EMG analysis.pdf
Band pass filtering removes low and high frequencies from the signal. Low frequency cutoff of band pass filter removes baseline drift sometimes associated with
AUDIO SIGNAL FILTERING WITH LOW-PASS AND HIGH-PASS
The audio signal's noise can reduce by using filtering methods and in this system high-pass and low-pass filters which are designed in Matlab with the sampling
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Feb 15 2015 In this paper FDA tool from Matlab mathematical computational package with digital signal processing tool- boxes is used to design filter ...
Chapter 1 Digital filters
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EE213 Lab 9: Type I Chebyshev High Pass Filter
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• Use Matlab to design a four pole elliptic analog low-pass filter with 0.5dB a 3-pole Chebyshev high-pass analog filter with cutoff frequency fc = 5KHz.
Lab 8: Filters in MATLAB ReadMeFirst
Recall that the cut-off frequencies are 1/RC in radians per second for both high pass and low pass filters. Also recall that the cut-off frequencies occur at -3
Design and Implementation of Low-Pass High-Pass and Band-Pass
15?/02?/2015 Digital Filters FIR Filter
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Key Words: Digital Filters DSP
EMG analysis
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In addition several types of filters can be designed such as lowpass
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work we implemented a digital FIR high pass filter using MATLAB program. (FDATools) using sampling and windowing methods then the design in the FPGA kit.
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ii). High Pass Filter iv) Band Reject Filter. 9. Write a MATLAB program to design following Chebyshev-I filters. i) Low Pass Filter iii) Band Pass
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In this paper FIR low-pass filter designed by MATLAB software is used to realize low-pass filtering of audio files and the high frequency is filtered to
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Active High-Pass Filter For each of the configurations you will Design the filter for a specified cut-off frequency Model the filter in MatLab Simulate the design with PSpice2 and Test the design in the Lab Steps 1 2 and 3 are Prelab activities Step 4 will be performed during the lab period
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Refs: Matlab Signal Processing Toolbox help; Hartmann notes Chapter 7 15 1 Introduction Filtering a time series means removal of the spectral power at some chosen frequencies while retaining other frequencies A high-pass lter retains higher frequencies while removing low frequencies; a low-pass lter does the opposite
Lab 4 Band Pass and Band Reject Filters - University of Texas
band-pass filter by cascading the high-pass and low-pass filters you designed in Lab 3 Then you will design and test a high-Q band-pass filter and a band-reject also called notch filter As a result of performing this lab the student will • demonstrate the function of band-pass and band-reject filters and
Low-Pass Filter - MATLAB & Simulink
Ahigh-pass lter retains higherfrequencies while removing low frequencies; alow-pass lter does the opposite band-pass lterremoves all frequencies outside a prespeci ed band Filters are widely used for digital signal processing (DSP) as well as timeseries analysis
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High pass filters Download the M-files IIR_Low m and IIR_High m from the ECE 2050 website and run them in MATLAB Now plot your IIR filter on the Z-plane using zplane(NumZDenZ); NOTE: In the Discussion Questions you are asked to sort out which poles and zeros belong to which stage of the analog filter using the transfer functions for each of
AUDIO SIGNAL FILTERING WITH LOW-PASS AND HIGH-PASS FILTERS
Figure 2 Design Low-pass Filter in Matlab 3 2 High-pass Filter The high-pass filter is passed high frequencies above the cutoff frequency and it can be used to remove low frequency content from the audio signal The process of high-pass filter and the design in Matlab are shown in Figure 3 and 4 Figure 3 Observation of High-pass Filter [4]
Digital Filter Design Using Matlab By Timothy J Schlichter
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BUTTERWORTH 4/19 •&To&design&the&nth&order&BuPerworth&?lter&use&butter&command:& [B A] = butter(N Wn ‘ftype’ ‘s’); Where*Nis?lterorder& Wn[ rad/s
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Both the low pass filter and the High pass filter were made using the function called “designfilt” in MATLAB All the parameters including stopband filter order design method passband weight stopband weight and the sample rate were set The user has the option to control the passband
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The high pass digital filters are used in image processing It is used to avoid an amplification of dc current which can makes amplifier to not work properly They are also used in rc phase shift and given input for the ac coupling vi RTL View: Fig 3 2 simulation result for high pass filter
Searches related to high pass filter matlab filetype:pdf
Use MATLAB to create three magnitude Bode plots for the high pass filter For the first and second plots use the transfer function of the circuit with the nominal and measured values from Table F respectively and a frequency range of 10? rad/s ? ? ? 20000? rad/s For the third plot use the empirical data generated by the Bode plot VI
What is a low-pass filter in MATLAB?
- Low-Pass Filter Design low-pass filters using MATLAB A low-pass filter is a filter that allows signals below a cutoff frequency (known as the passband) and attenuates signals above the cutoff frequency (known as the stopband).
What is a high-pass filter?
- High-pass filter? ?? ???? ??? ????, ??? ???? ?? ???? ????, ??? ????? ???? ??? ??? ???? ????, ??? ????? ???? ??? ?????? ???? ??? ???? ? ?????. ?? ??? high-pass filtering.
How to design a filter in MATLAB?
- In practical life, we use LRC circuits to design these filters. In MATLAB, we can design them in different ways. We can see the response of any filter by designing an LRC circuit in MATLAB Simulink, or we can just give a signal to a built-in filter in Simulink and use the scope to see it.
How do I know if a signal is filtered in MATLAB?
- The waveforms show the filtration clearly but we can also listen to the music to get the difference. To listen to the music in Matlab, we execute the sound command sound (xn, fs) for the noisy signal in the command window. We also do the same for the filtered signal sound (y, Fs).
Design of FIR Low-pass Filter Based on MATLAB
Weiwei Tian
a, Zixuan Ni b, *Lanzhou University, Lanzhou 730107, China;
a weiweitian96@163.com, bnzx18069832761@163.comCorresponding author: Zixuan Ni
Keywords: MATLAB, FIR, low-pass filter, audio processing. Abstract. With the rapid development of the global electronic market, the filter has been greatly promoted. And the development of digital technology also create a good technical foundation for thefilter. This paper introduces the basic concepts of digital filters, the brief introduction of MATLAB
Software, and the design method of FIR low-pass filter. Taking MATLAB as working platform anddevelopment tools, the design and simulation of FIR low-pass filter are realized by MATLAB
program. And the practicability of low-pass filter for audio processing is proved by tests.1. Introduction
Digital signal processing technology has entered the mature stage at present. In the process ofdigital signal processing, digital filtering is a very important part. The digital filtering converts the
digital signal through some arithmetic logic, and eliminates some frequencies or improves the relative
occupancy ratio of some frequencies at the same time, so as to eliminate the interference frequency [1].
Compared with traditional filtering, digital filter has great advantages both in stability and flexibility.
Therefore, the comprehensive analysis of digital filters is of great significance for the promotion and
application of digital signal technology. In this paper, FIR low-pass filter designed by MATLABsoftware is used to realize low-pass filtering of audio files and the high frequency is filtered to get the
bass part finally.2. Overview of Digital Filter
Digital filter is an algorithm or device made up of digital multiplier, adder and delay unit. Thefunction of digital filter is to handle digital code of the discrete signal input, in order to change the
signal spectrum. There are many kinds of filters. Functionally, filters can be divided into low pass,
high pass, band stop, and band pass. In terms of the unit impulse response, the digital filter can be
divided into finite impulse response filter and infinite impulse response filter. In the process of using
the filter, it is necessary to make a comprehensive analysis of the two filters before deciding what kind
of filters to use. In general, the FIR filter achieves the goal of filtering through the superposition
algorithm, so there is no ready-made formula for it to use in the design process. And the IIR filter can
complete the design work only through the analog filter design parameters, but in the flexibility, the
IIR filter has the obvious inferiority compared with the FIR filter.3. Brief Introduction of MATLAB Software
MATLAB language is an engineering application software which can integrate scientific computing, data visualization and program design. It has become the essential basic software for thesubject of computer to have aided engineering analysis, design, simulation as well as teaching, which
is composed of MATLAB main package, simulink module and toolbox with various functions [2]. MATLAB has efficient numerical computation and symbolic computation function which can relieve users from mathematical analysis in complex. And it has a complete graphics processing functionwhich can realize the visualization results and programming. Its rich function (such as signal 2nd International Conference on Computer Engineering, Information Science & Application Technology (ICCIA 2017)Copyright © 2017, the Authors. Published by Atlantis Press.
This is an open access article under the CC BY-NC license (http://creativecommons.org/licenses/by-nc/4.0/). Advances in Computer Science Research, volume 74460
processing toolbox, communication toolbox, etc.), provides a number of convenient and practical tools
for users.4. Design of The FIR low-pass Filter
4.1 The Design Method of FIR Low-pass Filter
Usually MATLAB is used to design and simulate the FIR filter coefficients, the usual way is optimization design, FDA TOOL design and window function design [3]. For the method of optimization design, Park-Mc-Clellan algorithm can be used to design linear phase FIR filter which makes the maximal error between actual frequency response and the expected response frequencyminimum, but this design method cannot directly set the filter resistance band attenuation parameter.
FDA Tool (filter design & analysis tool) is a tool specifically for the design and analysis of filter in the
MATLAB toolbox which provides an interactive design environment for the entire process. Users design almost all types of filters by setting amplitudes and zeros and poles, including type, order, frequency and other parameters. The window function method is to set the unit impulse response sequence h(n) by adding a window according to the given frequency response feature. The window functions commonly used in the project include rectangular Windows, hanning Windows, hamming Windows, cassett Windows, blackman Windows and bartlett Windows and so on. The function commands based on fir1 and fir2 can easily get the filter coefficients.4.2 Design and analysis of FIR low-pass filter based on MATLAB
The fast algorithm of FFT discrete Fourier transform in MATLAB can transform a signal into the frequency domain. After FFT, N sample points can get the FFT results of N points. In order tofacilitate FFT operations, N usually takes integer power of 2. Assuming that the sampling frequency is
sF, the signal frequency F and the sampling point is N, so the result of FFT is a complex number ofN points. Each point corresponds to a frequency point. The module value of this point is the amplitude
characteristic of the frequency. For example, the frequency of a point n is: NFnFs n)1(. If the sampling frequency is 1024Hz, the sampling point is 1024, then it can be identified by 1Hz. In other words, sampling the signal for a second time and conducting FFT, the result can be analyzed to 1Hzand if the sample is sampled for two seconds and the conducting FFT, the result can be analyzed to 0.5
Hz. If you want to increase the frequency resolution, you have to increase the number of samples, which is the sampling time. Frequency resolution and sampling time are reciprocal. The most common FIR implementation architecture is called the direct-form FIR [4]. Low-pass filter is to take low-frequency component out and abandon the high frequency part of thedigital signal, and the band-pass filter is take digital signal in a certain range of frequencies. So the
band-pass filter after the translation is a low-pass filter. The structure of the low-pass filter is shown in
figure 1.Figure. 1 Structure of the low-pass filter
Its function expression is: )()()(0ccffUffUfh (1)
And the low-pass filter is shifted by the band-pass filter. Its function expression is: )()()()()(1111fffUfffUfffUfffUfhcccc (2)
Advances in Computer Science Research, volume 74461 The Discrete Fourier Transform of the input digital signal is:NnnfjenxfX2)()( (3)
The Fourier transform of the output signal is:
NnccccnfjfffUfffUfffUfffUenxfHfXfY)]()()()([)()()()(11112 (4) Take the inverse Fourier Transform and we can get output signal: Nnc ntfntfntnxny)())(2cos())(2sin()(2)(1 (5)This is the output of the pass-through filter.
4.3 The filtering effect of audio signal
Because
cf and1f are normalized, the period of frequency is 1, so when the frequency is greater
than 0.5, it is a high frequency component, whereas less than 0.5 is the low frequency part. For a low-pass filter, the sum of cf and1f is less than 0.5. Taking a signal, the sample frequency of which
is 11400hz. First, the spectrum and digital signal of the experiment audio are obtained which were shown in figure 2.Figure. 2 The digital signal and its spectrum
Then obtained the following results after pass the designed low-pass filter in figure 3 and figure 4.
The analysis of the audio shows that the lower part of the audio is retained and higher part is removed
through the low-pass filter. cf is 1109/88200Hz, 1f is 1091/88200Hz) Figure. 3 Audio data Figure. 4 Spectrum And then obtained the following results after pass the designed band-pass filter in figure 5 andfigure 6. The analysis of the audio shows that the audio becomes thinner and deeper than before, as the
audio through the band-pass filter loses a portion of the frequency. cf is 1109/88200Hz, 1f is 10091/88200Hz) Advances in Computer Science Research, volume 74462 Figure. 5 Audio data Figure. 6 Spectrum5. Conclusion
MATLAB is a large and powerful software that is suitable for multi-disciplinary, multi-workingplatform. It has many functions include mathematical statistics, digital signal processing, time series
analysis, dynamic system simulation and so on. In this paper, the low-pass filter designed by
MATLAB can be used for signal selection. It can be used to select the signal according to the frequency, to avoid interference of high-frequency signals, and it also can be used in instruments, acoustics, signal processing and other circuits.References
[1]. Xuran LI. Design of FIR Low Pass Filter Based on Matlab. Computer Knowledge andTechnology. Vol. 10 (2014) No. 36, p. 8806-8807.
[2]. Sul Seung Ki, Kim.Jang Hwan, Choi. Jong Woo. Novel Rotor-Flux observer using observer characteristic function in Complex vector space for field-oriented induction motor drives. IEEETrans I. A. vol.38(2002), pp.1334-1343.
[3]. ZHANG Bo, PEI Dongxing. Design of FIR Low- pass Filter Based on DSP Technology. AudioEngineering. Vol. 38 (2014) No. 9, p. 33-35.
[4]. Fred J. Taylor.Digital Filters: Principles And Applications With Matlab.IEEE Press,2011, p.131-133.
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