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Feb 16, 2013

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power line communication report

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power line communication report

Attribution Non-Commercial (BY-NC)

- AK4520_datasheet
- Dsp Lab Sample Viva Questions
- Powerline Communication Seminar Report
- DsPIC Filter Design Sell Sheet_51438b
- Solution Chapter8
- iirfilterdesine
- Equalizer부분-rev2
- Complex Eigenfilter
- Advance DSP Ch8
- Filter Design Matlab
- Task 03
- Chap7 Filter Design FIR
- ICTSyllabus2010-2012
- ZCD
- dsp1
- CourseInfoCASE2013.pdf
- Mat Lab Intro
- MX7705
- Pre-processing of ECG Signals Using Filters
- Harold John Hayes2

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CONTENTS Sl no 1 2 3 4 5 6 7 8 9 10 11 Topic ABSTRACT INTRODUCTION FILTERING OF POWER LINE NOISE DLC NOISE CHARACTERISTICS ADAPTIVE FILTERS ADAPTIVE FILTER STURUCTURE ADAPTIVE FILTERING ADAPTIVE NOISE CANCELLER WIENER FILTER FIR-ALE Harmonic Noise Cancelling Filters IMPLEMENTATION OF DLC NOISE CANCELLERS 12 ADVATAGES OF POWER LINE COMMUNICATION 13 DISADVANTAGES OF POWER LINE COMMUNICATION 14 15 16 CONCLUTION FUTURE SCOPE REFERENCES 17 17 17 17 17 Page no 3 4 5 5 6 6 7 8 9 13 14

Page 1

1 2 3

8 9 13

4 5

ADAPTIVE LINEAR ENHANCER ADAPTIVE NOISE CANCELLER REQURING SIGNAL INTERRUPTION METHOD 1

14 16

15

Page 2

ABSTRACT

Power line communication (PLC) is a communications method where signals are carried across already existing power lines, rather than through other mediums such as optical fibres or radio waves. PLC has been a very important inter discipline topic for power, communications, industrial, and automation engineers and researchers since the 1980s. Designing and planning optimized or standard PLC systems require information and data regarding signal attenuations, load variations, and communication performance for various locations of the world which have different power networks. Power line communications provide a convenient and cost-effective solution for data transmission, because power mains are the most popular and widely distributed medium in the world. However, the time-variant and unpredictable channel environment for communication poses a major challenge.

Harmonic disturbances caused by non-linear loads such as power switching devices often occur in the domestic and commercial power distribution systems can affect the signal. So to avoid this Adaptive

LMS filters are applied to cancel harmonic noise. Two implementation methods are discussed.

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INTRODUCTION

We use several communication methods to exchange data. These communication means could either be uneconomical or could require rewiring scheme to build a LAN in an existing building. These problem can be solved by using existing power lines. This avoids rewiring . PLC techniques have the potential to economically install a domestic network into any building, new or old, and connect any room in that building that has a power point. But Power lines are a very noisy environment in which to transmit signals, and are divided into two broad categories by the utility industry: transmission and distribution power lines. Transmission power lines are those which carry power from the generator to the distribution substations where they branch out to individual homes; at this point they are called distribution power lines. All manner of devices are connected to the distribution power lines. These devices, such as light dimmers, universal motors, and other common household appliances might contain switching devices which can inject large voltage current spikes anywhere onto the distribution power lines. The large spikes, which are generated by switching the loads on the lines, are synchronous with the primary 50 Hz power signal such that they appear as 50 Hz harmonics. Generally, these harmonics are small compared to the 50 Hz fundamental. However, when it is desired to transmit signals over distribution power lines, these harmonics can be a serious problem, especially when signals are transmitted over a distance long enough to cause substantial attenuation of the signal. Transmission power lines do not have consumer or industrial switching devices connected to them because they are high voltage lines used exclusively to carry power from the generator to the distribution substations. They are therefore much cleaner than distribution power lines, and make a better transmission medium. However, the current interest in power line communications is directed toward the distribution power line so that the utilities may access individual homes and commercial customers

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Most distribution line communication (DLC) systems attempt to avoid the effects of the harmonic noise by placing carrier frequency between the spikes of the harmonic noise in the frequency domain. These synchronous DLC systems assume that the noise occurs at fixed harmonics (multiples of 50 Hz) by designing the signal spectrum to be zero at the noise harmonics .A serious problem that these systems encounter is that the characteristics of harmonic noise are affected by drifting power frequency and drifting harmonic bandwidth. The 60 Hz power signal can vary about plus or minus 0.05 percent, and the harmonics are not ideal sinusoids; their nonzero bandwidth can also vary. The variation of the power frequency becomes significant at high frequencies. That is for 200th harmonics noise (10khz) the signal varies from 9.99khz to 12.01khz that is bandwidth is 20.2hz. Thus, the DLC systems which use fixed filters are susceptible to variation in the harmonic noise .In order to overcome this adaptive filters are used.

The major characteristic of the harmonic noise is that the harmonics are multiples of 50 Hz. Not all signals have all harmonics present nor is their power evenly distributed. The main reason for this effect is deviation in the exact trigger times of some switching devices. A secondary characteristic of this type of noise is the deviation from exact 50Hz harmonics as the fundamental power frequency varies. This is quite noticeable at power line carrier frequencies typically above 4 kHz.

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ADAPTIVE FILTERS

An adaptive filter is defined as a self-designing system that relies for its operation on a recursive algorithm, which makes it possible for the filter to perform satisfactorily in an environment where knowledge of the relevant statistics is not available. Adaptive filters are classified into two main groups: linear, and non linear. Linear adaptive filters compute an estimate of a desired response by using a linear combination of the available set of observables applied to the input of the filter. Otherwise, the adaptive filter is said to be nonlinear. Adaptive filters may also be classified into: (i) Supervised adaptive filters, which require the availability of a training sequence that provides different realizations of a desired response for a specified input signal vector. The desired response is compared against the actual response of the filter due to the input signal vector, and the resulting error signal is used to adjust the free parameters of the filter. The process of parameter adjustments is continued in a step-by-step fashion until a steady-state condition is established. (ii) Unsupervised adaptive filters, which performs adjustments of its free parameters without the need for a desired response. For the filter to perform its function, its design includes a set of rules that enable it to compute an input-output mapping with specific desirable properties. In the signal-processing literature, unsupervised adaptive filtering is often referred to as blind de convolution or blind adaptation.

Adaptive-Filter Structure:

The adaptive filter can be implemented in a number of different structures or realizations. The choice of the structure can influence the computational complexity (amount of arithmetic operations per iteration) of the process and also the necessary number of iterations to achieve a desired performance level. Basically, there are two major classes of adaptive digital filter realizations, distinguished by the form of the impulse response, namely the finite-duration impulse response (FIR) filter and the infinite-duration impulse response (IIR) filters. FIR filters are usually implemented with nonrecursive structures, whereas IIR filters utilize recursive realizations. Adaptive FIR filter realizations: The most widely used adaptive FIR filter structure is the

DEPT OF ECE, PESSE Page 6

transversal filter, also called tapped delay line, that implements an all-zero transfer function with a canonic direct form realization without feedback. For this realization, the output signal is a linear combination of the filter coefficients, that yields a quadratic mean-square error function with a unique optimal solution. Other alternative adaptive FIR realizations are also used in order to obtain improvements as compared to the transversal filter structure, in terms of computational complexity, speed of convergence, and finite word length properties as will be seen later in the book. Adaptive IIR filter realizations: The most widely used realization of adaptive IIR filters is the canonic direct form realization due to its simple implementation and analysis. However, there are some inherent problems related to recursive adaptive filters which are structure dependent, such as pole-stability monitoring requirement and slow speed of convergence. To address these problems, different realizations were proposed attempting to overcome the limitations of the direct form structure.

Adaptive Filtering

Adaptive filtering can be considered as a process in which the parameters used for the processing of signals changes according to some criterion. Usually the criterion is the estimated mean squared error or the correlation .The adaptive filters are time-varying since their parameters are continually changing in order to meet a performance requirement. In this sense, an adaptive filter can be interpreted as a filter that performs the approximation step online. Usually the definition of the performance criterion requires the existence of a reference signal that is usually hidden in the approximation step of fixed-filter design. The general set up of adaptive filtering environment shown in Fig 1, where k is the iteration number, x(k) denotes the input signal, y(k) is the adaptive filter output, and d(k) defines the desired signal. The error signal e(k) is calculated as d(k)-y(k). The error is then used to form a performance function or objective function that is required by the adaptation algorithm in order to determine the appropriate updating of the filter coefficients. The minimization of the objective function implies that the adaptive filter output signal is matching the desired signal in some sense.

Page 7

d(k)

+

+

e(k)

x(k)

FILTER w(k)

y(k)

ADAPATIVE ALGORITHM

Adaptive filter is widely used as noise canceller. In an adaptive noise canceller (Fig. 2) two input signals, d(k) and x(k), are applied simultaneously to the adaptive filter. The signal d(k) is the contaminated signal containing both the desired signal, s(k), and the noise n(k), assumed uncorrelated with each other. The signal, x(k), is a measure of the contaminating signal which is correlated in sole way with n(k), x(k) is processed by the filter to produce an estimate y(k), of n(k). An estimate of the desired signal, e(k) is then obtained by subtracting the digital filter output, y(k), from the contaminated signal

Page 8

d(k) = s(k)+n(k)

++ -

e(k)

x(k)

Adaptive filters

y(k)

If d(k), n(k) & x(k) are jointly wide sense stationary processes, and if the autocorrelation rx(k) and cross correlation rxn(k) are know then wiener filter can be designed to find the minimum mean squared estimate

Wiener filter

In this section, we consider another filter design problem that involves removing the noise from a sum of a desired signal plus noise. In this case, the desired signal is a random process and the goal here is to estimate the desired part of the signal plus noise. In its most general form, the problem is stated as follows. Given a random process X(t), we want to form an estimate Y(t) of some other zero-mean process Z(t) based on observation of some portion of X(t). We require the estimator to be linear. That is, we will obtain Y(t) by filtering X(t). Hence, Y(t)=

h(t u)X(u) du

We want to design the filter to minimize the mean square error E[2(t)] = E[(Z(t) Y(t))2].

In this section, we will consider the special case where the observation consists of the process we are trying to estimate plus independent noise. That is X(t) =

Page 9

Z(t) + N(t). We observe X(t) for some time interval t (t1, t2) and based on that observation, we will form an estimate of Z(t). Consider a few special cases:

Case I. If (t1, t2) = (, t-t0), then we have a prediction problem in which we must estimate the present based on the entire past. We may also have (t1, t2) = (t to, t), in which case we have a prediction problem where we must estimate the present based on the most recent past. Case II. If (t1, t2) = (,), then we have a smoothing problem where we must estimate the present based on a noisy version of the past, present, and future. Case III. If (t1, t2) = (, t ), then we have a filter problem where we must estimate the future based on the past and present .

All of these cases can be cast in the same general framework and a single result will describe the optimal filter for all cases. In order to derive the optimal filter, it is easier to view the problem in discrete time and then ultimately pass to the limit of continuous time. Hence, we reformulate the problem in discrete time. Given an observation of the discrete time process X[n] = Z[n]+N[n] over some time interval n [n1, n2], we wish to design a filter h[n] such that the linear estimate

Y[n] =

h[n k]X[k]

The filter h[n] can be viewed as a sequence of variables. We seek to jointly optimize with respect to each variable in that sequence. This can be done by differentiating with respect to each variable and setting the resulting equations equal

Page 10

to zero:

E[2[n] ]= 2E[[n]

Noting that =

]=0

(Z[n]-Y[n])= -

= -X[n-m],

E[[n]X[n-m]]=0,

for m

[n-n1,n-n2]

E[[n]X[m]] = 0, for m

[n1, n2].

In summary, the filter that minimizes the mean square error will cause the observed data to be orthogonal to the error. This is known as the orthogonality principle. Applying the orthogonality principle, we have

E[[n]X[m]] = E

[(Z[n]

h[n-k]X[k] x[m]

) ]

=0;

h[n-k]Rxx[k-m]=Rzx[n-m], m [n1,n2],

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Or equivalently,

These equations are known as the Wiener-Hopf equations, the normal equations, or the Yule-Walker equations. The resulting filter found by solving this system of equations is known as the Wiener filter. A similar result can be found for continuous time systems by applying the orthogonality principle in continuous time. Given an observation of X(t) over the time interval (t1, t2), the orthogonality principle states that the filter that minimizes the mean square prediction error will satisfy

E[(t)X(s)] = 0,

for s

(t1, t2).

h(v)RXX( v) dv = RZX( ),

(t t2, t t1).

The techniques used to solve the Wiener-Hopf equation depend on the nature of the observation interval. In particular, however, a stationary assumptions is not generally appropriate and, even if it were, the required statics are unknown generally. Therefore, as an alternative to wiener filter, let us consider a adaptive noise cancellation filter as shown in fig 3

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X(n)=d(n)+v1(n)

+

+

e(n)=v^1(n)

-n0

Adaptive filter

d^(n)

_

fig (3) ADATIVE FILTER WITH NO SEPARAETE REFERENCE INPUT The adaptive noise canceller(ANC), unlike the Wiener filter, does not require any statistical information one of the advantages of this adaptive noise canceller over Wiener filter is that it may be used when the processes non stationary . Unfortunately, in many of the cases or applications reference signal is not available and another approach must considered. In some cases, however, it is possible to derive a reference signal by simply delaying the process X(n)=d(n)+v1(n) For example, suppose if d(n) & v1(n) are uncorrelated &

E[v1(n)v1(n-k)]=0; |k|>k0,

Then

E[v1(n)x(n-k)]=E[v1(n)d(n-k)]+E[v1(n)v1(n-k)]=0,|k|>k0

Therefore, if n0>k0 then the delayed process x(n-n0) will be uncorrelated with the noise v1(n),and correlation with d(n) (assuming that d(n) is a broad band process). Thus, x(n-n0) may used as a reference signal to estimate d(n) as in fig(3)

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An adaptive noise cancelling filter has two inputs. One is called the primary input which contains a noise corrupted signal. (Noise is defined as any undesired signal.) The other is called the reference input whose data samples are correlated with the noise in the primary input. Adaptive algorithms are used to adjust a set of filter taps (weights) which multiply data samples in the reference input to predict the noise in the primary input. The predicted noise is subtracted from the current data sample in the primary input, and hopefully the corrupting noise can be cancelled. For many adaptive noise cancelling applications, a reference input is not available However, if the primary input is composed of two types of data, one with long-term correlation and the other with short term correlation, a delayed version of the primary input can be used as the reference input. This so-called decorrelation delay should be long enough to remove any short-term correlation and still keep most of the long-term correlation. This adaptive scheme is usually referred to as the adaptive line enchancement (ALE). Its block diagram is shown in Fig. 4

s(n)=sl(n)+ss(n) s(n- )

ss^(n)

_

Adaptive filter

sl^(n)

Decorrelation Delay Sl(n) is the signal component with long-term correlation. Sf(n) is the signal component with short-term correlation fig(4) adaptive line enchancement

As was mentioned earlier, adaptive filters have long been used in noise cancellation. The emphasis in this work is to show how they can be used to eliminate harmonic noise in distribution power line communications. Several approaches are tested. Lets see only the most promising two approaches. If the data signal is properly scrambled and if the channel

DEPT OF ECE, PESSE Page 14

distortion is fairly small, only very short-term correlation exists among data samples. The most straightforward way (method 1) to implement the noise canceller in a distribution power line communication system is to simply use the ALE structure. However, if the data are not properly scrambled or if the channel distortion is severe which is the case in distribution power line communications, there is a need to alternate periods of signaling with periods of filter adaptation. The LMS filter is allowed to adapt only during periods of silence. The schematic diagram of alternating periods of signaling with periods of filter adaptation is shown in Fig. 5 more elegant method can be derived using the characteristic properties of the harmonic noise. A second method (method 2) is shown in Fig.6 . This method does not require alternating periods of signaling with periods of filter adaptation. Since the DLC noise is the result of periodic spikes in the time domain, it is clear that harmonics will occur over a wide frequency range. The harmonic noise which causes problems in the signal band is very much like the noise outside of this band. The LMS noise canceller in the second method is used to cancel the near-by out-of-band noise. The filter taps which are obtained by this adaptation are copied to the noise cancelling processor to filter the data in the signal band. If the out-of band demodulation frequency wi in Fig. 6 is properly chosen, input + output

LPF

Coswct

Copied filter

LPF

Coswit

Adaptive filter

Page 15

input

output

LPF

coswct

Adaptive filter

SW

the filter for one band should perform well in the other band. The choice for wi is a frequency which maps the harmonics in the demodulated out-of-band noise to the same frequencies as the demodulated harmonics in the communication band. This method is justified by an important result in the ALE: if a large number of filter taps is used, the converged ALE solution only depends on the frequency of the input sinusoid, the decorrelation delay, and the number of filter taps, Since the two ALE'S in this method have the same decorrelation delay and wi is properly chosen, this method should cancel the harmonic noise as efficiently as the method 1. In practice, the frequencies of the harmonics in these two near-by bands may differ by a small amount. However, if this amount is small compared to the bandwidth of the harmonics, this difference only slightly degrades the performance as to be shown in our results. There are some additional advantages of using the method In the real-time implementation of a fractionally spaced equalizer used in high-speed modems, the LMS algorithm could run into long-term instability problem . This has also been observed in the implementation of a real-time harmonic LMS noise canceller . A common technique to ameliorate this problem is to inject additional noise to the input. This noise can be introduced either by adding actual noise or by modifying the adaptive algorithm. Since in method 2 additional noise is injected onto the out-of-band input rather than the inband data, the performance degradation is small if there is any. Injecting additional noise can possibly provide an additional advantage. However, this is not feasible in method 1 since the

Page 16

additional noise is directly injected onto the signal itself. Finally, since the out-of-band noise is signal free, a larger step size can be used to achieve quicker convergence.

The advantage of power line communication is to avoid new installation of wire (communication channel) where the new installation imposes limitation due to place and cost. Also The availability of power line plugs extends the flexibility of power line as a communication channel.

This system requires high amount of isolation because of 230v power signal. Since distribution lines are metal wire at higher frequency they radiate which may interfere with the wireless signals and distorting them, therefore this is prohibited in military areas.

Conclusion

I hereby conclude that power line communication is one of the promising channel for communication in the presence adaptive noise cancellers

Future scope:

Noise is not the only entity which is responsible for affecting performance power line communication, impedance of the power line should also be considered, impedance is also time varying because of the channel time varying characteristics and any device which are plugged or unplugged to line comes in parallel to the hence varying the impedance.

_ Communications 2. S. Haykin, Adaptive Filter Theory, Prentice Hall, Englewood Cliffs, NJ, 4th edition, 2002. 3. The book Adaptive filter theory 4th edition by simon haykin .

DEPT OF ECE, PESSE Page 17

JULY 1988

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