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I have data in which the signal we want is a few gaussian peaks superposed on a noisy background in which some of the 'noise' is periodic. Simply suppressing the associated fourier modes is effective (e.g., a 'notch filter') but also suppresses the signal, since it has some power associated with those frequencies.
Wha... |
I'm a beginner with absolutely no experience in signal processing, but it has been a long-term goal of mine to do something that mixes music (a hobby of mine) and computer science (my "expertise"). I've been thinking of doing a project that involves learning notes from music, but I don't know where to begin.
For one, ... |
I am using the exponential moving average filter (IIR):
$$y(n)=\alpha{}\cdot{}x(n)+(1-\alpha{})y(n-1)$$
in a foward-backward filtering scheme, in order to eliminate completely any phase response phenomena. But I do not know how to eliminate the startup transients, ie. the transients from the initial state of the filte... |
We are looking for writing applications which would recognize Bird calls and Frog calls using Sound processing.
EDIT: I want to identify the species of bird / frog and do not want to differentiate whether the call is from a bird or a frog. There are about 1300+ bird species in India and about 300+ frog species which w... |
The continuous time Hilbert transform is
$$\hat x(t) := x(t) + j\left( p.v. \left\{\frac{1}{t\pi} \ast x \right\} (t)\right)$$
where
$$ \theta(t) = \tan^{-1}\left( \frac{ p.v. \left\{\frac{1}{t\pi} \ast x \right\} (t)}{x(t)} \right)$$
and
$$
\omega(t) = \theta'(t)
$$
and since $\theta(t)$ is smooth, you can ... |
I am working on an application to classify Talas(a piece of music with onsets). For that, I have the spectrogram of a Tala, which I want to compare with the test Tala, or compare 2 Talas.
For this I intend to find the onsets in the spectrogram and then form a classifier based on the frequencies that I get at the onset ... |
I'm having a little trouble plotting the DFT of a signal in dB. I can get the picture up nicely when using the magnitude but when I try converting the magnitude to decibels it all goes south. I've been using this for help: http://dadorran.wordpress.com/2014/02/20/plotting-frequency-spectrum-using-matlab/
Here is the co... |
I am trying to develop a classifier for Indian Taals, for which I would be focusing only on the Tabla.
Now once a extract a period of the Tala(a repetitive pattern), I want to develop some feature vectors for a Tala, which can be used to differentiate two Talas.
I am not sure what they should be. As of now, what I am... |
How can I find the frequency response of the following linear system
|
I have a vector with raw data (120 samples). From the frequency spectrum I can tell there is one dominant frequency component. Now I am supposed to use the fft data (magnitude and phase) to find a good sinusoidal fit and offset to fit the original data.
I am completely stuck on this one. Can anyone help?
|
The measure of a given frequency $\omega$ in a signal $x(t)$ is:
$\frac{1}{N}\sum\limits^N_{t=0}x\left(t\right)e^{^{-i \omega t}}$
This is basically an average of the correlation between the signal and a complex sinusoid rotating at $\omega$ frequency over some time range (N).
My goal is to realize this as a IIR filter... |
I have a frequency-domain representation $X(e^{i\omega})$ of the complex discrete one-dimensional signal $x[n]$: $X(e^{i\omega})=\mathcal{F}\{x[n]\}$. Is there a frequency-domain transformation of $X(e^{i\omega})$ into $\hat{X}(e^{i\omega})=\mathcal{F}(|x[n]|)$?
Obviously, $\hat{X}(e^{i\omega})=\mathcal{F}\left\{\left|... |
I am willing to overlay a equalized cdf with equalized histogram in matlab.
But it's really hard to find any recommends.
so if you can, would you please let me know about that?
|
I know that JPEG uses 2D dct and splits the images in 8x8 blocks. Why doesn't it simply split the image in one-dimensional vectors in $\mathbb{R}^{64}$? Wouldn't it simplify the math? My guess is that this way doesn't take advantage of pixel correlation. Is this right or is there any other reason?
|
My question involves proper FFT Technique for AM demodulation, in the special
case where the modulation rate exceeds the carrier frequency.
This question has a specific hobbyist application, the decoding of weather satellite pictures using minimal computing hardware and inexpensive receivers.
Warning: this is necessa... |
I have some noise measurements and 4 different models that can generate a similar noises using estimated parameters from the real measurements. To compare the estimated noises I calculated the PDF, the CCDF and the PSD.
To compare the results I used the Kull-Back Leibler divergence of the PDF, the MSE of the CCDF and t... |
I performed bicubic interpolation on a 256*256 image(img)
dest = interp2(img,'bicubic')
and i got a 511 * 511 image.I want to compute PSNR of a 512 * 512 image(original) and the 'dest' image as follows
original = double(original);
dest = double(dest);
[M N] = size(original);
error = original - dest;
MSE = sum(sum(er... |
When designing a root raised cosine filter on Matlab, there is an option to set its "roll-off factor". What does this mean? I have been looking around for a while but I can't seem to find a clear explanation. Any help will be appreciated.
|
I have implemented "normalized cut" segmentation. The change here is that, instead of giving an image as input, I have given an matrix of image. As my output looks odd to me. I need to know whether my implementation is correct or not.
Code:
clear all
tic;
% im = imread('lena.pgm');
im =[94 122 99 101 111 101;
99 92 10... |
I need to apply successive interference cancellation (SIC) to an RF signal which is the result of collisions between 2 or more unsynchronised packets. For this question let's just assume that the number of packets is 2.
In SIC, you store the collided packets until one of the packets are repeated and use the 'clean' pac... |
I have a simple, but interesting question for you.
I have to compute the RMS (root mean square or standard deviation) value of a time signal which represents in my case a velocity mesasurement over time, of course sampled an stored in a computer. In your opinion, does the sampling frequency have a strong impact on the ... |
I am working on a project, for which I am finding the onset times of an audio and then extracting the MFCC coefficients for all the times.
Thus if I have n onset times, I would have n feature vectors, corresponding to the MFCC coefficients at those points.
Now I want to characterize a "bol" of the tabla by the the wave... |
For those who have either used Picasa/Photoshop, they would see different image enhancement options such as:
Change exposure
Change shadow
Change highlights
Change vibrance
I have searched online a lot about algorithms for doing the above operations. I know that these can be achieved from basic operations such by cha... |
I'm trying to do the Hanning Window manually but didn't work, what I'm doing wrong?
t=[0:.1:10];
x=.5-.5*cos(2*pi*t/102);
plot(x);
Not sure why I cant obtain the full hanning window..
|
What equation best describes the alpha content of a black colored pixel applied to a white pixel when measuring the resulting pixel intensity?
For instance
given a white pixel: argb : 255, 255, 255, 255
drawing over it with a black pixel: argb : 78, 0, 0, 0
the result is argb : 255, 50.2, 50.2, 50.2
Measuring the pi... |
I have a RLC circuit with transformation $$H(s) = \frac{R}{Ls+\frac{1}{Cs}+R}$$ and I know that by setting $s=j\omega$ I can obtain the frequency response of this system. What I don't understand is how I can use the frequency response graph to decide which values I should set on R and f $(L=2\pi f, C=1\mu F)$ to filter... |
We are doing a test which consists in sending a waveform from Matlab to the signal generator SMB100A and sending it through a cable to the r&s spectrum analyzer just to have a channel. We set the trigger manually in the signal generator in the restart mode, then we set the frequency and the chip rate in our Matlab code... |
According to wikipedia Onset detection:
Onset refers to the beginning of a musical note or other sound, in
which the amplitude rises from zero to an initial peak. It is related
to (but different from) the concept of a transient: all musical notes
have an onset, but do not necessarily include an initial transient... |
I am using Audio Precision 2700 to test some test vectors.
I have connected resistor banks as class - d power amp's load, and tap from this resistor using a XLR cable to the AP.
When my power amp is powered up, but no song is playing(song is paused, not muted) i checked fft spectrum of the input channel , i get the fol... |
I'm reading a paper where image classification is done. Their approach is to use the discrete wavelet transform with bi-orthogonal wavelets of degree 3.5 and decomposition of level 3 on images. Based on the 2d coefficients gotten from the DWT they calculate statistical features such as standard deviation, skew, kurtosi... |
So if I were to have an $n$ length sample of a given sample rate and were to run it through an FFT, resulting in $n$ complex elements. How would I obtain the frequency bins, such that if I know the frequency of some noise, I can remove the noise?
|
Question:
Is there any software, source-code, or well documented and easily implementable research (I am not afraid for a little bit of Python) to do the following:
Take two similar videos from different sources (say a video file obtained from a VHS and a video file ripped from a DVD) and map an estimated time differen... |
The question says it all. In typical (wavelet-like) decomposition of a signal, why is only the low pass component chosen for successive decomposition ?
|
There are a plethora of tools, both commercial and free, which I have found online for designing filters. The ones I have tried (so far) prompt for frequency response, number of steps (for FIR), then generate coefficients and a frequency response plot.
Question: are there any particular sites or programs, which allow y... |
I have some MP3s that are encoded at 44.1kHz with a bitrate of 160kbps. I hook these up to my car using an FM transmitter and the sound quality isn't particularly great. Low and high notes sound distorted and crackly.
I suspect that I'll need to reduce the sampling frequency, but I am not sure. Reading up on FM broadca... |
I was trying to solve the following problem .
Given an input x[n], consider upsampling by 2, followed
by interpolation with a filter having z-transform H(z) for magnification of the signal.
Then, to recover the original signal size, apply filtering by a decimation filter G(z) followed
by downsampling by 2, in o... |
I am trying to generate a multitone signals with low crest factor and as suggested in the Boyd's paper I am using the quadratic phase distribution:
$Phase(k) = \pi(k - 1)(k - 1)/N$
where $N$ is the number of tones in the signal and $k$ their frequency indexes.
This formula works well, giving the expected crest factor (... |
Is there any way to use as single input an image (256x256 pixels) and get output of single value using Matlab neural network toolbox
|
Let $s(t)$ be a signal that can be approximated by a uniform spline
function of order $K$ (say $K=2$): $$s(t)\approx\sum_{n\in \mathbb{Z}}c_n\beta_+^{(K)}(t-n) $$
Suppose that we know the values of $s(t)$ at integer locations
$s(n)=x[n]$. Develop and implement a filter-based algorithm to recover
sequence $c_n$ f... |
So I was working on a program to comparing the similarity between the pronunciation of a word, First I got the audio file, and I use FFT on it to get the amplitude over frequency array, then I just square everything to get the Energy Spectral Density of it. The graph that can be drawn from those frequency of 4 sample o... |
I'm preparing interview and found this question. But I don't really understand what is the question. Does it ask about Fourier transform or Z transform?
How the simple identity
$$xy=\frac{1}{2}x^2 + \frac{1}{2}y^2 - \frac{1}{2}(x - y)^2$$
was used in developing a significant transform in digital signal processing?
|
A continuous signal $x_a(t)$ is a linear combination of sinusoids of frequency 250 Hz, 450 Hz, 1 kHz, 2.75 kHz and 4.05 kHz. The signal $x_a(t)$ is sampled at $f_s=$1.5 kHz, and the resulting digital signal is converted into analog using a reconstruction filter with a low pass filter with cutoff frequency of 750 Hz. T... |
The title says it all.
I have two waves, Master and Slave. Master is at 441 Hz and Slave at 700 Hz. The sample-rate is 48 kHz.
I want to know how to calculate the period of each wave in terms of samples. How many samples long is each wave's period?
Please show me how to calculate this
Thanks
|
I am working on Automated GUI components Segmentation problem. My Images are very simple as it has GUI components such as Buttons, text box and other polygons as you can see in the diagram (a).
Diagram (b) is the hypothetical and desired image of segmented portions.
image A and B respectively
My approach:
I have... |
This question is probably more for fun and theory, but how will the sound quality be, if you create a sound with really high sampling rate (like 192kHz) and really low bit depth (e.g. 3bit)?
Can you emulate 44.1kHz/16bit with 192kHz/3bit (+dithering)?
I can probably think this through and post an answer myself with, s... |
I want to make a BCI project, based on EEG signals, to control a robot using the brain's P300 wave. I have read a lot of papers, which give me a wide knowledge on how to deal with those signals, but I get confused, there are many approaches that papers follow which is DSP, statistical and AI approach (usually AI appro... |
As part of a project I am doing, I have to investigate how having a carrier frequency offset would affect the constellation pattern of the received signal.This question has got me baffled because to me a carrier frequency offset should not affect the constellation pattern because any offset would be dealt with on the r... |
I am processing rtl_fm output (in time domain), extracting Manchester encoded bits. I can detect preamble and, therefore, decode the message which follows the preamble.
This is how my signal looks like (here I have 2 messages, the rest is noise):
And this one of the messages zoomed in:
Now I want to know how strong w... |
I have the generic problem of analysing a discretly sampled signal, lets
say at 100Hz, and would like to compute features (ML application) of this signal at different scales:
An instantiation of this problem is computing the FFT on real-signal with
N samples:
Original signal: X[1],...,X[N]
Suppose I have already comput... |
Let $f=A\sin{\omega t}=x_1$ and $\dot{f}=A\omega\cos{\omega t}=\dot{x}_1=x_2$. Let the output be $y=cA\omega$, where $c=1$ is a constant. I want to represent this in a state space formulation:
$\dot{x}=\underline{A}x+\underline{B}u$
$y=\underline{C}x+\underline{D}u$
What will $C$ be? I tried: $A\omega=\frac{A}{t}\arcsi... |
in Digital Image Processing restoration ,from my understanding after reading articles , it's seems that the noise must be dominant in the histogram of the image , which is make it possible , to get the mean and variance of the noise from the histogram it self.
My question is if I have a little Gaussian noise in the i... |
Is there a method to find the relative direction of a Bluetooth device to the receiver?
A quick Google search on "Bluetooth positioning" returns several methods using triangulation, but I don't want to use multiple receivers. Also, I don't need to know the distance, only the relative angle to the receiver is enough.
|
I'm studying for my midterm , and I'm struggling with finding bandwidths.
I have this energy spectral density : $\Psi(f)= { 1 \over \sqrt { 1 + ({f \over B }) ^2} }$ , and I need to find the $3$ $dB$ band and the band with $90%$ % power.
About the $90%$ % case, do I need to find the total energy and then $\int_... |
I am currently working with a set of wavelet transforms that I came across in a paper. The paper suggests that I first apply the FFT on the source image and then perform the downsampling. So it says something like:
Consider a source image of size 128*128.
P = fft(source);
P(1:64,1:64) = P(1:64,1:64) + P(65:128,65:128);... |
Apologies if this makes no sense or seems rather simple, I'm a Matlab newbie.
Basically, I've got a .wav audio waveform which I am resampling to various different frequencies and then trimming the length to a second (44100 samples). However, when resampling to certain tones, such as the note C3 or A5, I'm not left wit... |
What do I need to know to difference a note for different musical instruments? for example, if they play the same note, (same frequency), how can I difference them if they have the same frequency?
|
I am working on modelling some FM transmitted signals, and I'm trying to figure out how to model the effects of multipath channels. The most obvious method is to delay the signal by some integral number of samples and then add it to itself (with some attenuation factor). But, this is obviously not very realistic, sin... |
I want to start studying image processing and need some tutorials for beginners. I am not bad at Mathematics and good when it comes to programming. I want to use the CImg library and write code in C++. Could you advice me some good tutorials or books to start with ?
|
I'm going through a Signal Processing lecture where the professor mentions this fact and the argument given is:
Suppose you have a sinusoidal signal: $Acos(\omega t)$
Now if you change the phase of the signal: $Acos(\omega t + p)$ then if it would correspond to a time shift, then
$Acos(\omega t + \omega t_0) = Acos(\om... |
I have this triangle wave as input:
and I want to derive a sine wave out of it. How do I go about it?
PS: I am doing this in Lua, so any piece of code close to that language would be appreciated.
|
I was trying to understand convolution from this book and came across this figure:
Here, the input signal is a combination of a sine wave and a ramp.
In fig a, the input signal is convolved with a special type of impulse response that is said to be a 'low pass filter' and the output is just the ramp.
And in fig b, imp... |
I have developed a spectrum scope that plots signal vs. frequency starting from IQ signals. I currently do not have a way to balance the IQ signals, so am looking at various ways to do so. I have come across this article, which is reasonably easy for me to follow since I am fairly new to all DSP. The one question I h... |
I have some low quality audio from my native tongue on a VHS. I am using this audio to dub HQ released of the same program (from another language source). The video frames/second ratio is 25:23.98 . I do not want to tamper with the fps ratio, since I am still using the other audio tracks as alternative languages in my ... |
I'm currently tagging insects with reflective tags and filming them in night-mode on my camera which has an infrared light. I have some markings on the tags which I need to catch on film. Unfortunately, when I film about 2 metres back, the the contrast between the reflective tag and the marking is too great, resulting ... |
I want to apply Hamming window function the Audio sample then find out its fft..
[y,Fs,bits] = wavread('file1.wav');
Nsamps = length(y);
ham= hamming(Nsamps);
ham_fft = abs(fft(ham));
ham_fft = ham_fft(1:Nsamps/2); % discarding Half points
plot(ham_fft);
I don't know whether i am getting the correct fft or not ..
|
I am using Excel. If I have an input and an output, and I know that a recursive band-pass filter produced the output, how can I can find the coefficients of that filter?
|
I understand the reasons for zero-padding a finite signal before FFT.
However, I was reading this book (http://www.dsprelated.com/dspbooks/sasp/Zero_Phase_Zero_Padding.html) where the topic is Zero Phase Zero Padding.
Top image: windowed finite signal.
Bottom image: Zero Phase Zero Padded version of the same signal... |
I have to estimate the parameters of a 1st order transfer function, namely, the coefficients, through experiment. I ran a few experiments and I have a bunch of input-output data vectors. The experiment was a standard frequency response test, i.e., the inputs were a bunch of different frequency sine waves, around the fr... |
Given a (FFT-sized) frame of data, and detection of a spectral component statistically above the noise floor in the FFT of this window, what characteristics or signal analysis could be used to determine that this spectral component is more likely to be a linearly swept sinusoid, rather than one that is stationary acros... |
IIR filters have feedback whereas FIR filters don't use feedback.
Then why is it said that FIR filters sometimes have the disadvantage that they require more memory and/or calculation to achieve a given filter response characteristic?
Can you please explain explain what is it that makes FIR filter computationally more ... |
A constellation diagram has points at the following coordinates: (2,2), (2,-2), (-2,2), (0,-2). What is the modulation used by the modem?
This is literally never explained in my textbook and I can't do it by logic, since I have NO clue what a constellation diagram is.
|
thanks for taking the time to help with this problem!
I have to prove the time-scaling property:
$$
x_{(m)}[n] =
\begin{cases}
x[n/m], & n=0,\pm m, \pm 2m,...\\
0, & otherwise
\end{cases}
$$
First, for part (a), Show that $$x_{(m)}[n]$$ has a period of $$ mN$$
Then, for part (b), Show that if $$\\x[n]=v[n]+w[n... |
Suppose I have a stationary process $\phi(t)$ with a known autocorrelation function
$$ A(\tau) \equiv \langle \phi(0) \phi(\tau) \rangle$$
and suppose I also know that $\phi(t)$ is Gaussian distributed.
If I have a particular realization of the process in which $\phi(0)=\phi_0$, what is the conditional distribution of ... |
I am given the following signal:
$$[e^{-at}cos(w_{o}t)]u(t),\ a>0$$
Then I am told to find the Fourier Transform, which tells me I need an answer of the form:
$$X(jw)=\int_{-\infty}^\infty \! x(t)e^{-jwt} \, \mathrm{d}t.$$
I know I can reset the bounds of my integral with the unit step function, so my equation becomes... |
This is part formula of command-response model (Fujisaki Model)：
Analysis and synthesis of fundamental frequency contours of Standard Chinese using the command–response model
http://www.sciencedirect.com/science/article/pii/S0167639305001706
Why they are critically-damped second-order linear systems?
|
I have been studying about the box window and am unable to understand one thing, do we first sample a signal and then window it resulting in up-scaling or downscaling, or is windowing a practical case of sampling itself which is performed on a continuous time signal?
|
I need to do an ir convreverb on a lot of wav files i got, always with impulse responses of different rooms. it does not have to be zero latency so even calculating the convolutional reverb would be fine. I'm just searching for a possibility to do this from the commandline. Does anybody know a tool that can do what i'm... |
So I have a signal which is caused by an impulse train. I can determine the impulse train with 500hz but sample my output signal with only 10hz. Then I want to calculate an impulse response function. So I have a binary "signal" for the impulses with 50 times more samplepoints then my actual signal. If I resample my bin... |
I know that to build Harmonizer I need pitch shift the signal in differents ways, my question is how know for what scale/factor/notes/frequency I need pitch shift to create harmony?
For example if i press a middle C piano key how create two or three harmonization for this key ?
I believe to create 3 harmonizer the outp... |
I'm trying to understand why Matlab function upsample causes contraction in the spectrum, like the following figure shows.
Also showed in mathwork online document.
enter link description here
Filtering after upsampling removes the imaging artifacts, but contraction still stays, as shown below. Could anyone help expla... |
I have an output signal $y$ which is an input signal $x$ convolved $\star$ with an impulse response function $h$ with some added noise $n$ :
$$y(t) = h(t) \star x(t) + n(t)$$
I know the input signal $x$ and output signal $y$ and would like to calculate $h$ the impulse response function. I found that deconvolution is no... |
I am going to plot the phase spectrum of a zero-phase ricker wavelet. I calculate the fft of the wavelet and use the phase or angle to calculate the phase spectrum but the result is not zero!
I use the following code:
Fs = 250;
t = 0:1/Fs:0.5*(Fs-1)/Fs;
f1 = 30;
w = (1-2*pi^2*f1^2.*t.^2) .* exp(-(pi*f1.*t).^2);
X = ff... |
Consider $N$ discrete signals $x_1(n), x_2(n), \ldots,x_N(n)$ each a bounded support of size $M$.
To convolve them, we can zero-pad each of them, multiply their FFTs, and take the result's inverse FFT.
I believe this requires taking $N$ FFTs of length $N(M-1)+1$ each, taking $O(MN\,\log(MN))$ time.
However, what I need... |
Theoretically in BPSK signal generation we will multiply the digital signal with the carrier and this output we will call as BPSK signl which is an analog signal.
But in OFDM if the data is modulated at the base band by using BPSK we will directly take that BPSK modulated data as 1,-1's which is not an analog sign... |
I'm trying to use a Butterworth filter in Python as described in this thread with these functions:
def butter_bandpass(lowcut, highcut, fs, order=5):
nyq = 0.5 * fs
low = lowcut / nyq
high = highcut / nyq
b, a = butter(order, [low, high], btype='band')
return b, a
def butter_bandpass_filter(data, ... |
could anyone explain why there is a need of negative exponent in fourier and laplace transform.I looked through the web but I couldn't get anything.Does anything happen if a positive exponent is placed in these transforms.
While looking through http://1drv.ms/1tbV45S it says that if $s>0$ it becomes a rapidly decreasi... |
I have a lot of songs I made over the 20 years and was thinking it would be interesting to see how the overall EQ changed on the various songs I made.
Any tool out there to help me with something like this ?
|
I am working in a fingerprint recognition project using C code running on embedded system. After processing the fingerprint image, I have the minutiae now. I tried to matching two sets of minutiae by calculate the distance between each pair of minutiae and check if it is less than a threshold. But it seems not accurate... |
I'm stuck on a final step in this problem. Essentially, there are the two systems above, which we'll call System 1 (Fig. 4.26, with ideal lowpass $H(jw)$) and System 2 (with $H_1(jw)$). The question proposes that the output y(t) of System 2 is identical to that which would be obtained by retaining $\Re\{f(t)\}$ of Sy... |
I am planning to use Pyramid Match Kernel method for object recognition from depth images: I am going to extract a set of feature vectors $x \in \mathbb{R}^d$ for each object instance and then I want to use the Pyramid Match Kernel as described in http://jmlr.csail.mit.edu/papers/volume8/grauman07a/grauman07a.pdf in Gr... |
I would like to write a 3 Band "EQ" with BiQuad Filters but for some reason, I get some phase distortion.
The problem already got discussed here.
I found out that I need to work with Allpass filters in order to compensate the phase response of my filters. Btw, the filters are Linkwitz-Riley 2nd order filters.
But when ... |
I'm trying to implement a simple Image super resolution algorithm (DWT-Based Resolution Enhancement
) in the following paper
Discrete Wavelet Transform-Based Satellite Image Resolution Enhancement
I tried to implement the algorithm in figure 3 of this paper using Matlab.Code is given below.
img1 = imread('lena1.jpg'); ... |
$H(z) = G\cdot\displaystyle\frac{2b_0\left(1-z^{-2}\right)}{1-2a_1z^{-1}-2a_2z^{-2}} + 1$
I have a data sheet with the above transfer function, I need to calculate its frequency response in code. I have the functions to calculate $b_0$, $a_1$, $a_2$ and $G$, but it has been a loooooong time since I have handled $H(z)$ ... |
I've got two signals which are consecutive received signals by a radar system. The signals' values are complex (in form $x+\textrm iy$).
I have generated a trasmitted signal using the info available.
Now, I have to calculate the range and motion(if any) of the objects using comparision of these signals in MATLAB. Comp... |
This is the impulse response:
Can anyone give a detailed method to prove that it is stable or not ?
|
While we look at Fourier Series there are both $\sin$ and $\cos$ components.But I think $\sin$ component is ony needed to describe wave.why there is also an $\cos$ component in Fourier Series?
$S_n(x)$=$A_0 + \sum_{n = 1}^{\infty} (A_n\cos(nx) + B_n\sin(nx))$
|
I have to find the gradient of the following term with respect to $X_{1}$:
$\|\Phi\circ(X_{1}-X_{2})-u\|_F^2$ ,
where $u\in\mathbb{R}^{n}$; $X_{1}, X_{2}\in\mathbb{R}^{N\times J}$ and $\Phi\in\mathbb{R}^{n\times NJ}$. The $\circ$ operation is defined as follows:
$\Phi\circ X=\Sigma_{i=1}^{J}\Phi_{i} x_{i}$,
where $x_i$... |
I tried to understand the following source code of a Phaser:
#define SR (44100.f) //sample rate
#define F_PI (3.14159f)
class Phaser{
public:
Phaser() //initialise to some usefull defaults...
: _fb( .7f )
, _lfoPhase( 0.f )
, _depth( 1.f )
, _zm1( 0.f )
{
Range( 440.f, 1600.f );
... |
Given the following signal I would like to calculate the rate of damping.
I presume that I need to find out where the peaks are using some kind of smoothing on the sample data. After performing an FFT I get the following…
I guess I could perform some kind of clipping (high/low pass) and then do an inverse FFT to get ... |
I am sending a chirp that sweeps form 20Hz to 500Hz to an audio codec, CS42L52, connected to an MCU, STM32F405, and I am receiving a "wavy" amplitude output. There is no processing on the MCU, I pass the signal straight from the DMA in to the DMA out. The DMA services the I2S lines. If I change the time that the chi... |
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