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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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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23 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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32 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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23 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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32 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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21 views

FIR vs. IIR for filter realization [on hold]

What are the pros and cons of using FIR versus IIR structures for the realization of a digital filter ?
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8 views

SRAD code in matlab it doesn't work [on hold]

i just applied the SRAD code in matlab and i get error in line 430 ...
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21 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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38 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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18 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
57 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
68 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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16 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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53 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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18 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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20 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
34 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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2answers
62 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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23 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
56 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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30 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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54 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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how to code fir filter in matlab using discrete equation?

I want to design a fir filter in matlab using the discrete equation. I won't be using any built in functions. For doing so what will be the coefficient? How can i compute it?
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85 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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167 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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88 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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35 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 ...
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22 views

Continuous wavelet function spanning piecewise functions

Is there a continuous wavelet function that spans the space of piecewise: linear functions (as Haar wavelet spans the space of piecewise constant functions). polynomial functions?
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62 views

Modulation and DeModulation

I have a question regarding Modulation and Demodulation. In question 5.21 (b) from the "Discrete-time Signal Processing" book by Oppennheim, he asks to find out the type of filter used in the attached ...
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33 views

Basic Fourier Transform filtering and LTI frequency response using Matlab

Not sure if this is the right place to ask... I'm new to Matlab for LTI signal processing and wondering if anyone can help with something that I'm sure is meant to be basic. I've spent hours and hours ...
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12 views

Butterworth filter for quadricorrelator

I am trying to implement a FM demodulator (quatricorrelator) according to this paper (3.1 & 3.2). I am now struggling to figure out the proper values for my filters, as described by the paper: ...
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1answer
29 views

Splitting a Complex FIR Filter into Real Filters

I'm working on a software defined radio project on a PIC32 MCU. I'm using a CMX 973 quadrature frontend to provide I and Q. I'm independently low passing I and Q with carefully matched analog filters ...
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42 views

Is it possible from 1 order cascading low pass filters to make a butterworth filter

Today I have heard, that is "impossible" to make low pass butterworth filter from cascading 1 order low pass filters(RC). I am trying to discover is that true, but I can find little to none ...
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Applying filter in scipy.signal: Use lfilter or filtfilt?

I saw in a SO thread a suggestion to use filtfilt which performs backwards/forwards filtering instead of lfilter. What is the ...
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41 views

Filter for audio sampler

An audio sampler (in "music" terms because I'm not experienced in dsp) is a software which loads an audio file and allows you to play it at different pitches by slowing it down and speeding it up. ...
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1answer
30 views

Orthornormal Wavelet scaling function coefficient calculation (filter design)

I stumbled upon an equation system as shown in the image below while designing an orthonormal wavelet filter. I tried solving the equation (5.63,5.64,5.65 => 5.66) but no avail. I'm not sure if ...
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44 views

3 Band Crossover with Linkwitz Riley Filter

I would like to write a 3 Band "EQ" with BiQuad Filters but for some reason, I get some phase distortion. The problem already got discussed here. I found out that I need to work with Allpass filters ...
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41 views

How do I take the real part of this bandpass filter system's output?

I'm stuck on a final step in this problem. Essentially, there are the two systems above, which we'll call System 1 (Fig. 4.26, with ideal lowpass $H(jw)$) and System 2 (with $H_1(jw)$). The ...
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2answers
70 views

Butterworth filter in python

I'm trying to use a Butterworth filter in Python as described in this thread with these functions: ...
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34 views

How to perform Wiener deconvolution in Matlab

I have an output signal y which is an input signal x convolved * with an impulse response ...
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60 views

FIR vs IIR complexity

IIR filters have feedback whereas FIR filters don't use feedback. Then why is it said that FIR filters sometimes have the disadvantage that they require more memory and/or calculation to achieve a ...
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31 views

How can you intutively figure what the impulse response will behave like?

I was trying to understand convolution from this book and came across this figure: Here, the input signal is a combination of a sine wave and a ramp. In fig a, the input signal is convolved with a ...
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40 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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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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54 views

MATLAB implementation Spline Fitting

Let $s(t)$ be a signal that can be approximated by a uniform spline function of order $K$ (say $K=2$): $$s(t)\approx\sum_{n\in \mathbb{Z}}c_n\beta_+^{(K)}(t-n) $$ Suppose that we know the ...
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1answer
40 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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107 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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88 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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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 ...