• 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