I.

ADAPTIVE ECHO CANCELLATION

ABSTRACT: Echo is the repetition of a waveform due to reflection from points where the characteristics of the medium through which the wave propagates changes. The perceptual effects of an echo depend on the time delay between the incident and reflected waves, the strength of the reflected waves, and the number of paths through which the waves are reflected. Echo can severely affect the quality and intelligibility of voice conversation in a telephone system. The perceived effect of an echo depends on its amplitude and time delay. In general, echoes with appreciable amplitude and a delay of more than 1 ms are noticeable. Telephone line echoes, and acoustic feedback echoes in teleconference and hearing aid systems, are undesirable and annoying and can be disruptive. Echo cancellation was developed in the early 1960s by AT&T Bell Labs and later by COMSAT TeleSystems. KEYTERMS: Echo; Acoustic echo; Telephone line echo; Adaptive echo cancellation; Adaptation method INTRODUCTION There are two types of echo in a telephone system. Fig 1 shows the 2 varieties of echoes that is possible in a mobile to landline system. (a) Acoustic echo due to acoustic coupling between the speaker and the microphone in hands-free phones, mobile phones and teleconference systems. Acoustic echo may be reflected from a multitude of different surfaces, such as walls, ceilings and floors, and travels through different paths. Acoustic echo results from a feedback path set up between the speaker and the microphone in a mobile phone, hands-free phone, teleconference or hearing aid system. (b) Electrical line echo or telephone line hybrid echo due to mismatch at the hybrid circuit connecting a 2-wire subscriber line to a 4-wire truck line in the public switched telephone network.

Fig 1: Illustration of echo in a mobile to land line system In the early days of expansion of telephone networks, the cost of running a 4-wire line from the local exchange to subscribers premises was considered uneconomical. Hence, at the exchange the 4-wire truck lines are converted to 2-wire subscribers local lines using a 2/4-wire hybrid bridge circuit. At the receiver due to any imbalance between the 4/2-wire bridge circuit, some of the signal energy of the 4-wire circuit is bounced back towards the transmitter, constituting an echo signal. If the echo is more than a few milliseconds long then it becomes noticeable, and can be annoying and disruptive. In digital mobile phone systems, the voice signals are processed at two points in the network: first voice signals are digitised, compressed and coded within the mobile handset, and then processed at the radio frequency interface of the network. The total delay introduced by the various stages of digital signal processing range from 80 ms to 100 ms, resulting in a total round-trip delay of 160200 ms for any echo. A delay of this magnitude will make any appreciable echo disruptive to the communication process. Owing to the inherent processing delay in digital mobile communication systems, it is essential and mandatory to employ echo cancellers in mobile phone switching centres.

ADAPTIVE ECHO CANCELLATION

Fig 2 below illustrates the operation of an adaptive line echo canceller. The speech signal on the line from speaker A to speaker B is input to the 4/2 wire hybrid B and to the echo canceller. The echo canceller monitors the signal on line from B to A and attempts to model and synthesis a replica of the echo of speaker A.

This replica is used to subtract and cancel out the echo of speaker A on the line from B to A. The echo canceller is basically an adaptive linear filter. The coefficients of the filter are adapted so that the energy of the signal on the line is minimised. The echo canceller can be an infinite impulse response (IIR) or a finite impulse response (FIR) filter. The main advantage of an IIR filter is that a long-delay echo can be synthesised by a relatively small number of filter coefficients. In practice, echo cancellers are based on FIR filters. This is mainly due to the practical difficulties associated with the adaptation and stable operation of adaptive IIR filters.

Fig 2: Block diagram illustration of an adaptive echo cancellation system Assuming that the signal on the line from speaker B to speaker A, is composed of the speech of speaker B A ( ), we have ( ) ( ) ( ) ( )

( ), plus the echo of speaker

In practice, speech and echo signals are not simultaneously present on a phone line. This, as pointed out shortly, can be used to simplify the adaptation process. Assuming that the echo synthesiser is an FIR filter, the filter output estimate of the echo signal can be expressed as ( )= ( ) ( )

where and

( ) are the time-varying coefficients of an adaptive FIR filter

( ) is an estimate of the echo of speaker A on the line from speaker B to

speaker A. The residual echo signal, or the error signal, after echo subtraction is given by e(m) = = ( ) ( ) ( ) ( ) ( ) ( )

For those time instants when speaker A is talking, and speaker B is listening and silent, and only echo is present from line B to A, we have e(m) = ( )= = where ( ) ( ) ( ) ( ) ( )

( ) is the residual echo. An echo canceller using an adaptive

FIR filter is illustrated in Fig 3 below. The magnitude of the residual echo depends on the ability of the echo canceller to synthesise a replica of the echo, and this in turn depends on the adaptation algorithm.

Fig 3: Illustration of an echo canceller using an adaptive FIR filter and incorporation a echo/speech classifier

ECHO CANCELLER ADAPTATION METHODS

The echo canceller coefficients ( ) and ( ) are adapted to minimise the energy ( )are uncorrelated, the energy on the ( ) is

of the echo signal on a telephone line, say from speaker B to speaker A. Assuming that the speech signals telephone line from B to A is minimised when the echo canceller output equal to the echo

( ) on the line. The echo canceller coefficients may be

adapted using one of the variants of the recursive least square error (RLS) or the least mean squared error (LMS) adaptation methods. One of the most widely used algorithms for adaptation of the coefficients of an echo canceller is the normalised least mean square error (NLMS) method. The time-update equation describing the adaptation of the filter coefficient vector is w(m) = w(m-1)+ where ( )=[ w(m)=[ ( ), ..., ( ), ..., (
( ) ( ) ( )

( )

)] and ( )] are the input signal vector and the

coefficient vector of the echo canceller, and e(m) is the difference between the signal on the echo line and the output of the echo synthesiser. Note that the normalising quantity ( ) ( ) is the energy of the input speech to the adaptive filter. The scalar is the adaptation step size, and controls the speed of convergence, the steadystate error and the stability of the adaptation process. SUMMARY Telephone line echo and acoustic feedback echo affect the functioning of telecommunication and teleconferencing systems. In general, line echo cancellation, is a relatively less complex problem than acoustic echo cancellation because acoustic cancellers need to model the more complex environment of the space of a room. For adaptation of an echo canceller, the LMS or the RLS adaptation methods can be used. The RLS method provides a faster convergence rate and better overall performance at the cost of higher computational complexity.

II. ADAPTIVE EQUALISATION

ABSTRACTThe recent digital transmission systems impose the application of channel equalizers with short training time and high tracking rate. Equalization techniques compensate for the time dispersion introduced by communication channels and combat the resulting inter-symbol interference (ISI) effect. Given a channel of unknown impulse response, the purpose of an adaptive equalizer is to operate on the channel output such that the cascade connection of the channel and the equalizer provides an approximation to an ideal transmission medium. Typically, adaptive equalizers used in digital communications require an initial training period, during which a known data sequence is transmitted. A replica of this sequence is made available at the receiver in proper synchronism with the transmitter, thereby making it possible for adjustments to be made to the equalizer coefficients in accordance with the adaptive filtering algorithm employed in the equalizer design. KEYWORDS- Channel Equalizer; Adaptive Equalizer; Least Mean Square; Recursive Least Squares. INTRODUCTION The digital transmission of information is accompanied with a phenomenon known as intersymbol interference (ISI). Depending on the transmission media the main causes for ISI are: Cable lines the fact that they are band limited; Cellular communications multipath propagation Equalizer is meant to work in such a way that BER (Bit Error Rate) should be low and SNR (Signal-to-Noise Ratio) should be high. ISI problem is resolved by channel equalization in which the aim is to construct an equalizer such that the impulse response of the channel/equalizer combination is as close to where as possible,

is a delay. Frequently the channel parameters are not known in advance and

moreover they may vary with time, in some applications significantly. Hence, it is necessary to use the adaptive equalizers, which provide the means of tracking the channel characteristics. The following figure shows a diagram of a channel equalization system.

Fig 1: Digital transmission system using channel equalization In the fig 1, s(n) is the signal that you transmit through the communication channel, and x(n) is the distorted output signal. To compensate for the signal distortion, the adaptive channel equalization system completes the following two modes: - This mode helps you determine the appropriate coefficients of the adaptive filter. When you transmit the signal s(n) to the communication channel, you also apply a delayed version of the same signal to the adaptive filter. In the previous figure, is a delay function and d(n) is the delayed signal, y(n) is the output signal

from the adaptive filter and e(n) is the error signal between d(n) and y(n) . The adaptive filter iteratively adjusts the coefficients to minimize e(n) .After the power of e(n) converges, y(n) is almost identical to d(n), which means that you can use the resulting adaptive filter coefficients to compensate for the signal distortion. -directed mode - After you determine the appropriate coefficients of the adaptive filter, you can switch the adaptive channel equalization system to decisiondirected mode. In this mode, the adaptive channel equalization system decodes the signal and y(n) produces a new signal, which is an estimation of the signal s(n) except for a delay of taps. Here, Adaptive filter plays an important role. The structure of the adaptive filter is as shown in following fig 2.

Fig 2: Adaptive filter To start the discussion of the block diagram we take the following assumptions: The input signal is the sum of a desired signal d(n) and interfering noise v(n) , x(n)=d(n)+v(n) The variable filter has a Finite Impulse Response (FIR) structure. For such structures the impulse response is equal to the filter coefficients. The coefficients for a filter of order p are defined as ( ) ( ) ( )

the error signal or cost function is the difference between the desired and the estimated signal e(n)=d(n)- ( ) The variable filter estimates the desired signal by convolving the input signal with the impulse response. In vector notation this is expressed ( ) Where x(n)= ( ) ( ) ( * x(n ) ) is an input signal vector.

Moreover, the variable filter updates the filter coefficients at every time instant

Where

is a correction factor for the filter coefficients. The adaptive

algorithm generates this correction factor based on the input and error signals.

ADAPTATION ALGORITHMS
There are two main adaptation algorithms one is least mean square (LMS) and other is Recursive least square filter (RLS). LEAST MEAN SQUARES ALGORITHM Least mean squares (LMS) algorithms are a class of adaptive filter used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean squares of the error signal (difference between the desired and the actual signal). It is a stochastic gradient descent method in that the filter is only adapted based on the error at the current time. LMS filter is built around a transversal (i.e. tapped delay line) structure. Two practical features, simple to design, yet highly effective in performance have made it highly popular in various application. It consists of 2 procedures: the filtering process and adaptive process. The main features are as listed below. i. LMS is the most well-known adaptive algorithms by a value that is proportional to the product of input to the equalizer and output error. ii. LMS algorithms execute quickly but converge slowly, and its complexity grows linearly with the no of weights. iii. Computational simplicity iv. In which channel parameter dont vary very rapidly RECURSIVE LEAST SQUARES ALGORITHM The Recursive least squares (RLS) adaptive filter is an algorithm which recursively finds the filter coefficients that minimize a weighted linear least squares cost function relating to the input signals. This in contrast to other algorithms such as the least mean squares (LMS) that aim to reduce the mean square error. In the derivation of the RLS, the input signals are considered deterministic, while for the LMS and similar algorithm they are considered stochastic. Compared to most of its competitors, the RLS exhibits extremely fast convergence. However, this benefit

comes at the cost of high computational complexity, and potentially poor tracking performance when the filter to be estimated changes. SUMMARY Bandwidth-efficient data transmission over telephone and radio channels is made possible by the use of adaptive equalization to compensate for the time dispersion introduced by the channel. Spurred by practical applications, a steady research effort over the last two decades has produced a rich body of literature in adaptive equalization and the related more general fields of reception of digital signals, adaptive filtering, and system identification. There is still more work to be done in adaptive equalization of nonlinearities with memory and in equalizer algorithms for coded modulation systems. However, the emphasis has already shifted from adaptive equalization theory toward the more general theory and applications of adaptive filters, and toward structures and implementation technologies which are uniquely suited to particular applications.