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Questions tagged [bandpass]

A band-pass filter is a device that passes frequencies within a certain range and rejects (attenuates) frequencies outside that range.

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How can I calculate how much of my signal at the beginning and end will be affected by the filtering process?

I'm using a Butterworth bandpass filter to process my signal (low-pass + high-pass, I am also applying a notch filter). I would like to calculate how much of my signal at the edges will be affected by ...
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Butterworth Bandpass Filter Design

Determine the transfer function for the Butterworth bandpass filter? (At the Transfer Function Factoring and Sp Substitution stage) The Parameter $$a_{pass} = - 1dB$$ $$a_{stop} = - 20dB$$ $$f_{pass1} ...
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How can I smoothen this Chebyshev type I bandpass filtered signal?

I'm unable to figure out the cause behind my noisy bandpass filtered signal, despite my adaptation of the paper's parameters, in which they specified it to be a 4th order Chebyshev type I bandpass ...
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What duration of a white noise burst is required for it to be "white" at a given frequency or frequency range?

I am running white noise bursts (with very short ramps on/off to prevent discontinuities) through underdamped resonant bandpasses which are tuned to any given $f_0$ and an underdamped $Q$. Continuous ...
mike's user avatar
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What is the cleanest, most transparent way to rapidly excite a resonant bandpass to a given amplitude (for musical synthesis)?

Background Music audio synthesis can be performed using underdamped resonant bandpasses as harmonic oscillators (set to harmonic frequencies of a given note). These can then be excited when input ...
mike's user avatar
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PSD of real and imaginary parts of complex envelope of a bandpass signal

I know that we can write a real bandpass signal x(t) as: $x(t)=x_c(t)\cos\omega_ct + x_s(t)\sin\omega_ct$ $x_c(t) = 2x(t)\cos\omega_ct$ for $|f|<W$ $x_s(t) = 2x(t)\sin\omega_ct$ for ...
Ang's user avatar
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2 answers
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Confusion on when a filter is considered band-pass

As part of an assignment, I was given a filter with a specific set of poles and zeros, namely poles at $0.51 \pm 0.68i$ and zeroes at $1.5$ and $−0.7$. This part of the assignment is titled as "...
Nyquist-er's user avatar
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Is a bandpass filter superfluous when the frequency band of interest after applying the FFT is below the Nyquist frequency?

I have time-series data with a sampling rate of 1 Hz, hence the Nyquist frequency is 0.5 Hz. I am interested in the frequency band 0.01-0.1 Hz and I would like to apply a Periodogram to transform the ...
Philipp's user avatar
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Use periods instead of frequencies in Comodulograms

I want to create a Comodulogram to assess PAC. I use the pactools library, which uses frequencies. Since I work with hourly data and want to look at multidien phases, it would be better to use periods ...
Rolandos's user avatar
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Is there any way of calculating the portion of signal lost as we keep on reducing the passband edge frequency of a band-pass filter?

Any kind of references would be appreciable.
Hardik Dey's user avatar
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Calculating the settling time of lowpass and bandpass SOS filters constructured using SciPy

I have constructed a high pass filter using scipy: ...
BeginnersMindTruly's user avatar
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Understanding the parameters of a notch filter

Adobe Audition has a very interesting notch filter that have been very useful to me. It has the following parameters According to their documentation, the chosen frequency is the center frequency and ...
TheGMX's user avatar
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Phase correction of a received signal

How is it possible to correct the possible phase distortion of a typical signal which has been received by an antenna and passed through different analog elements including an amplifier and a band ...
Fernando Sanz's user avatar
3 votes
2 answers
521 views

Given two low-pass digital IIR filters, find bandpass coefficients

In the case of FIR filters it is easy to get a band-pass filter by subtracting the coefficients of two low-pass filter filters or by convolving a high-pass and a low-pass filter: ...
nopeva's user avatar
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Reduce memory usage by bandpassing a signal

When recording an audio signal one can have a high sampling frequency. To reduce the amount of data that is stored, and thereby the power consumption, I head that it is possible to use a high-pass ...
3dSpatialUser's user avatar
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1 answer
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Can the standard deviation of the Gaussian window in a Gabor filter be made infinitesimally small?

My understanding is that the standard deviation of the Gaussian window in a Gabor filter dictates the temporal resolution. Wouldn't it always be better then to make the window smaller, thus achieving ...
SNIreaPER's user avatar
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What is the difference between STFT and Gabor filter?

My understanding is that a Gabor filter gives us information about when in the input signal does the frequency of interest (equal to the frequency of the sinusoid in the Gabor Filter) occur, thus ...
SNIreaPER's user avatar
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How to band pass only on positive frequency [duplicate]

I've posed the similiar question I appreciate that the former replier tells me the process is about Hilber transform,but I'm still confused with the positive frequency. The paper process a cardiac ...
subborrrr's user avatar
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1 answer
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fast fourier transform band pass filter only on positive frequency

Recently, I came across a paper that focuses on signal processing using lock-in amplification. the algorithm involves applying the Fast Fourier Transform (FFT) to a time-domain biosignal. A band-pass ...
subborrrr's user avatar
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What does a Gabor filter "filter" out?

Almost all of the searches online have returned results related to image-processing, but I am looking for a more rudimentary example with some physical intuition (eg; a high-pass filter only lets the ...
SNIreaPER's user avatar
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single tone -> multi bandpass filter -> incorrect output

background: Trying to filter out all other signal frequencies except -> signal whose frequency can be in multiple's of 16Hz {16,32,48.......160]. so started experimenting with a single tone [no ...
particlereddy's user avatar
2 votes
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Issues with ML Pattern Recognition After Bandpass Filtering

We've been working on a machine learning project for pattern recognition, using time-domain features such as kurtosis, mean, standard deviation, variance, skewness, and peak-to-peak values. Background:...
Tarek's user avatar
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How can I simulate the process of transmitting BPSK signal from start to end?

I would like to simulate the process of transmitting a binary phase shift keying (BPSK) signal from start to finish, going through each step over additive white Gaussian noise (AWGN) channel. I know ...
Math_Novice's user avatar
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Event-related EEG (epoch averaging) as a form of inter-harmonic bandstop filtering?

There is a matter that comes to my mind from time to time, now I describe it in detail. Anyway I am a first year PhD student, doing research in the field of cognitive neuroscience, but yet I only have ...
LimeAndConconut's user avatar
1 vote
2 answers
240 views

How to construct a bandpass filter with minimal loss in amplitude/power near the lower and upper frequency limit in Python?

I am analyzing functional MRI data with a sampling rate of 1 second (1 Hz). The frequency band that I am interested in is 0.01-0.2 Hz. Regarding measurements, I am interested in computing (1) the ...
Philipp's user avatar
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Hilbert transform with non-modulated signals

Is it correct to compute the Hilbert Transform and then the complex envelope, expressed as $v = z~ e^{-j 2\pi f_c t}$, where $z$ represents the analytic signal and $f_c$ represents the carrier ...
Asli's user avatar
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3 votes
1 answer
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Fitting high order rational function to frequency response measurement data

I need to model measurement data of a frequency response with physically meaningful band-pass filters. All Measurements happening in die range of 20Hz to 20000Hz I have to work with python. The ...
LorenzNew's user avatar
1 vote
1 answer
401 views

Narrow bandpass filtering

I want to plot a specific frequency (e.g., 1 kHz or 100 Hz) of an audio file (sampling rate 44.1 kHz). I have done fft and spectrogram analysis on the file, but I am now interested in plotting only ...
pbena's user avatar
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2 answers
101 views

Maximum bandwidth of radio-frequency filter to avoid the image frequency

Consider the following super-heterodyne receiver The carrier frequency of the local oscillator is chosen to be $$ f_{LO} = f_c + f_{IF} \tag{1} $$ where $f_c$ is the transmitted carrier frequency and ...
Rubem Pacelli's user avatar
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100 views

Bandpass filter with a changing passband

The problem: I have a signal that contains multiple, relatively stable frequency components and I want to extract only one of them. I attached a plot that illustrates the situation with a simple ...
weygoldt's user avatar
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1 answer
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Proof of the equivalence of the Band-RMS evaluated in time and frequency

Ignoring normalization factors, we can define the squared Root Mean Square (RMS) of a discret-time signal $x[t+n], n=0,1,...$, at time $t$ as: $$\mathrm{RMS}^2_t = \sum_{n=0} ^{N-1} |x[t+n]|^2$$ ...
AstoundingJB's user avatar
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445 views

Calculating the filter coefficients for 2nd order bandpass and bandstop Butterworth filter

I want to calculate the filter coefficients for both bandpass and bandstop filter for 2nd order Butterworth filter, like ...
Devil_____03's user avatar
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1 answer
126 views

Bandpass filtering EEG and getting weird distortions, parameter issue?

I am trying to bandpass filter an EEG signal, nothing fancy but it's coming out pretty distorted. EEG data is taken from forehead. Sampling rate is 250 Hz. Cutoff is 2.5 Hz & 120 Hz. Tried in both ...
fillifilterbutterfly's user avatar
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1 answer
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design of cascaded biquad filter

I have to compare the response of an 8th order IIR filter and its equivalent 4 stage cascaded biquad structure. A sinusoidal signal is given as input to both the systems and the output responses are ...
Deepa's user avatar
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Bandpass function in Matlab inverting the Frequency spectrum when using a hermitic transformed signal

I have a VNA response (S11 parameters) from 3.5 to 40 GHz. Then I apply a hermitic transformation (complex conjugate) to it since we only have the spectrum for the positive frequencies. Then I use ...
Gabriel Llorens's user avatar
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Matlab: How to implement a bandpass filter to isolate harmonic with 0 gain at this frequency? [closed]

I want to isolate the harmonic at 440Hz and I need to design a filter with zero gain at 440Hz. I tried this with matlab: ...
tonythestark's user avatar
1 vote
1 answer
129 views

Filter IIR, 1/3 Octave Corner with 2 Octave BW

I'm trying to design a new filter with IIR to filter out specific bands. Basically it's a bandpass filter with 2 Octave or 3 Octave bandwidth but the filter order should be equal to 1/3 Octave filter. ...
BugraKezan's user avatar
-1 votes
1 answer
245 views

Inverting (spectrogram vertical flip) a signal of WAV file

What I need is to invert a signal (on spectrogram it would look like a vertical flip) of a WAV file. Actually, of a bandpass of a signal - from 500 to 800 Hz, for example. This image will perfectly ...
Ivan G's user avatar
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2 answers
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frequency Domain with bandlimit

May I ask why does the frequency Domain Picture of a signal look like that? Is the signal built with many different frequencies?
kidox's user avatar
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Practical problems in down-converting via Hilbert filter

While teaching communication system to my students, I showed different ways to define the diagram block of the system responsible for transforming a bandpass signal, $x(t) \in \mathbb{R}$, into its ...
Rubem Pacelli's user avatar
5 votes
4 answers
326 views

Bandpass Stationary Stochastic Process

I was following this interesting post by a new user Rubem Pacelli and got stuck at Proakis' referenced definition (see Section 4-1-4 starting on page 159 here). The math, all repeated further below, ...
Dan Boschen's user avatar
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2 answers
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Designing a digital band pass filter with Sinc filter in time domain

I want to design a sinc filter in the time domain (a bandpass filter in the frequency domain). I wrote test code in python to investigate, and would appreciate help understanding it and answer my ...
Captain's user avatar
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Signal power after frequency shift with different frequency than the carrier frequency

I have this received signal in passband $$ \begin{split} r(t)=&\Re\left\{\tilde{x}_1(t)e^{j2\pi f_1t}+\tilde{x}_2(t)e^{j2\pi f_2t}\right\}\\ =&\Re\left\{e^{j2\pi f_1t}\underbrace{\left[\tilde{...
Math_Novice's user avatar
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1 answer
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Minimum bandpass bandwidth?

Is there any restriction on decreasing the bandwidth (beyond the sampling rate) that I should be aware of when filtering a signal with a bandpass? My intuition tells me that the quality of the result ...
Tojur's user avatar
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2 votes
1 answer
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Doubts about digital signal processing methodology

Disclaimer: I'm a beginner at all of this. I am working with vibration data from accelerometers recorded at a sampling rate of 1600 Hz and the vibrations are induced by a source within the range of 20-...
Teja Jupudi's user avatar
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1 answer
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Online algorithm to detect non-directional volatility?

I'm trying to write an online algorithm to detect when a time series goes sideways (i.e., its mean trend or mean drift is approximately zero) but is also unusually volatile. For example, in the red ...
Jase's user avatar
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1 vote
2 answers
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How do I use this bandpass filter? [closed]

I found this free DSP book by Rutgers University professor Sophocles J. Orfanidis. The book is called Introduction to Signal Processing and the link contains many different implementations of common ...
yellow_watermelon's user avatar
2 votes
3 answers
1k views

What is the relationship among digital filter cutoff frequency, signal frequency and filter settling time?

I have some general observations but I am asking if anyone can summarize more conclusively: (1) Filter settles slower if the ratio between signal frequency and filter cutoff frequency is smaller; (2) ...
John's user avatar
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Are there any recommended filter types for passband frequencies near Nyquist?

Is there a recommended filter type for frequency bands near the Nyquist frequency? I have been trying to design complex passband filter with passband $[0.8 ~~1]$ ( where $1$ corresponds to the ...
Hilbert's user avatar
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1 answer
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About the process to convert basedband signal into passband

Theoretically, the passband $y(t)$ signal equivalent into baseband signal $x(t)$ is given by: $y(t) = \Re \begin{bmatrix} x(t) e^{(-i2 \pi f_ct )} \end{bmatrix}$ where $\Re$ represents the real part. ...
Sajjad's user avatar
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