DocumentCode :
3048898
Title :
Speech enhancement using microphone array neural switched Griffiths-Jim beamformer
Author :
Yoganathan, V. ; Moir, T.J.
Author_Institution :
Sch. of Eng. & Adv. Technol., Massey Univ., Auckland, New Zealand
fYear :
2010
fDate :
21-23 Oct. 2010
Firstpage :
1
Lastpage :
5
Abstract :
There is a great need for speech enhancement in today´s world due to the increasing demand for speech based applications. These applications vary from hearing-aids, hands-free telephony to speech controlled devices. The main goal is to minimize the interference from an acquired speech signal. The interference we considered here could be from any noise source such as competing speaker, radio, TV and so on. This paper proposes a solution to improve the current design of the switched Griffiths-Jim beamformer structure. It introduces an adaptive nonlinear neural network algorithm for the noise reduction section. The network topology used here is a partially connected three-layer feedforward neural network structure. The error backpropagation algorithm is used here as the learning algorithm. Comparison analysis of the traditional four channel linear beamformer and the proposed four-channel neural switched Griffiths-Jim beamformer structure is discussed here. They are both tested with different types of interference signal from the Noise-X database. All the experiments are conducted in real-world surrounding. The nonlinear approach introduced here shows remarkable improvement over the previous linear adaptive beamformer approach.
Keywords :
array signal processing; backpropagation; feedforward neural nets; microphone arrays; speech enhancement; Noise-X database; adaptive nonlinear neural network algorithm; channel linear beamformer; error backpropagation algorithm; feedforward neural network structure; interference; learning algorithm; linear adaptive beamformer; microphone array neural switched Griffiths-Jim beamformer; network topology; noise reduction section; noise source; speech based application; speech enhancement; speech signal; Adaptive filters; Artificial neural networks; Microphones; Noise cancellation; Signal processing algorithms; Speech; Generalised sidelobe canceller; Nonlinear adaptive filter; Time delay neural network; multi-layer perceptron; noise reduction;
fLanguage :
English
Publisher :
ieee
Conference_Titel :
Wireless Communications and Signal Processing (WCSP), 2010 International Conference on
Conference_Location :
Suzhou
Print_ISBN :
978-1-4244-7556-8
Electronic_ISBN :
978-1-4244-7554-4
Type :
conf
DOI :
10.1109/WCSP.2010.5633568
Filename :
5633568
Link To Document :
بازگشت