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1.771 filters questions.

for a Project I need to build an Audio Modem in GNU Radio. I decided, that it would be the best to use DBPSK Modulation, because that is fast and relatively noise resistant. But as it seems, the ...

What happens if the noise has no zero mean? I mean, if the exercise is something like: $$y(k) = A x + \eta(k)$$ When I have zero mean, I start from: $$y = A x$$ $$\Rightarrow \hat{y} = A \hat{x}$$ ...

In OFDM system, I need to transmit a signal $x$ in SIMO channel $H$ with one $Tx$ antenna and 4 $Rx$ receiver's antennas. The initial equation for that convolution is $r=Hx+n$ where $n$ is the noise. $...

Let's say we want to isolate a band 1000 hz +/- 50 hz. Obviously, limiting the bandwidth by applying a passband filter will always destroy a bit the sharp transients (a Dirac or a rectangular ...

I wanted to know how I can estimate velocity of a particle (contained with a sensor) from the acceleration output of a MEMS sensor device in order to find the total kinetic energy of the particle? ...

a figure for instance of size 500*500 has half above part with black and below half white should result in a white line where the white meets the black (something like a single line at line 250 with ...

I've been trying to design a bandpass filter using scipy but I keep getting a LinAlg Singular Matrix error. I read that a singular matrix is one that is not invertable, but I'm not sure how that error ...

I have a blurred and noisy image $X$, I want to apply the Wiener filter on it and get a deblurred and denoised image $Y$ (i.e apply inverse of blurring filter while at the same time reducing some ...

For a given narrowband Gaussian filter with a specific central frequency and filter width, I need corners of a bandpass Butterworth filter whose amplitude response is close enough to the Gaussian ...

Good Morning! I am new in signal processing and I am trying to do a work in noise control of an electronic steering lock device (ESL). My aim is to calculate the loudness (Zwicker Method- ISO 532 B) ...

Consider the LTI system given by: $H(z) = 1 - \frac{1}{2}z^{-1}+\frac{3}{4}z^{-2}$ Let $x[n] = (\frac{1}{2})^nu[n]$ be the input to the system. We want to find the output for $n = 0,1,...,N_a$, using ...

We have a position sensor that under some conditions receives some high frequency noise. We can eliminate that very well with a simple mean filtering. Unfortunately this causes too much lag when ...

I am trying to make filters for days but now I just want to know the experience of others rather than go on to wrong way and finally decided to put my query over ...

With scipy I can use signal.butter and signal.lfilter on a time signal. Performing a Fourier ...

I have a question about matched filtering. Does the matched filter maximise the SNR at the moment of decision only? As far as I understand, if you put, say, NRZ through a matched filter, the SNR ...

I understand to an extent various filter like low pass filter, high pass filer, Wiener filter Kalman filter etc. I also understand some of this filter will decorrelate/uncorrelate the signal. The ...

I'm trying to use a Butterworth filter in Python as described in this thread with these functions: ...

Is anybody familiar with Gustafson's algorithm for minimizing transients in forward backward filtering [1]? I'm trying to implement it and my first guess was to check Matlab's filtfilt.m, since they ...

I'm having a problem with a signal, I need to leave only the low frequencies so smooth as possible, because then this signal will be the input of a neural network, so I need to reduce complexity. The ...

I would like to prove that convolution of an image $I \in \mathcal{M}_{m_1 \times n_1}$ with respect to a separable 2D filter $F$, (i.e., $F = F_1 F_2$, where $F \in \mathcal{M}_{m_2\times n_2}(\...

Could you please introduce me some references about edge detection of 1-D signals? I have a pulse similar to rectangular pulses which is immersed in noise. Would you please introduce me some filters ...

I am trying to extract features from a signal using filtfilt function in MATLAB. I implement the filtfilt function on my original signal using a high pass filter(second derivative) and use the peak of ...

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...

I'm currently working with a dataset of $5000$ pulses of $N=15000$ samples each. I managed to implement the RLS algorithms with a FIR M-Tap filter such that $M\leq 15000$ ($150$ seems to achieve the ...

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...

Weierstrass transform (Gaussian filtering), with infinitely wide Gaussian, is expected to give zero and if we start from the definition it is consistent with this expectation as $\lim_{\sigma \to \...

I would like to know if it's possible to reconstruct the original time domain signal from it's time-stretched version? Is there any algorithm out there that can do this? Python, Matlab, etc? I want ...

If a practical filter can't remove all unwanted frequency components like ideal filter, does that mean unwanted frequency component are still present after filtering? how can we use such distorted ...

I'm very new in digital signal processing. I have multiple sensors and the way I filters the signals in post processing is: take FFT of the signals. put zero on out range of interesting frequency (...

I work with depth time series data from electronically tagged fish. When the fish spend time on the bottom we get a prominent tidal signal (approx 1.96 cycles per day). This interferes with our ...

In steepest descent methods of minimizing a function $f(x), x \in \mathbb{R}^d$, it's common to approximate the gradient by finite differences: $\qquad\qquad \nabla f(x) \approx gradest( x; h ) \...

To introduce my situation: I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "...

I'm trying to understand how to show that with real coefficients, the phase response of a filter is 0. Here is the impulse response $h[n] = b_1d[n+1] + b_0d[n] + b_1d[n-1]$ How should I approach ...

I consider a signal of length $N = 2^n$ for some $n$. I want to derive two signal from it, one containing only the odd frequencies and one only the even frequencies. Each of these signals have length $...

I have a periodic signal(ECG) with period of ~1 seconds. It does not have features that are shorter than 0.04 seconds. For removal of 60Hz, I thought instead of implementing a notch filter, doing a ...

This is the simple code to find transfer function between sigout and sigin signals and then are the filter coefficients ...

Suppose I have point array $y_i$ of size $N$. How to implement moving average algorithm that conserves quantity $$ I = \sum_{i=1}^{N}y_i $$ NOTE: I don't want time shift so I would prefer to use ...

So I implemented the code for bilateral filter from here and ran it for various values of spatial sigma and intensity sigma. I noticed that when I add gaussian noise to an image with variance say 0.02,...

I would like to implement EM for blind channel estimation. The algorithm is briefly described in this paper, under the section "EM for blind channel identification". I'd like to derive the algorithm ...

I want to determine optimal initial states for a FIR/IIR filter to get rid of oscillations and shifts at the initial interval. I've found that these initial states depend on filter type in the ...

I have a weakly stationary signal $X(t), t\in R.$ The covariance function of $X$ is $r_x(\tau) = \frac{1}{1 + \tau^2}$ where $\tau$ is the time difference. I have a filter with "filter-function" $H(f) ...

How can we check whether the filter is realizable given its transfer function and What are the parameters the realization depends on? Here is an example: Show that a filter with transfer function ...

I am a noob to signal processing, but I need to create some audio files for IVR for Asterisk VOIP. So my lady friend with a nice voice records her message on a Windows machine and sends me a 44KHz, ...

A traditional IIR / FIR filter (lowpass to remove the high freq oscillations), e.g. moving average, or a Savitzky-Golay filter can all be useful to smoothen a signal, such as an envelope signal: ...

I have a question about the implementation of digital filters in CoreAudio. I'm in big trouble because it is a few weeks I'm trying to understand how to implement them. The basic idea is this: while I ...

I have an audio clip. Here is the clip in wave format https://www.dropbox.com/s/542jgpfk7bfkt01/sample.wav?dl=0 I chose the following area of time domain signal The spectrum of this signal shows ...

I sent a square wave signal (red) through a IIR filter (butterworth in this case), to make it more "realistic". Then there's a slight group delay in my output signal (blue). Is there a way to find out ...

I am a bit confused about the Kelly Lochbaum Ladder Filter. Image below taken from here. My question is, is this an LTI (linear time-invariant) filter? I thought that it was LTI, but I am not sure ...

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...

I'm trying to apply a sliding window minimum and maximum filter to an image of a certain window size. Actually, I'm trying to find the optimum window size for it. But I really haven't gotten the hang ...

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