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Dive into the research topics where Kazunobu Kondo is active.

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Featured researches published by Kazunobu Kondo.


IEEE Transactions on Audio, Speech, and Language Processing | 2011

Theoretical Analysis of Musical Noise in Generalized Spectral Subtraction Based on Higher Order Statistics

Takayuki Inoue; Hiroshi Saruwatari; Yu Takahashi; Kiyohiro Shikano; Kazunobu Kondo

In this paper, we provide a new theoretical analysis of the amount of musical noise generated via generalized spectral subtraction based on higher order statistics. Power spectral subtraction is the most commonly used spectral subtraction method, and in our previous study a musical noise assessment theory limited to the power spectral domain was proposed. In this paper, we propose a generalization of our previous theory on spectral subtraction for arbitrary exponent parameters. We can thus compare the amount of musical noise between any exponent domains from the results of our analysis. We also clarify that less musical noise is generated when we choose a lower exponent spectral domain; this implies that there is no theoretical justification for using power/amplitude spectral subtraction.


IEEE Transactions on Audio, Speech, and Language Processing | 2012

Musical-Noise-Free Speech Enhancement Based on Optimized Iterative Spectral Subtraction

Ryoichi Miyazaki; Hiroshi Saruwatari; Takayuki Inoue; Yu Takahashi; Kiyohiro Shikano; Kazunobu Kondo

In this paper, we provide a theoretical analysis of the amount of musical noise in iterative spectral subtraction, and its optimization method for the least musical noise generation. To achieve high-quality noise reduction with low musical noise, iterative spectral subtraction, i.e., iteratively applied weak nonlinear signal processing, has been proposed. Although the effectiveness of the method has been reported experimentally, there have been no theoretical studies. Therefore, in this paper, we formulate the generation process of musical noise by tracing the change in kurtosis of noise spectra, and conduct a comparison of the amount of musical noise for different parameter settings but the same achieved level of noise attenuation. Furthermore, we theoretically derive the optimal internal parameters that generate no musical noise. It is clarified that to find a fixed point in kurtosis yields the no-musical-noise property. Comparative experiments with commonly used noise reduction methods show the proposed methods efficacy.


international conference on acoustics, speech, and signal processing | 2009

Musical noise generation analysis for noise reduction methods based on spectral subtraction and MMSE STSA estimation

Yoshihisa Uemura; Yu Takahashi; Hiroshi Saruwatari; Kiyohiro Shikano; Kazunobu Kondo

In this paper, we reveal new findings about the generated musical noise in minimum mean-square error short-time spectral amplitude (MMSE STSA) processing. Recently we have proposed a objective metric of musical noise based on kurtosis change ratio on spectral subtraction (SS). Also we found an interesting relationship among the degree of generated musical noise, the shapes of signal-s probability density function, the strength parameter of SS processing. This paper is aimed to automatically evaluate the sound quality of various types of noise reduction methods using kurtosis change ratio. We give a mathematical analysis based on higher-order statistics viewpoint, and lead to a valuable relation in that MMSE STSA has a weakness in speech period distortion rather than noise period, and vice versa in SS.


EURASIP Journal on Advances in Signal Processing | 2010

Musical-noise analysis in methods of integrating microphone array and spectral subtraction based on higher-order statistics

Yu Takahashi; Hiroshi Saruwatari; Kiyohiro Shikano; Kazunobu Kondo

We conduct an objective analysis on musical noise generated by two methods of integrating microphone array signal processing and spectral subtraction. To obtain better noise reduction, methods of integrating microphone array signal processing and nonlinear signal processing have been researched. However, nonlinear signal processing often generates musical noise. Since such musical noise causes discomfort to users, it is desirable that musical noise is mitigated. Moreover, it has been recently reported that higher-order statistics are strongly related to the amount of musical noise generated. This implies that it is possible to optimize the integration method from the viewpoint of not only noise reduction performance but also the amount of musical noise generated. Thus, we analyze the simplest methods of integration, that is, the delay-and-sum beamformer and spectral subtraction, and fully clarify the features of musical noise generated by each method. As a result, it is clarified that a specific structure of integration is preferable from the viewpoint of the amount of generated musical noise. The validity of the analysis is shown via a computer simulation and a subjective evaluation.


IEEE Transactions on Audio, Speech, and Language Processing | 2011

Musical Noise Controllable Algorithm of Channelwise Spectral Subtraction and Adaptive Beamforming Based on Higher Order Statistics

Hiroshi Saruwatari; Yohei Ishikawa; Yu Takahashi; Takayuki Inoue; Kiyohiro Shikano; Kazunobu Kondo

In this paper, we propose a musical-noise-controllable algorithm for array signal processing with the aim for high-performance and high-quality noise reduction. Recently, many methods of integrating linear microphone array signal processing and nonlinear signal processing for noise reduction have been studied, but these methods often suffer from the problem of musical noise. In the proposed algorithm, channelwise spectral subtraction is applied before adaptive array signal processing. We also introduce a new automatic control algorithm to obtain the subtraction strength parameter used in the spectral subtraction, which depends on the amount of generated musical noise, measured by higher order statistics. We confirm the effectiveness of the proposed algorithm via objective and subjective evaluations.


international workshop on machine learning for signal processing | 2010

Theoretical analysis of iterative weak spectral subtraction via higher-order statistics

Takayuki Inoue; Hiroshi Saruwatari; Yu Takahashi; Kiyorhiro Shikano; Kazunobu Kondo

In this paper, we provide a new theoretical analysis of the amount of musical noise generated via iterative spectral subtraction based on higher-order statistics. To achieve high-quality noise reduction with low musical noise, the iterative spectral subtraction method, i.e., recursively applied weak nonlinear signal processing, has been proposed. Although the effectiveness of the method has been reported experimentally, there have been no theoretical studies. Therefore, in this paper, we formulate the generation process of musical noise by tracing the change in kurtosis, and conduct a comparison of the amount of musical noise for different parameter settings under the same noise reduction performance. It is clarified from mathematical analysis and evaluation experiments that iterative spectral subtraction with very weak processing can result in the generation of less musical noise.


IEEE Transactions on Audio, Speech, and Language Processing | 2015

Multichannel signal separation combining directional clustering and nonnegative matrix factorization with spectrogram restoration

Daichi Kitamura; Hiroshi Saruwatari; Hirokazu Kameoka; Yu Takahashi; Kazunobu Kondo; Satoshi Nakamura

In this paper, to address problems in multichannel music signal separation, we propose a new hybrid method that combines directional clustering and advanced nonnegative matrix factorization (NMF). The aims of multichannel music signal separation technology is to extract a specific target signal from observed multichannel signals that contain multiple instrumental sounds. In previous studies, various methods using NMF have been proposed, but many problems remain including poor separation accuracy and lack of robustness. To solve these problems, we propose a new supervised NMF (SNMF) with spectrogram restoration and a hybrid method that concatenates the proposed SNMF after directional clustering. Via the extrapolation of supervised spectral bases, the proposed SNMF attempts both target signal separation and reconstruction of the lost target components, which are generated by preceding directional clustering. In addition, we experimentally reveal the trade-off between separation and extrapolation abilities and propose a new scheme for adaptive divergence, where the optimal divergence can be automatically changed in each time frame according to the local spatial conditions. The results of an evaluation experiment show that our proposed hybrid method outperforms the conventional music signal separation methods.


international conference on acoustics, speech, and signal processing | 2009

Musical noise analysis based on higher order statistics for microphone array and nonlinear signal processing

Yu Takahashi; Yoshihisa Uemura; Hiroshi Saruwatari; Kiyorhiro Shikano; Kazunobu Kondo

In this paper, we conduct an analysis for reduction of musical noise in integration method of microphone array signal processing and nonlinear signal processing. In these days, for better noise reduction, integration methods of microphone array signal processing and nonlinear signal processing have been researched. However, non-linear signal processing causes musical noise. Since such musical noise make users uncomfortable, it is desired that musical noise is mitigated. Moreover, in these days, it is reported that higher-order statistics is strongly related with the amount of generated musical noise. Thus, we analyze the integrated method of microphone array signal processing and nonlinear signal processing, based on higher-order statistics. Also, we propose an architecture for reducing musical noise based on the analysis. The effectiveness of the proposed architecture and analysis correctness are shown via a computer simulation and a subjective evaluation.


international conference on digital signal processing | 2013

Music signal separation by supervised nonnegative matrix factorization with basis deformation

Daichi Kitamura; Hiroshi Saruwatari; Kiyohiro Shikano; Kazunobu Kondo; Yu Takahashi

In this paper, we address a music signal separation problem, and propose a new supervised algorithm for real instrumental signal separation employing a deformable capability for a spectral supervision trained in advance. Nonnegative matrix factorization (NMF) is one of the techniques used for the separation of an audio mixture that consists of multiple instrumental sources. Conventional supervised NMF has the critical problem that a mismatch between the bases trained in advance and the target real sound reduces the accuracy of separation. To solve this problem, we propose a new advanced supervised NMF that employs a deformable capability for the trained bases and penalty terms for making the bases fit into the target sound. The results of the experiment using real instruments show that the proposed method significantly improves the accuracy of separation compared with the conventional method.


Signal Processing | 2014

Musical-noise-free blind speech extraction integrating microphone array and iterative spectral subtraction

Ryoichi Miyazaki; Hiroshi Saruwatari; Satoshi Nakamura; Kiyohiro Shikano; Kazunobu Kondo; Jonathan Blanchette; Martin Bouchard

Abstract In this paper, we propose a musical-noise-free blind speech extraction method using a microphone array for application to nonstationary noise. In our previous study, it was found that optimized iterative spectral subtraction (SS) results in speech enhancement with almost no musical noise generation, but this method is valid only for stationary noise. The proposed method consists of iterative blind dynamic noise estimation by, e.g., independent component analysis (ICA) or multichannel Wiener filtering, and musical-noise-free speech extraction by modified iterative SS, where multiple iterative SS is applied to each channel while maintaining the multichannel property reused for the dynamic noise estimators. Also, in relation to the proposed method, we discuss the justification of applying ICA to signals nonlinearly distorted by SS. From objective and subjective evaluations simulating a real-world hands-free speech communication system, we reveal that the proposed method outperforms the conventional methods.

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Kiyohiro Shikano

Nara Institute of Science and Technology

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Daichi Kitamura

Graduate University for Advanced Studies

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Ryoichi Miyazaki

Nara Institute of Science and Technology

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Satoshi Nakamura

Nara Institute of Science and Technology

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Takayuki Inoue

Nara Institute of Science and Technology

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