Advantages And Disadvantages Of Resonance

Advantages And Disadvantages Of Resonance
Advantages:
Harmonic reduction is possible.
Undesirable harmonics are absent.
Reactive power compensation is possible.
Disadvantages:
Resonance with line impedance occurs.
Tuning frequency is less accurate and requires a lot of calculations.
It cannot be used when harmonic components vary randomly.
Heavy and bulky.
3.1.2 Active Filters: It inject equal and opposite harmonics onto the power system to cancel those generated by other equipment.
Remarkable advances in the field of power electronics had sparked interest in APF for harmonic distortion mitigation. The basic
technology of APF is to use power electronics technologies to produce the harmonic current components such that the source will
supply only the fundamental part of ... Show more content on Helpwriting.net ...
(4) Fast acting.
(5) Good filtering action for large range of frequencies.
APF is basically of two types. They are (i) Shunt APF (ii) Series APF
(i) Shunt APF
This type of configuration is widely used in active filtering applications. A shunt APF has a controllable voltage or current source.
Fig. 3.4 demonstrates principal configuration of a voltage source inverter based APF which the most common type of shunt APF
today. It is based on the principle of injection of harmonic current equivalent to the distorted current waveform, thus eliminating the
original distorted current. This is connected in shunt with the distribution line through an interfacing inductor. Fig. 3.4: Basic
configuration of Shunt APF
(ii) Series APF
Basic principle of this type of configuration is isolation of harmonics in between nonlinear load and source. The basic configuration
is similar with shunt APF, except that interfacing inductor is replaced by an interfacing transformer, where the injection of harmonic
voltage occurs. It is otherwise known as a harmonic isolator, which offers zero impedance to the fundamental and infinite impedance
to the harmonic frequency components. Fig. 3.5 demonstrates the basic configuration of a series APF. Series APF is suitable for
improving the quality of the distribution source
... Get more on HelpWriting.net ...

The Importance Of Light Channels For RZ And WDM...
At the transmitter side, our system consist of WDM transmitter for RZ and NRZ modulation, and a CW laser array is used for the
radiation of light waves. It is one that emits radiation continuously rather than in short bursts, as in pulsed laser. Externally modulated
subsystem consists of Pseudo Random Bit Sequence (PRBS) generator, NRZ Pulse generator, two Mach–Zehnder modulators & a
sine carrier for CSRZ modulation. WDM multiplexer multiplexes 8 number of WDM signal channels. The transmission channel
consist of single mode fiber (SMF), dispersion compensation fiber (DCF) and EDFA. EDFA is used to compensate the attenuation
losses occurring in SMF and DCF. Then a demultiplexer multiplexes 8 number of WDM signal channels. The receiver ... Show more
content on Helpwriting.net ...
The performance of spectrally efficient WDM system is analyzed in terms of Q–factor, BER and optical spectrum of the signal. Q–
factor is given as:– (4)
Where and are the average values of signal strength depending on whether the bit corresponds to 1 or 0 in the bit stream [9]. The
relationship between Q–factor and bit error rate is given as:– (5) Where 'erfc' denotes the complementary error function [10].
III. RESULTS & DISCUSSIONS
We have one 8 channel DWDM system operating at 10 Gbps with channel spacing of 100 GHz and the transmission distance of 86
Km. We have made a comparative analysis of the system by using RZ, NRZ and CSRZ modulation schemes and at various power
levels. As the signal passes through the fiber the nonlinear effects such as XPM, FWM, SPM, SRS and SBS contribute to nonlinear
changes like increase in channel power for some channels while decreasing power for the other channels. When transmitter consist of
8 DWDM channels, 3 dBm power and NRZ modulation then eye diagram & bit error rate, quality factor at receiver is shown in fig.
3, at 3 dBm power level we are achieving a quality factor of 5.99039 with BER of 9.69935e–010, when we increase the power level
to 7 dBm quality factor is

X Common Filter Issues
X Common Filter Issues You Need to Know About Everyone who has a pool in their yard knows the importance of their filter.
Without it you end up with a disgust stew of warm green slim and bacteria filling your pool. Filters are essential to discarding
unwanted particles and debris that can create a hazardous environment for you and your swimming buddies. Filters not only allow
your pool to maintain it's spot as your backyard's number one, aesthetically pleasing, focal point, but keep they also keep your pool
safe for swimming. Without proper maintenance a filter can fail or break down. Pool owners know that filter problems can occur
quite often but many don't know why they occur or how to fix them. To enjoy your pool to the fullest and to be a responsible pool ...
Show more content on Helpwriting.net ...
The amount of pressure in the pool's filter is a common problem among pool owners and can lead to expensive repairs if not properly
treated. It's important to note that while the pressure gauge can give you a good idea of the pressure in your filter, it is not an exact
number. The pressure gauge can break or malfunction quite easily which can make determining the filter pressure a little difficult. On
all filter systems, the gauge should return to zero when the pump goes off but can get stuck. If this is the case, a careful tap on the
glass should help. The pressure on your filter can mean a series of issues. If the gauge reads to low it could mean that there is a
blockage before or after the filter. In order to find the root of the problem, check your pump basket and pump impeller for any
blockage. If the pressure is too high then there is a blockage at or beyond the pool filter. In this scenerio, there are three things that
you need to look out for. A clogged filter A partially closed return valve An obstruction inside of the heater Or, an obstruction inside
the return

Edge Detection And Frequency Domain
Edge Detection in Frequency Domain
A comparison between different Filtering techniques in Time and Frequncy Domain
Adithya R H
Department of Electronics and Communication
Sri Jayachamarajendra College of Engineering
Mysore, Karnataka, India rhadithya@gmail.com Akash Sharma
Department of Electronics and Communication
Sri Jayachamarajendra College of Engineering
Mysore, Karnataka, India akashshrm02@gmail.com Abstract–This paper presents a comprehensive comparison of different high pass
filtering techniques for Edge detection in both time domain and frequency domain. The paper examines various kernels and compares
the efficiency of the filtering technique against the computation time for various sizes of images with various sizes of high–pass filter
kernels. We have made use of Sobel filter as the standard filter kernel against which other techniques are compared.
Keywords–frequency domain; edge detection; sobel filter; laplacian; gaussian;
I. INTRODUCTION
Image sharpening and smoothening is perhaps one of the most ubiquitously used techniques in any image processing application.
While these are considered inexpensive operations, multiple use of the same may become computationally intensive, and hence may
make the overall application slow. In this paper we examine and compare the performance of sharpening filters in time–domain and
frequency domain. While there are many apparent computational advantages in converting an image into frequency domain to apply
a

Discrete Wavelet Transform For Compressing And...
Discrete Wavelet Transform for Compressing and Decompressing the Speech Signal Bhavana Pujari1, Prof. S.S.Gundal2 Abstract:
The original digital speech signal contains tremendous measure of memory, the main concept for the speech compression algorithm
is presented here, in which bit rate of the speech signal is reduced with maintaining signal quality for storage, memory saving or
transmission over the long distance. The concentration of this project is to compact the digital speech signal using Discrete Wavelet
Transform and reconstruct same signal using inverse transform, in .NET. The algorithm of Compression is oriented in three basic
operation, they are apply the DWT, Threshold, Encode the signal for transmission Analysis of compression procedure is done by
comparing the original speech and reconstructed signal. The main advantages of DWT provides variable compression factor.
Keywords: Bit rate, Compression, Decompression, Discrete Wavelet Transform (DWT), .NET, Threshold. Introduction The Speech is
finest effective medium for viewpoint to face communication and telephony application. Speech coding is the process of obtaining a
compact representation of audio signals for efficient transmission over band–limited wired and wireless channels and/or storage. The
procedure of Compression is done by transferring an original signal to alternate compressed signal that consist of small amount of
memory. Compression is conceivable simply because information in input data

Using Brute Force Algorithm For The Quality Of Ultrasound...
Abstract: In general, the main problem in ultrasound images is reduction of noise and problem is more severe when the noise is
multiplicative in nature. In this paper, we present a novel method to combine different filter using brute force algorithm to enhance
the quality of ultra sound image. This method is divided into two parts i.e. pre–processing and brute force algorithm. In pre–
processing median filter ,fuzzy filter which will evaluate the different characteristics as noisy, homogeneous, and edge and Savitzky–
golay method were applied which gives smooth image while preserving shape and height of the Ultrasound image. Secondly,
Savitzky–golay filtered image and fuzzy filtered image were decomposed into different wavelet sub bands using directional
dependent mask and finding out the better threshold value of each band by using brute force algorithm. Evaluation of the method
based on different parameters show that proposed method can improve the quality of Ultrasound image. Key Terms: Directional
dependent mask, non–decimated wavelet transform, fuzzy filter, Savitzky–Golay filter.
INTRODUCTION
Unlike many other imaging applications where image quality get enhanced and nice visual interceptions giving from the human eye
but in case of medical applications are limited, such as the generation of the device which can be interpreted as clinically interesting
functions to achieve to need the best possible diagnosis is important that is in ultrasound medical images are crisp

Audio Electronics : Battery Powered Guitar Amplifier
AUDIO ELECTRONICS
Battery Powered Guitar Amplifier
Contents
Abstract 2
Current Products 3
Definition of Performance Targets 4
Device Build 4
Power Amp Section 4
Tone Control 5
Transistor Amplification Stages 6
LM386 Power Amplifier 8
Physical Build 9
References 10
Abstract
This paper looks at the research, design and computer aided circuit design required to create and manufacture a 9V battery powered
guitar amplifier. This task is aided through the use of such circuit design and testing programmes such as Multisim and KiCad. All
equations used throughout this paper were created using LaTex. At the end of the paper, reflections are given in how the physical
build of the circuit could have been improved. Current Products
The basic principle of the device created In this module is not a new one, battery powered guitar amplifiers are popular among many
guitarists and have been for many years. Popular products currently in the market place range from cheap, products featuring low
quality questionable design, through to expensive, boutique and custom work, created by audio engineers across the UK and USA.
Such devices are as stated below:
Marshall MS–2
Orange Micro Crush
Roland AC33
Definition of Performance Targets
The plan for this project, is to create a battery powered guitar amplifier that is capable of creating tones similar to the style on tones
available on a valve powered electric guitar amplifier.
It is planned for the amplifier to be used as a

A Short Note On Internal And Internal Assessment
IB Mathematics HL Internal Assessment
Judy Taylor
IB HL Mathematics
Word Count: 1311
Table of Contents Abstract.......................................................................................Page 3
Introduction...................................................................................Page 4
Exploration....................................................................................Page 5
Applications...................................................................................Page 8
Conclusion.....................................................................................Page 9
References....................................................................................Page 11 Abstract The sine integral, written as Si (x), is important
factor in signal processing as it causes overshoot, ringing artifacts, and is considered as the convolution of the cardinal sine function
(represented as sin (x)/ x or sinc (x)) and the Heaviside step function that corresponds to shortening the Fourier series, which causes
the Gibbs ... Show more content on Helpwriting.net ...
After some research, I found out that cardinal sine could not be solved by elementary means and that its integral was, in fact, a
crucial part of the Gibbs phenomenon, which will be expanded upon in the Applications section (Pg. 8) of this investigation. My aim
for this investigation is to prove the integral of cardinal sine to be sine integral and to approximate five (5) values of the sine integral
function. Afterwards, I will show how this is relevant in both mathematics and other, more practical, applications. Exploration In
order to organize the mathematical exploration most effectively, the exploration will begin with the following problem: Prove that
∫▒sin⁡
x/x is equal to Si (x). Approximate Si (0), Si (π/6), Si (π/4), Si (π/3), and Si (π/2) using a Taylor polynomial of order 3 at 0 for
f(x) = Si (x). In order to verify the results of the approximations, Wolfram Alpha, an online calculator capable of processing the sine
integral function, will be used.
To begin with, the Taylor series representation of the sine integral function is as such:
Si (x)=∑_(n=0 )^∞▒((–1)^n x^(2n+1))/(2n+1)(2n+1)! At this point, it is necessary to find the Taylor series representation of sin⁡
x/x.
First, the Taylor series representation of sin (x) is:
∑_(n=0)^∞▒(〖(–1)〗^n x^(2n+1))/(2n+1)!
Therefore, sin⁡
x/x is equal to:
1/x ∑_(n=0)^∞▒(〖(–1)〗^n x^(2n+1))/(2n+1)!
Which is equal to:
∑_(n=0)^∞▒(〖(–1)〗^n x^(2n+1))/x(2n+1)!

Advantage And Disadvantage Of Rectenna
Today's world scenario there is an necessity for an alternative energy source which replaces the on hand source of energy. These
include rectenna's, solar cells, harvesting human energy and wind power. A new imminent rectenna system design incarnates an
antenna, a Schottky diode, a matching circuit and a DC filter. The harmonics rejection of the antenna is primarily used to reduce the
rectenna size. The filter implemention provides the various advantages in the antenna structure such as Miniature of rectenna size,
hence its become suitable for several applications where the size and weight are well thought–out as critical aspects. There is
maximum energy conversion efficiency in this configuration when compared to the conventional type design. ... Show more content
on Helpwriting.net ...
INTRODUCTION Various energy harvesting techniques which form best alternative to the existing energy resources. These include
energy harvesting from wind energy, hydro power, tidal power, geothermal and solar power. Solar rectennas are used to harvest solar
power but those rectennas are operates at GHz/THz and finds many disadvantages such as power fading, complicated design
procedure and high manufacturing technology. Thus, it has been recommanded to use rectennas operating at low range of frequencies
which consists of antennas not to harvest solar power but to harvest electromagnetic waves. Instead of all those energy harvesting
techniques used to harvest resources like solar, wind and tidle, rectennas are simple and inexpensive to construct. Voltage harvested
can be doubled with help of DC–up convertors
PN junction diode and transistor are used as rectifier for low frequency signal (low frequency RF signal) but for microwaves (1 GHz
and higher), Schottky diodes (si or GaAs ) with shorter transit times are required. In the present case, we have chosen silicon based
on availability, low cost, and simulated performance. Similar to low–frequency, high–power applications, the diode is driven as a
half–wave

Mini Project
EE2031 Circuit and Systems Design Lab
Mini–Project Report
Voice Disguiser
Presented by:
Friday Room 3 Group 7
Wang Ke (A0105276J)
Wu Jianxiao (A0099910W)
1.Introduction
1.1 Aim
During the 6 hands–on individual labs before this mini project, we have learnt the design and implementation of filter and oscillator,
together with applied characterization of different components. The objective of Mini–Project was for students to demonstrate their
innovative design skills and implement knowledge learnt in lectures to real–life product design. Therefore, we aimed to integrate
what we learnt in individual labs into one single product, which is a Voice
Disguiser that can solve practical problems of witness protection.
1.2 ... Show more content on Helpwriting.net ...
Figure 1: Project Flowchart
2.2 Microphone Unit
The microphone was implemented to allow the user to transmit speaker's voice as analog voice signal from the audio socket. The
analog voice signal captured by the microphone is then needed to pass through an amplifier circuit. The amplitude of this voice signal
can be amplified from millivolts to volts so that the signal becomes easier to for us work with in following implementations. As
figure 2 shown, this amplifier circuit is built by firstly connecting the microphone input through the positive input terminal of the
LM358 chip. Secondly, a feedback loop is built through the output and negative terminal. Then the analog voice signal captured by
the microphone is, therefore, amplified.
Figure 2: Schematic Of Microphone Unit
The resistance values of R4 and R3 is purposely chosen for an amplifier gain of 5.7. As shown in Figure 3, a significant amplifying
effect of the microphone voice signal can been seem afer passing through the amplifier.
Figure 3: Waveform Of Voice Signal Before (Blue) And After (Yellow) Amplifier
Based on oscilloscope detection, the peak–to–peak value of microphone voice signal before amplifier is 220mV, while the one after
the amplifier is 1.28V. The actual amplifier gain is therefore 5.8 (1.28/0.22), which is very close to the designed value of 5.7.
2.3 Frequency Modification Unit
This

Analysis and Design of Fm Transmitter
CHAPTER THREE
ANALYSIS AND DESIGN
3.1 INTRODUCTION
This chapter deals with the process of obtaining the design specifications of the wireless Frequency Modulated Transmitter with a
frequency of 100MHz in a range of 100m under accurate conditions.
From the block diagram shown in FIG. 2.1, various circuit components are now chosen and appropriate approximate values are now
chosen to match its properties as well as perform duties of the stages.
The entire system could be grouped into two major stages The Q1 stage consisting the oscillator, the audio amplifier and the
modulator The Q2 stage which is made up majorly of the RF amplifier and the antenna
3.2 Q1 STAGE
One transistor is used as the audio frequency (AF) amplifier.
The ... Show more content on Helpwriting.net ...
For effective coupling, Grob (1977) states that the reactance of the capacitor, XCC should be less than or equal to 10% of the
resistance, R1.
XCC < R1; but R1 = 100KΩ
Let XCC = 10KΩ and f = 20Hz (cut off frequency for the HPF)
But XCC = 1 = 10KΩ 2пfCc
Cc = 1 = 1 a = 0.796 x 10–6F 2пfXCc 2п x 20 x 10 x 103
= 0.796µF Chosen value of CC = 1µF.
C1 also forms a low pass filter with R1.
XC1 < 10%R1
But R1 = 100KΩ, hence letting XC1 = 10KΩ and using a cut off frequency for the LPF = 20KHz,
XC1 = 1 2пfCc
C1 = 1 i
2п x 20 x 103 x 10 x 103 = 0.796 x 10–9F
The design value chosen for C1 = 1nF.
Similarly,
XC4 < 0.1R2, R2= 1KΩ, f = 100MHz
XC4 = 0.1KΩ
C1 = 1 i = 15.9 x 10–12F ≈ 16pF
2п x 100 x 106 x 0.1 x 103
Design value of C4 = 18pF.
3.2.3 DETERMINATION OF PARAMETERS FOR THE FREQUENCY DETERMINING NETWORK (FDN OR TANK CIRCUIT)
At oscillation, C2 and C2 are voltage devices. C2, C3 and L serve as oscillation frequency determining network for the colpitt
oscillator. The equivalent circuit is shown in FIG. 3.4 C2 C3 L FIG. 3.4: the FDN.
C = C2C3 ; where C= equivalent capacitance of the tank circuit. C2 + C3 C

Nt1310 Unit 3 Assignment 1
Student Name:– Sunil
Student id: – 11587717
Subject :– SITC513_201560_SM_I (Wireless Networking Concepts)
Answer 1.
In elecronic, broadcast, transportation, filters are used to select the desired signal, and deny or block unwanted signals. Or put in
other words, they may be the only way we have to block.The doing so, we want to signal and other frequency or with a device, is
frequency–selective – it behave different towards different frequencies. Such device is called a filter. (Bertrand, 2002)
Irrespective of the mechanism by which the filler does its job, filters can be broken down into four basic types.
1. Low–pass filter
2. High–pass filter
3. Band–pass filter
4. Band–stop filter (Bertrand, 2002)
Low–pass filter – It passed all the ... Show more content on Helpwriting.net ...
Mixer:
Mixer is an active or passive device, from a switching signal Frequency to another. It can be adjusted or the demodulated signal. It
has three signals Connection, which is called a port in the language of radio engineers. There are three ports. It is the RF input, a local
oscillator input, and intermediate Frequency output. (NA, NA)
Here Figure (a) show up–conversion and figure (b) show down–conversion:
Dual band (DSB) signal comprising on the lower sideband, and a single sideband (SSB) signal may be generated by filtering or by
using a single sideband mixer.
Single sideband modulation may be viewed as the removal or reduction of the amplitude modulation signal component. In order to
see how the SSB is created, it is necessary to use an amplitude modulated signal as a starting point. (Rosu, NA)
Amplitude modulated carrier
It shows sidebands on either side of the carrier
From this it can be seen, the signal having the two sidebands, each other's mirror, and a carrier. To improve the efficiency of the
signal, both in terms of power and spectrum use, carriers can be removed, or at least reduce it, and remove one sideband – one is a
mirror. (Poole,

Essay On Digital Filter
CHAPTER 4
DESIGN AND ANALYSIS OF LOW PASS FIR DIGITAL FILTER USING ARTIFICIAL NEURAL NETWORK
In proposed work, low pass FIR digital filter has been designed and analyzed using artificial neural network (ANN) with Bartlett–
Hanning and Blackman–Harris window function.
4.1 Introduction In proposed work, three neural network algorithms, namely feed forward distributed time delay (FFDTD), feed
forward back propagation (FFBP) and radial basis function(RBF) is preferred for estimation of the cut off frequency of low pass FIR
digital filter and is established using a Bartlett–Hanning window and Blackman–Harris window function. FFDTD, FFBP and RBF
neural network algorithms have been used to implement the ANN model for ... Show more content on Helpwriting.net ...
 Response type: low pass
 Design method: FIR window technique
 Filter order: 10
 Scale pass band: Bartlett–Hanning and Blackman–Harris window
 Frequency specifications: normalized (0 to 1)
 Cut off frequency: 0 to 1
 Magnitude specifications: the attenuation at cut off frequencies is fixed at 6dB (half the pass band gain) as shown in figure 4.1 and
4.2. Figure 4.1: Design of Low Pass FIR digital filter Using Bartlett–Hanning Window in FDA Tool of MATLAB
Figure 4.1 shows the design of low pass FIR digital filter using Bartlett–Hanning window function in FDA tool of MATLAB. Figure
4.2: Design of Low Pass FIR digital filter Using Blackman–Harris Window in FDA Tool of MATLAB
Figure 4.2 shows the design of low pass FIR digital filter using Blackman–Harris window function in FDA tool of MATLAB.
4.2.1 Design and Data Generation of Low Pass FIR Digital Filter Based on Window In proposed work, different coefficients values
have been generated by using the FDA tool of MATLAB software based on Bartlett–Hanning and Blackman–Harris window in order
10. The filter coefficients have been estimated from ranges 0.025 to 0.975 cut off frequencies. At each cut off frequency, it gives a
total 38 samples in order 10. Basically design of filter means to calculate the filter transfer function coefficients. After simulating
design of low pass FIR digital filter in the FDA tool, filter coefficients are exported to the workspace

Obtaining Motion Blur Parameters Form The Frequency Spectrum
Obtaining Motion Blur Parameters Form The Frequency Spectrum
Fourier transform is applied on digital images to interprets their content in terms frequency information. To illustrate, Flat areas,
where the intensity is slowly changing, result in low frequencies. Rough areas, on the other hand, result in high frequencies because
of the dramatic change in the intensity value. this paper discusses the impact of manipulating the frequency information of digital
images and how the frequency spectrum can be used to address a real world situation.
Filtering an image in the frequency domain is usually composed of three steps. First, the Fourier transform is calculated (DCT or
DFT). Then, a certain operation is performed on the frequencies ... Show more content on Helpwriting.net ...
Huei and Kun [2] proposed that the speed of moving objects in a single image can be estimated using the blur parameters, the camera
parameters and imaging geometry. To illustrate, the displacement of a moving object (d) can be determined using similar triangles
according to the blur length (k) Figure2. And by knowing the shutter speed of the camera (T), the speed of the object is v =d/T(1).
Since this paper is concerned with Fourier Transform, the details of equation (1) are omitted and calculating the blur parameter is
detailed below.
The blur parameters, including the blurs' direction and length, can be determined by examining their impact on the Fourier spectrum.
As shown in figure, the Fourier spectrum of the motion blur contains strips of dark lines that are parallel and uniformly separated.
Note that the Fourier spectrum depends on the objects' orientation; so the direction of the motion blur could be extracted from the
frequency spectrum. Also, as the blurs' length increase, the edges get smoother, resulting in lower frequency response. These
observations suggest that the frequency spectrum has all the needed information to determine the blurs parameters. In particular, the
object shown in the figure is moving along the horizontal axis, and this caused the vertical orientation of the dark lines in the
frequency spectrum. Furthermore, increasing the blurs length, which can be achieved by increasing the object's speed, lessens the
distance

Neural Recording And Processing Of The Neural Signals
Neural Recording and Processing
The most critical element of a Brain Machine Interface (BMI) is the recording and processing of the neural signal. We use an
invasive neural signal recording to achieve higher performance of the BMI and to obtain better resolution. We will be recording the
neural signal on central sulcus located in the cortex region of the brain. This recording is referred to as Electrocorticography (ECoG).
We will be using subdural grid electrodes (surface electrodes) with 48 contact points spanning across the upper limb region of the
central sulcus. The signal will be picked up by the electrodes and it will be transferred to the signal processing unit through the
encapsulated leads. The leads will be connecting the ... Show more content on Helpwriting.net ...
The signal gets amplified, filtered and then get digitised. The digitised data is fed into the decoder.
Signal Decoding
We intend to design a signal decoder as an ASIC. It will perform the decoding of neural signals and predict the brain's intention to
perform arm movements. Though we use multiple electrodes to record the neural signal, the number of features that can be extracted
from the recorded signal is very less. The neurons in the brain perform a task by sequential generation of action potentials which are
then transmitted to other neurons through the synapse. These action potentials are referred to as spikes.
The first step in decoding is to extract features from the recorded neural signal. The features we will be interested are the spikes. The
efficient method to detect spikes is threshold crossing. This is a simple approach where we will setup threshold amplitude and count
the number of threshold crossings across a time bin. Whenever a spike or set of spikes cross the threshold value it is detected and
assigned to a single or a group of neuron. This method does not affect the decoding performance and has gained acceptance widely.
We will adopt the threshold crossing technique to extract features from the neural signal.
We will identify the spikes detected based on their frequency, timing and location. The information found is referred to as rate
coding, temporal coding and population coding respectively. We will then classify the spikes based on this

Low-pass Filter and Critical Frequency Essay
Name:
Alfredo del junco
Date:
2/22/2014
ET 1410 Final Exam
1. 1. An IC op–amp has ________.
a. two inputs and two outputs
b. one input and two outputs
c. two inputs and one output
d. one input and one output
2. In the common mode, ________.
a. both inputs are grounded
b. the outputs are connected together
c. an identical signal appears on both inputs
d. the output signals are in phase
3. If Av(d) = 3500 and Acm = 0.35, the CMMR is ________.
a. 1225
b. 3500
c. 80dB
d. 90dB
4. A certain op–amp has bias currents of 50 μA and 49.3 μA. The input off–set current is ________.
a. 700 nA
b. 99.3 μA
c. 49.65 μA
d. None of these
5. A certain non–inverting ... Show more content on Helpwriting.net ...
When 2 low–pass filters are cascaded, the roll–off rate becomes ________.

a. –60 dB
b. –20 dB
c. –10 dB
d. –40 dB
24. When a low–pass and a high–pass filter are cascaded to form a band–pass filter with a gain of 0 dB at center frequency, the
critical frequency of the low–pass filter must be ________.
a. equal to the critical frequency of the high–pass filter
b. less than the critical frequency of the high–pass filter
c. greater than the critical frequency of the high–pass filter
d. adjusted in a way that Q becomes 10
25. When the gain of a filter is minimum at its center frequency, it is a ________.
a. band–pass filter
b. band–stop filter
c. Sallen–Key low–pass filter
d. 2nd order high–pass filter
For questions 26 through 30, refer to Figure 3.
26. The circuit in Figure 3 is a ________.
a. band–pass filter
b. low–pass filter
c. high–pass filter
d. band–stop filter
27. The critical frequency is ________.
a. 1.48 kHz
b. 8.84 kHz
c. 491 MHz
d. None of the above
28. If C1 is accidentally replaced with a value 0.15 μF instead of a 0.015 μF, the bandwidth will be ________.
a. doubled
b. divided by 2
c. remain the same
d. divided by 10

29. If C1 is accidentally replaced with a value 0.15 μF instead of a 0.015 μF, the number of poles will ________.
a. increase by 1
b. decrease by 1
c. remain the same
d. double
30. If

Unit 1 Comm-3011
The list of tasks to be completed is: COM–3011 Verification; Sample Rate Decimation; VITA–49 Packaging; Ethernet Transmission;
and Software Functionality. COM–3011 Verification
The COM–3011 operates by first mixing the received signal with 125 MHz, thus the data is now contained on it. The centre
frequency is chosen by mixing a user specified radio synthesiser frequency with the 125 MHz together. For example to obtain a
centre frequency of 100 MHz, the user would specify 225 MHz or –25 MHz. The bandpass filter will reduce the bandwidth to 40
MHz or 20 MHz either side of centre frequency [12]. This sample is the duplicated and the second part delayed by 90 degrees to form
an in–phase and quadrature features. Sample Rate Decimation
For the ... Show more content on Helpwriting.net ...
This packet contains information such as the frequency of the samples and bandwidth. This is only transmitted once until the
information changes. For the implemented solution, the system will only transmit this context packet at the beginning of the
transmission and will not transmit unless the system is rebooted. Ethernet Transmission
Using the COM–5102, the SDR will transmit the radio signal, this pulg in will add associated MAC addresses and UDP headers onto
the project automatically [15]. The VHDL code has values that can be changed slightly to accommodate UDP or TCP transmission.
For the project, UDP headers will be used. Software Functionality
This is the section of the work being undertaken by the software engineer. The software must contain a custom driver to unpack the
VITA–49 packet outlined before. This driver will be written to work with a Ubuntu Linux distribution. The unpacked data will then
be handed to GNU Radio Companion and the signal processed. The FFT, dynamic peak algorithm, spectrum display and list of
signals are default tools that can be downloaded from the file repository [16]. The module will require a custom module called an
"out of the tree module" to use that data brought in from the VITA–49 packet: "out of the tree" modules are custom modules made by
a user for the graphical

What´s a Digital Filter?
Introduction
In signal processing, the purpose of a filter is to remove unwanted portions of the signal, such as random noise, or to extract useful
parts of the signal, such as the constituents lying inside a precise frequency range.
Figure 1.1 A block diagram of a basic filter
There are two main kinds of filter, analog and digital. An analog filter uses analog electronic routes made up from constituents such
as resistors, capacitors and op amps to produce the needed filtering effect. Such filter routes are extensively utilized in such requests
as sound reduction, video signal enhancement, graphic equalizers in hi–fi arrangements, and countless supplementary areas.
A digital filter uses a digital processor to present numerical calculations on sampled values of the signal. Processor may be a general–
purpose computer such as a PC, or a enumerated DSP (Digital Signal Processor) chip.
As contrasted to analog filters, digital filters are possessing higher precision, larger stability and reliability. Digital filters do not have
matching problems. Digital filters embody period separation multiplexing that can finished a little filtering tasks that a number of
analog filters are incapable of doing. We can accomplish digital filters by hardware circuits. Additionally we have an alternative way
as computer multimedia programming.
The frequency reply of the filter depends on the worth of its coefficients, or taps. Each multimedia plans can compute the benefits of
the coefficients

What Are Two Different Ways To Filter Data And Reduce Data...
There are two different ways to filter data and reduce data noise: through hardware changes, through software calculations. Hardware
alterations include: grounding and shielding, difference and instrumentation amplifiers, analog filtering, and modulation. Grounding
and Shielding– Noise from environment can often be dramatically reduced by shielding, grounding, and minimizing wire lengths.
Shielding is surrounding a circuit or the wires in the circuit with a conducting material then attaching that to ground. Environmental
EM radiation can't penetrate the conductor where it gets absorbed. Difference and Instrumentation amplifiers– Any noise in
transducer is bad because this is the signal that gets amplified and interpreted by the rest of ... Show more content on Helpwriting.net
...
Once the signal is in a high frequency, it is in a region that has less background noise, and so the signal can be amplified without
amplifying the noise. Then demodulate and record the data. The Software methods for data acquisition are useful for instruments that
are computer controlled. They include averaging, exponential filter, and running average/moving average. Averaging– It filters data
by adding together many measurements, then dividing the total by the number of measurements you added together. The more
measurements you include in the average the more noise gets removed. There is a diminishing return though: the average of 101
measurements isn't going to be much less noisy than the average of 100 measurements. Exponential filter– it is a recursive filter. A
recursive filter is just one that calculates a new, smoothed value (yn) by using the last smoothed value (yn – 1) and a new
measurement (xn): yn = w × xn + (1 – w) × yn – 1 The amount of smoothing is controlled by a weighting parameter (w). The weight
is a value between 0% and 100%. When the weight is high (say 90%), the filter doesn't smooth the measurements very much but
responds quickly to changes. If the weight is low (say 10%), the filter smooths the measurements a lot but doesn't respond very
quickly to changes. Running Average/Moving Average– An alternative to making all the measurements at once then averaging them
is to make one measurement at a time and add it

Comparison Of Mean Square Error ( FFBP )
comparison of mean square error using feed forward back propagation (FFBP) and radial basis function(RBF) neural network
algorithms are given in table 5.2 for analysis of band pass FIR filter with hanning window.
Table 5.2 Comparison of mean square error using FFBP and RBF neural network algorithms used for cut off frequency calculation of
band pass FIR digital filter with hanning window
Test input
(Filter
Coefficient) hanning
Window (actual cut off
Frequency) Output of Artificial Neural Network
(calculated cut off frequency) Mean Square Error
(MSE)
FFBP RBF FFBP RBF h(n) fc1 fc2 fc1 fc2 fc1 fc2 fc1 fc2 fc1 fc2 h1(n) .155 .355 .15528 .35528 .155 .355 .0000000784
.0000000784 0 0 h2(n) .215 .415 .21459 .41459 .215 .415 ... Show more content on Helpwriting.net ...
Table 5.5 Comparison of mean square error using FFBP and RBF neural network algorithms used for cut off frequency calculation of
band pass FIR digital filter with kaiser window
Test input
(Filter
coefficient) Kaiser
Window (actual cut off
Frequency) Output of artificial neural network
(calculated cut off frequency) Mean Square Error FFBP RBF FFBP RBF h(n) fc1 fc2 fc1 fc2 fc1 fc2 fc1 fc2 fc1 fc2 h1(n) .155 .355
.15567 .35567 .155 .355 .0000004489 .0000004489 0 0 h2(n) .215 .415 .21418 .41418 .21498 .41498 .0000006724 .0000006724
.0000000004 .0000000004 h3(n) .275 .475 .27404 .47404 .27499 .47499 .0000009216 .0000009216 .0000000001 .0000000001
h4(n) .335 .535 .33475 .53475 .335 .535 .0000000625 .0000000625 0 0 h4(n) .395 .595 .39593 .59593 .395 .595 .0000008649
.0000008649 0 0 h5(n) .455 .655 .45541 .65541 .455 .655 .0000001681 .0000001681 0 0 h6(n) .515 .715 .5145 .7145 .515 .715
.00000025 00000025 0 0 h7(n) .575 .775 .57232 .77232 .575 .775 .0000071824 .0000071824 0 0 h8(n) .635 .835 .63531 .83531 .635
.835 .0000000961 .0000000961 0 0 h9(n) .695 .895 .69439 .89439 .695 .895 .0000003721 .0000003721 0 0
Table

Blood Pressure Measurement Is A Most Common Test Performed...
Blood pressure measurement is a most common test performed for diagnosis. This paper presents a digital blood pressure
measurement device which possesses a non–invasive measurement of blood pressure without the help of a well–trained physician.
The proposed device measures systolic and diastolic pressure based on Oscillometric method which eliminates the need of
stethoscope. The pressure into the cuff is converted to electrical signal using a pressure transducer. The systolic and diastolic pressure
from the output oscillation of the pressure sensor is detected using microcontroller. This device facilitates with portability and
diagnosis of hypertension at patient's home. Keywords–Blood pressure, Oscillometric method, pressure transducer, cuff, Systolic
pressure, Diastolic pressure. I. INTRODUCTION Hypertension is a leading cause of death in developing countries [1]. Hypertension
can accelerate vascular weakness, making blood vessels more prone to rupture. Untreated hypertension can cause heart attack, stroke,
kidney disease, eye disease and many more which eventually leads to economic burden in developing countries [1]–[3]. In
developing countries with large population due to less healthy life style, the number of patient affected by hypertension is growing
[4]. So, for a developing country like Bangladesh where the number of patient affected by hypertension is large, Blood pressure
machine needs to be inexpensive and user friendly. The heart pumps blood through arteries

Relationship between Cognitive Disability and Mechanics of...
The purpose of the study is to establish the relationship between cognitive disability and mechanics of timed up and go(TUG) in the
individuals with multiple sclerosis (MS). Cognitive abilities directly influence the performance of functional abilities; independence
and social life of the individuals with MS.1 TUG is an important measure of functional measure in healthy and diseased population.6
Cognitive disability such as memory attention, execution of task, fluency and information processing speeds are affected in the
people with MS that affects their activities of daily living.2 We assume that variability in the mechanics of TUG will increase with
increase in cognitive disability. On the basis of result of the present research study, a protocol combining drug treatment, cognitive
rehabilitation along with physical rehabilitation can be designed for improving quality of life of people MS.3 Methods: Participants:
Ten men and women with a medical diagnosis of multiple sclerosis will be invited to participate in the study. Individuals with MS
must be able to walk up to a distance of 15 meters without an assistive device and have a self–reported expanded disability score
(EDSS) ≤ 6.5. Participants will be recruited from the state of South Dakota. The individual with orthopedic disability, pregnancy or
recent exacerbation of symptoms in past 3 months will not be included in the study. I. First session: During the first session all
individuals will complete the following

Detection Of PAnd T
Detection of P and T – Waves in ECG Signal
Abstract– Electrocardiogram (ECG) analysis plays vital role in diagnosis of heart diseases. In order to avoid erroneous conclusions, it
is of utmost importance that all the features of ECG waveform are extracted accurately and only sinus rhythms are present in the
ECG Signal. Significant features of the ECG include the P wave, the QRS Complex, and the T wave. Determining the position of P
wave and T wave is a complicated task due to low amplitude. In this paper simple Low pass differentiation (LPD) based method is
proposed for estimation of P and T wave. Locating all the features accurately is essential for the performance of other ECG
processing such as signal Analysis, Diagnosis, Authentication and Identification. Performance evaluation of this algorithm is done on
recorded ECG signals by holter recorder and MIT–BIH database. Validation of proposed algorithm indicates that it performs
satisfactorily with approximate accuracy of 92.46 %.
Keywords: ECG, P wave, T wave, QRS Complex, Digital Filter.
I. INTRODUCTION
The electrocardiogram (ECG) is an important tool for providing information about functional status of the heart. Analysis of ECG is
of great importance in the detection of cardiac abnormalities. The ECG is characterized by a recurrent wave sequence of P, QRS and
T– wave associated with each beat. The QRS complex is the most striking waveform, caused by ventricular depolarization and atrial
repolarization of the human

V14 System Report and Components
Topic : VI4 System Report
The VI4 system consists of three main components. They are the local rack, the stagebox and the control suface. The inputs of the
local rack of the VI4 system consist of 3 analogue mic/line inputs, 16 analogue line inputs, 1 talkback mic input, 64 channel optical
MADI in and 8 pairs of AES/EBU inputs.. The outputs of the local rack consists of 8 pairs of AES/EBU, 16 analogue line outputs,
TB line output , LCR local monitor A analogue line outputs and LR local Monitor B analogue line outputs. The inputs and outputs of
the stagebox consists of 48 analogue mic/line inputs and 24 analogue line inputs.
Now I shall describe the features of control surface. The control surface in the VI4 system has 24 faders that allow 48 mono inputs to
be mixed simultaneously into 27 outputs by default with the use of 2 fixed layers. The input faders can be assigned to the 4 mute
groups and/or 16 VCA masters. Input faders can be switched to control an individual Aux send mix or control all 24
Group/Matrix/Aux outputs by using the "Follow Solo" function that is switchable. And also the soundcraft FaderGlow will show
when faders are not controlling inputs by utilizing colours. Furthermore the fader trays accommodate motorized faders to allow fader
automation when changing scenes or memory recalls.. The outputs of the control surface compromises of 8 assignable Output faders,
including exclusive LR and C Master faders plus 16 assignable rotary output

Images And Images Of Images
INTRODUCTION
1.1 IMAGE
An image is an artifact that depicts or records visual notion, for example a two–dimensional picture, that has a comparable
appearance to some subject – normally a physical object or a person, for this reason offering an outline of it. Photos may be –
dimensional, which includes a photo, display screen show, and as well as a 3–dimensional, inclusive of a statue or hologram. They
may be captured by optical devices – such as cameras, mirrors, lenses, telescopes, microscopes, and so on. And natural items and
phenomena, consisting of the human eye or water surfaces.
The phrase image is also used inside the broader sense of any –dimensional discern along with a map, a graph, a pie chart, or an
summary painting. on this wider experience, photos also can be rendered manually, along with by using drawing, portray, carving,
rendered mechanically through printing or laptop pics generation, or evolved with the aid of a combination of techniques, mainly in a
pseudo–photograph.
1.2 IMAGE FUSION
In pc vision, Multi sensor image fusion is the system of combining applicable statistics from two or extra snap shots right into a
single photo. The ensuing picture might be more informative than any of the enter pictures. Image fusion is the method that mixes
statistics from a couple of snap shots of the identical scene. Those pix may be captured from specific sensors, obtained at unique
times, or having distinctive spatial and spectral characteristics. The item of the

What Is A Four-Distributed Static Compensator
This paper deals with a four–leg distributed static compensator (DSTATCOM) which is used to solve the current related power
quality. An amplitude adaptive notch filter (AANF) is employed for the reference current extraction of DSTATCOM control scheme
because of its simplicity, capability of frequency and amplitude measuring, appropriate extraction of the fundamental signal, and
insensitivity to the amplitude variation of the input signal. To generate the gate pulses of the switches, an adaptive hysteresis band
current controller is used. In addition, fuzzy logic controllers are used for a better performance of DSTATCOM under dynamic
conditions. The proposed control algorithm is robust to power disturbances, especially when the main voltage ... Show more content
on Helpwriting.net ...
However, these algorithms generally demonstrate a slow time response. Time–domain algorithms are based on the instantaneous
extraction of harmonic currents/voltages. The common time–domain control strategies are instantaneous reactive power theory
(IRPT) [6] and synchronous reference frame theory (SRF) [7]. Calculation of active and reactive powers by transforming three–phase
voltages and currents into two phases is the principle of PQ theory, which does not work properly under non–sinusoidal supply
conditions [8]. SRF theory is based on conversion of three–phase quantities into their corresponding dc components, and low–pass
filters (LPFs) are employed for harmonic filtering which contain a time delay and deteriorate the performance of the controller [9].
Most of the advanced control and signal processing methods are accurate and show a better dynamic response than the FFT, but a
large amount of calculations is required, which does not demonstrate an excellent performance in frequency–varying conditions [10–
13]. Adaptive notch filter (ANF) is another advanced algorithm which has been introduced as an effective control technique for
extracting reference sinusoidal components from distorted input signal. ANF is capable of changing the notch frequency suitably by
tracking the frequency variations of the input signal [14–16].
To the best knowledge of the authors, the modified notch filter in a

Improvement Of Power Quality Using Pq Theory
IMPROVEMENT OF POWER QUALITY USING PQ–THEORY
SHUNTACTIVE POWER FILTER B. Udayasri 1 P.A Mohan Rao 2 Dasumanta Kumar Mohanta3
PG Student, MVGR College of Engineering, Vizianagaram, Andhrapradesh,India Assistant Professor of EEE, MVGR College of
Engineering, Vizianagaram, Andhrapradesh,India Professor of EEE, MVGR College of Engineering, Vizianagaram,
Andhrapradesh,India(Formerly Professor, BITS, Mesra)
Abstract: This paper deals with the staging the evaluation of a three– phase Shunt Active Power Filter for Power Quality
improvement. Power Quality problems can be take up using Shunt Active Power Filters. A simulation model based on mathematical
model of a three–phase shunt active power filter has been developed. The results show an inherent power quality improvement in
terms of reduced Total Harmonic distortion (THD) of source current. It also betrays power factor correction (PFC) properties to a
large extent as the source current and source voltage are in phase. Keywords– shunt active power filter (SAPFs), Total harmonic
distortion (THD), Proportional integral (PI) and Hysteresis current controller (HCC). INTRODUCTION In the present era of
advanced technologies, power electronics devices have wide applications. These devices are important for industrial and domestic
uses but they generate the power quality problems [1–2].these problems are created on account of these devices

Nt1210 Lab Report Paper
The fig.3 and 4 above is a simulated result showing the red is return loss (S11) and blue is insertion loss (S21) obtained for model.
Figure 4 shows the reflection and transmission factor measured for the range 1 GHz to 4GHz. This result gives that the bandpass
characteristics are valid for a wide range of EM spectrum for communication. In order to get detailed characteristics in the
neighborhood of 2.45 GHz, Figure 4 gives more information needed to the measurement results. With the 3 dB boundary, we get
about 260 MHz bandwidth, and insertion loss of –4.3 dB, with parametric dimension as given in Table.1. Table1– Comparision table
of coupling gap As shown from table.1, if the resonator gap between two parallel couple microstrips transmission lines is small then
the ... Show more content on Helpwriting.net ...
It shows that where the current is minimum & maximum. The red colours indicates current is maximum along the all resonators. 4.
Conclusion The simulation result exactly shows better insertion loss and return loss curves which define the characteristics of a
bandpass filter which operates at 2.45GHz. The pass–band of the filter is fine and free from noise disturbances such as higher order
harmonics, ripples, EMI etc. One of the techniques for BPF designing has been presented here. By varying the resonator length from
30mm – 34mm, but get better result at the 32mm as it gives much more enhanced bandwidth response over it.The resonator width
also concurrently varied by length from 0.6mm – 1.0mm, but get better result at 0.8mm width as it improves the more bandwidth.
The proposed filter is an efficient, high–performance and flexible filter. Because of frequency selective nature of BPF, it is used
mainly in reception device for electromagnetic interference reduction. Also the nearer frequency channels are separate out by proper
filtering at particular frequency

How Does The Band Pass Digital Filter Design Using Kaiser...
5.6 Band Pass FIR Digital Filter Design using kaiser Window The band pass FIR digital filter has been analysed with hanning
window by using FDA tool in the MATLAB. The cut off frequency has been estimated by using nn tool. 5.6.1 Training using Feed
Forward Back Propagation (FFBP) artificial Neural Network Algorithm for kaiser Window Figure 5.11: trained network for kaiser
window with FFBP Figure 5.12: performance plot for kaiser window with FFBP In above figure 5.12, it shows the performance plot
of a FFBP neural network. This plot is result of MLP training algorithm. Here after 7 epochs the mean square error is almost 0. That
means the network is trained. Figure 5.13: regression plot for kaiser window with FFBP Here figure 5.13 ... Show more content on
Helpwriting.net ...
Figure 5.16 shows an error graph between hamming window output obtained from the FDA tool of MATLAB and feed forward back
propagation algorithm output obtained from nn tool of MATLAB. Figure 5.16: Error graph between desired cut–off frequencies and
obtained cut–off frequencies for hamming window with FFBP Figure 5.17 shows an error graph between hamming window output
obtained from the

Design Of Low Pass Fir Filter
DESIGN OF LOW PASS FIR FILTER THROUGH
PARTICLE SWARM OPTIMIZATION
Abstract: In the common case, the impulse response is finite because there is no feedback in the FIR. A lack of feedback guarantees
that the impulse response will be finite. Therefore, the term "finite impulse response" is nearly synonymous with "no feedback".
However, if feedback is employed yet the impulse response is finite, the filter still is a FIR. In this paper FIR filter design through
particle swarm optimization.
Key words: FIR filter, particle swarm optimization
INTRODUCTION
FIR filters can also be designed from a frequency response specification. The equivalent sampled impulse response, which
determines the coefficients of the FIR filter, can then be ... Show more content on Helpwriting.net ...
It is developed from swarm intelligence and is inspired by social behaviour of bird flocking or fish schooling. PSO is an optimization
algorithm with implicit parallelism which can be easily handled with the non–differential objective functions. PSO algorithm uses a
number of particle vectors moving around in the solution space searching for the optimist solution. Every particle in the algorithm
acts as a point in the N–dimensional space. Each particle keeps the information in the solution space for each iteration and the best
solution is calculated that has obtained by that particle is called personal best (pbest). This solution is obtained according to the
personal experiences of each particle vector. Another best value that is tracked by the PSO is in the neighborhood of that particle and
this value is called (gbest) Figure 1 PSO flow chart.
SYSTEM DOMAIN
An RLC circuit (the letters R, L and C can be in other orders) is an electrical circuit consisting of a resistor, an inductor, and a
capacitor, connected in series or in parallel. The RLC part of the name is due to those letters being the usual electrical symbols for
resistance, inductance and capacitance respectively. The circuit forms a harmonic oscillator for current and will resonate in a similar
way as an LC circuit will. The main difference that the presence of the resistor makes is that any oscillation induced in the

Essay about Acoustic Theory and Synthesis
Acoustic Theory and Synthesis
Frequency:
Frequency means the number of cycles per second and depending on the amount of cycles per second determines how high or low
pitched the sound is and the time that it takes to complete one cycle is called the period. Frequency is measured in Hertz (Hz). And
An average human is able to hear sounds between 20Hz and 20,000Hz.
As the cycles per second increases, the smaller the wavelengths become, therefore there is a higher frequency which will cause the
pitch of the sound to increase or get higher.
If there is frequency of say 20KHz then it is going to be a much higher pitch than a 20Hz because there is a lot more cycles per
second.
Fundamentals:
A fundamental is the lowest frequency ... Show more content on Helpwriting.net ...
Sustain: This is how long the volume is sustained for during the main event of the sound. Release: The is how long it take for the
sound to fade out. An Example of a slow and fast release can be found on track 7 and 8 of my logic session which is with my
assessment. Filters: Filters can either emphasize or reduce some frequencies from a signal. There are different types of filters that can
be used. There are different types of filters that can be used. There are also filters that can reject frequencies within a certain
frequency band e.g. band bass or band reject.
High Pass Filter: A High Pass filter lets through all the frequencies above the cutoff and eliminates the ones below the cutoff
frequency. An example of a HPF can be found on track 10 of my logic session. !
Low Pass Filter: A Low Pass filter lets through the frequencies below the cutoff frequency and eliminates the ones above it. An
Example of a low pass filter can be found on track 9 of my logic session which is with my assessment.
!
ESM
The ESM synthesizer is a very basic synth to create simple bass and lead sounds that can emulate the Roland – TB 303 synth because
it is a monophonic synth and you cant play chords . With the ESM you can only choose between a saw tooth and square wave and it
only

Uses And Functions Of Oscilloscopes And The Methods Used...
Statement of Objectives
The objective of this lab was to gain understanding of the use and functions of oscilloscopes and the methods used to measure a
system's time constant. Several methods were used throughout the laboratory in order to gain familiarity with the equipment as well
as investigate characteristics of an RC high pass filter. The ability to problem solve and critically think was necessary for completion
of this laboratory and therefore an objective.
Apparatus
Table 1. List of Equipment Used During Session 1 and 2 of Laboratory
Device
Manufacturer
0.301μF Capacitor
Unknown
Variable Resistance Box
Omega Engineering
NI Soft Front Panel Oscilloscope
National Instruments
LabVIEW Software
National Instruments
DAQ ... Show more content on Helpwriting.net ...
For each resistor/capacitor pair, times required to achieve 63.2 percent of maximum voltage change and 86.5 percent of maximum
voltage change were recorded. For the third method, the oscilloscope was next set to produce a 300mV sine wave. The oscilloscope
was initially set to produce a very low frequency and amplitude was recorded. Frequency was increased slowly and amplitude was
recorded at each interval. Output amplitude was plotted vs frequency. 휏 was calculated using the amplitude and frequency results
and equation 3. 휏=½*흅*fc (3)
For methods 4 and 5, LabVIEW software was used to record full charge or discharge cycles. The data recorded was then normalized
using equation 4. The natural log of the normalized data results was plotted against t. For method 4, the linear portion of the result of
the plot was –1/휏. For method 5, the normalized data was used without a logarithmic function, plotted against time. An exponential
line of best fit was created resulting an equation equal to e–t/휏. Vnormalized = (4)
For session 2 a low pass filter was set up using the wiring diagram described in the apparatus section of this report. The same
capacitor and resistance values were used as in session one, along with the associated theoretical 휏 values. Theoretical gain values
were calculated using equation 5.
Vout/Vin = (5) The appropriate output signal, as

A Planar Dual-Band Filter based on the Reduced Length...
A Planar Dual–Band Filter based on the Reduced Length Parallel Coupled Lines
Seungku Lee and Yongshik Lee, Member, IEEE
Sowmya Ravichandran
Review paper summary: A Dual–Band Filter is obtained by cascading the reactively loaded parallel coupled lines of reduced length
(< 〖90〗^0).The proposed filter have same electrical performance at both the passbands at frequencies f1 and mf1, where m is the
frequency of separation between the two passbands. A complete set of design equations are derived for arbitrary number of Filter
stages using standard filter synthesis technique. This provides simple, low cost and high performance filter with multiple bands.
Introduction: Multiple channels are used by most of the modern communication devices. So ... Show more content on
Helpwriting.net ...
The ports 2 and 3 can be made open or short. This design operates only at one frequency (f1).The equation for designing this filter is
derived by even and odd mode analysis. The bandwidth of the filter with dual band is less compared to the conventional filter with
single frequency of operation. This reduction in bandwidth is due to the Shunt stubs which are used to load the parallel coupled lines.
Novel Design: The electrical length of the coupled line is reduced to 〖34.84〗^0 . This decrease in length can be obtained only by
increasing the frequency of separation m to 4.167.Hence the passband frequencies thus obtained are 2.4GHz and 10GHz.The even
and mode impedances of the coupled line and the impedance of the shunt stub are calculated using the filter design equations. The
shunt impedance value is found to be decreased when compared to filter with length 〖56.8421〗^0.The bandwidth at both the
frequencies are increased to 216MHz for a filter with 9% fractional bandwidth at 2.4GHz and 2.16% fractional bandwidth at
10GHz.Thus a reduced length filter would have increased bandwidth and decreased stub impedance. A graph is plotted for frequency
Vs insertion loss. The insertion loss is found to be 0.66dB at 2.4GHz and 19.8dB at 10GHz.A better insertion loss of 2.9dB is
obtained with a little shift in the frequency at 12.1GHz.
Table 1: 2.4/10GHz Dual band filter

Questions On Properties And Properties
1
2
/
5
/
2
0
1
6
F
i n a l p r o j e ct 1
2
/
5
/
2
0
1
6
F
i n a l p r o j e ct Filter Examples
f i l e :
/
/
/
D
:
/
M
a i n
/
L a k e h e a d /
E
N
G
I
5
6
3
1
F
A
_
B i o m e d i c a l /
P
y t h o n
N
o t e b o o ks
/

S u b m i ss i o n s /
F
i n a l P r o j e c t s /
E
_
M
a t h e w
_
P a u l j i /
F
i n a l + p r o j e c t . h t m l
1
/
1
7
1. Introduction
Filter is a device used that removes unwanted noise in the signal.
Classification of filter
Filters are classified into different categories depending on their transfer function.
Linear and nonlinear
Time variant and time invariant
Casual and non casual Analog and digital Passive a ndactive filter
Infinite impulse response (IIR) or finite impulse response (FIR) filter.
IIR(Infinite impulse response
IIR filters are digital filters with infinite impulse response. They have the feedback (a recursive part of a filter) and are known as
recursive digital filters. They are the time invariant systems. IIR filter has better frequency response than FIR filters.
FIR(Finite impulse response) The transfer function of FIR filter approaches the ideal as the filter order increases, thus increasing the
complexity and amount of time needed for processing input samples of a signal being filtered.
IIR filters require more multiplication and addition when compared to FIR, because FIR is of a higher order in comparison to IIR,
which is of lower order
Digital filter
A digital filter uses a digital processor

Reaction Paper On ECG Waveform
BioMedical Matlab Thesis – ECG Waveform Jessica Garcia,
BioMedical MatLab,Thesis Final Prof. Jonathan Ibera 11–9–17
An ECG, or electrocardiogram, is a waveform generated by the natural sinus rhythm of the heart. It is an electrical representation of
the heart beat in form of what is known as a "PQRST" Complex. The physical generation of this sinusoidal heart rhythm waveform is
generated by the expansion and contraction of the four different chambers of the heart that work collectively to pump blood through
the human body.
The biomedical piece of equipment used to measure this electrical activity of the heart is an Electro Cardiogram machine, also widely
known as an ECG. This electrical signal is acquired by electrodes placed on the human body. On screen, the heart rate appears in
form of a wave in which each pulse and different peaks if the waveform, known as the PQRST waveform, represents a small amount
of an electrical signal.
These signals have different peaks, which are representations of different levels of electrical intensity, are detected by the electrodes
that then transmit the energy to a sensor that helps the processor measure the actual value of this peak. Below is a representation of
the heart rate waveform and its named peaks.
Upon analysis of these waveforms collected by medical professional, they are able to tell many things. They are able to analyze and
assess regular heart rhythms as well as any other abnormal

Advantages And Disadvantages Of Power Filter
CHAPTER # 02
PASSIVE FILTERS
2.1 INTRODUCTION
2.2 FILTERS
2.3 ACTIVE FILTER
2.4 PASSIVE FILTERS
2.5 CLASSIFICATION OF PASSIVE FILTER
2.4 DISADVANTAGES OF PASSIVE FILTER
2.6 CONCLUSION
2.1 INTRODUCTION
The increase of electronic equipments has caused the increase in the nonlinear loads and due to these loads power quality is being
deteriorated. The supply voltage is distorted due to harmonic currents drawn from the supply by nonlinear loads. Distorted currents
and voltage cause the user's equipment to not work properly, conductors will overheat and may decrease the efficiency and life of the
equipment. Usually a passive LC power filter is employed to wipe out the current harmonics ... Show more content on
Helpwriting.net ...
The parallel LC circuits create the very high impedance to the unwanted signal and a series circuit forms very low impedance
passage to ground at same frequency.
2.4 DISADVANTAGES OF PASSIVE FILTER
Following are the disadvantages of the passive filters:
Filter impedance should be smaller than the source impedance for mitigating the source current harmonics and hence compensation
characteristics depend on system impedance.
Overload condition can arise in the passive filter because of harmonics circulation arising from nonlinear loads which are connected
at the connection point of passive filters.
They are not recommended for the variable loads because their design is for some specific reactive power and changing the load can
detune the filter.
Resonances either in series or in parallel may appear in the system. [7]
Passive filters do not use active elements and hence signal gain is not provided by them.
Tuning of passive filters requires a lot of time and they are expensive when produced in large quantities.
Passive filters higher than the 2nd order are complex and very difficult to design them.

Essay On Wireless Filter
|| Volume 2 ||Issue 4 || 2017 ISSN (Online) 2456–3293 COMPARISON OF MICROSTRIP BAND PASS AND BAND REJECT
FILTER DGS TECHNIQUE BASED IN GSM BAND (1850MHZ) Richa Chaturvedi11, Dr. Laxmi Shrivastava2 Department of
Electronics and Communication Engineering, Madhav institute of technology and science, Gwalior1 2
richa.chaturvedi98@gmail.com
––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––––
Abstract: In microwave communication filters are more essential high frequency component. To improve the insertion loss the DGS
technique is used. DGS i.e, (defected ground structure) is a technique where cuts are drawn on ground to improve the results and
reduce the size of filters. ... Show more content on Helpwriting.net ...
The filters using DGS technique has an attractive feature, which are– 1) As compared to the conventional low pass filters, the stop
band is very wide and deeper. 2) It has a very simple structure and cost is very low 3) The losses will be reduced 4) Easy to fabricate
[5] II DESIGN 2.1 Design of Hairpin band pass filter–In the band pass filter by without DGS the dimension of the filters are having
4cm×3cm.The insertion loss of this filter is –25.1850 dB, which will be improved in DGS technique Figure 1 Hairpin band pass filter
Figure 2 Insertion loss of band pass filter WWW.OAIJSE.COM 8 || Volume 2 ||Issue 4 || 2017 ISSN (Online) 2456–3293 OPEN
ACCESS INTERNATIONAL JOURNAL OF SCIENCE &ENGINEERING In this figure the simulated insertion loss of band pass
filter without DGS is shown i.e., ––28.197 dB. N g1 g2 g3 g4 g5 g6 1 0.3052 1.0 2 0.8431 0.6220 1.3554 3 1.0315 1.1474 1.0315 1.0
4 1.1088 1.3062 1.7704 0.8181 1.3554 5 1.1468 1.3712 1.9750 1.3712 1.1468 1.0 2.2 Design of Band Pass filter with DGS
Technique (Hexagonal ring shape) Where FL=lower frequency, FH=Higher frequency, FC=cut off frequency.[6] M12=M45=0.3538
M23=M34=0.205 Qe1=Qe5=2.0625 By varying the spacing between the resonators, the coupling coefficient can be varied, by using
the formula, K= (f22–f12) / (f22+f12) Where f1

Advantages And Disadvantages Of Resonance

Recommended

Recommended

More Related Content

Similar to Advantages And Disadvantages Of Resonance

Similar to Advantages And Disadvantages Of Resonance (19)

More from Carolina Cardona

More from Carolina Cardona (20)

Recently uploaded

Recently uploaded (20)

Advantages And Disadvantages Of Resonance