Use of Microphone Array and Model Adaptation for Hands-Free Speech Acquisition and Recognition

Jen-Tzung Chien*, Jain Ray Lai

*Corresponding author for this work

Research output: Contribution to journalArticlepeer-review

3 Scopus citations


This paper presents a combined microphone array and model adaptation algorithm for hands-free speech recognition. Our purpose is to remove the inconvenience of using head-mounted/hand-holding microphone in conventional speech recognizer. To improve the speech quality with car noise interference, a linear microphone array is applied and acted as robust acquisition system. A time-domain coherence measure (TDCM) is applied to reliably estimate the time delay for speech signals collected by different microphones. The estimated delay is adopted in a delay-and-sum beamformer for speech enhancement. Further, we adapt the speech hidden Markov models to get close to the acoustic conditions of the enhanced test speech for robust speech recognition. In acquisition and recognition experiments using connected Chinese digits, we found that TDCM can effectively estimate the time delay. The increase in the speech sampling rate is helpful to determine the time delay. Incorporating the model adaptation scheme significantly reduces the recognition errors with moderate computation overhead.

Original languageEnglish
Pages (from-to)141-151
Number of pages11
JournalJournal of VLSI Signal Processing Systems for Signal, Image, and Video Technology
Issue number2-3
StatePublished - 1 Feb 2004


  • Coherence measure
  • Delay-and-sum beamformer
  • Microphone array
  • Model adaptation
  • Speech enhancement
  • Speech recognition

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