An efficient pre-processing scheme to improve the sound source localization system in noisy environment

Sheng Chieh Lee, K. Bharanitharan, Bo Wei Chen, Jhing Fa Wang, Chung Hsien Wu, Min Jian Liao

Research output: Contribution to journalConference article

1 Citation (Scopus)

Abstract

In this study, we introduce an efficient pre-processing scheme for direction of arrival (DOA) estimation, which is capable of reducing the noise and reverberation effects in speech sound source localization. Furthermore, this presented system is also suitable for far-field speech localization. The adopted method of this proposed system can be simply subdivided into three stages: Linear phase-difference approximation, covariance matrix reconstruction, and frequency bin selection. The first two stages can initially decrease the influences of noise and reverberation; the last stage is used to filter the noise frequency bands according to the eigenvalue decomposition (EVD) of the covariance matrix. The experimental results show that our proposed system has effective performance of detecting different directions of speeches. For different signalto- noise ratios (SNRs) speech signals, the average estimation errors can be decreased by about 5 to 7.5 degrees.

Original languageEnglish
Pages (from-to)2493-2496
Number of pages4
JournalProceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH
Publication statusPublished - 2011 Dec 1
Event12th Annual Conference of the International Speech Communication Association, INTERSPEECH 2011 - Florence, Italy
Duration: 2011 Aug 272011 Aug 31

Fingerprint

Source Localization
Preprocessing
Acoustic waves
Covariance matrix
Reverberation
Processing
Acoustic noise
Eigenvalue Decomposition
Difference Approximation
Direction of Arrival
Speech Signal
Phase Difference
Estimation Error
Far Field
Direction of arrival
Bins
Error analysis
Filter
Frequency bands
Decrease

All Science Journal Classification (ASJC) codes

  • Language and Linguistics
  • Human-Computer Interaction
  • Signal Processing
  • Software
  • Modelling and Simulation

Cite this

@article{3864092052354558af76ef2399f2603f,
title = "An efficient pre-processing scheme to improve the sound source localization system in noisy environment",
abstract = "In this study, we introduce an efficient pre-processing scheme for direction of arrival (DOA) estimation, which is capable of reducing the noise and reverberation effects in speech sound source localization. Furthermore, this presented system is also suitable for far-field speech localization. The adopted method of this proposed system can be simply subdivided into three stages: Linear phase-difference approximation, covariance matrix reconstruction, and frequency bin selection. The first two stages can initially decrease the influences of noise and reverberation; the last stage is used to filter the noise frequency bands according to the eigenvalue decomposition (EVD) of the covariance matrix. The experimental results show that our proposed system has effective performance of detecting different directions of speeches. For different signalto- noise ratios (SNRs) speech signals, the average estimation errors can be decreased by about 5 to 7.5 degrees.",
author = "Lee, {Sheng Chieh} and K. Bharanitharan and Chen, {Bo Wei} and Wang, {Jhing Fa} and Wu, {Chung Hsien} and Liao, {Min Jian}",
year = "2011",
month = "12",
day = "1",
language = "English",
pages = "2493--2496",
journal = "Proceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH",
issn = "2308-457X",

}

An efficient pre-processing scheme to improve the sound source localization system in noisy environment. / Lee, Sheng Chieh; Bharanitharan, K.; Chen, Bo Wei; Wang, Jhing Fa; Wu, Chung Hsien; Liao, Min Jian.

In: Proceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH, 01.12.2011, p. 2493-2496.

Research output: Contribution to journalConference article

TY - JOUR

T1 - An efficient pre-processing scheme to improve the sound source localization system in noisy environment

AU - Lee, Sheng Chieh

AU - Bharanitharan, K.

AU - Chen, Bo Wei

AU - Wang, Jhing Fa

AU - Wu, Chung Hsien

AU - Liao, Min Jian

PY - 2011/12/1

Y1 - 2011/12/1

N2 - In this study, we introduce an efficient pre-processing scheme for direction of arrival (DOA) estimation, which is capable of reducing the noise and reverberation effects in speech sound source localization. Furthermore, this presented system is also suitable for far-field speech localization. The adopted method of this proposed system can be simply subdivided into three stages: Linear phase-difference approximation, covariance matrix reconstruction, and frequency bin selection. The first two stages can initially decrease the influences of noise and reverberation; the last stage is used to filter the noise frequency bands according to the eigenvalue decomposition (EVD) of the covariance matrix. The experimental results show that our proposed system has effective performance of detecting different directions of speeches. For different signalto- noise ratios (SNRs) speech signals, the average estimation errors can be decreased by about 5 to 7.5 degrees.

AB - In this study, we introduce an efficient pre-processing scheme for direction of arrival (DOA) estimation, which is capable of reducing the noise and reverberation effects in speech sound source localization. Furthermore, this presented system is also suitable for far-field speech localization. The adopted method of this proposed system can be simply subdivided into three stages: Linear phase-difference approximation, covariance matrix reconstruction, and frequency bin selection. The first two stages can initially decrease the influences of noise and reverberation; the last stage is used to filter the noise frequency bands according to the eigenvalue decomposition (EVD) of the covariance matrix. The experimental results show that our proposed system has effective performance of detecting different directions of speeches. For different signalto- noise ratios (SNRs) speech signals, the average estimation errors can be decreased by about 5 to 7.5 degrees.

UR - http://www.scopus.com/inward/record.url?scp=84865703432&partnerID=8YFLogxK

UR - http://www.scopus.com/inward/citedby.url?scp=84865703432&partnerID=8YFLogxK

M3 - Conference article

AN - SCOPUS:84865703432

SP - 2493

EP - 2496

JO - Proceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH

JF - Proceedings of the Annual Conference of the International Speech Communication Association, INTERSPEECH

SN - 2308-457X

ER -