Questions tagged [lowpass-filter]
A low-pass filter is an electronic filter that passes low-frequency signals but attenuates signals with frequencies higher than the cutoff frequency.
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How to export raw audio from audacity?
Request to Experts:
I am a DSP novice. I would like to experiment with some methods for low pass filtering in the time domain.
To that purpose, I would like to generate a tone in audacity and export ...
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Which audio smoothing algorithm has the lowest latency?
I am working on an audio smoothing project. My inputs are from a record scratching MIDI controller, and I am trying to produce a smooth output (the raw data creates noisy audio).
I'm using a simple ...
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Relation between discrete wavelet transform and filter banks
I have approached wavelet transformation from a projection perspective. Specifically, we can show for a certain class of functions (in the continuous setting) that they can be written in terms of a ...
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How to measure aliasing?
Aliasing is bad, and we want good filters when downsampling. While what qualifies as aliasing is well-defined mathematically, and we can manually design filters with various tradeoffs, how do we ...
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NUFFT/FFT interpretation
I know this has been covered thousands of times on the internet, but I'm still struggling with it.
My end goal is to cancel out the noise in an Inertial Measurement Unit chip readings so I can double-...
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Why does the discrete bode plot look like the following and if possible explain the black vertical line at the end for an averaging filter
Why in the attached image for a simple 3 point moving average that has been converted into a TF (z domain) is there a wired dip? It seems that when I change the sampling time, the dip shifts to the ...
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Is this how to show decimation in the frequency domain?
Let's say I have a discrete-time signal $y[n]$ that is passed through an Ideal Lowpass Filter (impulse response $h[n]$) to get $y_1[n]$. The filter has cutoff frequencies at $\pm \pi/N$ rad.
Now this $...
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Why is implementing a digital LPF with low cutoff frequency but high sampling frequency infeasible?
I am new to signal processing and I want to implement an LPF using SciPy. In order to do so, I used the following python code from: here
The code itself looks like:
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Is it possible to lower the slope of a biquad lowpass? [duplicate]
I'd like to implement a lowpass with arbitrary slope. I'm currently cascading biquads in order to get a higher order, hence higher slope. But is it possible to do the opposite - lower the slope of one ...
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Scenarios where Phase Response is (non-)problematic
In order to get a better feel for phase response and how it applies across the field of dsp, I'm looking for example scenarios where a filter's phase response is a concern and where it is not.
Take ...
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Frequency components in Bin averaging
I am back with another question.
Context for the question: I am trying to smooth out the angular velocity data from an encoder. The encoder has 720 ppr and the rough angular speed of the wheel is ...
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Output Precision for Filter
I am designing a low pass filter which takes a 19 bit input.
\begin{align}
\log_2(\max(\mbox{LPF_COEFF})) &= 16.64\\
\log_2(\mbox{sum}(\mbox{LPF_COEFF})) &= 17.87
\end{align}
Now should I take ...
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Subtract Lowpass from Wire to Get Allpass
One technique to make a high pass filter is to subtract a lowpass filter from a wire (meaning subtract the output of the filter from the input as I have demonstrated in this post).
What lowpass can ...
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What parameter to use to decide if accelerometer data are noisy?
I would like to do regression thus I need to know if my input data are noisy.
In this case, I generate accelerometer data [x, y, z] from the sensor.
I have a look ...
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Multiple questions about how to implement practical resampling
I am learning resampling theory, and for the time being I am specifically interested in downsampling. I have a textbook that is not a dsp textbook but has a section on resampling. The way they put it, ...
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"There's no ideal lowpass filter" - really?
Sinc is $\propto 1/t$. If $x(t)$ is bounded, then there exists $t: |x(t)/t| < \epsilon_M$, where $\epsilon_M$ is machine epsilon. If $x(t)$ is also time-limited, it also means there's $\tau$ such ...
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transformation methods for digital filters
i have the following question:
in digital filter design what's the difference between the methods of transformations :
bilinear vs impulse invariance vs Euler vs step invariance.
thank you!
Here is a ...
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FIR Filter for complex float
I have been using FIR filters for float variables, can I know how can I pass complex float variables ( complex float buf[1024] ) to the filter equations? I mean pointers to the variables of type ...
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How we determine type of filter with pole(s), zero(s)? [duplicate]
Let's say we have this Laplace transform:
$$H_{1}(s)=\frac{1}{(s+1)(s+3)}\;, \; \Re{e} (s)>-1 $$
So, we know that there is a poles at $s=-1$ and $s=-3$.
With these informations, we found that to be ...
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Phase vocoder has crackling artifact
I built a phase vocoder in JavaScript based on this paper. It works great, except for a crackling artifact that is present in the rebuilt signal.
I recorded this clip if you want to hear what it ...
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LTI Filter for DAC Reconstruction
$\textbf{Question:}$ An analog-to-discrete is designed as,
$$x[n] = x_a(nT)$$
In an attempt to recover the analog signal from its samples x[n], a D/A converter is designed as ,
where $x_1(t)$ is ...
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What does frequency filtering actually practically mean?
I've got an electronics circuit which was filtering between a Passive HPF stage (1uF and 330K) and an Active LPF stage (4.7uF and 10K). As you can see it was filtering between 0.48Hz and 3.38Hz, and ...
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Trying to implement a digital LPF
So I took this digital LPF from some website:
And tried implement it first in MatLab to see how it goes, here's the code used:
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Why the sum of filter coefficients of an FIR filter does not add to 1?
In MATLAB, I have generated an FIR low pass and high pass filter of order 16. The code is as follows:
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Is f/fL a high pass filter where fL is a low-pass version of f?
Let there be a signal f and its low pass filtered signal be fL. Then what can we say about the spectrum of f/fL ?
To be specific I am obtaining fL simply using gaussian blurring in the spatial domain.
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Why are moving average filters and integrators considered low pass filters?
Why are digital moving average filters and integrators considered low pass filters in baseband signal processing applications?
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Intuition for how much ringing a particular filter will induce
I've seen figures in various books about the the tradeoff between aliasing and blurring when using a gaussian-like filter: the narrower it is, the more it cuts off low frequencies and thus blurs it, ...
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Why is LPF amplifies the baseband signal?
I am working with a PDM digital microphone and I've sampled the data using a FPGA.
For the beginning I am working in python to do all the algorithm, then I will implement it in FPGA.
I am using two ...
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Derivation of a non-ideal low-pass rectangular windowed FIR filter
I have been trying to understand certain aspects of FIR filter design which have frankly annoyed me for some time such as exactly why the critical frequency $\omega_c$ in a low-pass FIR filter is ...
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Why couldn't I reapply a LPF to remove more noise?
I have a piece of raw accelerometer data consisted of 10,000 samples to be processed offline. Since the movement or rotation of the object is moderately slow and smooth, I would think that I can ...
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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 ...
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low pass filtering for smoothing
I am using the following code in Matlab, from some source (sadly I cant remember).
The code is used for smoothing signals using a low pas filter.
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Deriving the impulse response of an ideal low-pass filter
The impulse response of an ideal low-pass filter can be determined by setting $H(\omega)=1$ in the Fourier-representation $$h(n) = \frac{1}{2\pi}\int_{-\omega_c}^{\omega_c} H(\omega)e^{j\omega n}d\...
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FIR-filter output gain
Lately, I've been doing some FIR-filter simulations as a part of a programmatic circuit design course. One task was to use different constants $b_i$ for the multiplying stage of the filter and see how ...
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Why 5Hz aliasing with LPF implemented
I have a simple signal chain with a differential sinusoid as the input. This passes through the first amp which converts the signal to a single ended signal. The next block is a 4 pole low pass ...
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How to apply a filter that prevents aliasing when reindexing a dataframe to a new datatime index
I have two dataframe and I want to reindex the second dataframe to the index of the first dataframe by downsampling it. This however creates alliasing and as a result an artificial flattening of the ...
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digital low pass filter time constant and the number of datapoints acquired
In digital lock-in, one can set the time constant, $Tc$, which determines the frequency cutoff.
Given the sampling rate of your data acquisition tool, does the $Tc$ of your digital implementation of ...
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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 ...
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How does a CIC filter output meet Nyquist?
In this image from Understanding cascaded integrator-comb filters
The bottom image shows the output response, with the entire range as passband:
Even the CIC compensation filters are all passband to $...
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I just computed an DFT and FFT of a sinc function, but when I visualise them the results are strangely different? what am I missing
The idea here is to improve a Sinc function by manipulating its Fourier transform.
As you can see, the tophat has ripples in the bands of interest and I want to eliminate them by smoothing them with a ...
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Matching outputs of FIR filters based on time domain convolution method and overlap-save method
I'm trying to match(to the best extent possible) the outputs of FIR filter based on Frequency domain overlap-save method and time domain convolution method resp.
I was able to get most of the output ...
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Low-pass filtering and computing integral spectral power of non-uniformly sampled data
I'm not too experienced in signal processing, so I apologise in advance if I get some of the basics wrong.
I have some data (human behaviour) collected over multiple months. The data is sampled non-...
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Why does applying fir filter in R (gsignal package) increase the magnitude of all values?
I am new to signal processing and I am sure this is a very rudimentary question. However I have done various searches and I cannot find an explanation (possibly because my knowledge is insufficient to ...
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Decimate a signal whose values are calculated, not stored
tl;dr I'm looking for something like Scipy's decimate function, but one that takes in a generator, rather than a Numpy array.
I am turning on and off a light using ...
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Decimation filters: IIR vs. FIR
I just read this article on decimation filters. In particular, in section 2.3.3 it is strongly suggested that FIR filters are advantageous, because you only need to calculate every $M$th sample, where ...
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Interpolating Filter for Quadrature Detector with one ADC
Single ADC is sampling at 16000 Hz the I and Q outputs of quadrature detector. So each I and Q signal is sampled at 8000 Hz, with phase offset 90 degrees between them. Bandwidth of interest is 3400 Hz....
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Research on SDR security Gnu Radio and legal transmission with filters
In fact, I am doing research around SDR and possible dangers. Then I stumbled upon the most well known "replay attack". Via Black hills, I came across the following diagram in gnuradio:
The ...
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Cascading IIR filter with All pass filter (APF) for constant group delay and linear phase over the passband using MATLAB not working?
I've designed an IIR Low pass filter in MATLAB with the following parameters.
...
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Low pass filter behavior in frequency domain
I have a relatively simple task, which is to implement a low pass filter for a sum of exponentially dampen complex sinusoids in random Gaussian noise.
To showcase my problem I have generated data ...
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Does LPF limit the maximum sampling frequency of the input signal? [closed]
Consider the system where the signal passes through analog LPF with cutoff frequency of 50 Hz, so the maximum sampling frequency of the input signal could be 100 Hz (according to the theorem). But can ...
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