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.

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Implementation of basic 2-pole low-pass filter

I know this question has been asked to hell and back from many people, and that it will probably get downvoted until it's deleted, but please bear with me. I am designing a sound engine for one of my ...
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49 views

voice with a random noise redction - MATLAB

I have this assignment that I can't figure how to solve. I have a sound.wav human-voice file, and a matlab add-random-noise (monotonic) Goal: remove the noise from the sound (and the sound from the ...
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DSP MATLAB example question about notch filter!! [closed]

![this is the example question from the dsp using MAtlab edition 2, using the sources I have it was quite hard to answer it correctly. Needs you guys help!!][1] [1]: ...
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60 views

Step response overshoot in IIR filters

Is there an analytic expression by which it is possible to calculate overshoot of step-response of second order IIR digital filter? I have tried to search through Mitra's signal processing book and ...
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70 views

bandpass implementing with DWT in matlab

In matlab help in find the implementation of bandpass as following ...
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1answer
120 views
+50

'Best fit' motion curve for an unachievable profile

I have a discrete function that represents a 1d position in space over time. I have a motion system that would like to achieve this curve, but, due to constraints on jerk (3rd derivative) and ...
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1answer
103 views

DC blocking IIR filter clipping?

I'm processing audio data for voice input from a mic. The data arrives in 32 bit floats [-1 ~ +1]. My first filter is to remove DC: ...
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2answers
93 views

Understanding matched filter

I have a question about matched filtering. Does the matched filter maximise the SNR at the moment of decision only? As far as I understand, if you put, say, NRZ through a matched filter, the SNR ...
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1answer
64 views

Conversion between DCT to DFT frequency domains

It is well known that the DFT can be interpreted as a filter bank. The same can be done for the DCT, for instance Fig. 3.29, pag. 129 in [1] shows a plot of the frequency response of a DCT filter ...
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73 views

What is the most efficient type of software bandpass filter?

I'm writing a C++ application to extract four modulated carrier signals at four separate frequencies, from a single input signal. I've designed four FIR filters at t-filter.appspot.com but they run ...
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23 views

algorithm for audio and video separation using matlab [migrated]

sir i want to design an algorithm which can take a movie clip or a video signal having both audio and video and extract the audio and video part separate using the help of filters, matlab and signal ...
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3answers
55 views

How to tell the effect of kernel of an image filter (matrix) on an image?

I am going through an old exam paper and trying to understand this question where they ask us to tell them what effect the following image filter would have if its convolved with an image: ...
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58 views

Simulink continuous to discrete question

I'm new to Matlab/Simulink and would like to convert the continuous system below to discrete for eventual conversion to C code. Matlab has a c2d function, but I don't know how to apply it to a ...
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68 views

Delta-limiting filter

Consider a filter which tracks $x_i$, clipping $|x_i - x_{i-1}| >$ some limit, say 1: ...
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5answers
210 views

Why do we use window in time domain rather than do FFT modify the spectrum and than inverse FFT

I thought that DSP would be done by using FFT of portions of a signal, modify the samples that result from FFT (since they represent the spectrum of our signal + noise) and remove any unwanted signals ...
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3answers
209 views

Why does adding a time delayed version of a signal to itself create a filtered signal?

I was asked this question and couldn't come up with an answer on the spot which didn't involve the frequency domain (basically that the co-efficients of the delay sequence are the impulse response of ...
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29 views

Signal level estimation, matlab

Three methods of estimating maximum signal level at a node: Worst Case Min-Max (L1-Norm) Mean Square (L2-Norm) Frequency Domain (Linfinity-Norm) In matlab i found the code for L1, L2 ...
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1answer
63 views

Direct Form II , IIR filter

I have read in my textbook and i don't understand this part, that the Direct Form II implementation of an IIR filter is a bit tricky in memory addressing.There is need of clever memory addressing. ...
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3answers
109 views

“Subtracting” one audio stream from another audio stream

I am a long time software engineer but have practically zero experience with signal/audio processing. I am interested in learning about signal processing via a use-case we have for one of our audio ...
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1answer
26 views

Second order sections and normalized factor

I use matlab function tf2sos to convert digital filter transfer function data to second-order sections form: [sos,g] = tf2sos(b,a) Taken from the help menu of the matlab: G is a scalar which ...
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1answer
65 views

Feature Selection by Filtering

so there is this paper I'm reading and trying to understand fully: Towards Practical Identification of HF RFID Devices http://dl.acm.org/citation.cfm?id=2240276.2240278 I don't want to link the PDF ...
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1answer
47 views

How to find the normalized coefficients?

I use the below matlab function to design the unquantised coefficient set: [b,a]=ellip (4,2,60,2*[0.2,0.3]); and then i use the matlab finction "tf2sos" to convert the direct form coefficients ...
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2answers
72 views

Help designing Butterworth filter

I am trying to bandpass filter a signal using a butterworth filter, but I am getting the following result: http://oi46.tinypic.com/3450ps5.jpg The original signal is the blue and the filtered one ...
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1answer
94 views

Hilbert transform filter for audio applications: Using IIR half-band parallel all pass structure

Does anyone have any experience designing wideband IIR Hilbert transform filters with audio applications in mind? I am using the filter for single side band modulation audio effects discussed in this ...
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2answers
72 views

Filter order vs Number of taps vs Number of coefficients

I'm learning DSP slowly and trying to wrap my head around some terminology: Question 1: Suppose I have the following filter difference equation: ...
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1answer
42 views

Housekeeping operations in filters

I have to implement a second order filter in DF1 or DF2. I should avoid housekeeping operations by suitable addressing strategies. I don't understand what "housekeeping operations" means. Anyone ...
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2answers
69 views

Interpolation vs Interpolation Filter?

Hi guys, I've been reading some papers on - how to remove ghosting artefacts from the Fourier Slice theorem applied to a 3D discrete image volume. The papers mention that in order to remove ghosting ...
3
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1answer
43 views

How to filter pulse like jumps in a signal?

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. ...
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1answer
45 views

Detection of regions of voice in music

I want to detect the regions of voice in a song. Currently, I am using the fact that the vocals in the song are recorded as center panned. So, I am removing all center panned audio from the song. ...
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2answers
76 views

IIR Filter structure implementation dynamic range

I want to implement an IIR filter. The filter is going to implement by cascading second order sections. Below there are 3 pictures that implement this filter using: Denominator - Numerator using ...
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2answers
128 views

Pole-Zero Plots Diagram Explanation

I experiment with the coefficient quantization in an IIR filter. I change some values like the passband ripple and also how many bits i want for the quantization method. In the first picture i have ...
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1answer
47 views

what is window size?

i am working on simulation of an article "A Real-time ECG Data Compression Algorithm for a Digital Holter System". This algorithm has five steps "Difference=> ECG Signal=>Classification=> DCT=> Window ...
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2answers
74 views

What is the difference between a simple gaussian filter and gaussian filter multiplied by its sum of it elements?

1 2 1 1 2 1 2 4 2 (A) (1/16) * 2 4 2 (B) 1 2 1 1 2 1 Both matrices are the same ...
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2answers
64 views

Types of rounding in coefficients quantization

Suppose we have create an IIR filter with matlab function "ellip", and then we want to quantize the coefficients using: \begin{align*} bq=Quantize('round',b,2^8); \cr aq=Quantize('round',a,2^8); ...
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1answer
35 views

Fill matlab “ellip” function through transfer function magnitude response

The matlab "ellip" function can be used to design the unquantised coefficient set. From matlab website: $ [b,a]=ellip(n,Rp,Rs,Wp) , $ n: order of filter Wp: normalized passband edge ...
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42 views

object level sensor fusion for multiobject tracking

I want to fuse objects coming from several sensors, with different (sometimes overlapping!) fields of view. Having object lists, how can I determine whether some objects observed by different sensors ...
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38 views

Intensity change after filtering this image?

I need to filter the image raster below with the following mask: 0 1 -1 0 and the Image Raster: ...
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1answer
75 views

information filter instead of kalman filter approach

I read many sources about kalman filter, yet no about the other approach to filtering, where canonical parametrization instead of moments parametrization is used. So I would like to learn on examples ...
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48 views

How to implement structures for difference equations?

I want to learn how to design structures that could model difference equations. I don't know how we call this things but i can show a picture: For example i want to understand the procedure that ...
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1answer
57 views

How to show that this filter is a HP filter

i got a system with transfer function given by: $$H(\omega)=1-e^{-j\omega}$$ I already plot it, and that seems to be a periodic function with $H(0)=0$, $H(\pi)=2$, , is that enough to show that this ...
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1answer
57 views

Determine the type of filter from the transfer function

i got that system $x_n \to x_n-x_{n-1}$, so $h_n=[.....,0,1,-1,0,...]$, with $h_0=1$ and $h_1=-1$, so the transfer function given by: $$\sum_{i=-\infty}^{\infty} h_ne^{-jwn} = ...
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34 views

How to find the group delay of IIR filters in Python/SciPy

I sent a square wave signal (red) through a IIR filter (butterworth in this case), to make it more "realistic". Then there's a slight group delay in my output signal (blue). Is there a way to find out ...
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1answer
53 views

How can frequency dispersion in a vibrating string be modelled using an LTI filter?

According to JOS on stiff-string synthesis, stiff strings (like on a piano) introduce inharmonicity (i.e. the harmonics of the tone are not all in tune) due to different frequency components of the ...
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2answers
160 views

Simulating analog filter using convolution or converting with fft

My task is simple; I want to simulate analog low-pass filtering of an input signal, using Python. Note that the input signal is an array of values, not an analytical function. My first question is if ...
4
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1answer
74 views

Dynamically changing cut-off and sampling frequency of a digital filter

I have designed a low pass filter that smooths the output coming from an accelerometer attached to a vibrating machine. I designed this assuming cut-off frequency $f_{c1}$, and sampling frequency ...
3
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1answer
52 views

difference between Bilateral and Gaussian Filter

I would like to know the basic differneces between Gaussian filter and Bilateral filter in digital image processing... I just know that both are used to smooth out an image but I would really ...
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2answers
128 views

How to separate the upwards-propagating from the downwards-propagating waves?

I've got a real 2D image of propagating waves in both directions. How can I separate this image into one with the upward-propagating waves and one with the downward-propagating ones? This example has ...
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3answers
110 views

Is this system an FIR filter or an IIR filter?

Consider this system: y[n]=y[n-1] + x[n] It can be verified easily that this system has a finite impulse response (by putting x[n]= delta[n] i.e impulse, and ...
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88 views

Goertzel algorithm: Relationship of magnitude

I've written a quick test app that uses the Goertzel algorithm to determine if a given frequency is present in the signal. This is to pick up DTMF tones and various other signals. The app appears to ...
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85 views

How do I find the Hilbert transform (qudrature) of a discrete signal by convolution with a Hilbert kernel?

Given a discrete real signal $ f_{re}(t) $ the (complex) analytic signal is given by: $ x(t) = f_{re}(t) + j*f_{im}(t) $. I want to calculate $f_{im}(t)$: the quadrature by convolving with a Hilbert ...

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