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Dereverberation by Using Time-Variant Nature of Speech Production System

EURASIP Journal on Advances in Signal Processing volume 2007, Article number: 065698 (2007) Cite this article

Abstract

This paper addresses the problem of blind speech dereverberation by inverse filtering of a room acoustic system. Since a speech signal can be modeled as being generated by a speech production system driven by an innovations process, a reverberant signal is the output of a composite system consisting of the speech production and room acoustic systems. Therefore, we need to extract only the part corresponding to the room acoustic system (or its inverse filter) from the composite system (or its inverse filter). The time-variant nature of the speech production system can be exploited for this purpose. In order to realize the time-variance-based inverse filter estimation, we introduce a joint estimation of the inverse filters of both the time-invariant room acoustic and the time-variant speech production systems, and present two estimation algorithms with distinct properties.

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Authors and Affiliations

  1. NTT Communication Science Laboratories, NTT Corporation 2-4, Hikaridai, Seika-cho, Soraku-gun, Kyoto, 619-0237, Japan

    Takuya Yoshioka, Takafumi Hikichi & Masato Miyoshi

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  1. Takuya Yoshioka
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  2. Takafumi Hikichi
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  3. Masato Miyoshi
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Correspondence to Takuya Yoshioka.

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Open Access This article is distributed under the terms of the Creative Commons Attribution 2.0 International License (https://doi.org/creativecommons.org/licenses/by/2.0), which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.

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Yoshioka, T., Hikichi, T. & Miyoshi, M. Dereverberation by Using Time-Variant Nature of Speech Production System. EURASIP J. Adv. Signal Process. 2007, 065698 (2007). https://doi.org/10.1155/2007/65698

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