Questions tagged [filters]

In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

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Direct Digital Synthesis - Super Sampling - Unknown Harmonics

I'm using a Xilinx FPGA (Virtex) with 4 DDS cores (each supplied a 250MHz clock) used in parallel to provide a samples to a DAC38J82IAAV from TI, 16 bit DAC running at 1 Gsps. The four cores super ...
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1answer
27 views

Linear phase FIR filter for impulse responses that don't appear symmetric

I would like to clarify some confusion I have about linear phase FIR filters of which do not seem to have symmetric impulse responses. Starting with a simple case, a delay, $ h[n] =\delta(n-n_0) $ ...
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Wiener Filtering to perform minimum mean square estimation

As the Title suggests, I am trying to design a Weiner Filter for getting the lower bound on my LTE channel estimation algorithm. To give you a high level on the chain, First after generating the ...
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1answer
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Are MATLAB function zp2tf() and tf2zp() are complementary or not?

I was under impression that given, pole, zero and gain the transfer function (filter coefficients b and a) is fixed. Therefore, ...
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1answer
31 views

Zone Plate Reconstruction

I am trying to reconstruct the Zone Plate image and am struggling to remove the last remaining aliasing. ShaderToy: https://www.shadertoy.com/view/wdGGWK In the shader above you can see that I am ...
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Passing a sampled signal through a filter

I was wondering why is it wrong to use a band-pass filter on a sampled signal? If the signal we want to sample has frequencies up to fmax, we sample it with frequency fs = 2fmax (so that Nyquist ...
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4answers
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Can every type of linear filter be modelled by a convolution?

I have an input time series going through a filter that creates another time series as output. If I assume in first approximation that my filter is linear, does it necessarily mean that I can model ...
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1answer
51 views

How to smooth filter response?

I have a filter designed in matlab with the function cheby2( N, Rs, Ws, 'stop'). The filter would give nice frequency response with for a given parameter set when the filter order is 2 or 4 (N=4). But ...
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1answer
36 views

How to filter very low amplitude staircase signal from a background noise

I have a staircase signal of a specific pattern that is drowned in a background noise. The ratio of signal to noise amplitude is nearly $10^{-4}$. I have attached the image of both with this post. I ...
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2answers
71 views

Convolution of Signal with a Gaussian Filter / Kernel

Following up on Analytical Solution for the Convolution of Signal with a Box Filter, I am now trying to convolve a Gaussian filter with the sine signal by hand. My method is to use the definition of ...
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1answer
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Bessel filters: linear passband phase characteristic only for low pass filters?

I've found a test (internet) from the contest to signal processing teacher and it has a question (true) or (false). I cant find any book or paper to argue this statement. Is it really true?
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Values of angular frequency of discrete-time sinusoidal signal

Why are the values of angular frequency of discrete-time sinusoidal signals in between -pi and pi?
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RSP1A sample rate vs if bandwidth

I'm using the RSP1A SDR and I want to understand how to choose the best configuration for receiving a signal of 125 khz bandwidth. I would like to know if when choosing the option 200 KHz IF ...
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How to filter out noise in high frequency signal?

I am trying to filter this signal (download-zip): ...
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FIR bandpass filter for complex baseband signals

I am DSP newbie and I need to apply a bandpass filter to the complex baseband signals of bandwidth 20MHz and sampling rate 61.44 MHz. Below is the octave code for the same and I have obtained ...
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1answer
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How would I calculate the DC Gain for a Chebyshev Type 2 filter?

I'm currently working on an implementation of a Chebyshev Type 2 filter but the gain is all over the place when playing around with the order and ripple. I haven't found anything online but (as far as ...
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1answer
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How to perform spectral inversion in the frequency domain to convert a low-pass filter into a high-pass filter?

To convert a linear-phase FIR low-pass filter into a high-pass filter with the same cut-off frequency, we can invert the sign of the low-pass filter's impulse response $h(n)$ and then add one to the ...
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2answers
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Are Chebyshev Type 1 filters very unstable at very high orders?

I am currently experimenting with a digital implementation of a Chebyshev Type 1 filter of mine, for audio purposes. I have written the code myself so I was wondering if the behavior I've noticed is ...
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Low pass filter output showing instability

I am fairly new to DSP, and I'm trying to implement the Pan Tompkins algorithm for QRS detection of ECG signals in MATLAB. The first stage of the algorithm consists of a second-order low-pass filter. ...
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What Is the Bilateral Filter Category: LPF, HPF, BPF or BSF?

I`m reading about Bilateral filter with images, I knew that it uses Gaussian filter (which is a LPF) as a domain filter plus a range filter and it is used to preserve edges(high freq component). Can ...
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Using a Multirate FFT system to analyse audio

I am trying to implement a multirate fft system to try and achieve frequency windowed analysis of audio. Say I have managed to get an impulse response of a system. This impulse may be say one ...
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1answer
67 views

Analytical Solution for the Convolution of Signal with a Box Filter

I have an exercise in which I am trying to filter an input signal $y(x) = \sin(x)$. Ideally, I would like to apply a box filter to this signal. Previously, I successfully convolved the input signal $...
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Calculating output of a digital system signal with a filter that introduces delay

We have a step function at the entrance that goes through the following system. The $D(z)$ digital filter introduces a 0.25s delay. Output as $y(kT)$ must be calculated. How can we tackle this problem?...
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1answer
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Understanding a lowpass - comb filter implementation

I try to understand the implementation of the low-pass comb filter of the Freeverb reverberation algorithm: https://ccrma.stanford.edu/~jos/pasp/Lowpass_Feedback_Comb_Filter.html The original ...
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1answer
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Stability of the Low pass filter after Spectral Transformation

Using a spectral transformation a filter $G(z^{cap})$ is designed to be a low pass filter from an existing lowpass filter $G_{lp}(z)$ via $G(z^{cap}) = G(z)z=H(z^{cap})$ . Which property should H(z) ...
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Can Kaiser's formula be used to estimate the filter length of an FIR filter for all windowing methods?

I came across this problem while designing a FIR filter for given $\omega_p = 0.4\pi$ and $\omega_s = 0.6\pi$ along with the minimum stop-band attenuation $\delta_s$. I have worked out some examples ...
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2answers
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Lattice structure realization of an all-pole system

Given an IIR system with the system function, $$ H(z) = \frac{1}{2 + 1.4z^{-1} + 1.8z^{-2}} $$ I need to draw the lattice structure realization. I have drawn the lattice structure for $\frac{1}{1 + ...
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O-MOMS3 interpolating function?

I've been reading the following paper on a B-spline based interpolation technique: https://pdfs.semanticscholar.org/fd55/61d561fa2bf1124959cba0d4abfd5d81a784.pdf The paper derives the so-called O-...
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1answer
32 views

Calculate transfer function of two parallel transfer functions in a feedback loop

I have rational transfer functions: $$H_1(z) = \frac{ b_0 + b_1z^{-1} + b_2z^{-2} }{a_0 + a_1z^{-1} + a_2z^{-2}}$$ $$H_2(z) = \frac{ q_0 + q_1z^{-1} + q_2z^{-2} }{p_0 + p_1z^{-1} + p_2z^{-2}}$$ And ...
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How to reduce noise in plot of velocity calculated from GPS points?

I'm interested to see if it's possible to compute the velocity given only GPS data and so, I calculated the distance between GPS points and computed the velocity by dividing it by the time interval ...
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1answer
53 views

Compensate delay of non-symmetric (non-linear phase) filter

Lets assume I want to apply a matched filter h which is non-symmetric to my signal x and the output is ...
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2answers
56 views

How to filter data with given numerator and denominator coefficients of an IIR Filter, in MATLAB Implementation

Coefficients are: b = [9.02201251375918 0.485095309463300 -0.571759308660462 3.63774928924158 4.42592447788520 1.30093840271037 -0.898592924453941 -9.08965268561448 -12.4939309921467 ...
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2answers
76 views

How to do FFT fractional time delay (SOLVED)

I am trying to time-shift a signal using the FFT, however I have encountered some strange effects that depend on the size of the time-shift. I need to be able to shift the time by an arbitrary amount -...
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38 views

Magnitude attenuation by a filter

I have a filter, with coefficients given by W=sin((2*pi*2*Ff*k/Fs)/(2*pi*2*Ff*k/Fs))*h(k), where Ff is the filter reference frequency, k runs from -N/2 to N/2 (N is the order of the filter), Fs is the ...
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1answer
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How to choose order and cut-off frequency for low-pass Butterworth filter?

When using IMU data (accelerations and angular velocities), many papers use a 4th order low-pass butterworth filter with a cutoff frequency of 5 Hz. However, none of them really justify why those ...
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2answers
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Distinguishing Spoken Speech from Played-back Recording of Speech

Do any general-purpose techniques exist to distinguish live spoken speech from played-back speech samples? For example, if I were to deploy a speaker verification system to authenticate users by voice,...
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Design of low pass filter

I'm designing a low pass filter to identify the DC content of a signal. The most important aspects of my application are the accuracy and the speed of the identification. I would like to know if the ...
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Vary poles/zeros of digital IIR filter

Many applications that produce sound, such as software synthesizers, are able to apply a filter that varies with time, such as applying a low pass filter that varies with an LFO. I currently have a ...
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Determining window Size for Savitsky-Golay filter (Savgol_filter)

First, I am aware that an ideal optimal window task is really signal/task-specific, i.e. nature of signal/sampling rate/characteristics of interest etc. What I am interested in: "Is there an optimal ...
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1answer
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how do I derive the system equation for a simple delay with feedback?

I am a software engineer, and just learning digital signal processing formally, though I've hacked around before a fair amount. I'm implementing a delay audio VST and I'm trying to wrap my head ...
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Inconsistant Variation in BLE Beacon RSSI Values for Distance Measurement

I am developing an application to estimate the distance to a BLE Beacon using its RSSi values measured from a Mobile Phone. But when I started to collect data I could see that they varied so much that ...
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1answer
45 views

Localized Gamma Correction

Question: Is there an algorithm that does (what I would call) "localized" gamma correction? That is, applying different gamma corrections on different parts of an image, using a masking filter or ...
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1answer
23 views

Impulse response of a 3x3 PSF - how to find analytical expression for fourier transform of a 3x3 matrix?

I have a filter $\mu[n_1, n_2]$ with taps: $$ (1/8) (1/4) (1/8)$$ $$ (1/4) (1/2) (1/4)$$ $$ (1/8) (1/4) (1/8)$$ How do I find an analytical expression for $\hat\mu(w_1, w_2) $? Since it looks so ...
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Is the typical implementation of low pass filter in C code actually not a typical low pass filter?

Quite often in C code you see something along the lines of: $$ y = (1-a) y_{prev} + input * a $$ ..where $a$ is some small number. This equation then runs in discrete intervals at the loop rate. ...
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Downsides of higher order filtfilt's

I understand that with a standard filter, higher order will yield sharper frequency selection at the cost of increased phase delay. However, with so-called zero-phase-distortion filters like that ...
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Avoiding transients during filtering: Padding or Gustafsson's method?

I'm pretty uneducated when it comes to filtering, more of a stats guy, but I have to bandpass some signals and after a bunch of reading today I understand that the ...
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1answer
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Automotive Audio Question: IIR Filters and minimum phase

In a car, does matching the EQ of both the left and right side using IIRfilters mean that I'm matching the phase of each side? I have someone telling me that the car's environment is mostly minimum ...
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Deriving transformation from input and output images

I will preface this by saying that my undergrad signal processing course is very far back... Context: I am comparing images of global illumination (GI) passes. I want to be able to change the ...
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1answer
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FIR Filter Type - MATLAB

A question about using fir1/fir2/firpm to design FIR filters in MATLAB. The first parameter is the 'filter order'. For my filter design I pass in 250 as this argument. However, when looking at the ...
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What is wrong with band passing by inverse Fourier transforming a section of a signal's frequency spectrum? [duplicate]

If the question is unclear, here is what I mean in Python using scipy.fftpack: ifft(fft(arr)[x:y], len(arr)) What is wrong with this method of band passing?