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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Find impulse response and frequency response through given filter

I cannot find anything truly helpful in my book, and I find mostly theoretical answers and not practical. So I am posting the exact exercise I want to solve, and through it I hope I'll work some ...
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38 views

Combine multiple FFT

I have been acquiring data through ADC and into memory, the input signal are square waves that step down in frequency as time goes by. Due to memory, I have a fixed maximum number of samples per FFT. ...
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60 views

Question regarding filter implementation - Audio EQ Cookbook

Some of you may know the infamous Audio EQ Cookbook. You can take a look here. We tried to implement the functions there to achieve a implementation for a Lowpassfilter. However, when we try to ...
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2answers
38 views

Is DCT basicly a filter?

I am reading about audio processing for my university project and I found out that DCT basicly filters out some frequencies. I wanted to use for example high-pass filter and then DCT for compression, ...
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1answer
25 views

Concept of combining multiple FIR Filters into 1 FIR filter

I try to combine my 7 band FIR filters into 1 FIR in order to process an audio signal. I use convolution to combine their coefficients but the effective is not the same as I apply 7 different FIR ...
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1answer
76 views

Duration of unknown rectangular pulse with additive white Gaussian noise

Problem. There is a discrete signal $f[i]$ (example below). It is known, that $f[i]$ have a form of rectangular pulse with additive white Gaussian noise. $f[i] = s[i] + n[i]$, $s[i] = ...
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21 views

Noise Removal from an image

What type of filters can be used to remove Poisson noise in an image? I tried applying Gaussian filter to few points in the spectrum.
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1answer
35 views

All pass systems- Poles and Zeros

If an All Pass system has poles at $0.8\exp\left(-j \dfrac{\pi}{4}\right)$ and $0.8\exp\left(j \dfrac{\pi}{4}\right)$, There would be $2\pi$ jumps in the phase (principal value) at angular frequency ...
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30 views

Applying a custom filter to FFT results?

I think I am missing something here and I do not quite understand why. Background: I am collecting quadrature signals from a software defined radio. There are two signals, the I, and the Q which is ...
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3answers
52 views

In filter design, why isn't it possible to have a frequency response with phase 0?

A linear phase filter will cause a delay in the output signal of the filter. If I'm not mistaken, a filter with phase equal 0 would cause no delays in the output signal. Can someone explain me why ...
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38 views

How to deal with the implications of sample Size for the FFT in Low Frequencies

I'm trying to detertmine presence of low frequencies, but I'm working from the tone 40 (82,4 Hz) until tone 84 (1046,5 Hz). With difference between one tone and its adjacent of Pow(2,1/12). Then ...
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1answer
53 views

Allpass filter feedforward/feedback design and code

I want to design/implement a simple feedforward/feedback all-pass filter, and I am having some troubles and questions with it. From Shroeder design (based on J. Smith article here): I came up ...
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40 views

pre-emphasis filter - parameter a

Can someone give my a link to study to solve this problem. I can not find any formula for this problem... Suppose we filter singals with the below pre-emphasis filter: $y(n)=x(n) - ax(n-1)$ How can ...
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1answer
28 views

Fast/efficient way to compute Laplacian edge enhancement filter

I would like to implement a somewhat smarter Laplacian edge enhancement convolution. Right now it is implemented as (generic 3x3 convolution): ...
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17 views

Why Matlab fdatool designed butterworth filter and butter command's co-efficients are different

I met an great confusion. Using below command I have got co-efficents for a 300Hx cut-off high pass filter of order two. [z p k] = butter(2,(300/44100),'high') [sos,g] = zp2sos(z,p,k) Now ...
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1answer
30 views

Filtering Passband Signals using Complex Baseband Filtering

So I'm given a pass band filter with specific transfer function $H_p(f)$, I want to implement this via baseband processing. I already know how to take the input signal $u(t)$ and process it such that ...
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1answer
28 views

Linear Regression Filter Properties

I have a filter to detect when a slow moving signal (oven temperature) reaches steady state. I am using a linear regression of a moving window and looking at the slope. Based on the equations for ...
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1answer
19 views

Problems filtering image with prewitt

I'm trying to filter an image vertically to get the edges, in MATLAB, but I get very different results from convultion and correlation. CODE: ...
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1answer
20 views

Estimating Velocity from Fuzzy Position Data with Known Uncertainty

I was wondering if someone could maybe clarify or direct me to the best answer for this question. I want to estimate velocities from position data. However, the position data is fuzzy but I have an ...
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60 views

bandpass FIR filter

I need to make simple bandpass audio filter. Now I use this simple C++ class: http://www.cardinalpeak.com/blog/a-c-class-to-implement-low-pass-high-pass-and-band-pass-filters It works good and cut ...
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25 views

Filter sharpening; a stub

Filter sharpening is a technique to sharpen the magnitude response of a digital filter. Finding information online about it is problematic since almost all google hits will be about image sharpening ...
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1answer
49 views

Is this how the lowpass butterworth analog filter can be implemented?

Is this how the lowpass butterworth analog filter can be implemented?
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13 views

Higher dimensional Kaiser formula

Kaiser formula gives filter length for FIR filters. Is there a generalization of the formula to $2D, 3D, 4D, 5D,\dots$ FIR Filters?
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1answer
64 views

How can we calculate the frequency response of each filter of a Discrete Wavelet Decomposition filter bank?

I am using the Multilevel Wavelet decomposition. The decomposition results in a filter bank such as the following: Specifically, I am using one of the Daubechies wavelets, ...
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1answer
45 views

applying convolution theorem swaps quadrants

For educational purposes I implemented the DFT and inverse DFT for images using OpenCV. Applying the DFT to an image and taking the inverse DFT of the spectrum yields the original image, so this ...
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2answers
50 views

FIR Filter length Upper Bound

Say I want to design a low pass filter with stop and pass band ripple constrained to $\epsilon$ and transition region $\delta$, what is the smallest FIR Filter length that suffices? Can you please ...
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1answer
68 views

Beat Detection with Arduino

I'm trying to make a robot dance on several songs using Arduino. I need to understand how to get the rhythm from the songs, something like detecting the beats, calculating the bpm.. I'm not expert but ...
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59 views

How to determine cut-off frequency using power spectral density?

I am new to DSP and my question is a very simple one. It is about determining the right cut-off frequency. I calculate the power spectral density using FFT by Welch's algorithm. However I am confused ...
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44 views

BandPass and StopBand FIR filters (Hamming) is not working

I have two groups of 3 frequencies of tone (music), First Group Tones: 39 (77.7817 Hz), 40 (82.4069 Hz) and 41 (87.3070 Hz). Second Group ...
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1answer
51 views

Why oversample data before using raised cosine filter?

In the process of creating a baseband signal, the complex symbols are filtered using a raised cosine filter to perform pulse shaping (I'm following some literature as a guide). The pulse shaping is ...
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27 views

Choosing the frequency for a comb filter

I have an audio sample. From doing FFT on it it looks as if I need to use a comb filter to remove resonant frequencies. This is what I generate: I honestly have no clue where to start ...
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1answer
46 views

Frequency of a noisy signal with fixed width pulses

My electricity meter has a front panel LED which pulses in proportional to the current energy rate. The frequency changes but the pulse width is set at 250ms. Due to environmental conditions there's ...
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1answer
28 views

coded filter vs fdatool : inconsistency in the results for same parameters

I tried creating a filter in 2 different ways in Matlab and I obtain different results: one filter is stable, whereas the other one is unstable. Coded filter ...
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1answer
64 views

Filters performance analysis

I am working on some experimental data which, at some point, need to be time-integrated and then high-pass filtered (to remove low frequency disturbancies introduced by integration and unwanted DC ...
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2answers
77 views

Questions about Active Noise Cancellation

Being new to ANC, this is my understanding of how ANC works: Ambient noise (n) is entering a geographically small zone. To suppress the noise, an interfering (corrective) sound is injected ...
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21 views

How to plot/graph frequency response of a continuous time comb filter?

How can I plot the frequency response of a comb filter, and how can I plot it so every octave from nyquist (normalized at x=0) is 1 unit on the graph? Sorry for the poorly phrased question, I am not ...
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3answers
64 views

Filter order and cut off frequency

How can I know exactly what frequencies are not influenced by the filter? I mean, I have a signal in the range of frequencies $5-100 Hz$ and I apply a high-pass, $4^{th}$ order Butterworth filter ...
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24 views

Convolution / Correlation correct implementation in java

I have a java programm which is manipulating an RGB picture which comes in a 3dimensional matrix. I tried to implement an apply Convolution and an apply Correlation function, which applies a kernel ...
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21 views

Implementing a Sum of Sincs filter in Matlab

I am trying to implement a filter that looks like: $$G(\omega)=\dfrac{\tau}{\sqrt{2\pi}}\sum_{k\in\mathcal{K}}b_k\mathrm{sinc}\left(\dfrac{\omega}{\frac{2\pi}{\tau}}-k \right)$$ in Matlab. I have: ...
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2answers
44 views

Guitar Effects/Filters in real time using low cost DSP?

I have an c5502 texas low cost DSP. I can capture sound trough its P2 input and process every separated sample after sending it to the P2 output. Im trying to implement some kind of effect in the ...
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72 views

Removing 'high-frequency' components from an array?

I am new to the stream of digital signal processing, and hence my doubt may seem extremely stupid, but: In tools such as matlab, signals are stored in a single array. If i convolute it with an LPF ...
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49 views

Filtering of accelerometers data for double integration

This question in strictly related to this one http://scicomp.stackexchange.com/questions/16057/numerical-integration-and-filtering-of-acceleration-experimental-data but in this case I would like to ...
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2answers
63 views

Why is this MATLAB butterworth filter giving me wrong results

Given a 128 Hz signal, in MATLAB I entered the following command ...
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1answer
34 views

How to find roll-off from its transfer function equation?

I want to find roll-off (unit dB/decade) of highpass filter $H(z) = 1 - a \cdot z^{-1}$. I derived magnitude response to be $1+a^2 - 2a\cdot \cos(\omega)$ To find out roll-off factor from above ...
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Mel vs. Gammatone filterbank

After some research about the problem of segmenting voice from unvoiced sounds I found that most of the papers deal with an operation to simulate the cochlea filtering of the signal. I found two ...
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57 views

How to Estimate Attenuation Rate ($\alpha$) of band limited filter impulse response

For 2nd order systems, we could find Impulse Response Attenuation (Damping) Rate ($\alpha$) from Q-factor equations, combining $Q = \frac{f_0}{\Delta f}$ and $Q = \frac{2\pi f_0}{2\alpha}$ for 2nd ...
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1answer
125 views

Design a Filter with Resonance

I want to design a Filter with a variable Cutoff Frequency and Resonance. However in the source I'm using there is a Bi-Quad Filter with two IIR and two FIR Filters designed like this : ...
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1answer
207 views

Is it necessary to downsample data for filtering of low frequencies?

My data is sampled at $f_s = 32kHz$, and for a particular analysis I'm only interested in signals with certain frequencies from $f_{highpass} = \text{0.1Hz}$ to $f_{lowpass} = \text{300 Hz}$. When I ...
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1answer
97 views

Question on Chebyshev filter

What does the output signal look like when you insert $2000 Hz$ square pulses to chebyshev filter with parameters: (Matlab syntax). I tried to do it like this but I don't understand where it goes ...
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48 views

Generate band limited white noise to be used as excitation in ODE

I am trying to repeat an example I found in a paper. I have to solve this ODE: $25 \ddot{x} + 15 \dot{x} + 330000 x = p(t)$ where $p(t)$ is a white noise sequence band-limited into the 10-25 Hz ...