Tagged Questions

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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32 views

What are the frequency components of a sampled signal after low-pass filtering?

A continuous signal $x_a(t)$ is a linear combination of sinusoids of frequency 250 Hz, 450 Hz, 1 kHz, 2.75 kHz and 4.05 kHz. The signal $x_a(t)$ is sampled at $f_s=$1.5 kHz, and the resulting digital ...
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0answers
65 views

DSP interview question: use of the identity in development of a significant transform

I'm preparing interview and found this question. But I don't really understand what is the question. Does it ask about Fourier transform or Z transform? How the simple identity $$xy=\frac{1}{2}x^2 ...
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29 views

MATLAB implementation Spline Fitting

Check the attached problem please. I am a beginner in spline fitting and have a few questions: 1) How to find the coefficients $c[n]$. Is it by DTFT? 2) I understand how to find the derivative but ...
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1answer
29 views

Perfect reconstruction ( Single Channel )

I was trying to solve the following problem . Given an input x[n], consider upsampling by 2, followed by interpolation with a filter having z-transform H(z) for magnification of the signal. ...
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0answers
27 views

Filter design using inverse dft

How do I get the cut-off frequency? A filter is needed to recover $x_2(n)$ from the signal: $$x(n) = x_1(n) + x_2(n) + x_3(n)$$ where: $x_1(n) = \sin(0.5n)$ $x_2(n) = \sin(1.4n)$ ...
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1answer
25 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 ...
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1answer
76 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} ...
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44 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 ...
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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 ...
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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 ...
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0answers
17 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? ...
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0answers
33 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 ...
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1answer
72 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 ...
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2answers
58 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?
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1answer
27 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 ...
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2answers
35 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) ...
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2answers
45 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 ...
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2answers
41 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 ...
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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 ...
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0answers
44 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 ...
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0answers
23 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?
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24 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 ...
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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 ...
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1answer
14 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 ...
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1answer
63 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 ...
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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 ...
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2answers
55 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 ...
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1answer
93 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 ...
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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 ...
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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 ...
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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 ...
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1answer
51 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 ...
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3answers
61 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 ...
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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 ...
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0answers
36 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 ...
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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 ...
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0answers
26 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
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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: ...
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1answer
46 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 ...
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4answers
92 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 ...
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3answers
61 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
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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
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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
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0answers
51 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 ...
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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 ...
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1answer
193 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 ...
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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 ...
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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 ...
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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 ...
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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 ...