? Is it a good idea?
|
I am reading the lecture slides from this site http://luthuli.cs.uiuc.edu/~daf/courses/CS-498-DAF-P/Lecture%2018%20-%20Time%20series,%20DTW.pdf
In slide 24, the lecture suggest using metric of the following
d(i, j) = ||f(i) - f(j)||
select top frequency in i and top frequency in j,
find the Euclidean distance bet... |
Depending on the size of the original image and depending on the kind of padding used during 2D discrete wavelet transform and reconstruction, the reconstructed image has a larger size.
Is there a general method given a mxn matrix and given some padding mode to crop the obtained pxq matrix to the mxn matrix (without tr... |
Recently, I am thinking about a problem regarding image projection. I have a larger picture as bellow.
I want to find the transformation matrix, x'=Hx. When the above image pass the transformation matrix H, I will get another smaller picture as bellow.
I have tried to use SIFT to find four major points and use these... |
I am new to the stream of digital signal processing, and hence my doubt may seem extremely stupid, but:
In tools such as matlab, signals are stored in a single array. If i convolute it with an LPF array, i get another array/signal, that has certain elements attenuated, can i find out which elements these are and comple... |
I have a data vector which consists of calibrated pressure at each sampled point in time. I have also taken this signal, applied a filter bank to it using a butterworth filter with passbands according to third octave (i.e. band center frequency at 25, 31.5, 40,...,20,000), and for each band calculated $20\log(p_{rms}/p... |
Is there any way to detect water diseases using image processing ? and is there any way to develop it in android or in matlab thank you ?
|
What I've learnt so far is that, if we change a time-domain signal into frequency domain, its bandwidth shows the range of frequency of radio waves that required to generate such signal.
However, for example in fiber optic,a 1550nm light is used to be the source and BPSK is used to modulate it, but if there is only a ... |
I have an c5502 texas low cost DSP.
I can capture sound trough its P2 input and process every separated sample after sending it to the P2 output.
Im trying to implement some kind of effect in the sound, but this DSP has very low memory so i cant storage the signal to process it as a whole after.
There is any way of im... |
As the title says,w hat happens to the fourier transform of the sample output of $30KHz$ when sampled with sample frequency $44.1KHz$? I do not get how alias can appear, because fourier transform of pure sinusoidal signal contains impulse, not shape...
|
For the last week or so I have been trying to understand how quantization error results in the noise floor outside of a mathematical perspective and I haven't really had any luck finding a source that discussed quantization noise without using equations to show where quantization error comes from. I certainly have not... |
I have a set of data, in the freqeuncy domain, corresponding to a structural test.
I have amplitude, phase and frequency for both the measured output and the input signal (accelerations are measured in the test). The input covers the frequency range $5-100 Hz$.
I can compute the frequency response:
RE = abs(amplitude).... |
I am doing some project on medical image processing and I need some uncompressed medical images especially magnetic resonance angiography, vessel and so on. I have searched a lot but most of the available images are in JPEG format which is not a suitable format due to its lossy compression. Do you know any free dataset... |
I have a series of images. I want to be able to draw a curve on the image and track the curve in the series. For example, in the figure below, the left image illustrates the original image at time 0 and the one on the right is the curve I want to track in the next images in later times $t>0$. In fact, the edges that ar... |
Lets consider a pure sine signal at $\nu$ that is chopped using square pulses (like a burst mode on signal generators). My understanding is that instantaneous frequency is $\nu$ when oscillations are ON and 0 when they are OFF. On the other hand fourier spectrum is constant over time and contains also other frequencies... |
I didn't mess around with the code here, it was straight from their website, I just want to see if it would work. However, whenever I run it, it has feedback in it, sometimes I can't hear anything at all. It is not the speaker because I can record and play things separately. Here is the code if you want.
I tried changi... |
There are many tutorials and guides explaining how to create a "robotic" voice by using a vocoder or a mix of filters that change pitch, tempo and maybe add a slight echoed delay. However, is it possible to do that opposite? How would you take a "robotic" sounding voice and make it sound more natural? I know certain ef... |
The generic IFFT OFDM transmitter takes $N$ serial input data symbols, parallelizes them, maps them to their signal constellation points, and processes all $N$ resultant complex values using the IFFT to produce $N$ subcarriers (assuming no virtual carriers).
The IFFT takes $N$ input values and produces $N$ output value... |
When attempting to take the Short Time Fourier Transform of a pulse, at the end of the pulse I'm running into problems.
The signal looks like this at the end (it is a simple $sin^2$ pulse envelope, with a carrier frequency $\omega$. The only frequency that I care about in the STFT is $\omega$):
And when I perform a S... |
I have been given a sound file of a plane passing over a rain forest filled with birds. I am supposed to filter out the sound of the plane as it flies over. I've accomplished this with various types of filters in MATLAB, but I always run into one problem. I can either cut out all of the plane and lose some of the back... |
I want to know more about the latest attempts to create synthesizers for singing. I would ideally like to know more about both the most widely respected algorithms/models as well as mainstream software or programs that implement them (whether free or proprietary). Can someone offer me resources that would give me a dec... |
I am studying for my DSP final and I came across this question from the Oppenhiem and Schafer book 3rd edition. The question says
3.18 A casual LTI system has the system function
$$H(z)=\dfrac{1+2z^{-1}+z^{-2}}{\left(z+\frac{1}{2}z^{-1}
> \right)\left(1-z^{-1} \right)} $$
(a) Find the impulse response of the system, $... |
How can I know exactly what frequencies are not influenced by the filter?
I mean, I have a signal in the range of frequencies $5-100 Hz$ and I apply a high-pass, $4^{th}$ order Butterworth filter with cut-off frequency of $8 Hz$. The filter will not cut sharply at $8 Hz$, instead it will have a slope related to the fil... |
I assume that the sampling rate is w_s = 2w_max if we have an ideal low-pass filter that nulls frequencies whose absolute value is higher than that of w_max. Therefore, that the only reason for oversampling is to compensate for the imperfect filters whose magnitude at the stop-band isn't exactly 0, etc.
Consequently, o... |
Pose Estimation of single camera and planar pattern:
If I have an appropriate camera matrix with intrinsic parameters that allows me to calculate translation of the camera relative to a chessboard, for example, and it does so very accurately, does that mean my rotation calculations will also be accurate?
This may also... |
Suppose we have an added discrete noise signal defined as:
$$y[n]=x[n]+w[n], $$ where $w$ is zero mean white Gaussian noise.
Question:
When we say white noise, is it sufficient to say that it is always be uncorrelated with signal $x$, no matter what the distribution of signal is? (I think yes)
Can we conclude that bo... |
I have this filter:
filter_2 = firceqrip(2,0.6,[0.05 0.03]);
I want to convert it to the frequency domain to multiply it by a signal (i.e filter in the frequency domain)
I have a signal and I have a filter.
I converted the signal into frequency domain and now I need to filter it with my filter.
Converting the signal i... |
I have a x[n] = $\delta$[n].
By formula is should be
$$
X[k]= \sum_{n=0}^{N-1} \delta[n]W_N^{kn}
X[k]= \sum_{n=0}^{N-1} e^{-j2*pi*kn/N}
$$
The formulae isn't showing for some reason. I took a screenshot of what I got here: http://imgur.com/6j0Ibgu
basically why is the summation of an exponential term going to 1. $W_N=... |
I have a sinusoidal signal namely:
signal = sin(2*pi*0.08*t) + sin(2*pi*0.2*t) + sin(2*pi*0.32*t) + sin(2*pi*0.4*t);
so the segment of the code becomes:
freqs = [0.08, 0.2, 0.32, 0.4];
periods = 1./ freqs;
% Let the max value of t be 4 times the largest period
t_max = 4 * periods(1);
% Let us have 500 samples over the... |
I am trying to understand the limits of down sampling on a signal, i.e. how far can I go without loss of information or overlapping. It says in the solution manual of the Oppenhiem and Schafer book (on a question where down sampling is 2) that
"There is no loss of information if X(e^(jw/2) ) and X( e^(j(w/2) - pi) ) d... |
(cross post of SO: https://stackoverflow.com/questions/27345861/extending-1d-function-across-3-dimensions-for-data-windowing )
For the sake of image (volume) registration, I'd like to apply a windowing function to input data, such that the non-periodic image boundaries do not cause streaking in the FFT. I'm using the e... |
I am trying to find an example of a continuous system that is homogeneous, but not additive.
So far, the only example I could find was an example from this page, which describes system as $y(t) = \overline{x(t)}$, where (as I understand it) $\overline{a}$ is the conjugate of a complex number $a$.
This, however, will wo... |
I am trying to figure out what seems very trivial thing about down sampling and up sampling. The problem that I am facing is how to find the effects of up sample on a signal followed by a filter and/or a filter followed by down sampling. Here is a problem related to this from the Oppenhiem and Schafer book.
I am tryi... |
I know that a general purpose microprocessor requires 18 operations to perform each butterfly of the FFT, but I do not understand how does using using a DSP or FFT chip improve FFT processing?
Any thoughts?
|
I need to construct Fourier transform of non-equispaced data.
That is, I have signal $s(t)$, $t\in[0,T]$ sampled at non-equispaced points $t_k$, $k=0...N-1$ with sample values $s_k = s(t_k)$. For Fourier transform I use approximation of integral:
$$
\hat S(\omega) = \int\limits_{-\infty}^\infty s(t)e^{-i\omega t}dt... |
Suppose there are two vector signals, $u$ and $v$, which are functions of time. Given, $E[uv^T]=0$, what is the physical significance of this statement. For clarity, assume $u$ and $v$ are noises in a system like in kalman filter, control noise and measurement noise. E is the expected value and assume that the vector d... |
Being new to ANC, this is my understanding of how ANC works:
Ambient noise (n) is entering a geographically small zone. To suppress the noise, an interfering (corrective) sound is injected acoustically into the zone by a speaker. The interfering sound from the speaker is adaptively generated so that it - in theory - en... |
During MFCC process FFT is done to get the power spectrum but last step is DCT in order to get back to time domain. Cannot DCT be used in the first case to convert the signal to frequency domain?
|
have two User one of them sending over freq (0-B] and the other transmit over (B-2B]. Both of them are using OFDM and each one has Nc subcarriers. I wanted to simulate the received signal using Matlab.
One way to implement, I beleive is that I say user one have 2*Nc subcarriers and only use the first Nc subcarriers an... |
I am working on some experimental data which, at some point, need to be time-integrated and then high-pass filtered (to remove low frequency disturbancies introduced by integration and unwanted DC component).
The aim of my work is not related to filtering, but still I would like to analyze more in detail the filters I ... |
I tried creating a filter in 2 different ways in Matlab and I obtain different results: one filter is stable, whereas the other one is unstable.
Coded filter
delta_t = 1.53846e-04;
Fs = 1/delta_t;
cut_F = 8;
Wn = cut_F/(Fs/2);
ftype = 'high';
[b,a] = butter(8,Wn,ftype);
fvtool(b,a)
The result is an unstable filter
... |
I have a question that might be stupide !
Supposing that I try to detect object moving in a video or human action.
Many works are based on optical flow computation.
My question is why using OF is the best for motion detection. A tracking of pixels or region of interest will not be as informativa as OF .?
|
This is a general question on multitaper spectral analysis of signals and stationarity
I work with time series which are non stationary. In this regard I have two questions
Is weak sense stationarity of second order sufficient to apply multitaper spectrum to compute valid estimates from contiguous time series objects ... |
In frequency domain $\zeta>0.707$ implies no resonant peak in frequency response. But I am unable to correlate what is the exact time domain effect of this? If $\zeta=0.707$ acts like an 'important damping value' in the frequency domain, shouldn't its importance be reflected in time domain as well. But in time domain ... |
Can someone please help me with this Fourier series question:
Determine the Fourier series coefficients of $x(t)$ given as $x(t) =
> \cos4t +\sin8t+3$?
|
Evaluating a volume flow incremental sensor does not deliver expected results.
The pump of which the volume flow is to be measured has a hand-calculated flow rate of
$$Q = \dfrac{n}{ 60 \;\mathrm{s/min}} e \pi\left(\frac{d}{2}\right)^2 \approx 2 \;\mathrm{cm^3/s}$$
where $n$ is the rotational speed of $2000 \;\mathr... |
Can anyone please explain about Cepstral Mean Normalization, how the equivalence property of convolution affect this? Is it must to do CMN in MFCC Based Speaker Recognition? Why the property of convolution is the fundamental need for MFCC?
I am very new to this signal processing. Please help
|
I have a 1st order lowpass IIR filter of the form
$ H(z) = \frac{b_{0} +b_{1} z^{-1}}{1+a_{1} z^{-1}} $
implemented using the direct method my code looks something like this
state2=state1;
state1=signal x-state2*a1;
output=b0*state1+b1*state2;
that has a cutoff frequency X that has been calculated from the fundame... |
Is it necessary to use filter bank in MFCC process? Can anyone explain what is the Mel filter? I know that frequency in hertz is converted into Mel scale but is this formula can be directly applied after the Fourier transformation of the speech signal? Why the filter bank is needed?
|
I am having troubles applying the IFFT function to some data I have. I already had a look here Complex conjugate and IFFT and here What assumptions should be used to invert spectrum into time domain data? but it still does not work.
This is the code:
% amplitude: data related to the signal amplitude
% phase: data relat... |
What are the implications of:
If $x(t)$ is real and $x(-t) = x^*(t)$, then $X(-j\omega) = X^*(j\omega)$ and $X(j\omega)$ is real.
I am trying to understand it and I would like to research it further (what's the theorem that guarantees this?).
I'd like to understand why it's important aside from the fact that it gua... |
I gotta wrap my head around to design CIC compensation filter
I'm studying by referring these materials:
Altera, "Understanding CIC compensation filters
Hardware Efficient FIR Compensation Filter
For Delta Sigma Modulator Analog to Digital Converters.
Circuits and Systems, 2005. 48th Midwest Symposium on
Saiyu Ren ;... |
I am reading this paper about speech enhancement using a minimum mean square error short time spectral amplitude estimator for our speech enhancement project. Basically in the paper they mix a clean speech with noise and try to recover the clean speech.
The paper keeps mentioning a priori SNR and a posteriori SNR. I am... |
In overlap add method for convolution a filter $x$ of length $M$ will get convolved with a signal $y$ of length $N$ where $M << N$, i.e., $z = x * y $.
Here the signal $y$ is sliced into subparts each of length $M$.
Can I slice $y$ into subparts of size $P < M$ and then do the convolution by overlap add method. What sh... |
I'm trying to match a picture saying "this is ducks, this is towers" by giving votes from the hough transform of a picture.
I've found this article http://ttic.uchicago.edu/~smaji/papers/hough-cvpr09.pdf, but I didn't find a implemented code (and I'm trying several things).
Do you know anything similar to that, for ex... |
Let's suppose I have an ADC running at 81.824MHz. I am quadrature subsampling a signal that is at 143.5MHz using a sampling rate of 81.824MHz/4 = 20.456MHz = Fs. I expect the signal to appear at baseband with a small offset of 143.5MHz-7*Fs = 308KHz. But it is showing up at 320KHz when I run the code below. Why is that... |
I get confuse with the basic theory of particle filter for tracking single object..
What I want to ask is:
In some paper, they explained about velocity in state vector.. Are we
randomly put number to velocity or get it from some formula?
In pdf by eiji ota, he desrcibe system model as Xn=f(Xn-1)+Wn where Wn is noise... |
I saw a code in Python in the scipy.org (http://docs.scipy.org/doc/scipy-dev/reference/tutorial/fftpack.html) and found this:
First question: Is it right that T = 1/800 <==> fs = 800 Hz fulfill the condition fs = 2*fmax, because the maximum frequency of the funtion contains 80 Hz?
Second question is in the row: xf = n... |
Is there any way how to compute convolution matrix for Nearest Neighbor (bicubic, bilinear) image scaling (upscale/downscale)?
|
What are some of the reasons why DC offset is considered undesirable for a timeseries signal other than wastage of power during transmission ? Specifically, How does it cause spectral leakage and out of band emissions ?
|
I'm trying to filter out line noise(60Hz and 120Hz) from a EEG signals received over a bluetooth link. I'm proposing to use a IIR notch filter to filter out the line-noise which varies w.r.t to distance between the bluetooth transmitter and receiver. I'm not able to establish a clear relationship between the Notch filt... |
I am getting I/Q data from a software-defined radio. I want to do some stuff on signals in the data, but only if it exceeds a certain range. What is the general procedure to get dB (dBm, or anything) from this kind of data? Programs such as SDR# do it, but I don't know what exactly they do so that I can imitate them... |
I have an application that fills a buffer from my computer's microphone and does an FFT. My objective is to figure out what frequencies are in the sample.
I've tested the application by whistling as well as by playing this video from my cellphone into my computer's mic (I've tried two different mics to be sure). What ... |
My electricity meter has a front panel LED which pulses in proportional to the current energy rate. The frequency changes but the pulse width is set at 250ms. Due to environmental conditions there's plenty of noise. Currently I'm sampling this at 10kHz, the pulses may be a few seconds apart.
I would like to filter the... |
I have an audio sample. From doing FFT on it it looks as if I need to use a comb filter to remove resonant frequencies. This is what I generate:
I honestly have no clue where to start unfortunately.
|
I am reading a recent paper titled "Phase Estimation of Speech Enhancement - unimportant, important, or impossible?" by Grekmann et al. published in the 2012 the IEEE convention. The paper proposes an algorithm for estimating the phase of the noisy speech.
can anyone help me please on understanding what is meant by ... |
I'd like to implement freuqency sampling method for linear phase FIR filter design using IDFT transform.
My procedure goes like this :
determine desired magnitude response value in frequency points
I add linear phase response function with group delay (N-1)/2 to get complex frequency response values and equidistant po... |
I am trying to shift an image on a subpixel level, with the method described here:
http://www.mathworks.com/matlabcentral/fileexchange/18401-efficient-subpixel-image-registration-by-cross-correlation/content/html/efficient_subpixel_registration.html
The results are however disappointing. Since I am actually programmin... |
Using the matrix equations found on Wikipedia I wrote some Python code to compute the Savitsky-Golay coefficients for $m$ data-points, using a polynomial of degree $k$. Here it is:
from numpy import *
from numpy.linalg import *
k = 4 # Degree of polynomial to use for regression
m = 25 # Window size, that i... |
I have a filter.
It has two poles and two zeros.
I found the state space equations and the matrices A, B, C, and D
Now. I have 9 samples that I need to process with my filter.
How do I use A,B,C,D matrices to process the samples?
|
I am trying to do an image analysis problem on some images, and I want to calculate some things in the frequency domain which are giving me problems. Say I have an (discrete) image $I(x,y)$ of size $N_x\times N_y$ (in practice this size is about $150\times40$ pixels). I want to calculate
$$DFT\left[\frac{\partial I}{\... |
In the process of creating a baseband signal, the complex symbols are filtered using a raised cosine filter to perform pulse shaping (I'm following some literature as a guide). The pulse shaping is performed with the code:
%Create the filter
over=16;
pulse=rcosine(1,over,'normal',0.35);
[trash, pos]=max(pulse);
%Overs... |
I have a N point sequence x. Let X be the fft(x). (Assume x is arbitrary)
where,
x=[x1 x2 x3 x4 x5 x6 x7 x8]
fft(x)=[X1 X2 X3 X4 X5 X6 X7 X8]
How do I get the result of a 8 point DFT by doing cascading 4 point DFT and 2 point DFT?
Suppose I reshape the signal 'x' (of dimensions '1 x 8') to 'x1' (of dimensions '2 x 4... |
I need to clarify this and rather confused by this.
Lets say:
$x = h * g$
$x$ - measured data
$h$ - raw data
$g$ - instrument response function (convolution kernel)
So lets say I have $x$ and $h$. Can I say if I were to deconvolve $x$ with $h$ I will be able to obtain $g$?
Or if I have $x$ and $g$, deconvolving $x$ w... |
I am trying to evaluate my processed speech for my DSP project. My project is an implementation of an MMSE algorithm proposed by Y.Ephraim and D. Malah, "Speech Enhancement Using a Minimum Mean Square Error Short-Time Spectral Amplitude Estimator," in the IEEE Transactions on Acoustics, Speech, and Signal Processing, V... |
I have two groups of 3 frequencies of tone (music),
First Group
Tones: 39 (77.7817 Hz), 40 (82.4069 Hz) and 41 (87.3070 Hz).
Second Group
Tones: 83 (987.7666 Hz), 84 (1046.5022 Hz) and 85 (1108.7305 Hz).
These groups work independently (the frequencies aren't mixed between them)
ToneB: Tone Back 39 or 83
ToneH: Tone... |
This isn't my field, so please forgive any misnomers on my part.
I'm looking into buying a amateur radio receiver, and I'm thinking about what sampling rate I need. I'm most interested in satellite signals, which transmit on ~400MHz (for example). So that would imply I have a nyquist frequency of ~800MHz, or 800 millio... |
I'm trying to design a lowpass IIR filter with 5Hz cutoff frequency (sample frequency is 200Hz). i know how to create it with fdatool, or with functions such as 'butter' or 'ellip'. However, i think that all the filters created are causal. Is it possible to create also noncausal filters? Thanks in advance
|
Subsets and Splits
No community queries yet
The top public SQL queries from the community will appear here once available.