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ADAPTIVE DEREVERBERATION OF SPEECH SIGNAL USING MULTI CHANNEL

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Praveen P.B, Dr. Aravind H.S
» doi: 10.48047/ecb/2023.12.si4.1320

Abstract

The quality of the speech signal degrades mainly due to additive noise and/or reverberation. This in-turn degrades the performance of many speech-based applications like automatic speech recognition, telecommunications, speech recordings, etc. This paper deals with reducing the reverberation effect without prior knowledge of room impulse response (RIR) using multiple channels. In most cases, it is unknown, therefore, much research is done on blind dereverberation [2][3]. This paper proposes an adaptive method to perform de-reverberation of an unknown RIR using 2 or more channels. This method first estimates the impulse response of the echo path for each channel using the method described in [1], using it finds the transmission delay at 2nd mic, 3rd mic and so on with respect to 1st mic and finally cancels the echo signal. This method doesn’t extract the clean signal, but the direct signal as explained in [1]. Performance and convergence of the estimated weights/coefficients of an adaptive filter purely depends on the type of adaptive filter used to estimate filter co-efficient, numbers of echo path and number of channels. This method is implemented and tested using the MATLAB tool

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