AMENDED ADAPTIVE ALGORITHM FOR CORPUS BASED IMPROVED SPEECH ENHANCEMENT

SELVA NIDHYANANTHAN, Shanmuga Priya, Shantha Selva Kumari

Abstract


Speech enhancement objective is to improve the noisy speech signals for human perception. The intention of speech enhancement algorithm is to improve the performance of the communication, when the signal is occluded by noise. The quality and intelligibility of speech is reduced because of the presence of background noise. There are various adaptive filtering algorithms for speech enhancement. The existing least mean square and normalised least mean square algorithms have the problem of choosing the step size that guarantees the stability of the algorithm. To overcome this problem, we focus on speech enhancement by amended adaptive filtering. The proposed algorithm follows blind source separation strategy using adaptive filtering. Comparison of existing adaptive filtering algorithms with proposed algorithm justifies the amendment incorporated in this paper. Taking the objective criteria into account the algorithms has been tested for segmental signal to noise ratio (SegSNR), segmental mean square error (SegMSE), signal to noise ratio and mean square error. The proposed algorithm can be used for hand-free cell phone, hearing aids and teleconferencing systems.


Full Text:

PDF

References


Loizou PC. Speech enhancement theory and practice.1st edition. CRC Press; 2013.

Cohen I, Gannot S. Spectral enhancement methods, Springer handbook of speech processing.1st edition. Springer; 2008.

Djendi M, Gilloire A, Scalart P. Noise cancellation using two closely spaced microphones experimental study with a specific model and two adaptive algorithms. In IEEE International conference. ICASSP, 2006 May; vol. 3; pp. 744–8.

Steven F. Boll. Suppression of Acoustic Noise in Speech using Spectral Subtraction. IEEE Transaction on Accoustics, Speech and Signal Processing 1979 April; vol. 27; pp.873-902.

Chin Tuan Tan et.al. A parametric Formulation of the Generalised Spectral Subtraction Method. IEEE Transaction on Speech and Audio processing, 1998 July; vol. 6; pp. 328-337.

Suhadi, Carsten Last, and Tim Fingscheidt. A Data-Driven Approach to A priori SNR estimation. IEEE Transactions on Audio, Speech, and Language Process 2011 January; vol. 19; pp. 186-195.

Djendi M, Gilloire A, Scalart P. New frequency domain post-filters for noise cancellation using two closely spaced microphones. In Proc

EUSIPCO, Poznan, 2007 Sep; vol. 1; pp. 218–21.

Djendi M, Henni R, Sayoud A. A new dual forward BSS based RLS algorithm for speech enhancement. In International conference on

engineering and MIS, ICEMIS , 2016. May; vol.5; pp.302-309.

Henni Rahima, Djendi Mohamed, Mustapha Djebari. A Dual Backward Adaptive Algorithm for Speech Enhancement and Acoustic Noise Reduction. Association for Computing Machinery, 2017 June; vol. 978; pp.4503- 6392.

Akila Sayoud, Mohamed Djendi, Soumia Medahi, Abderrezak Guessoum. A dual fast NLMS adaptive filtering algorithm for

blind speech quality Enhancement. Applied Acoustics, 2018 June; vol.135, pp.101–110.

Jyoti Dhiman, Shadab Ahmad, Kuldeep Gulia. Comparison between Adaptive filter Algorithms. International Journal of Science, Engineering and Technology 2013 May; Research vol. 2; pp.503-515.

Djendi M, Gilloire A, Scalart P. Noise cancellation using two closely space Microphones experimental study with a specific model and two adaptive algorithms. In IEEE International Conference. ICASSP, 2006 May; vol. 3; pp. 744–8.


Refbacks

  • There are currently no refbacks.


International Journal of Electronics and Telecommunications
is a periodical of Electronics and Telecommunications Committee
of Polish Academy of Sciences

eISSN: 2300-1933