filter-design's questions - English 1answer

701 filter-design questions.

I was planning to develop Android / iOS applications that enable users to measure 3D length using their smartphones. According to this question, you need to know at least the time-varying bias that ...

What happens if the noise has no zero mean? I mean, if the exercise is something like: $$y(k) = A x + \eta(k)$$ When I have zero mean, I start from: $$y = A x$$ $$\Rightarrow \hat{y} = A \hat{x}$$ ...

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 ...

User-definable functions may be used, instead of the predefined frequency response functions for @fresp. The function is called from within cfirpm. I 'm using the handle function to adding the group ...

For a given narrowband Gaussian filter with a specific central frequency and filter width, I need corners of a bandpass Butterworth filter whose amplitude response is close enough to the Gaussian ...

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 ...

I'm looking construct a stable pole-only filter where the feedback coefficients start with a block of zeroes, i.e. \begin{align} a_0 &= 1\\ a_i &= 0, \textrm{ for}\ 1 \le i \lt k\\ a_i &\...

I hope to design 1st order highpass filter from transfer function. In the example of 1st order lowpass filter, I first get the coefficients of numerator and denominator in the variable 'b' and 'a'. In ...

I understand to an extent various filter like low pass filter, high pass filer, Wiener filter Kalman filter etc. I also understand some of this filter will decorrelate/uncorrelate the signal. The ...

Could you please introduce me some references about edge detection of 1-D signals? I have a pulse similar to rectangular pulses which is immersed in noise. Would you please introduce me some filters ...

Two 1st order filters : ...

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 ...

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...

If a practical filter can't remove all unwanted frequency components like ideal filter, does that mean unwanted frequency component are still present after filtering? how can we use such distorted ...

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.

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 "...

I noticed, when I try to fit filter coefficients to a given complex transfer function with the output error method, implemented for example in the MATLAB function ...

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: ...

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 ...

I'm working with signals with variable sampling rate (the time space between samples is not constant). I know the delay between samples but I don't wont to interpolate the signal. Is there a way to ...

I come from Computer Science so please pardon for my possibly wrong terminology. I need to design a filter which has coefficients $$h_0, h_1, \ldots, h_n, \ldots \quad\text{such that}\quad h_0 > ...

I'm trying to apply a filter to an audio signal in MATLAB and having some trouble processing it. So far, I have a transfer function that describes a K-weighted filter, and I am able to create a bode ...

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 ...

I have a noisy signal with jumps to high value and back to normal for limited time bins (40 for example) which I want to filter to have a smooth signal with no jumps. Here is the graph, the blue one. ...

The wavelet transform has a problem as it gives poor time resolution for low frequencies and poor frequency resolution for high frequencies according to uncertainty conditions. This appears well ...

I'm trying MIM (Magnitude Invariance Method) and PIM (Phase Invariance Method) for to improve biquad LPF response at low sampling rates. I'm looking some help and examples of usage if available. ...

It's a beginner question, but useful to users from python - signal.lfilter, I was using lfilter from Find reverse one pole ...

I know this may be a kind of basic question, but I have my head wrapped over it for some time and hadn't found a solution. I want to generate the frequency response of a 4th order butterworth filter ...

I have studied convolutions and filters a long time ago. Today, I am trying to review the basics using some notes of mine, but I am finding difficult to solve easy problems. Since I don't have ...

I'm wondering if we can measure 3-D length only by our smart-phone's accelerometer. And we all know these low cost IMUs are not accurate. You can model accelerometer's error this way: $$ a = f*a' + g ...

Where to learn about "analog prototype filters"? I've heard about them, but I'm unsure about what they really are and how they're constructed.

I ran a finite-difference simulation and the behavior of an output signal, $s$, in time, $t$ (sampled with period $\Delta t$) behaves approximately as in the figure below. It is well-described by a ...

i have downloaded magdwick fliter AHRS written in C: http://x-io.co.uk/open-source-imu-and-ahrs-algorithms/ // Reference direction of Earth's magnetic field hx = mx * q0q0 - _2q0my * ...

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}}| \...

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 ...

I have to filter eeg data (500 samples) offline using a FIR bandpass filter for alpha rhythms (8-12 Hz) providing the fixed order of filter (i-e., 128). Can some one explain what should be the ...

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 ...

I would like to implement in software the same isolators that the Pioneer EFX-500 features: The manual says the following about the isolators: High-performance 3-band Isolator Function [Fourth-order ...

I have images with systematic noise caused by the instrument in different subdomains of the image, and I'd like to design a filter to - on average - improve the quality of the images. Since the noise ...

@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 ...

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 (...

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 ...

I would like to approximate a moving average filter with an IIR filter of much lower order than the tap-length of the moving average filter. Optimality shall refer to the $L_2$ norm of the impulse ...

I am looking for methods to enhance noisy images, where: some pixels in the image are very noise, some other pixels do not contain so much noise. My first thought is to build an adaptive Gaussian ...

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 ...

suppose I know the frequency response of a (linear) model approximating a real physical system but only at a specific frequency $f_0$ (so basically I have a complex number whose module is the ...

I have designed a 6 order low pass filter of 5 Hz whose baseline (dc level) is set at 3.3Volt but its output baseline changing very often, sometimes it even reaches 4 volt. Why is it so? As expected ...

I have raw values that are coming directly from a sensor (on the fly). These values are coming from a wearable sensor installed on a cows neck. When the cow ruminates (meaning that it chews food that ...

I'm hoping to use an implementation of a frequency-locked loop for rough frequency synchronization in a PSK31 demodulator. The approach is to define a filter that is the derivative of the matched ...

Question 1 $$ H(e^{j\omega})=\sum_{n=0}^{N-1}h[n]e^{-jn\omega} =\mathbf{c}^H(\omega)\cdot \mathbf{h} \tag{1} $$ $$ =\mathbf{h}^H\cdot\mathbf{c}(\omega) \tag{2} $$ $$H(\mathbf{h})=\sum_{k=1}^Kh[k]e^{...

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