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Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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What frequency each bin of a Discrete Cosine Transform represents?

A practical example: I perform a DCT on a time series of discrete values that are spaced in time by 1/30 of a second. What frequencies each bin of this DCT represents? What is the formula to find that?...
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43 views

How should I modify my filter coefficients for the filter to be unaltered, when I change my sample rate

I am guessing this is a commonly encountered problem, but I am lost on how to really solve it. I have FIR filter coefficients that are used to match the frequency response of data sampled at 3 kHz. I ...
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0answers
62 views

The non-recursive filter [closed]

An FIR digital filter of length N with impulse response ${b(n)}, n=0,1,...,N$. Therefore, the complex-valued, multiplicative function $H(f)$ is the filter's frequency response.It is defined by a ...
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1answer
44 views

Digital filter design of time series for specified frequency response function

I am currently working with vibration measurements in structures. In the netherlands there is a guideline for verifying vibration measurements for damage to machinery. This is the so-called "SBR ...
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32 views

Accelerometer signal processing

I'm new signal processing (this is my first pratical application), so I ask sorry if my question is simple. My challenge is to see the displacement of accelerometer when I have vibration too. I do ...
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1answer
54 views

how can I design Chebyshev filter with 5th-order polynomial and 0.001%-allowed ripples?

I'm very beginner in signal processing. I need to design a Chebyshev filter with 5th-order polynomial and 0.001%-allowed ripples by using matlab as I found in a paper. I'm filtering pressure data ...
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1answer
48 views

Is it possible to design a band-pass filter with fewer taps than a series HPF + LPF?

I am digitally filtering a signal that is streaming live. The goal is to reduce the filter delay as much as possible while maintaining the quality of some measurements based on the filtered signal. ...
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1answer
45 views

How to filter an OFDM signal? [closed]

Across the channel bandwidth there shoudd be 100 RBs (central 1200 sub carriers in a 2048 bin). Max power normalized to 0dB, the first OOB lobe should be -34dB. This 1st lobe on either side is at 5 ...
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0answers
60 views

Practical implementation of IIR/biquad filters following non-textbook transfer function of real-wold analog filter

I am trying to program a DSP filter to study prototypes of loudspeaker filters, which will be implemented as passive analog circuits in the final speaker system. This process involves simulation of ...
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1answer
27 views

Defining Model in Kalman for Getting Position Using

I am stuck at modeling a system model, i.e. getting my state vector and input vector for navigating just using navaid and ins (tactical). My guess is that position is my only state vector and INS ...
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3answers
313 views

Why is it easier to design an IIR filter in the continuous time domain than in the discrete time domain?

Given that the goal is to approximate the ideal filter with an IIR filter, why would it be easier to design it in the continuous time domain than the discrete? This paper says: Discrete time ...
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2answers
41 views

What should be the cut off frequency of a Filter operating on an OFDM signal?

I have a time domain OFDM signal and to this I need to apply different linear filter however I am not sure on the cut off frequency. I am simulating for multiple symbols where each symbol has 1200 ...
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19 views

Constructing a lowpass digital differentiator such that maximum error is $n$ dB for passband

I want to construct a lowpass digital differentiator with uniform-interval samples such that error from ideal differentiator is $n$ dB in magnitude at maximum for the passband up to frequency $\...
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3answers
69 views

How to filter my noisy square wave so that i can recover my signal using the Chebyshev Filter

I have a noisy square signal as input but it has a lot of noise. I should use a Bandpass filter to recover my signal. I know that the Chebyshev Filter is a bandpass filter; but it doesn't work. the ...
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2answers
74 views

Frequency response of numerical differentiation by polynomial interpolation / finite difference

One can use polynomial interpolation (or finite difference) to do numerical differentiation. However, there seems to be a surprising lack of interest in obtaining frequency response of this numerical ...
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1answer
31 views

Using Parks-Mcclellan (or others) for FIR differentiator?

I have never seen the use of Parks-Mcclellan for differentiator - most uses tend to be lowpass filters. Is this because Parks-Mcclellan implementation of FIR differentiator has numerical problems and ...
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2answers
41 views

Is Parks-Mcclellan algorithm guaranteed to stop?

Is there any guarantee that Parks-Mcclellan eventually stop? If so, is there any formula that shows number of maximum interations that ensure that the algorithm stops?
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1answer
40 views

FIR butterworth also possible or just IIR?

can we use butterworth and other such filters like chebyshev,elliptic etc with FIR or they can be only used with IIR?
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1answer
35 views

How to apply two notch filters simultaneously

I have a sound file that I need to apply a notch filter to in Matlab. comparing to original file I have two frequencies which are the noise. Applying a notch filter like this one: ...
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1answer
59 views

Design of FIR and IIR filters in Matlab

I need to create 4 different pass-band filters. What remains the same throughout is the following: Sampling Frequency = $3000Hz$ Pass-band Frequency = $900Hz-1200Hz$ Now, these are the filters I ...
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1answer
48 views

Deciding an EQ's Center Frequencies

I am currently making a relatively simple EQ of a music app, for which we decide to have 5 bands. Other than that, we want to have a bass boost knob, which, as I view it, is just another Peak EQ ...
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1answer
59 views

Upsampling signal using convolution-based interpolation filters

I am currently reading this paper which discusses several image interpolation methods, such as nearest neighbor and linear interpolation, using convolution filters. I first want to do this in 1D with ...
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1answer
188 views

Single-pole IIR low-pass filter - which is the correct formula for the decay coefficient?

A single-pole IIR low-pass filter can be defined in discrete time as y += a * (x - y), where y is the output sample, ...
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1answer
79 views

Orders of complimentary butterworth filters for IMU

I'm currently trying to design a complimentary filter for deployment on a drone, as part of an assessed semester project. Even though processing power is not an issue (we have GHz, lots of pipelining ...
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7answers
2k views

Is sound the best way to transmit a signal under water?

Could sound be used in transmitting sound waves under water digitally? Could the sound then be converted by a repeater to transit by radio waves? What other ways can a signal be transmitted other than ...
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1answer
28 views

EEG Sampling, Which Order should cover my result?

I have EEG data sampling rate is 250 Hz and I want to band pass this data in ( 0.5-3 Hz & 3-8 Hz & 8-13 Hz & 13-30 Hz & 30-40 Hz). Currently, the data is in the frequency range of ...
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1answer
52 views

Doubts about Notch filter design

I am following the process that is described in this question: Transfer function of second order notch filter , I want to create a notch filter with the band suppressed equal to $f_c = 4000$ Hz, so ...
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1answer
32 views

How do I add more resolution / taps to an analog -> digital filter?

(I am trying to create an IIR audio filter that adds reverb to an initial sample) Say I designed an analog filter to model acoustic attenuation based on the following mathematical model: $$ I = I_0 ...
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2answers
63 views

Practical Least Squares FIR filter design: How to find a realistic desired response

I'm using Matt. L.'s lslevin.m function to design long ($N\approx 400$) filters to an arbitrary phase and magnitude specification. The specification is the result of a numeric optimization and does ...
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1answer
68 views

Denoising a signal using eigendecomposition

I have a complex observable series $Y(t)$ which is the result of summing two complex r.v $X(t)$ (unobservable) and a $\epsilon(t)$ (unobservable). $$Y(t)=X(t)+\epsilon(t)$$ Assume that $X$ and $\...
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1answer
31 views

“Side Lobes” When Using a Butterworth filter

So I am attempting to filter a signal containing frequencies at 7Hz, 11Hz, and 74Hz. The signal lasts for 30s and looks like this between -1s and 1s: Using a butterworth filter of order 2 with lowcut=...
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1answer
118 views

FIR design for arbitrary magnitude and phase: Why can I separate real and imaginary parts like this?

In this question about the design of a FIR filter with arbitrary magnitude and phase specifications user robert bristow-johnson suggested to split the desired complex frequency response $H$ into its ...
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0answers
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Using a Passive 1st Order RC filter to Extract DC Component

I am trying to extract the DC component out of a PWM signal produced by the MCU using a LPF. I understand that 1st order RC filter is looked upon as being amateurish because the gain in the passband ...
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2answers
44 views

Digital Filter Design using difference equations

I am reading a chapter on digital filter design from analog filter design using difference equations. What they do first of all is that they map $s$ (Laplace variable) to $z$ ($z$-transform) by the ...
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1answer
49 views

Signal reconstruction from sampled signal? [closed]

Can we construct exact analog signal at receiver as same as the source from the sampled signal?
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0answers
66 views

Generation of Noise-shape filter from Power Spectrum Density

This is my very first time in dealing with signal processing, so I am sorry if I will not use a rigorous terminology. I am dealing with some issues about noise modeling in matlab. I'm trying to ...
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1answer
71 views

Find filter coefficients to model a device using its measurement

I am trying to realize a digital filter that has the same freq. response of an existing speaker. I have fed an audio sine sweep to the speaker and measured the speaker output, both at 48kHz. Then I ...
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1answer
63 views

Why must the Region Of Convergence (ROC) contain infinity and the system function be a right-sided sequence for it to be causal?

I need the simple logic on the condition a system becomes causal. We know that causal contains only past values. I can't relate this with the Region Of Convergence (ROC) concept.
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1answer
44 views

4-phase FIR LPF in MATLAB

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...
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3answers
540 views

Savitzky–Golay filter vs. IIR or FIR linear filter

A traditional IIR / FIR filter (lowpass to remove the high freq oscillations), e.g. moving average, or a Savitzky-Golay filter can all be useful to smoothen a signal, such as an envelope signal: ...
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2answers
108 views

Bandpass filter that automatically adapts its bandwidth when a transient is detected (to avoid to smoothen the transient)

Let's say we want to isolate a band 1000 hz +/- 50 hz. Obviously, limiting the bandwidth by applying a passband filter will always destroy a bit the sharp transients (a Dirac or a rectangular ...
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2answers
108 views

How to compare many bandpass filters (for their signal propogation delay)?

Let's say we have an audio signal sampled at 96 kHz, and we want to compare several bandpass filters to find the one with the lowest rectangular-envelope-propagation-delay (see graph below). The band ...
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1answer
89 views

Design of a complex FIR filter

For apply least-squares linear-phase FIR filter design,with frequency domain specification is not symmetrical. The pass-band error function, $$E(\mathbf{h})_p=\int_{\omega_{p_1}}^{\omega_{p_2}}| \...
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0answers
32 views

How far can numerical differentiation be accurate for sinusoids?

Suppose you have samples of uniform interval of a sinusoidal signal that is some sum of sinusoids. Forget about aliasing, as the signal is bandlimited. If we have infinite number of samples at ...
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1answer
109 views

How does a digital filter work?

To introduce my situation: I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "...
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4answers
774 views

Why is the first value in digital first-order IIR filter impulse response not the largest?

When I create a digital first-order IIR low-pass filter with scipy (code below), I get the following coefficients: B: [ 0.1367 0.1367] A: [ 1. -0.7265] The ...
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2answers
231 views

Alias-free digital nonlinear filter design

@Jazzmaniac has a good answer to the question of how to design an alias-free digital nonlinear time-invariant filter here: https://dsp.stackexchange.com/a/28787/18276 Basically, according to that ...
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1answer
56 views

Problem with 1st order Massberg LPF [duplicate]

I'm trying to implement this filter in title by following the book "Designing Audio Effect Plug-Ins in C++" By Will Pirkle Problem I'm facing is the magnitude response error when comparing against ...
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0answers
107 views

Issue with the Massberg Analog-Matched Low Pass Filter

Implemented the filter mentioned in title using Octave and couple sources Vicanek and Zeitlin (LPF.cpp). Got filter working otherwise but, ... when tested the responses against 'analog model' of the ...
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1answer
152 views

Advancing a causal IIR filter's impulse response

Given an infinite impulse response (IIR) filter as coefficients in a topology of your choice, what is a straightforward way to modify the filter so that the impulse response is shifted in time (...