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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EQ - Signal attenuation with IIR bandpass filter bank

I've been charged to make an audio equalizer in c++, and I've been learning DSP during the last months but I'm still a newbie, so the title may not explain exactly my problem... I was able to make it ...
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31 views

Hilbert Transform, filters - two different phase graphs

Relationship between magnitude response and phase response for minimum phase is: Phase plots are obtained from Hilbert transform of real magnitude values. Input to Hilbert transform was a vector ...
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21 views

Understand the assumption of the ideal bright line in this paper

hope I ask my question on the right place. I am a student and my favorit subject is computer vision. I try to understand some apects of the paper "Three-dimensional multi-scale line filter for ...
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2answers
50 views

derivative filter transfer function

In many of the papers it is said that the derivative filter transfer function is given by: $$H(z) = \dfrac{1}{8T}\left(-z^{-2} - 2z^{-1} + 2z + z^{2}\right)$$ But no one gave the detailed information ...
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42 views

EM algorithm and Kalman filter

I am new to the subject of Kalman filtering and therefore my question might seem trivial. I see that there is a tight connection between Kalman filter and EM algorithm when one wants to predict the ...
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1answer
38 views

For linear IIR digital filter, what happens for negative frequencies?

By negative frequency, I refer to Fourier transform. Often, the frequency response of a digital filter is only displayed for positive frequencies. For a linear IIR digital filter, what happens for ...
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47 views

Filters - Hilbert Transform and Absolute Value of Magnitude Response

Following relationship between magnitude response and phase response for minimum phase: I have implemented in C++ code using the Hilbert Transform C code from file found online ht.c. However, I am ...
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38 views

Hilbert Transform - C or C++ code implementation

For a minimum phase system relationship of magnitude response to phase response is: In the search for C code implementation I have found a ht.c file on www.physionet.org. However, this code ...
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30 views

ITU-R 2k filter implementation

How to generate the coefficients of an itu-r 2k filter when all I know is my 0 dB point is at 2 kHz and my maximum of 6 dB is at 7 kHz
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1answer
23 views

Visualisation of bandwith of filter with sampling

I'm having some trouble visualising (and understanding) the filtering of a sampled signal. If I understand correctly, after sampling a signal, it is desired to have it go through a lowpass to ...
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2answers
58 views

Raised-Cosine Filter - phase response graphs for three kinds of filter

In the previous post it has been mentioned that the phase response of Raised-Cosine filter is linear and is given with ϕ(ω)=−ωT, where T is the delay. But, I have also read in literature that it is ...
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23 views

What is pulse train filter?

I have recently seen some code written in modelling tool like Simulink, ASCET which uses PT1 filter? Why do we use a pulse train filter? What is the difference between FIR, IIR, Chebyschev, ...
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36 views

Raised-Cosine Filter - calculating magnitude response

Here is the formula to compute magnitude of a raised-cosine filter: I know that 0 <= Beta <= 1. I am not sure what are the values for f (frequency) and Ts. I assume that some normalization ...
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31 views

Filter implementation and alternatives

I'm learning of digital filters and found a question asking why a filter with a impulse response $$h[n]= \begin{cases} \frac{1}{n^2+3n+2} & n\geq 0 \\ 0 & n<0 \\ ...
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1answer
63 views

Raised-Cosine Filter - calculating phase response

I want to compute phase response of an raised-cosine FIR filter. I have trouble finding the complex transfer function of the filter in literature (there is a real valued transfer function of ...
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3answers
118 views

IIR Filter - calculating the phase response

Using formulae from Audio EQ Cookbook from here it was easy to implement biquad IIR filters in C++, i.e. to calculate coefficients b0, b1, b2, a0, a1, a2. Now, I want to calculate the phase response ...
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27 views

SDR Signal processing - noise filtering

I'm looking for an advice with noise filtering using digital singal processing. I've been pointed here in my original question posted on Stack Overflow. So whats the problem: 1. I am using my SDR ...
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1answer
30 views

Frequency filtering with a DFT and meaning of removing complex conjugates

I've seen many implementations to filter out frequency components of some time domain signal by performing a DFT, zeroing the unwanted frequency bins, and performing the IDFT to get the filtered ...
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48 views

Finding filaments in high dimensional space

I have unit vectors in a high-dimensional space (~300) that are non-uniformly distributed over this hypersphere. There is physical meaning when the vectors are clustered together, but I'm ...
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26 views

How are filters for polyphase decimation like?

I know in naive decimation the signal should filtered by a low-pass filter with cutoff frequency equals to Nyquist rate.I don't know how the filter banks like for polyphase decimation. Are them still ...
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1answer
33 views

Removing a non linear noise trend from data

I have a dataset that looks as follows Now, this is from an actual measurement (of the reflection coefficient of a microchip with four LC resonators connected in a periodic fashion) so there is ...
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1answer
36 views

Possible to implement simple IIR filter with greater than 6 db/octave slope?

I am using the following IIR code as a low-pass filter in a software project, and it works great. However, I would like the slope of the filter to be much steeper as the one here seems to be very ...
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1answer
31 views

What is the mathematical interpretation of using direct inversion around a single pole/zero

Let $F[z]=N^2\frac{z(z-(3+j\sqrt{7})/2)(z-(3-j\sqrt{7})/2)}{(z-1)^3}$ This has 3 poles at $z=1$; one zero at $z=0$; and a conjugate pair of zeros at $z=\frac{3\pm{}j\sqrt{7}}{2}$ Assuming a contour ...
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39 views

Conditions on $v \in \mathbb{C}^{n}$ such that the complex to real DFT is sensible

I'm using FFTW to compute some DFTs, and as a memory space optimization, you can give it a $v \in \mathbb{R}^{n}$ and it will return a $\hat{v} \in \mathbb{C}^{\lfloor n/2 \rfloor + 1}$ via the call ...
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2answers
94 views

Which filter for an audio equalizer

I’m new to DSP and I have to do an audio equalizer in c++. I did a lot of research about it and tried some stuff in the last month, but I’m a little overwhelmed with all those informations and it’s ...
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3answers
70 views

Pre-processing EEG signals

Good Afternoon Everybody. I am developing filters for EEG signals in these days. During the Develop I find some problems: The filter is FIR (band-pass using a convolution of low-pass filter and ...
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19 views

what is flicker remove/reduction technic?

I'm looking a method about flicker remove or reduction. In image processing, what is the flicker remove/reduction technic for ISP engineers? Typically, 50,60Hz's camera data input has flicker. So I ...
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1answer
29 views

Smooth (not) equally-spaced signal strength data

I'm currently trying to reduce the noise in/smooth sampled signal strength data. The context is that I collected data from a transmitting device using bluetooth - the available data is basically a ...
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1answer
59 views

How to get main frequency from digital signal

I have sampled digital signal with noise. I want to get main frequency of this signal how to do it? Currently it is smoothed by moving average. Is good way to make Fourier Transform an take the ...
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1answer
32 views

Help with Hankel Tranform filters

I found a filter for Hankel Transform . As far as i understand this is effectively used as an approximation for J0(x) and J1(x) . But how will the change in input affect these filter terms (eg J0(r) ...
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1answer
60 views

Why are ideal filters not particularly useful/ hard to do?

Why is a LPF ideal filter: a) Not useful? b) Hard to make? I was asked to make a Low pass filter (simulink block) that will take an input of a cutoff frequency, and for a signal with a frequency ...
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1answer
29 views

Transfer function, amplitude response and difference equation for a filter

I've found a paper with a filter described in terms of transfer function, amplitude response and difference equation: transfer function of the second-order low-pass filter: $$ H(z) = ...
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1answer
50 views

how to extract meaning from signal (distance estimation)?

I have 3 times series representing detected rssi signal power from 3 emitting devices. The devices are at 1 meter distance from the receptor, plotting the 3 time-series gives the following results : ...
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67 views

Zero-frequency response of an IIR transfer function

I have a question about the zero-frequency (DC) response of a IIR filter. The coefficients of the 3rd order IIR filter are as follows, ...
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2answers
30 views

Is the common FIR Filter delay line (see pic) also valid on purely logical FPGA implementations?

Lets say we want implement simply the mean with constant coefficients. So all b's are set to e.g. b=0.5. In this case I do not understand the delay line in the picture below. I mean I understand it ...
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4answers
124 views

What does it mean to say that “sinc” filters are ideal?

I suggested to someone that one can create a sinc filter that is ideal enough to be indistinguishable from the ideal, given certain limitations with the data in the first place. Is that true? It was ...
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1answer
42 views

Kalman filter - understanding the noise covariance matrix

What is the significance of the noise covariance matrices in the Kalman Filter framework? I am referring to: process noise covariance matrix Q, and measurement noise covariance matrix R at any ...
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40 views

Raised-Cosine filter - design

I have a requirement to design bandpass Raised-Cosine filter with the magnitude and phase response as outlined in the picture below. I have read some literature online (still learning DSP) and I have ...
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2answers
50 views

Help for baseline drift removal/correction

I have signal as seen in the picture. It has high frequency noise and drifting baseline. I am able to remove the noise in MATLAB but I am not sure what can I do about the drift. I would like the ...
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60 views

Spectral Whitening: Approches

Given the absence of information about the algorithms of spectral whitening functions in software packages: How many ways do you know for doing spectral whitening? What are the differences?
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Biquad high-pass filter - calculating filter slope

I have to design high-pass IIR Biquad filter with cutoff frequency of 80Hz, sampling frequency 24kHz, and filter slope of 24dB/oct @ 80Hz. I have used formula for HPF biquad coefficients from here. ...
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Applying a FIR filter to floating point array c# [closed]

I have designed using MATLAB a FIR filter with some coefficients. I have an array that is normalized to -+1. The array is float. I used this method to apply the FIR to the array but I'm not sure if ...
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30 views

Effect of Pulse shaping on BER

am trying to implement OQPSK modulation with half sine pulse shaping. If I simulate the OQPSK modulation without pulse shaping, the simulated curve is similar to the theoretical value. However, when I ...
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1answer
55 views

Sampling frequency and oversampling factor

A quite naïve question. I need some clarification regarding choosing the sampling frequency and oversampling factor. The scenario is as follows: I have an OQPSK modulated sequence with symbol rate 2 ...
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21 views

Can't make a Wiener filter work

I have two wav signals, one desired and the other one an unwanted signal perturbing the first one. I'm trying to implement a Wiener filter to reduce the second one, but when using the sound function I ...
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1answer
43 views

I'm looking for the concept about estimated standard deviation

I am looking for the concept about estimate standard deviation. Actually I'm not sure how can I get a concept the estimate standard deviation ? If you know the concept, then would you let me know ...
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57 views

FIR Filters: determining optimal number of taps and phase slope

As a design input for an low-pass FIR filter I have 6dB@1.6kHz, 24 dB/oct, sample frequency 48kHz. I need to specify phase response, and for that I need a phase slope. I have found, from Julius O. ...
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How modulation spectrum of zebra finch song is calculated

I read this advanced signal processing technique in Neuroscience Journel. Modulation spectrum is a spectral modulation vs temporal modulation plot. Please tell procedure of plotting it step by step. ...
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2answers
50 views

Find Stopband Edge Frequency

I tried to find the stopband edge frequency.I know that the signal amplitude falls to -3 dB below the passband edge frequency.I examine a lot of DSP books so i don't any information about how to ...
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65 views

MATLAB, recover a signal that has a cosinusoidal interference and that has been AM modulated

I have a signal which has been AM modulated but the channel also added a cosinusoidal interference A cos(2 * Pi * f0 * n * Tc + Phase), at the same ...