Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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135 views

Binomial Approximation of Gaussian Distribution

It is said that we may use the binomial coefficients ( a layer from Pascal's triangle) to approximate the 1-D Gaussian kernel with certain $\sigma$, where $\frac{n}{4} = \sigma$ and $n$ is the index ...
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351 views

Alternative to Extended Kalman Filter when prediction function is not differentiable

I am looking at a tracking problem. It can be modelled similarly to the Extended Kalman Filter: $$ \begin{array}{rcl} \mathbf{x}_k &=& \mathbf{f}(\mathbf{x}_{k-1}, \mathbf{u}_k) + \mathbf{w}...
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413 views

Help with audio EQ cookbook BPF filters and Q

I am fairly new DSP, and am using maths that I have not used in 20-30 years. I have been working with Cookbook formulae for audio EQ biquad filter coefficients by Robert Bristow-Johnson and am having ...
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1answer
129 views

Can I have some feedback on this research paper?

I have submitted this paper "Effect of Transducer Positioning in Active Noise Control" to arXiv. I would like to get some feedback on the methodology of the same. Inputs of any sort are welcome. Can ...
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2answers
77 views

Digital filter simulating hydroacoustic signal distortion

Preambule: I'm designing a sound model for my small submarine game. Model is running on the server, and I want to present the client with a mono-channel wav-stream from his hydrophone (20kHz ...
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1answer
87 views

Is this sampling frequency right?

I designed a Butterworth second order bandstop filter on Matlab with the following parameters: Notch frequency: 150 Hz (the frequency I want to remove) Bandwidth = 8 Hz, that is Fc1=146 Hz Fc2=154 ...
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1answer
115 views

Wavelets and bank filters

The wavelet transform has a problem as it gives poor time resolution for low frequencies and poor frequency resolution for high frequencies according to uncertainty conditions. This appears well ...
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1answer
79 views

Measure the length of curves via a smartphone's accelerometer

Is it possible to measure the length of curves by moving a smartphone along them and processing the accelerometer's data? For example, measure the circumference of a circle of radius $8 \,\rm{cm}$. ...
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2answers
1k views

What is the difference between the sampling frequency of signal and sampling frequency of filter

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...
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141 views

About how to increase the FIR filter sampling frequency in FPGA? And what is the tradeoff of increasing the sampling frequency?

I am trying to implement a FIR filter on FPGA and trying to have a solid understanding of the FIR filter tap delay and sampling frequency. Does the “one tap” delay equal to “1/Fs (sampling frequency)”...
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2answers
97 views

Can I combine an IIR (an EWMA) with a maximum?

I'd like to have a exponentially weighted moving average (EWMA) which is raised to the value of the input signal whenever this is higher than the filter output. This taking of the maximum shall take ...
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4answers
79 views

Can I use IIR or FIR to get a sliding mean?

My output signal shall be computed by (meta code): o = 0 # or some arbitrary initial value for i in input: o = (o * 99 + i) / 100 print o I call this a "...
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1answer
300 views

Design Lowpass Filter With -7.5 dB/Octave Rolloff

I am working on a project that needs a lowpass filter with a -7.5 dB/Octave rolloff. This is somewhere between the rolloff of a one-pole and two-pole filter, so I wanted to ask what some standard ...
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1answer
273 views

Butterworth low pass filter zeros location after bilinear transformation explanation

I am studying in a text book the transformation of a continuous time Butterworth low pass filter into a discrete time filter by means of bilinear transformation: $$ s = \frac{2}{T_d}*\frac{1-z^{-1}...
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44 views

Phase response approaches towards the zeros

Phase response is the relationship between the phase of a sinusoidal input and the output signal passing through any device that accepts input and produces an output signal, such as filter.https://en....
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803 views

Transition bands and passband gain in digital filter design

In an ideal design, a digital filter has a target gain in the passband and a zero gain (−∞ dB) in the stopband. In a real implementation, a finite transition region between the passband and the ...
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1answer
453 views

“fred harris rule of thumb”:

I would like to cite "fred harris rule of thumb" for estimating filter order. Is there a book or paper where I can find this rule of thumb? Thanks a bunch.
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64 views

Optimization‏ FIR filter

I have a doubt about this : Iterative methods for optimization‏ in FIR filter (on the one hand traditional like window,frequency sampling ,WLS,Remez Exchange...,on the other hand evolutionary methods ...
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1answer
244 views

Least Squares Linear Phase FIR Filter Design

In explication ''the geometric interpretation of least squares'' Typically, the number of frequency constraints is much greater than the number of design variables (filter coefficients). In these ...
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388 views

Get difference equation of a filter from MATLAB transfer function coefficients

I need to get the difference equation of a specific elliptic filter. I calculated the transfer function coefficients in MATLAB with: ...
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3answers
102 views

different moving average filters differences

Suppose we want to use a moving average filter (3-point moving average) for $X[n]$ (where $Y[n]$ is output). What are the differences among $Y[n]=\frac13X[n-1]+\frac13X[n]+\frac13X[n+1]$, $Y[n]=\...
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1answer
115 views

Optimal Chebyshev FIR filters

I have many information i want to be clear very well . The defining characteristic of FIR filters optimal in the Chebyshev sense is that they minimize the maximum frequency-response error-magnitude ...
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1answer
174 views

Design single pass band FIR filter

I'm new in DSP. I want to design single pass band FIR filter using LS method. I have this information : one pass band $[-0.1\pi , 0.3\pi]$ two stop bands $[-\pi , -0.18\pi]$, $[0.38\pi, \pi]$ ...
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27 views

Direct form I and Direct form II

I was studying about realization structures of digital filters. Is it mandatory to have the order of numerator must be less than that of denominator of transfer function for realization of filter ...
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2answers
822 views

How to Design a Pre-emphasis filter knowing the gain, a zero and a pole?

I am simulating an FM radio signal in order to use it as a waveform in passive radar; for this purpose I'm following an article: Analysis and Emulation of FM Radio Signals for Passive Radar In a ...
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2answers
4k views

Determining the Filter Coefficients of an FIR Filter

I need to find the filter coefficients of an FIR filter that will block sinusoids of frequency $200\ \rm Hz$ if the sinusoid is sampled at $1.2\ \rm kHz$. I feel like this is a fairly simple problem,...
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1answer
2k views

Pre-emphasis filter for speech recognition

I'm trying to design a: $$H(z)=1 - 0.95z$$ FIR filter, for speech signal. I am using this arm_fir_example_f32.c code shown below. And I need to change the filter ...
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1answer
170 views

Biquad Treble Shelf filter gain being weird

Trying to compute Biquad Coefficients for use in a codec chip TLV320AIC3100. Bass Shelf(boost and attenuation) work as expected. Treble Shelf attenuation works as expected but Treble Shelf boost is ...
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3answers
166 views

Why make the phase of an FIR filter linear?

I am new to DSP and filter design. It is easy to design an FIR filter with linear phase by making the coefficient sequence symmetric. However, why make the phase linear?
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41 views

Discrete Hilbert transform enhancements?

Consider the Discrete Hilbert Transform, which takes a sequence $u[n]$ defined on the interval $-M\leq n\leq M$ and returns the discrete convolution $$\tilde u[n]=\sum_{i=-M}^Mu[i]h[n-i]$$ where the ...
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1answer
110 views

Filter bank for complex spectrum

I'm using a complex-valued baseband signal as my input for subband adaptive filtering. Therefore i have a complex input spectrum. My problem is now, how to split my complex spectrum into $N$ equally ...
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1answer
693 views

Implementation of the filter function (Matlab) C/C++

I am trying to apply some filtering in C/C++ code, I have designed an algorithm using Matlab and it works well, so now I am moving it to native code. I am ...
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1answer
170 views

Best way to process this signal

I am novice in signal processing, and i have to aplogise for my english level. The problem is the next: I work for foundries, in this sector the cast parts are manufactured pouring liquid metal in ...
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2answers
111 views

Stable digital filter with leading zeroes

I'm looking construct a stable pole-only filter where the feedback coefficients start with a block of zeroes, i.e. \begin{align} a_0 &= 1\\ a_i &= 0, \textrm{ for}\ 1 \le i \lt k\\ a_i &\...
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1answer
89 views

Design of a filter with given bandwidth and sidelobe level using Parks–McClellan algorithm

I want to design a filter (beamformer) with a given bandwidth (beamwidth) and sidelobe level using Parks-McClellan algorithm. I use "firpm" function of MATLAB. however, I cannot reach a solution for ...
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1answer
332 views

When does a Window function's sidelobe level become out of band attenuation?

Another question I'm struggling to answer by myself. For a Window function, say Blackman (response pictured), what is the out of band attenuation? If I'm designing a filter with out of band ...
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160 views

What is the meaning of a inner system/transfer function?

In an academic paper I'm currently reading [1], the notion of an inner system is central. It is defined as follows A (linear) system $G$ (with $m$ inputs and $p$ outputs) is called inner if it ...
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1answer
113 views

Filter response is changed according to its order

I want to build a band-reject filter for the 60 Hz noise. So, I tried to use Butterworth filter to implement this specification. However, when I chose an order of 8, I got the following response, ...
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1answer
130 views

Properties of Optimization Techniques in Filter Design

I have this design: Can you tell me about the type of filter and the algorithm just viewing this design, or should there be other information?
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226 views

Least Squares Approximation for FIR Filter Design

Following this paper , I am trying to make a least-squares algorithm in MATLAB, but for type I (I know about firls()). ...
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252 views

Applying Jake's remez exchange algorithm for filter design

I have found C implementation for remez exchange algorithm(https://github.com/janovetz/remez-exchange/blob/master/remez.c) and I want to use this to implement a generic prototype lowpass filter. ...
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1answer
276 views

Is there a $z$-transform like for variable sampling rate signals?

I'm working with signals with variable sampling rate (the time space between samples is not constant). I know the delay between samples but I don't wont to interpolate the signal. Is there a way to ...
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1answer
38 views

Determine proper coefficients for the transfer function

The problem says that we have the information (actually a .mat file that contains the data) of 1 pixel from an IR camera from an object with 26° of temperature. Because of the noise, the output is ...
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1answer
323 views

DWT versus band-pass filter

Which technique gives better result in extracting a band from a signal, let's say the alpha band (8-13 Hz): using a band-pass-filter or a DWT (Discrete Wavelet Transformation)? P.S.: I am working in ...
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1answer
369 views

Frequency Translation in MATLAB

I have to design a generic baseband filter. I have a C++ code which gives me filter coefficients based on the input parameters such as no. of taps, no. of bands, band edges, amplitudes and weights(...
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2answers
392 views

Looking for a simple and effective notch filter

I ran a finite-difference simulation and the behavior of an output signal, $s$, in time, $t$ (sampled with period $\Delta t$) behaves approximately as in the figure below. It is well-described by a ...
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1answer
137 views

Is initializing a digital filter's output with no “momentum” a non-trivial task?

I'm working on implementing a filter with a very slow step response. This filter is implemented as a cascaded second-order-section filter (transposed direct form 2). I'm using the output of this ...
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2answers
207 views

Is there anything called uncorrelated filter?

I understand to an extent various filter like low pass filter, high pass filer, Wiener filter Kalman filter etc. I also understand some of this filter will decorrelate/uncorrelate the signal. The ...
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1answer
68 views

can $\frac{1}{H(z)}$ be causal and stable? [duplicate]

if we have linear phase FIR filter $H(z)$ which is causal and stable can $\frac{1}{H(z)}$ be causal and stable ? can it be causal without been stable ?
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147 views

Speaker equalization for white noise

I need a speaker to play broadband white noise for my scientific application. I would like to know whether my relative naive implementation is correct or not. The speakers I use have a large frequency ...