In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

learn more… | top users | synonyms (1)

0
votes
0answers
22 views

Can we eliminate transient response by pre-calculating predicted transient response for some IIR digital filters?

So for IIR digital filters that have rational transfer function (assume bilinear transformation and low-pass filter), transient response can be calculated easily for inputs, because each input is ...
0
votes
1answer
26 views

Minimum Energy Delay Property

I have 4 systems which are represented in z domain like this: Where I know the poles and zeros. Then I compute Partial Energy Sequence of impulse response of all these system using this formula: ...
0
votes
1answer
33 views

Slight crackle with overlap-add frequency-domain convolution

I'm trying to make an FFT audio filter using overlap-add but there's some slight rhythmic distortion/crackle in the output. I have an irSize (impulse response size) of 512 samples and a blockSize of ...
1
vote
1answer
54 views

What is the low-pass filter that gives minimal transient response?

As title says, what is the low-pass filter that gives minimal transient response? By minimal transient response, I mean minimal settling time with reference to very tiny amount of divergence of ...
0
votes
0answers
39 views

MATLAB, recover a signal that has a cosinusoidal interference and that has been AM modulated

I have a signal which has been AM modulated but the channel also added a cosinusoidal interference A cos(2 * Pi * f0 * n * Tc + Phase), at the same ...
1
vote
1answer
104 views

What do we espect Likelihood function used for?

In my understanding, I let make a example. For example, you can see in the following picture. Consequently, We want to find ^sigma. and We have already known the observation data(which is random ...
0
votes
1answer
32 views

Filtering at frequencies above sampling frequency in Equalizer

I've created an equalizer in C++ by using 6 peak filters to attenuate or amplify at that particular frequency. They are 60 Hz, 170 Hz, 600 Hz, 3000 Hz, 8000 Hz, and 16000 Hz. When I try to run an ...
0
votes
0answers
11 views

Gaussian filter at different order?

What effect does the Gaussian have at different order.. I mean comparing a Gausian Kernel with at 2. order Gaussian kernel.. What benefits, or when would it be appropriate to take the derivative.. ?? ...
1
vote
1answer
34 views

FIR filter phase response

With reference to the diagram below, is it correct to say that at the pass band the phase response is linear. It also remains linear in the stop band but it get chunks of π. These jumps happen when ...
0
votes
0answers
21 views

time domain non stationary time series individual colored noise identification algorithm

One of the methods for identification of modulated colored (stationary) noises is that of lag 1 autocorrelation power law noise identification algorithm. This method can determine the dominant noise ...
0
votes
2answers
30 views

How to determine the filter type of a discrete frequency spectrum

I have the discrete impulse response of a filter and i want to determine which kind of filter it is by using the DFT. The impulse response is: h0 = [0,1,2,1,0] ...
0
votes
0answers
35 views

FFTW - Inverse FFT input data

I am trying to use FFTW function for IFFT: fftw_plan_dft_c2r_1d(size, out_cpx, out, FFTW_ESTIMATE). My goal is to input desired frequency response data and ...
1
vote
1answer
38 views

FIR Filter Design - obtaining FIR coefficients from frequency response

I would like to obtain FIR filter (low-pass) coefficients from a given frequency response by using C++ to develop algorithms. I have read at http://iowahills.com/B2PolynomialFIRFilters.html, that, if ...
0
votes
2answers
37 views

FIR Filter phase shift and stability

I designed an FIR highpass filter which is causing a phase shift to the output. The output signal is also not stable (?) on start. Are these normal or am I doing something wrong? If these are normal, ...
0
votes
0answers
28 views

FIR Filter Design - from slope and cutoff frequency [duplicate]

I am stuck with the following design requirements for FIR filter: Low-pass -6dB @ 1.6kHz Slope: 24 dB/octave. Ideally I would like to get also FIR coefficients, but any help how to design this ...
3
votes
2answers
49 views

Create Matched Filter Reference Signal when Scaling Unknown

I am working on a problem which raised what should be a simple question, but I'm stuck on it. The objective is to detect a shifted and scaled version of a known signal in the presence of Additive ...
0
votes
2answers
94 views

FIR Filter Design - input parameters conversion

I am new to DSP and filter design. I have developed a code in C++ to calculate FIR coefficients using Parks-McClellan algorithm. The inputs to calculations are: Filter type (Low-Pass, High-Pass) ...
0
votes
3answers
46 views

MATLAB Haming window

I have been trying to achieve a haming window length of 23 using the equation. I tried several coding, one of the is the one below: ...
2
votes
2answers
55 views

How do can we cancel out noises while listening voice on headphones?

It's a common problem that while talking with the another person on Smartphones with the help of regular headphones,we get very annoying sounds at background . The opposite person's voice gets mixed ...
1
vote
1answer
22 views

Filter impulse response order calculation

I want to generate the following periodic sequence h[n]={1,2,3,4,0,0,0,0,1,2,3,4,0,0,0,0,1,2...}. I am planning to generate this as an impulse response of a filter with an order no more than 8. I ...
0
votes
1answer
38 views

Help choosing coefficients for a simple FIR filter to extract DC component

I am a beginner in design of digital filters... I have just completed Coursera's Linear Circuits course! I have a signal which is sampled at 50 Hz where the DC component has a value of 800000 and ...
0
votes
1answer
30 views

What is sound output delay of a band pass filter at different freqencies?

I measured that a 2-pole band pass filter, given imput audio of different frequencies, outputs an audio signal with different time delay. At 440Hz, the time delay between of the input audio and the ...
1
vote
2answers
43 views

Find the Differential Equation of an Analog Band Pass Filter

I don't even know how to start. We're supposed to find the differential equations that links $V_{in}$, $V_{out}$, and $V_{io}$, as well as the values of $C_1$, $C_2$ and $R$ so the filter works in the ...
0
votes
1answer
18 views

transient response of simple digitalized RC low-pass filter

In http://en.wikipedia.org/wiki/Bilinear_transform#Example, digital version of simple RC low-pass filter is presented: $$\frac{1 + z^{-1}}{(1 + 2RC / T) + (1 - 2RC / T) z^{-1}}$$ where $T$ is ...
0
votes
1answer
56 views

digital low pass Butterworth filter

A digital low pass Butterworth filter that has been designed using Bi-linear transformation has been a pole at $z=0.6$. It is also known that the filter's attenuate (at digital frequency) $\omega = ...
0
votes
2answers
42 views

Has this 2D filter for enhancing circular dots in images a name?

I came across this 2D filter for enhancing circular dots in images, for example to enhance a dot with a diameter of 5 pixel, the filter is: $\frac{1}{336}\left( \begin{array}{ccccccc} 0 & 0 ...
1
vote
0answers
30 views

How to apply a Butterworth filter to data of varying sample rate?

I am trying to apply a Butterworth bandpass filter to accelerometer data of my smartphone. However, the accelerometer samples I receive do not come at regular intervals. Sample frequency varies ...
0
votes
2answers
36 views

Filtering Random Signal

My question is easy one actually. First, I generate a random signal using randn() function of MATLAB like this: Then, I design a FIR filter of order 200 of ...
1
vote
1answer
36 views

What is the cause of slow roll off of windowed filters

Exactly why using the of windowing function, such as Hamming or Blackman, causes a slower transition to stop band? Everywhere it is mentioned just as an obvious fact that this happens but I haven't ...
0
votes
0answers
29 views

Filter signal1 from evolving frequency response of signal2

I'm a composer (of music) starting out with signal processing. I'm also reasonably proficient in C at this point. I'm not even sure if what I'm trying to do is possible, because I can't yet find an ...
0
votes
2answers
59 views

Median filter algorithm in opencv

I was wondering about the implementation of the Median Filter in opencv because I am suffering form its slowness. So, I intend to implement another one on my own but first I want to make sure about ...
0
votes
1answer
73 views

FIR filter coefficients from frequency responses

I have frequency responses of FIR filter of order 13 from obtained 50 frequency samples. How can I get FIR filter coefficients of order 13? For example, in MATLAB a lot of possible ways I tried ; I ...
1
vote
5answers
141 views

Noise Removal from an image using Opencv

I tried removing noise from the image shown below using Median Blur in OpenCV. But i'm not able to remove the colour noise completely as it is done in Neat Image. Any suggestions.? 1. Original Input ...
0
votes
0answers
7 views

Determining if an unknown signal has “converged” or become quasi-steady

Relevant background: I am trying to simulate unsteady flow in a converging-diverging nozzle, and trying to see how the flow shedding downstream of the nozzle affects the wall shear stress ...
0
votes
0answers
25 views

system identification with delayed desired signal

I have a white noise data for 10 sec ( with 2 min silence or zero data at start and end)that is captured by sensor after delay t1( which I dont know). same data is captured after delay t1 and t2 by ...
1
vote
2answers
49 views

Wiener filter optimal coefficients and the input signal

I have this question: "Show that for the optimal filter coefficients the error $e[n]$ is orthogonal to the input signal $w[n]$." anybody have an idea?
0
votes
1answer
71 views

Applying Gaussian Blur in frequency domain

Please see second update I'm trying to implement gaussian blur in frequency domain using c#. Those are steps that I'm doing: Load bitmap. Split bitmap into separate color channels. Convert every ...
0
votes
1answer
36 views

Filter wrap-around to avoid dealing with settling time? Using a Butterworth

I've been using a Butterworth high pass filter to correct the low frequency distortion of an acceleration waveform integrated to a velocity waveform. I throw out a small number of the first data ...
-1
votes
1answer
68 views

Spectral leakage issue cannot solved using windowing

I am designing a frequency lowpass filter using FFT&IFFT. My test data is a time series with a large DC component (the pink curve). Before the FFT, the time series was detrended (curve fitting) to ...
0
votes
2answers
47 views

Melody/Vocal extraction

I'm new to the whole DSP area, but I've built a program which plays music and does FFT to produce some cool visuals for the music in realtime. The FFT algorithm I chose was from a C# library called ...
0
votes
0answers
9 views

implementing bandwidth filter in R

I am reading this "how to" in R: http://www.inside-r.org/howto/time-series-analysis-and-order-prediction-r I am confused about the tutorial's implemented band pass filter. In particular, here's the ...
0
votes
2answers
93 views

Are SOS structures better for linear phase FIR filters?

I've seen many discussions suggesting that SOS's are often/usually the preferred structure for fixed point hardware filter implementations, since they preserve conjugate pairs, etc., but I've never ...
0
votes
1answer
30 views

transfer function of wiener filter

I am trying to understand the function of a Wiener filter. I get that the Wiener filter minimizes the mean square error between the estimated random process and the desired process. Does this mean ...
0
votes
0answers
15 views

Wavelets and filters?

I've tried using the search function to answer this question, but haven't found any explicit answers. I'm trying to study Gabor and log-Gabor filters. Most of the content I find is on Gabor wavelets ...
0
votes
1answer
50 views

How to account for sample rate in a FIR Digital Filter?

As a learning exercise I am trying to simulate a simple notch filter with Matlab. For test purposes I use Matlab to simulate sinewaves of various frequencies and plot frequency vs. gain so I can see ...
1
vote
0answers
23 views

scaling issue when using scypy's firwin2 to get coefficients for a filter

I am trying to use SciPy's firwin2 command to get the coefficients that correspond to a filter of my creating. I took white noise in Audacity, applied an ...
0
votes
1answer
32 views

Tradeoff between Filter Order and Sections

I have designed a bandpass IIR butterworth filter with desired response using the MATLAB FDAtool with the configuration shown in the figure below Since this is multistage implementation, when I ...
0
votes
2answers
41 views

what is the theory behind matlab butter function?

I would like to know the theory behind matlab butter function. Is there any book which gives clear idea on how it works. In future I would like to calculate the coefficients manually.
0
votes
0answers
9 views

Gabor filter bank for gist features

I was reading a paper about Gist features. I was interested about its code implementation which i found it here). Into LMgist.m there is a function called createGabor, this is the code: ...
0
votes
1answer
45 views

1D filter for speed input and noisy position sensor

I was tasked with designing a filter to smooth out a translating table. The system, we command a speed and a desired position, and measure it with two noisy position sensors. The commanded speed is ...