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Are there ways to increase computational performance of a normalized least squares (NLMS) filter? Multidelay block frequency-domain (MDF) filters have been proposed to do this, but they also take away from convergence speed and accuracy, because they only update the estimated impulse response once every block, not once... |
When developing products I do all of my algorithm design in Matlab. They are usually pretty basic, an IIR or FIR filter or two, a couple FFTs, etc. When it comes time to move it to an embedded environment I always have a hard time deciding what platform to run it on. I usually think of these (broad) platforms:
DSP cor... |
I have a project where I would like to image an object and be able to derive the heights of features in this image to sub-millimetre precision (exactly how precise is still yet to be determined, but let's say 100ths of a millimetre for now).
I have been previously advised that direct laser ranging techniques will not b... |
We can easily design interpolation filters that obey certain frequency-domain constraints using the Parks-McClellan algorithm. However, it's not immediately clear how to enforce time-domain constraints; in particular, I'm interested in generating Nyquist filters. So if I'm oversampling by a factor of N, I want the fi... |
I know that a Hidden Markov Model (HMM) is used in speech recognition and understand it to some degree. However, what I don't know is how input (speech) is "transformed" to a vector which in later used in HMM.
How do you get a vector from a sound input? Is this vector readable by a human?
|
In speech recognition, the front end generally does signal processing to allow feature extraction from the audio stream. A discrete Fourier transform (DFT) is applied twice in this process. The first time is after windowing; after this Mel binning is applied and then another Fourier transform.
I've noticed however, tha... |
I have a 2D image, which I want to lowpass filter, with these constraints / quality metrics:
I can not "add" light to the image, so each pixel in the result
should be <= the corresponding pixel in the input.
The lowpass cutoff frequency should be a parameter, to experiment with
Applying this filter repeatedly should n... |
I have 100,000 samples of a signal $x[n]$ that was sampled at 20kHz. The data is vibration data from a rotating machine, and contains a significant spectral component related to the speed of the machine's rotation.
Because the speed of the machine varies over the duration of the sample, using the peak of the FFT does ... |
For microscopy, we're frequently testing cameras. Since my applications involve very low signal-to-noise ratio, it becomes important that the noise is free of correlations and patterns, because local correlation is all that really distinguishes signal from background.
To test the noise, I usually acquire a series of ~1... |
In most audio processing tasks, one of the most used transformations is MFCC (Mel-frequency cepstral coefficients).
I mostly know the math that's behind the MFCC: I understand both the filterbank step and the Mel frequency scaling.
What I don't get is the DCT (Discrete Cosine Transform) step: What kind of information d... |
I have been looking at stereo vision applications, and one working example involves using two USB web cams, mounted together and calibrated.
Is there a way to estimate the theoretical spatial resolution of such a set up? The goal is to avoid purchasing and testing blindly but rather to be relatively confident ahead of... |
In the comment to this answer using a laser diode and diffraction grating to provide texture on a surface was suggested to aid in height calculations in a computer vision system.
I believe the pattern that I'm familiar with is projecting a chessboard pattern on the object. I was under the (incomplete) understanding th... |
IIR filters may be implemented as direct form 1 or 2. But how do you decide which form to use? What are the pros and cons of each?
|
I need to implement an approximation to the inverse of $x^x$, i.e. the square super-root (ssrt) function. For example, $\mathrm{ssrt}(2) \approx 1.56$ means that $1.56^{1.56} \approx 2$. I'm not as interested in any particular accuracy/bit-depth as I am in understanding what my options are in contrast to more straight... |
I have a wireless communication system that I am simulating in Matlab. I am performing some watermarking through slightly adjusting the phase of the transmitted signal. My simulation takes the original I (inphase) and Q (quadrature) values and adds in the watermark. I then have to simulate the resulting bit error rate ... |
Given a recording I need to detect whether any clipping has occurred.
Can I safely conclude there was clipping if any (one) sample reaches the maximum sample value, or should I look for a series of subsequent samples at the maximum level?
The recording may be taken from 16 or 24-bit A/D converters, and are converted ... |
Everyone discusses the Fourier transform when discussing signal processing. Why is it so important to signal processing and what does it tell us about the signal?
Does it only apply to digital signal processing or does it apply to analog signals as well?
|
In Image/Signal processing context we have seen/we know that if there are say 4 samples and if we take an average of those 4 samples, then we say that the result sample is a low pass filtered output sample. This is very relevant in Image processing context. Now then, what is the operation which implies a high pass filt... |
I've learned about a number of edge detection algorithms, including algorithms like Sobel, Laplacian, and Canny methods. It seems to me the most popular edge detector is a Canny edge detector, but is there cases where this isn't the optimal algorithm to use? How can I decide which algorithm to use? Thanks!
|
I can't think of a better way for asking this question so I will start with an example. Suppose that I have an input signal with a max frequency of 50Hz (sampled at 100Hz). Now the signals of interest lie in the range 0-5Hz, so I can add a low-pass filter with a cut-off of 5Hz, and use the resulting signal for further ... |
What exactly is meant by "stochastic sampling" and is it profoundly different from the regular Nyquist-Shannon sampling theorem? Is it related to sampling a stochastic process?
|
What are the proper steps for preprocessing my waveforms in order to perform an independent component analysis (ICA) later? I understand the how, though further explanation of that doesn't hurt, but I'm more interested in the why.
|
One of the standard ways to implement a Butterworth filter is with a cascade of second-order sections, each corresponding to a pair of complex-conjugate poles. For a fourth-order filter, for example, there would be two second-order sections. If we consider how the pole locations for a lowpass filter change in the z-p... |
Background: Often times I am doing some sort of signal processing task that requires a unique filter. Usually at this point I go to MATLAB and generate a new unique filter using $\tt firpm()$. The MATLAB firpm() function implements that Parks-McClellan algorithm. Now I have a filter and I put the filter into a hardcode... |
From what I understand, the DCT has half the bin size as a DFT of the same size N. The DFT also includes phase information, but often this is not needed when only the magnitude spectrum is desired.
Could the DCT be used to provide a magnitude spectrum with twice the density (half the bin spacing) of the DFT or woul... |
Huffman coding is a widely used method of entropy coding used for data compression. It assumes that we have complete knowledge of a signal's statistics. However, there are versions of Huffman coding that are used with streaming media and cannot possibly know everything about the signal's statistics. How do these adapti... |
Suppose we have a signal $n$ that consists of Gaussian white noise. If we modulate this signal by multiplying it by $\sin 2\omega t$, the resulting signal still has a white power spectrum, but clearly the noise is now "bunched" in time. This is an example of a cyclostationary process.
$$x(t) = n(t) \sin2\omega t$$
Su... |
Let's say that I have a 1 dimensional signal that has data which comes from one of 2 different spectral bands. The measurements come evenly mixed between the two bands. I'm interested in a method of extending the frequency calculation by combining information from both signals.
To show an example, let's just say that ... |
I have a signal of some length, say 1000 samples. I would like to extend this signal to 5000 samples, sampled at the same rate as the original (i.e., I want to predict what the signal would be if I continued to sample it for a longer period of time). The signal is composed of several sinusoidal components added togethe... |
I have an image and I would like to measure the amount of detail in it. Another way to look at it is to measure how blurry an image is. One way is to analyse the high frequency components in the Fourier transform of the image.
Are there any other/better methods?
|
In Five short stories about the cardinal series $[1]$, the author makes the following comment:
Interestingly enough, Shannon goes on to mention that other sets of
data can also be used to determine the band-limited signal--for
example, the values of ƒ and its first derivative at every other
sample point, the val... |
I have a set of contours (set of line segments) grouped in the following way:
$S_i = \{I^0, I^\frac{\pi}{4}, I^\frac{2\pi}{4}, \ldots, I^\frac{7\pi}{4} \}$
where
$S_i$ denote sequence of photos of one concrete object.
$I^j$ denote an image, with $j^{th}$ point of view ($j=0$ means front view).
Here is example of ... |
When designing a digital filter based on an analog filter we usually use the bilinear transform. To approximate a discrete transfer function $D_a(z)$ from analog (continuous) transfer function $A(s)$ we substitute
$$z = \frac{1+sT/2}{1-sT/2}$$
where $T$ is the sampling period. Alternatively, to approximate an continuo... |
What are less computationally demanding alternatives to the Viterbi Decoder?
Ideally what I would like is a list of the most commonly used approximate methods, along with brief pros and cons.
|
Given a filter used to shape the digital signal, $p(x)$, and given that we do not want the filter combination to cause any ISI, what "matched" filter, $q(x)$ will maximise the SNR?
Matched filters are used in digital communications to maximize the signal to noise ratio. Often a root-raised-cosine filter is used to sha... |
I have a floating-point digital signal processing system that operates at a fixed sample rate of $f_s = 32768$ samples per second implemented using an x86-64 processor. Assuming that the DSP system is synchronously locked to whatever matters, what is the best way to implement a digital oscillator at some frequency $f$... |
The same floating-point AMD X86-64 digital signal processing system mentioned in my previous question has a problem where it sometimes slows down substantially when signals attain values very near (but not exactly) zero.
The problem is that denormalized floating point values require special processing by the CPU which ... |
I would like to compute a power spectrum in which the frequencies are logarithmically spaced.
In Welch's method there is a trade-off between the frequency resolution of the resulting power spectrum and the number of averages (i.e. error in the result). I would like this trade-off to be dynamic, i.e. do fewer averages ... |
Suppose we have a linear represented in the standard state space notation:
$$ \dot{x}(t)=Ax(t)+Bu(t)$$
$$y(t) = Cx(t) + Du(t)$$
In order to get its impulse response, it is possible to take its Laplace transform to get
$$sX=AX+BU$$
$$Y=CX+DU$$
and then solve for the transfer function which is
$$\frac{Y}{U}=C(sI-A)^{-1}... |
Is there a generically accepted method of removing motion blur from an image. For a simple case, we can assume that motion happens in a straight line. I assume that it's a two part process composed of motion estimation and then deconvolution, but how is it actually done?
|
Suppose I've got an audio signal sampled at $48000$ Hz, and I'd like to design a low-pass filter that isolates everything below ~$60$Hz. In the digital world, this is a low-pass filter with the passband at $[-\frac{\pi}{400} , \frac{\pi}{400} ] $. Also, the transition band should be reasonable as well. Building a FIR f... |
In trying to understand this answer to my earlier question, can someone explain why it is important that the cameras are not in the same plane when the focal distance is large compared to the separation between the cameras and the object?
What is meant by "looking in"?
Does this change if the number of views/cameras in... |
The following paper describes an application of the Teager-Kaiser energy operator to x-ray image enhancement:
Reinhard Bernstein, Michael S. Moore and Sanjit K. Mitra, "Adjustable
Quadratic Filters for Image Enhancement" Proc. IEEE International
Conference on Image Processing (ICIP), Santa Barbara, CA, vol. 1, pp.... |
What techniques might one use to estimate the onset time of a sinusoidal tone burst in a noisy signal?
Assume the tone burst has a known fixed frequency (but unknown phase) and a very sharp rise time, and that the goal is to estimate the onset time within better than half the rise time, and/or one period of the freque... |
Compared to standard (conventional) beamformers, adaptive beamformers such as minimum variance distortionless response (MVDR) have much better resolution and interference nulling capabilities. Briefly, the MVDR beamformer is as follows:
If $\mathbf{a}_\theta$ is the look-vector in the direction $\theta$ and if $\wideha... |
In Hamming's book, The Art of Doing Science and Engineering, he relates the following story:
A group at Naval Postgraduate School was modulating a very high
frequency signal down to where they could afford to sample, according
to the sampling theorem as they understood it. But I realized if they
cleverly sample... |
We know that in quantum mechanics, momentum space is the fourier transform of position space (and vice versa)
And also, in time-series analysis, that frequency (of cycles) is the fourier transform of the distribution of all cycle lengths.
What about electromagnetic radiation? Is the distribution of frequencies the Four... |
A signal might have both continuous and discrete parts (where the "discrete" parts are regions where a sinusoidal Fourier transform would be subject to unnecessary Gibbs Noise). So I would think that it could be useful, even if it would require an entirely different implementation strategy.
If it is possible, what are ... |
I am referring to the the following paper : Non-contact, automated cardiac pulse measurements using video imaging and blind source separation
In the above article, the authors are able to extract cardiac pulse signal out from RGB components. I try to visualize the process as follow.
R' = R + cardiac pulse
G' = G + card... |
When decimating a narrow-band signal with a cascaded integrator-comb (CIC) filter, which FIR filter is more suitable to compensate the CIC frecuency response?
|
Polyphonic music transcription does not currently appear to be a solved problem.
How about the inverse of a small portion of the problem. Are there any kind of spectral characteristics (from an STFT) that can be used to eliminate some musical chords from the probability space? (e.g. this snippet of sound most likely ... |
Actually after reading the data from ADC is an $Q15$ format (ie. ADC supports 0-3.3v, ADC is an unipolar type). then the $Q15$ data is converted to $q31$ format ($Q31$ value = $q15_{data}$ <<16). this $Q31$ value is fed in to FIR filter (FIR filter is an $Q31$ format) then filter output is fed in to the fftr(real fft).... |
Does anyone know where there are any benchhmark timings for the DSP functionality in the ARM CMSIS V2 running on a Cortex-M4 processor. I am trying to work out whether this processor is powerful enough to meet my requirements, and to compare it to other microcontroller devices with DSP extensions.
edited to add
I will... |
A while ago I was trying different ways to draw digital waveforms, and one of the things I tried was, instead of the standard silhouette of the amplitude envelope, to display it more like an oscilloscope. This is what a sine and square wave look like on a scope:
The naïve way to do this is:
Divide up the audio file ... |
I usually use the mean squared error (MSE) or peak signal-to-noise ratio (PSNR) to compare two images, but this isn't good enough. I need to find a formula that returns a very big distance between an image A and its pixellated (or blurred) version B, but I don't know how to proceed. What would be a good metric for my n... |
For someone who has some experience in software development, loves his synths and knows the [very] basics of digital audio representation, what are the first steps he should follow in order to program a music synthesizer, DSP-wise?
|
I'm trying to implement an 8th order IIR filter and every application note and textbook I've read says that it is best to implement any filter of order more than 2 as second order sections. I used tf2sos in MATLAB to get the coefficients for second order sections which gave me a 6x4 coeffs for 4 second order sections, ... |
I am completely new to image analysis. Do you know how to binarize this image in such a way to get the fibers only?
A step by step procedure would be awesome, but any hint is appreciated as well. I can use Matlab, .NET and I am open to any other tool.
PS: At this address you can find the same question
|
I'm studying some DSP and I'm having trouble understanding the difference between phase delay and group delay.
It seems to me that they both measure the delay time of sinusoids passed through a filter.
Am I correct in thinking this?
If so, how do the two measurements differ?
Could someone give an example of a situ... |
If one wants to smooth a time series using a window function such as Hanning, Hamming, Blackman etc., what are the considerations for favouring any one window over another?
|
I have a 3D volume image that consists of 2D tomographic images. In MatLab, for 2D images, we can use bwmorph and call for 'skel' operation to infinity to get a pixel thick overall outline of the image. bwmorph cannot deal with 3D images, however, so I am looking around for implementing "skel" operation for such tomogr... |
A question recently came up regarding Parks-McClellan and some of the comments point out that the wikipedia article on Parks-McClellan states,
...the Parks-McClellan algorithm is a variation of the Remez algorithm or Remez exchange algorithm.
Is there a difference in performance? Is the remez function in octave e... |
I am looking for the fastest available algorithm for distance transform.
According to Image Processing Learning Resources - HIPR 2 (HYPERMEDIA IMAGE PROCESSING REFERENCE) - Morphology - Distance Transform:
The distance transform can be calculated much more efficiently using
clever algorithms in only two passes (e.g. R... |
Am new to signal processing and was wondering when given two signals, what are the widely used statistical analysis methods to understand the relationship between them?
|
I already posted this question on StackOverflow. I got a suggestion to go for a better answer here.
To make the question more concise I'm interested in an introduction to digital filters, any resource is good.
But first I will be satisfied if someone could point me to some resources explaining this filter (used in And... |
I've a question about the effect of aliasing on the magnitude of autocorrelations. From a simulation in MATLAB, I don't see any effect of aliasing or any need to anti-alias filter when I take the magnitude of the autocorrelation. Which means I can undersample the data and then take the autocorrelation. There is a paper... |
If a device generates clicks and pops at start and end of audio file playback, even when those audio files are completely zeroed out and dc-offset corrected, what could've caused them?
|
I have many EEG signals and I want to analyze them using linear methods such as STFT (Short Time Fourier Transform). In STFT , How can I optimize the analysis window length, to reflect the frequency spectrum of each analysis window in a proper way?
|
[Thank you, @leftaroundabout for your helpful comments to my related question. I'm trying to ask a less confusing question here.]
$0\leq\phi\leq 2\pi$ is a parameter that theoretically describes a system. $\alpha=\sin(\phi)$ and $\alpha' = \cos(\phi)$ are observables.
In a measurement, $\phi(t)$ is a continuous periodi... |
When a digital device (PC, portable media player, etc.) plays a digital audio file (ogg, mp3, flac, etc.) the audio signal output is always the same, regardless of the device type/brand, am I right? Thus the sound quality should not differ. For instance, I have an iPod that plays an mp3 file. If I play the same file on... |
When downsampling an image by an integer factor $n$, the obvious method is to set the pixels of the output image to the average of the corresponding $n \times n$ blocks in the input image.
I remember vaguely having read somewhere that this method is not optimal (sorry I don't remember any details)
Is it true that there... |
Back in the days of 1 MHz 8-bit CPU personal computers (Apple II, Atari 800, et.al.), there were software programs that could do comprehendible arbitrary text-to-speech synthesis on those PCs. What published speech synthesis algorithms might be suitable for implementation on micro-controllers with similarly limited pe... |
If I have the details of a scene and a camera, how can I tell how much of my camera's image sensor will be taken by a certain object in space?
For example, if I have a painting that is 1m by 1m, it is straight ahead 10 m away from my camera. If my image sensor is 5 mm by 5 mm, and I fix the focal length of the camera,... |
I was watching the American and Indian idol auditions on youtube earlier in the day. I watched both successful and failed contestants and I was able to distinguish the really bad from the good singers even if I did not recognize the song or had ever heard the melody. Among the good singers, however, my limited knowledg... |
If a wide-sense stationary signal $X$ is fed to an LTI filter with the transfer function $H$, the power spectral density (PSD) of the output $Y$ can be expressed as:
$$R_Y(f) = \left|H(f)\right|^2R_X(f)$$
where $R_X$ denotes the PSD of $X$.
Does this relation have a common name?
|
A generic implementation of ICA for the separation of a mixture of $N$ signals into their $M$ constituent components requires that the signals be assumed to be a linear instantaneous mixture of the sources. Every description of ICA that I have come across seems to take for granted the fact that all $M$ sources are pre... |
Propose that we have a signal with a model of
$$
f_{4at0} = v_0(1 - \epsilon + \epsilon \exp(-c_l(t_{t2d}-t_0)))
$$
$f_{4at0}$ is dependent variable while $t_{t2d}$is independent variable,others are parameters.
It is a representation of the relationship of biological material concentration-time that is inside the blo... |
I asked a question over on stack overflow.
I'm having a slight problem however. As suggested by Paul R I am mirroring my lower $n/2$ bins into the upper $n/2$ bins.
I have a few questions however.
I'm assuming $n$ is $2^{fftOrder}$? Thus for an order 10 IFFT I would be mirroring the bottom 512 frequency bins in to t... |
I'm new to DSP, and just discovered this StackExchange, so apologies if this isn't the right place to post this question.
Is there a resource that describes genres in a more mathematical terms? For example, if I've performed an FFT on the signal on this section of the song (2:09 if the link doesn't start there), is th... |
I'm trying to get my head round the operation of the Wiener filter for the purpose of image noise reduction. In my case I'll have used another noise reduction filter first and will then use the result of this as an approximation of the noise characteristics for the Wiener filter.
Regarding information on the Wiener fil... |
A filter that can be used for digital signals like audio, video or image processing can be defined using a matrix ("kernel") that weights the surrounding area (this is a description I read in lecture notes from someone else).
The kernel
$$\left(\begin{array}{ccc} 1 & 1 & 1 \\
1 & 1 & 1\\
1 & 1 & 1\end{array}\right)$$... |
I'm trying to calculate the loudness of an audio track I have stored in a buffer. The buffer contains PCM data of the signal and I want to get how 'loud' it is by using Root Mean Squared. I assume I can do this in the time-domain instead of having to switch to the frequency domain. What would be the pseudo-code for doi... |
Just out of curiousity, does anyone know how does google map put together all those massive quantity of satellite pictures with with high accuracy? What advanced computer science technology has it employed?
|
I always see the Kalman filter used with such input data. For example, the inputs are commonly a position and the correspondent velocity:
$$
(x, \dfrac{dx}{dt})
$$
In my case, I only have 2D positions and angles at each sample time:
$$
P_i(x_i, y_i) \qquad \text{and} \qquad (\alpha_1, \alpha_2, \alpha_3)
$$
Should ... |
It has been shown that 'diagonal loading' a covariance matrix derived for an adaptive beamformer can improve robustness of the beamformer when the antenna array is perturbed, albeit at the expense of background noise power.
The question is why does this work? Is the diagonal loading stopping the adaptive filter from r... |
I recently implemented an MMSE channel equalizer, and I have read about many other types of channel estimation/compensation algorithms out there. However, one thing that is never talked about it how to estimate the number of taps in your model of the channel you are trying to estimate. How is this accomplished?
For exa... |
I need some help clarifying FFTs and what they represent. I have a buffer containing compressed audio. Due to limitations, I can't handle the full uncompressed audio but can decompress small segments at a time.
Lets say I take 10 seconds of uncompressed samples, I would need to store this in PCMaudiobuffer of size 10... |
My system is the following. I use the camera of a mobile device to track an object. From this tracking, I get four 3D points that I project on the screen, to get four 2D points. These 8 values are kinda noisy, due to the detection, so I want to filter them to make the movement smoother and more realistic. As a second m... |
What are some algorithms for generating a good pseudo-random approximation to $1/f$ (pink) noise, yet suitable for implementation with low computational cost on an integer DSP?
|
I'm looking for an algorithm to store small dimensions of some pictures and later search on the stored data for matches similar to a new picture. Maybe something with Self Organizing Map in machine learning ?
|
I am trying to teach myself more about image compression using the wavelet transform method. What is it about certain wavelets that make them preferable when compressing images? Are they easier to compute? Do they produce smoother images?
Example: JPEG 2000 uses the Cohen-Daubechies-Feauveau 9/7 Wavelet. Why this one... |
Suppose we wish to interpolate a periodic signal with an even number of samples (e.g. N=8) by zero-padding in the frequency domain.
Let the DFT X=[A,B,C,D,E,F,G,H]
Now let's pad it to 16 samples to give Y. Every textbook example and online tutorial I have seen inserts zeros at [Y4...Y11] giving
Y=[2A,2B,2C,2D,0,0,0,0,... |
Background
I am designing a system that will have a single small microphone and speakers for use in a phone type setting. Easiest example I can give is a Skype conversation where you are using your computers speakers and a desktop microphone.
I am worried about the audio from the speakers getting picked up by the micro... |
I am looking for information on how to decode slow scan TV signals, which is a method of transmitting static images using everyday FM radio.
I have seen software but I don't understand how to start. Any open source would help as well. To very specific I want to know how to decode the sound into an image.
|
Is it possible for two different decoders to output two different sample results for the same MP3? I'm decoding an mp3 on Matlab and it produces two columns (presumably one for each channel). When I extract the sample data from the same file using AVAssetReader (iOS), I get different results. When I tried the same code... |
I can calculate RMS in frequency domain as derived from Parseval's Theorem.
But what if I have applied a windowing function before doing the FFT (in my case a Hann window)?. Now the RMS values are not correct any more. How do I calculate the correct RMS values?
|
It's well known that the bilinear transform is also known as Tustin's Method. As far as I know, Arnold Tustin really did introduce the idea into the control systems literature, so the name isn't just a case of Stigler's Law. For example, I managed to find the following reference:
Tustin in the UK developed the bilin... |
Is there a data decomposition method similar to eigenvalue that estimates projection matrix to reduce dimensionality but does not project similar vectors too far away in euclidian distance terms from each other if original data from the same class varies a little in scale, shift and rotation (2D case).
$$y = E x;$$
e.... |
I'm looking at decoding an OFDM signal which consists of 6 carriers (or tones) which are BPSK modulated and a pilot tone which aids tuning. This is the first time I have worked with OFDM so I need to know if I'm approaching this in the right way.
The way I'm thinking of decoding it is to use the pilot tone to calibrate... |
We know that the under-sampling results in aliasing and frequencies higher than half of the Nyquist rate is not distinguishable. I've a base band signal that I want to use the higher frequencies which are higher than the half of the Nyquist rate (Nyquist frequency) as well as low frequencies(all parts). I've a special ... |
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