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.

learn more… | top users | synonyms (1)

0
votes
1answer
16 views

What is the Roll-Off factor of a Root Raised Cosine Filter?

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 ...
1
vote
1answer
65 views

How to extract IF (instantaneous frequency) from Discrete Hilbert Transform?

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} ...
0
votes
0answers
34 views

Transient cancellation in non causal filtering

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 ...
0
votes
0answers
26 views

moving average filter vs up sampled low pass filter

I'm experimenting on different low pass filters in matlab. I've considered a vector x=[1 0 1 0] ; an up sampled low pass filter by a factor of two. And the frequency response turned out to be And in ...
0
votes
0answers
24 views

How to remove modulation frequency and its harmonics

I have amplitude modulated sound file signal and after demodulation using FFT I got fundamental frequency and its harmonics and of course noise. How to remove that fundamental modulation frequency ...
0
votes
0answers
15 views

Does a Laplacian Pyramid and a Gaussian Pyramid use the same Kernel?

Does a Laplacian Pyramid and a Gaussian Pyramid use the same Kernel? I have to program this for an assignment, but it seems like the kernel used in both cases is a gaussian filter. Is that the case? ...
1
vote
0answers
32 views

Direct Form I Filter from Direct Form II

I am looking a way to make outputs match. For len 160 wouldnt the error be significant? For 40 ? I have included some sample output showing the error. Direct Form 2 Original ...
-2
votes
0answers
26 views

Enhancement Of Ecg Signal [on hold]

Load, display and manipulating of ECG signals Design notch filters both in FIR and IIR for 50 Hz and 100 Hz respectively. Design a band pass filter to eliminate muscle noise. Plot the spectrum of ...
0
votes
1answer
70 views

Strange Results from Biquad Peaking EQ Filter

I have a signal that I am filtering on an MCU (STM32F405) that I get from a codec (CS42L52). It is being sampled at a sampling frequency of 48kHz (actually between 48400 and 48600). I have a ping ...
0
votes
2answers
55 views

FIR vs IIR filter

In my filter design, FIR gives order 14 and IIR gives order 8 or 4. so what are the disadvantages of IIR design?
0
votes
1answer
26 views

Why bother impluse invariance, bilinear transform?

Now that we can operate in the frequency domain by way of FFT. We can implement ideal lowpass, highpass and bandpass digital filters. then why there many textbooks talk about designing digital ...
0
votes
2answers
34 views

multiplications per second for FIR filter

In my filter the fs=300MHz. and no. of coefficients is 31. I have following queries. (1) what will be the number of multiplications per second? (2) can multiplication be done in one clock cycle? (3) ...
0
votes
1answer
35 views

A filter in IOS to eliminate a certain frequency range

I am making an application where I record sound on an IOS device using the built-in microphone. I want to process the sound in such a way that only frequencies in the 16-18 kHz range remain; the other ...
0
votes
0answers
12 views

Reference request in Circuit and Signals for Computational Neuroscience [on hold]

In the area of computational neuroscience, there are basic theories from electric circuits and signal processing to be applied. For background study, which reference will be more suitable ? ...
1
vote
2answers
34 views

Signal corrupted using peak (biquad) filter at low frequencies

I'm trying to process an audio stream through a series of peak filters at static frequencies that range from 0 to nyquist. If I start the range from anything below about 25Hz, the audio gets corrupted ...
0
votes
1answer
52 views

FIR filter design

I am designing an FIR filter.my specs are fs=300MHz, Fc=45MHz, Fs=75MHz, passband gain=3dB , stopband attn=>40dB. what parameters I values of a have to provide in firpm for the minimum order filter ...
0
votes
0answers
43 views

DFE for QPSK in matlab

I need to do a DFE for QPSK simulation in matlab. The system, consist in QPSK signal transmitted with power = 1. Then pulse shapping with a square raised-cosine filter. Add AWGN and a filter receptor ...
0
votes
0answers
21 views

PLL as bandpass filter?

I'm working on writing the synchronization for MSK and I need to create two PLL's as band pass filters. How would I design the filter part of the PLL to make the system act as a band pass filter?
0
votes
0answers
22 views

How to smooth out a noisy signal?

I've been waiting for an answer to my question here therfore I decided to ask the question in a different way. Could anyone please help me to smooth out the signal below!!. I've no experience of ...
0
votes
0answers
26 views

Unstable Cascade of Allpass Filters

I'm working with cascaded second-order allpass filters. Each filter is characterized by conjugate pole/zero pairs. In a long (>50) cascade of identical filters, the pole locations of the overall ...
0
votes
1answer
13 views

Iterative Kalman filters and system parameters estimation

i am working recently on a project in which i want to implement a Kalman filter as being an observer, and i couple this observer with a state feedback controller that produces control actions ...
1
vote
1answer
62 views

Discrete algorithm for low pass filter

I am working on a position controller for a marine vessel. I have an measurement signal containing the y-position of the vessel that consists of both low frequency (<.1 rad/s) and high frequency ...
-1
votes
0answers
26 views

Best way to clean up spectra [on hold]

I have a spectrum made up of several peaks. I would like to keep these peaks but zero out the noise in between so I can then apply the inverse foruier transform to my data. Which is the best way to do ...
-1
votes
1answer
39 views

Can all pass filers have a pole/zero at the origin of the root locus

I came across this root locus diagram in my DSP class. It has a pole at the origin, another pole within the unit circle and a zero just outside the unit circle. I understand that the poles and zeros ...
2
votes
2answers
53 views

Discrete approximation to Gaussian filter

One method of producing a discrete approximation kernel to a Gaussian filter of variance $\sigma^2$ is to assume a cutoff of $5\sigma$. This seems to suggest the following procedure: Create a ...
0
votes
1answer
83 views

Picking out a signal that appears to be noise inside a large signal

I have a signal from a photodiode sensor that has two types of noise. One type of noise is ambient light white noise that gets introduced just from the surrounding environemnt. The other type of noise ...
0
votes
0answers
24 views

Gabor filter bank spatial frequency parameters

I've been trying to learn about using Gabor filters to extract features from an image. There are some research papers that list their spatial frequency parameter in the form of x:y cycle per pixels in ...
0
votes
3answers
66 views

Appying a Filter Several Times on Data

I have data from an experiment and apply a lowpass filter (digital Butterworth filter, in Matlab). My question ist what happens if I apply the same filter e. g. 2 times on the data? What is ...
1
vote
2answers
96 views

Filtering data so that only rising edge is left

I have data that looks like this: Sometimes the data has a higher point in the middle of the shallow slope I want to find a way to filter the data such that it smooths it and leaves the first ...
0
votes
1answer
50 views

Largest possible downsampling factor

With the intention of converting a single rate LPF FIR filter to multi rate equivalent: what is the largest possible downsampling factor? I found one source that said it was the sampling frequency ...
-1
votes
3answers
57 views

FIR / IIR and Wavelets

I am a self-learner of DSP by reading various books. I have accomplished basic understanding of Signals - CT and DT and a few transforms. I recently started to learn FIR / IIR Filters. The part that ...
1
vote
0answers
21 views

Variation in Signal Power in Mimo system

I am exploring methods for reducing computations required by MUSIC (Multiple Signal Classification) Algorithm. Most of the computational power is spent on calculating SVD. My approach is to take ...
1
vote
0answers
32 views

GCC-PHAT cross correlation

I am trying to implement GCC-PHAT in python. The approach is similar to the following two links: link1 and link2 It seems the only difference between GCC-PHAT and normal cross-correlation using FFT ...
0
votes
0answers
40 views

Filtering and decimating input as sections in multiple stages

What would be my options to decimate a large input signal available only in sections (or blocks) due to DMA type of interface? The FIR should be about order of 18 only and the total decimation factor ...
0
votes
0answers
25 views

How to handle encoder wraparound in an alpha-beta filter

I want to use an alpha beta filter to estimate speed from an encoder that's mounted on a motor shaft. The encoder position wraps around to zero every 8192 counts; it gives the absolute angle of the ...
2
votes
2answers
76 views

Reconstruct FIR frequency response from the coefficients

There is a lot of tools for designing FIR filters. Using one such tool I have designed a filter that satisfies my needs. It's coefficients are as follows: ...
1
vote
1answer
44 views

correct for shift from butterworth filter?

I have implemented a python butterworth filter but it seems to shift the data a lot, the x location of my data is very important to me for my analysis. Is there a way to correct for this shift ...
1
vote
4answers
91 views

The reasons for filter explosion

Assume signal $x$, sampled at $f_s = 44100\; \mathrm{Hz}$. I tried to filter it using the Butterworth bandpass filter ( $30\; \mathrm{Hz} - 70\; \mathrm{Hz}$) of order $8$. However, as a result I get ...
0
votes
3answers
58 views

comb filter - attenuation

I have a signal and from 'standard information' I know the predominant frequency is at 1Hz and then there should be three harmonics at lower amplitudes. My signal however has a lower amplitude at 1Hz, ...
1
vote
2answers
33 views

variable length no-refrence filter algorithm

I'm searching for a an algorithm or an idea of implementation. I need to filter a signal in real time, and want to change the length of it (number of taps) in order to change the delay. In low ...
0
votes
1answer
50 views

Problem with a code of FIR Lowpass filter

I'm making an interpolator (filter) to remove aliasing but this doesn't work. I see also some strange behaviour: x[i - j], but what if ...
2
votes
0answers
49 views

Designing a practical integrator

One way to describe a practical integrator ("leaky integrator") is $$ H(s) = \frac{g R}{1 + sRC} $$ I am trying to understand how to choose the values $g$, $R$ and $C$ because in practice, I will ...
0
votes
0answers
23 views

What filter should I pass to these signals to make them constant on every piece

I'm analysing the frequencies of something I'm recording (whistling), which is resulting like this (yes, it's super mario bros' beginning song: I'd like to filter it to make it remain like: in ...
0
votes
1answer
183 views

Real and Imaginary Frequency Responses of a single complex pole

I'm filtering a real signal with a single complex pole with a complex coefficient (a_re [real part] and a_im [imaginary part]), I also have a gain coefficient but I'm gonna leave it out for the sake ...
1
vote
1answer
61 views

How to reconstruct the signal which is subjected to FIR filtering - Matlab

I have a signal $x$ which is corrupted by Gaussian noise. The signal $x$ is convolved with a filter with coefficients $h = [1, -2, 1]$, to generate signal $y$. Now the signal is taken to Discrete ...
0
votes
1answer
46 views

Difference Equation - Time Response analysis of Discrete Systems

We know that that the difference equation formula for computing an output sample at time $ n$ is based upon its past/present input samples and past output samples in time domain: $\displaystyle ...
1
vote
3answers
72 views

Are reconstruction filter always needed?

in mixed (digital and analogue application) where you have to convert a signal from continuous to digital time domain and then back again to continuous time an anti-aliasing LPF is needed and in ...
0
votes
1answer
42 views

How to calculate the time delay of a signal by a first-order filter

lets assume following signal $$x=a_1 \sin(2\pi f_1 t) + a_2 \sin(2\pi f_2 t).$$ This is passed through a simple single-pole filter with cut-off frequency $f_c$. How is $x$ delayed by the filter (peak ...
0
votes
0answers
12 views

Gaussian kernel standard deviation and spacing

Suppose I know the normalized gaussian weights for a kernel of standard deviation 1.0 evaluated in the sample positions $O = \{-2, -1, 0, 1, 2\}$. Let's call these G. I know that the same values G ...
0
votes
1answer
49 views

Advice on filtering a CW signal for processing

This question is related to my other recent questions. In the last question, I learned that I should filter about 4 * WPM (words per minute) around the carrier frequency if I want to recover Morse ...