Masato Miyoshi

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Inverse filtering of room transfer functions (RTFs) is considered an attractive approach for speech dereverberation given that the time invariance assumption of the used RTFs holds. However, in a realistic environment, this assumption is not necessarily guaranteed, and the performance is degraded because the RTFs fluctuate over time and the inverse filter(More)
This paper proposes a statistical model-based speech dereverberation approach that can cancel the late reverberation of a reverberant speech signal captured by distant microphones without prior knowledge of the room impulse responses. With this approach, the generative model of the captured signal is composed of a source process, which is assumed to be a(More)
A speech signal captured by a distant microphone is generally smeared by reverberation, which severely degrades automatic speech recognition (ASR) performance. One way to solve this problem is to dereverberate the observed signal prior to ASR. In this paper, a room impulse response is assumed to consist of three parts: a direct-path response, early(More)
In this paper, we discuss the numerical problems posed by the previously reported LInear-predictive Multi-input Equalization (LIME) algorithm when dealing with dereverberation of long room transfer functions (RTF). The LIME algorithm consists of two steps. First, a speech residual is calculated using multichannel linear prediction. The residual is free from(More)
A speech signal captured by a distant microphone is generally smeared by reverberation, which severely degrades automatic speech recognition (ASR) performance. In this paper, we propose a novel dereverberation method utilizing multi-step forward linear prediction. It precisely estimates and suppresses the late reflections, which constitute a major cause of(More)
This paper proposes a method for performing blind source separation (BSS) and blind dereverberation (BD) at the same time for speech mixtures. In most previous studies, BSS and BD have been investigated separately. The separation performance of conventional BSS methods deteriorates as the reverberation time increases while many existing BD methods rely on(More)
It has recently been shown that the use of the time-varying nature of speech signals allows us to achieve high quality speech dereverberation based on multi-channel linear prediction (MCLP). However, this approach requires a huge computing cost for calculating large covariance matrices in the time domain. In addition, we face the important problem of how to(More)
This paper presents a new method for dereverberation of speech signals with a single microphone. For applications such as speech recognition, reverberant speech causes serious problems when a distant microphone is used in recording. This is especially severe when the reverberation time exceeds 0.5 of a second. We propose a method which uses the fundamental(More)
This paper proposes a method for adaptive speech dereverberation and speaker-position change detection, which have not previously been addressed. Signal transmission channels in rooms are modeled as auto-regressive systems in individual frequency bands. The proposed method adaptively estimates the regression coefficients of this model, which are called room(More)