Questions tagged [filter-design]
Fiter design is the process of designing a filter.
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1answer
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Change durration/tempo of a .Wav file without introducing pops or pitch shift?
I am attempting to develop a similar function as Audacity's Effects>Change Tempo effect.
Or more specifically: I am in need of a way to create a funciton in my ...
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0answers
20 views
FIR filter window method [duplicate]
In the window method for filter design it is explained that we do an ideal filter and then we pass it to time domain, but it will be time limited and then...
Why don't we filter directly in the ...
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0answers
16 views
Signal ECG , collecte & visualize data
I am currently working on a sensor that visualizes an ECG signal and visualizes its data. Its principle is that it has 3 electrodes that must be placed in specific parts of the body .... I would like ...
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0answers
10 views
Frequency warping while creating filter coefficients
If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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1answer
39 views
Filtering on time or frequency domain?
I'm applying an ideal filter on the frequency domain. My code below:
...
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0answers
15 views
Need help with detecting onset of audio in wav file
I'm new to DSP and I'm learning as I go. I have an audio file with a chirp signal(2kHz). I played and recorded this chirp signal and the recording has a lot of environmental sounds.
Here is my code:
2
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1answer
37 views
Electret microphone RC filter design contradiction among information sources
I'm building a circuit for an electret microphone and I want to build a bandpass filter around the op-amp.
I was using this source until I have found that most sources ( e.g. here and here) indicate ...
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1answer
40 views
How to find lattice coefficients of all pole IIR filter with filter coefficient $k_m$ is 1?
Given the following system function
$$ H(z) =\frac{1}{1+\frac{2}{3}z^{-1}+\frac{5}{8}z^{-2}+\frac{2}{3}z^{-3}+z^{-4}} \tag{1} $$
We have to draw lattice structure for above system function
The ...
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0answers
33 views
Filtering out different narrow-band frequencies at once
I have asked a closely related question on SO at https://stackoverflow.com/questions/55168460/python-implementation-for-filtering-out-multiple-distinct-narrow-band-frequencie
but I am still unclear ...
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0answers
11 views
Filtering frequencies in diferent ranges in Python
I would like to filter frequencies: Alpha, Beta and Gamma. These bands have a relation with the task that I want to classify,
I am using the Gumpy python tool which has a method to apply Butterworth ...
6
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3answers
289 views
Protect an IIR filter from being reverse-engineered
I created a somewhat unique IIR filter and I want to protect the filter from being reverse-engineered
I think you all know it is quite easy to get all the different weights of an IIR by using impulse ...
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0answers
25 views
Python scipy firwin2 has a low frequency error when designing arbitrary amplitude filter
I've been trying to design an arbitrary amplitude FIR filter which has a specific attenuation for each frequency. However I get an error at low frequency when comparing the results to my expected ...
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0answers
25 views
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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0answers
48 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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1answer
71 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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0answers
42 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
58 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
54 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. ...
0
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1answer
54 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 ...
2
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0answers
70 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 ...
0
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1answer
28 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
354 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
44 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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0answers
20 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
87 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 ...
2
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2answers
93 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
57 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
58 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
148 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
70 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:
...
1
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1answer
76 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 ...
0
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1answer
54 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 ...
0
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1answer
71 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
468 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, ...
1
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1answer
83 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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8answers
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
30 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
59 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
33 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:
$$
I = I_0 ...
5
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2answers
71 views
Practical Least Squares FIR filter design: How to find a realistic desired response
I'm using Matt. L.'s lslevin.m function to design long ($N\approx 400$) filters to an arbitrary phase and magnitude specification. The specification is the result of a numeric optimization and does ...
2
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1answer
79 views
Denoising a signal using eigendecomposition
I have a complex observable series $Y(t)$ which is the result of summing two complex r.v $X(t)$ (unobservable) and a $\epsilon(t)$ (unobservable).
$$Y(t)=X(t)+\epsilon(t)$$
Assume that $X$ and $\...
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1answer
38 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=...
4
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1answer
148 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
28 views
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 ...
-1
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2answers
47 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
55 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
81 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
83 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 ...
0
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1answer
89 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
48 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 ...
