4 votes

How to calculate the total time delay of a first order linear filter on a signal?

The filter will introduce phase shift to the signal that will appear as a total time delay. Not really. For an IIR filter the phase shift will be a function of frequency and it will (in general) not ...
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Invertible low-pass (Butterworth) filter in python (scipy)?

Is there an invertible low-pass filter No is there something particularly difficult about inverting a low-pass filter? Yes. Digital low pass filters (in the most common sense) have a zero at ...
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Difference equation from a transfer function of a low pass filter

$$ \begin{align*} \frac{Y(z)}{X(z)}&=g\frac{1+a_1}{1+a_1z^{-1}} &\text{...given transfer function}\\ Y(z)(1+a_1z^{-1})&=X(z)g(1+a_1) &\text{...via cross multiplication}\\ Y(z)+a_1z^{-...
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4 votes

filter noisy angle signal

The difference equation you're using for your lowpass is $$x_n = (1 - \alpha) x_{n-1} + \alpha u_n \tag 1,$$ where $\alpha$ is your "forgetting factor", $u_n$ is your input, and $x_n$ is ...
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4 votes

Frequency response of an ideal low pass filter

There is no good answer to your question. What you're basically asking is What is the value of a discontinuous function at its discontinuity? It is up to you to define a value of the function at the ...
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Impulse response of IIR low-pass filter

In discrete time, a filter's impulse response can have a finite length (FIR) or an infinite length (IIR). The impulse response is just what it says it is: a system's response to an impulse at the ...
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Requirements on Signal Spectrum for Fractionally Spaced Equalization

As far as equalizer performance and possible limitations I provide two key points below about the span of the equalizer and the number of samples per symbol to use. The equalizer duration in time is ...
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Why is the ideal low pass filter not achievable?

General Background: Representing a signal numerically in a computer requires the signal to be discrete. That means it's periodic in the other domain. When you use something like a DFT you need the ...
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Amplitude / gain of difference of 2 known FIR low-pass filters

Of course. The difference is linear operation so you simply get $$H(z) = H_1(z)-H_2(z)$$ The Z transform of a moving average filter of length N is simply $H_N(z) = \frac{1-z^{-N}}{1-z^{-1}}$ so in ...
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Ideally Lowpass-filtered White Gaussian Noise: question about a derivation of variance and covariance

The inverse Fourier transform of $H(f) = \operatorname{rect}\left(\frac{f}{2B}\right)$, the transfer function of the ideal lowpass filter of bandwidth $B$ Hz, is, as the OP says, $h(t) = 2B\...
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Butterworth Filter Transfer Function

Your first equation is the magnitude of the frequency response. So the squared magnitude of the transfer function becomes $$\big|H(s)\big|^2=\frac{1}{1+\left(\frac{-s^2}{\Omega_c^2}\right)^{N}}\tag{1}$...
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How to know if filter is lowpass or highpass?

This is more a quick examination test or a rule of thumb than an actual proof that a filter has a "low-pass" or "high-pass" behavior. For a low-pass filter, it is expected that: a ...
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Convert matlab line to scipy signal processing line

Matlab comes up with a 15th order Chebycheff Type 1 filter so can probably poke the same filter specs into https://docs.scipy.org/doc/scipy/reference/generated/scipy.signal.cheby1.html If the filter ...
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Frequency response of an ideal low pass filter

This is mostly an academic question since both a sine wave and ideal low pass filter are infinite in time and hence can't exist in a real world application. A quick numerical hack Matlab hack would ...
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2 votes

Matlab: How to design digital equivalent for a lowpass Bessel filter (Thiran filter)?

I don't have Matlab but the coefficients for the Thiran filter are given by the Gaussian hypergeometric function. If you have wxMaxima there already is a built-in function. If you'll run these two ...
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A real-time (causal, non-symmetric) lowpass FIR filter with minimal passband ripple and phase

Consider using a minimum phase filter, which will have the least delay for a given magnitude response. The following response by MattL at the first post below details this further in using 'Leja ...
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Low-pass vs. windowing function in front of FFT

There are two questions here: when do we need an "anti-alias" filter and when do we need to window the signal? ANTI-ALIAS FILTER The need for an anti-alias filter is associated with the ...
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filter noisy angle signal

Tim's answer is great, but I'll give another option. Try turning the angles, $\theta_n$ into a complex number: $$ y_n = \cos(\theta_n) + j \sin(\theta_n) $$ and then applying the smoothing filter to ...
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I do not understand the frequency calculation used in the construction of this IIR digital Butterworth Low pass filter using MATLAB

EDIT: FOUND THE SOLUTION I looked into buttord.m in MATLAB and found out that it converts the normalised frequencies given to it by pre-warping those frequencies and converts it into analog domain. It ...
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GNU Radio Low Pass Filter doesn't work as expected with different transition widths

To align the samples (and realigning with different filter implementations) consider implementing actual timing and carrier recovery loops, or using those discriminators and approaches to manually ...
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1 vote

Choosing a good low pass filter in the wavelet scattering transform

The choice of $\phi$ affects: Time-shift invariance: slower decay in time will increase it Time-warp stability: slower decay in time will increase it mainly for deformations along time (but not only ...
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1 vote

Find the output signal in a Low Pass Filter for a given cutoff frequency range

The ideal (and unrealizable) low pass filter is a brick-wall filter that will pass all frequencies below it’s cutoff with a gain of 3 and reject everything above cutoff. The formula is simply a ...
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Find the band pass signal for a given low pass range and cutoff frequency?

First to clarify, a bandpass signal need not be real, but this would be the case when the positive and negative frequencies are complex conjugate symmetric. We can and do have bandpass signals where ...
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1 vote

Why the received SNR increases with bandwidth?

The chirp signal bandwidth is [-B, B] and your cut-off frequency is $ L≤B$, so your filter is removing part of the desired signal. By increasing $L$ you will increase the power due to the signal ...
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Why an does an integrate-and-dump stage in a CIC filter provide the same functionality as the integrator and comb in series?

As you already now a 1st order CIC filter is identical to a moving average filter. Lets consider the decimation factor to be 2 and having the following time series: ...
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1 vote

Do we guess the cut off frequency when passband edge frequency and stop band edge frequency are given? FIR Filter design-LOW PASS FILTER

In the windowing design method for FIR filters, the impulse response for the desired frequency response filter is selected with a window. Typically the desired frequency response is a rectangular ...
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How to identify low pass, bandpass, band reject, high pass filter etc from given $|H(e^{j\omega})|$?

Here is the trick-: I just found it out somewhere
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Question about impulse response of an ideal lowpass filter

Matlab uses the normalized sync function. So we have $$ h[n] = \frac{\omega_c}{\pi}\mbox{sinc}( \frac{\omega_c}{\pi}n) = \frac{\omega_c}{\pi} \frac{\sin(\pi \frac{\omega_c}{\pi}n)}{\pi \frac{\omega_c}...
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Question about impulse response of an ideal lowpass filter

There are two common definitions of the Sinc function: the unnormalized Sinc function: $\textrm{sinc}(x)=\displaystyle\frac{\sin x}{x}$ the normalized Sinc function: $\textrm{sinc}(x)=\displaystyle\...
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1 vote

Cut-off frequency for Low-pass filters in Arctangent Demodulation

The primary purpose of those filters is to pass the desired signal bandwith and reject the carrier feed-through at $\omega_h$ and the double frequency component at $2\omega$. Given that, I would ...
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