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    ACOUSTIC ECHO

    CANCELLATION FOR DOUBLETALK

    BYS IMRANA 07H81A0417

    FATIMA TAYYBA

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    INTRODUCTION

    The purpose of this project is to cancel

    Acoustic echo.Before implementing in real

    time on DSP kit.

    we have simulated in Matlab and found theresponse for different speech signals.

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    BACKGROUND

    SPEECH SIGNAL.

    ECHO SIGNAL

    Echo is a phenomenon in which a delayed and

    distorted version of an original sound or electrical

    signal is reflected back to the source.

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    Adaptive Filter

    The general adaptive filter ,filters the input signal

    so that it resembles (in some sense) the desired

    signal input. An adaptive FIR or IIR filter designs

    itself based on the characteristics of the input

    signal to the filter.

    It iteratively alter its parameters.

    It minimizes the error between some desiredsignal and some reference signal.As shown in

    figure.

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    Block Diagram Of AdaptiveFilter

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    Diagram Explanation

    we are considering that the desired signal(d) is corr-

    upted and our aim is to eliminate the undesired part to

    get signal.For this purpose we are using adaptive

    filter.It uses algorithm which provides a set of taps.The tap

    weights could then be implemented as the correspond-

    ing filter, inversed and applied to the received signal

    thus producing the original signal in its original

    form.

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    Adaptive Filtering System

    Configurations

    There are four major types of adaptive filtering

    configurations.

    Adaptive system identification. Adaptive echo/noise cancellation.

    Inverse system identification.

    Linear predictive coding.

    Note:

    All of the above systems are similar in the implementation ofthe algorithm, but different in system Configuration.

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    ACOUSTIC ECHO

    Acoustic echo occurs when an audio signal is

    reverberated in a real environment, resulting in

    the original intended signal plus attenuated, time

    delayed images of this signal.

    When a signal is received by the system, it is

    output through the loudspeaker into an acoustic

    environment. This signal is reverberated within

    the environment and returned to the system via

    the microphone input.

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    Cont..

    These reverberated signals contain time

    delayed images of the original signal, which are

    then returned to the original sender. The

    occurrence of acoustic echo in speechtransmission causes signal interference and

    reduced quality of communication.

    The method used to cancel the echo signal is

    known as adaptive filtering.

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    SOURCES OF ACOUSTIC

    ECHO

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    ACOUSTIC ECHO

    CANCELLER

    The echo canceller mimics the transfer function of theecho path (or room acoustic) to synthesize a replica ofthe echo, and then subtracts that replica from the

    combined echo and near-end speech (or disturbance)signal to obtain the near-end signal alone.

    However, the transfer function is unknown in practice,and so it must be identified. The solution to thisproblem is to use an adaptive filter

    An adaptive filter algorithmically alters its parametersin order to minimize a function of the differencebetween the desired output d(n) and its actual outputy(n). This function is known as the cost function of theadaptive algorithm.

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    ACOUSTIC ECHO

    CANCELLER

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    CONT

    Here the filter H(n) represents the impulse response of

    the acoustic environment, W(n) represents the adaptive

    filter used to cancel the echo signal. The adaptive filteraims to equate its output y(n) to the desired output d(n)

    (the signal reverberated within the acoustic

    environment). At each iteration the error signal,

    e(n)=d(n)-y(n), is fed back into the filter, where thefilter characteristics are altered accordingly.

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    Conventional block diagram

    microphone ejnj

    uj

    dj pn

    n

    xjloudspeaker

    Adaptive

    filter

    Step size

    control

    +

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    `block diagram

    Main

    ADF

    StepSize

    Control

    +

    + +

    Sub adf

    microphone

    Loud speaker

    nj

    pn

    n

    dj

    UbjUcj

    xj

    ej

    j

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    Contd..

    In addition to the main adaptive filter in our project

    we are using the sub adaptive filter to cancel the

    echo.

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    LMS ALGORITHM

    The LMS algorithm is a linear adaptive filter algorithm,which in general consists of two basic processes.

    1. A filter process, which involves

    a.Computing the output of a linear filter inresponse to an input signal.

    b.Generating an estimation error bycomparing this output with a desiredresponse.

    2. An adaptive process, which involves theautomatic adjustment of the parameters of thefilter in accordance with the estimation error.

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    IMPLEMENTATION OF LMS

    ALGORITHM

    Each iteration of the LMS algorithm requires 3 distinct

    steps in this order:

    1. The output of the FIR filter, y(n) is calculated

    using equation

    2. The value of the error estimation is calculated

    using equation

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    CONT..

    3. The tap weights of the FIR vector are updated in

    preparation for the next iteration, by

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    REAL TIME

    IMPLEMENTATION

    Real Time Processing:

    Real-time processing means that the processingmust keep pace with some external event .

    Real Time Implementation involves:

    A/D converter is used to take the input Thisinput signal is processed by the processor and

    the output is send through D/A.To read and write data multi channel buffer

    serial port is used

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    Thank you