# 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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### Basic FIR Filtering with audio buffer in C++

I am willing to implement a basic FIR filter using C++ and JUCE. I wrote this simple algorithm using the information I found in the DSPGuide, but I only seem to get a variation in volume for all the ...
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### Response of cascading FIR filters

I can't seem to get the right answer for this. Could someone show me the procedure? As shown in the figure a system is constructed by cascading four FIR filters having the following frequency ...
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### How to plot magnitude and phase response by hand if I have the Transfer Function?

I have the transfer function of the system, which is: $$H(z) = \frac{1-z^{-1}}{5(1+2z^{-1})}$$ How do I sketch the magnitude and phase response? I'm sorry for the bad formatting, it's my first time ...
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### Bandpass Filter Order vs Sample Numbers of Noisy Signal

I'm working on signal filtering in MATLAB. i wrote sample code that generate 50 Hz signal with some noisy frequency and use Kaiser bandpass filter to get only 50Hz frequency. this code work great! but ...
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### How are these pole zero plots created

Back around 5 BC, where C stands for the programming language, engineers would photograph their oscilloscope's display to record their test results. This appears to be such a photograph, taken from ...
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### Calculate Q factor of a Low Shelf and High Shelf filter

I'm developing an app that has an EQ using eq10q plugin. My eq has 10 bands. I'm using peaking filters for the middle bands and a low shelf filter for the first band and a high shelf filter for the ...
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### Plotting band-pass filtered signal of an audio in MATLAB [closed]

I am new in MATLAB and try to plot some form of an audio file and I have managed to open and plot in time domain and FFT using the following code: Step I: Opening the Wave File ...
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### How to set a Carrier Wave for DBPSK Modulators in gnuradio?

for a Project I need to build an Audio Modem in GNU Radio. I decided, that it would be the best to use DBPSK Modulation, because that is fast and relatively noise resistant. But as it seems, the ...
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### Should the order of a filter be proportional to the sampling rate?

I have designed a fir filter with my desired frequency response at a specific sampling rate and filter order. If I were to redesign my filter for a new higher sampling rate, should I change the filter ...
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### What dictates the phase response of FIR filters when considering its zeros [closed]

I assumed an even tap filter and i got a pole at $z = -1$ and an odd tapped filter (assuming the middle coefficient is zero) gave me poles at $z = -1$ and $z = +1$. This was confirmed by the theory ...
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### Can we determine if a filter is butterworth or chebyshev from its physical topology?

Can all analog filters be classified as Bessel, Elliptic, Butterworth or Chebyshev? Given a physical ladder topology of several stages of {L, C or LC} in {series or parallel}, is it possible to ...
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### How can a signal have two maximum frequency components?

$x(t)$ can be exactly reconstructed from its samples at $\omega_s = 10 \textrm{ rad/sec}$. My conclusion is that the maximum frequency component in $x(t)$ is $5\textrm{ rad/sec}$. But I'm being told ...
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### EQ filter implementation

I am writing a simple EQ filter for my audio player. I got one working with Biquads (single stage 2nd order or double stage 4th order). The app is written with Qt so I also checked QMMP (Qt based) ...
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### Are wavelets for audio spectrograms available as a simple FIR filter?

I have read that "Debauchies Wavelet is simply a short FIR" so i am searching for a single FIR that i can make into a filterbank, a precise "bandpass filter" that detects narrow frequencies, i.e. 1Hz ...
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I would like to try optimize a bit more the biquad filters indended for RIAA/non-RIAA equalization at low samplerates (44.1/48kHz). Best fit I get now is little below ±0.3dB for 44.1kHz and below ...
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### What filter to use in audio analysis filterbank instead of FFT?

Standard bandpass filters can make super precise analysis filterbanks with 1024 to 4096 filters, on reaktor4. I tried in code to used cookbook BandPass and the result was aweful. Does someone know a ...
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### How can I shift this signal to it's midpoint in order to decode it

I'm wondering if there's a simple way to decode this pulse signal as I'm reading it in. I'm thinking the simplest way to decode this is if i can shift the whole signal down so that it's centered ...
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### Real digital filter property

I am a beginner to study about the filter notion and property Being a real digital filter, (here "real filter" I means that its impulse response is real-valued) this formula is established. But I ...
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### How to design very narrow, sharp lowpass filters - Only DC needed?

I asked exact the same here one year ago: One year ago question I asked this question again here, because after one year, I have learned more about DSP and am thinking ways to improve it. Filter ...
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### s_to_z (Pupalaikis)

Paper: Bilinear Transformation Made Easy - http://documents.mx/documents/easybilinearpdf.html Example of implementation - http://codepad.org/u3tvKn0S I get the same output for Butterworth lp example ...
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### How does an Image filter really sweep an Image?

Suppose, we have the following image: Now, we want to apply a 5x5 filter to the image. How would the filter sweep the image? Like this? Or, like this? Or, anything else?