研究者詳細 - 水町　光徳

論文 - 水町　光徳

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Restoration of intermittent speech based on human auditory capability and temporal characteristics of speech 査読有り

Mizumachi M., Motomura S., Takakura T., Horiuchi T.

20th International Congress on Sound and Vibration 2013, ICSV 2013 3 1896 - 1902 2013年01月

担当区分：筆頭著者記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Recently, a new obstruction to speech communication, that is, the short-term intermittence of the speech signal, is occurred due to the congestion in the digital communication network. Mobile phones and smart phones have spread rapidly, and then the communication lines are occupied by the heavy traffic of the received/transmitted packets. Therefore, the packet associated with speech can also be lost in the congested communication line. It is not advisable to require the re-transmission of the lost packet in speech communication, because the re-transmission of the lost packet disturbs smooth speech communication and causes the further traffic in the communication line. The restoration of the intermittent speech should be completed on the user's client equipment. The copy of the previous packet should be the easiest approach in restoring the intermittent speech, although the restored speech must be perceived slowly and lazily. The authors have proposed a novel approach to the restoration of the intermittent speech relying on the human auditory capability [Proc. ICSV19, 2012]. This is not an exact physical restoration of the intermittent speech signal, but a perceptual restoration. When the gap of the intermittent speech signal is filled up with an extremely noisy noise signal, we can listen to the intermittent speech smoothly even if some segments of the original speech signal are lost completely. The authors have proposed a practical scheme of the restoration based on our phonemic restoration capabilities. The previously proposed method has employed the less harsh noise and the speech-like harmonic component designed in the frequency domain as the gap-filling signal. Furthermore, in this paper, the temporal characteristics of speech are introduced into the gap-filling signal. It is confirmed that the amplitude modulation and the temporal decay in the gap-filling signal improve the auditory impression of the restored speech signals.

Scopus

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ハイレゾリューションオーディオの音質評価査読有り

山本竜太, 金只直人, 水町光徳

産業応用工学会論文誌 ( 一般社団法人産業応用工学会 ) 1 ( 2 ) 52 - 57 2013年01月

記述言語：日本語掲載種別：研究論文（学術雑誌）

In this paper, sound quality of high resolution audio (HRA) is investigated in the view point of auditory perception. Perceptual characteristics of HRA have been examined by listening tests compared with the standard audio CD and the compressed MPEG audio layer-3 (MP3) qualities. The listening tests were carried out by the paired comparison, and the participants were asked to discriminate among HRA, CD, and MP3. Jazz music was selected out as the original HRA. Both CD and MP3 quality audio files were prepared from the original HRA file by the authors. It is found that the participants prefer the audio format with richer information, that is, HRA is most desirable and CD is better than MP3. The questionnaires filled in by participants indicate that subjective characteristics of HRA include much presence, high spatial clarity, and wide spaciousness. Additionally, perceptual discrimination in quantization distortion is also focused on in between 16 bits and 24 bits resolution. As the result of the listening test with 28 participants, the difference of the quantization resolution was discriminated with the accuracy of 60.3 %. Concerning the perceptual discrimination of HRA, quantization resolution is an important factor as well as its wide frequency range.

DOI： 10.12792/jjiiae.001.02.003

Kyutacar

CiNii Article

その他リンク： https://ci.nii.ac.jp/naid/130005263583

Relationship between subjective and objective evaluation of noisereduced speech based on temporal variation of speech distortion 査読有り

Mizumachi M.

41st International Congress and Exposition on Noise Control Engineering 2012, INTER-NOISE 2012 1 109 - 117 2012年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

It is indispensable for designing a noise reduction algorithm to evaluate the quality of the noise-reduced speech signals accurately and efficiently. Subjective evaluation is an ideal approach, but requires a time-consuming listening test with a number of subjects. Then, objective distortion measures are often employed for efficient evaluation. However, almost all the distortion measures do not consider the temporal variation of the speech distortion. In this paper, the temporal aspect of the short-term speech distortion is investigated based on higher-order statistics such as variance, skewness, and kurtosis for short-term segmental SNR distribution. It is interesting that the skewness of the segmental SNR gives a good explanation for the discrepancy between subjective and objective evaluation.

Scopus

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Relationship between subjective and objective evaluation of noise-reduced speech based on temporal fluctuation of speech distortion 査読有り

M. Mizumachi

Proc. Inter-noise CD-ROM (Paper ID: 808) 2012年08月

Relationship between subjective and objective evaluation of distorted speech 査読有り

Mizumachi M.

131st Audio Engineering Society Convention 2011 1 351 - 356 2011年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

It is important for designing a noise reduction algorithm to evaluate the quality of the noise-reduced speech signals accurately and efficiently. Subjective evaluation is ideal, but requires a time-consuming listening test with a lot of subjects. Then, objective distortion measures are often used for efficient evaluation. However, almost all the distortion measures do not consider the temporal variation of the speech distortion. In this paper, the temporal aspect of the short-term speech distortion is investigated based on higher-order statistics, that is, variance, skewness, and kurtosis. It is interesting that the skewness of the short-term SNR gives a good explanation for the discrepancy between subjective and objective evaluation.

Scopus

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Versatile beamforming for speech enhancement in non-stationary acoustic environments 査読有り

Mizumachi M.

Proceedings of Forum Acusticum 311 - 316 2011年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is indispensable for designing high-quality speech interfaces. Various kinds of beamformers have been proposed for speech enhancement by spatial filtering. Delay-and-sum beamforming is usually expensive for speech enhancement with numbers of microphones, but provides less-distorted speech signals due to linear signal processing. Null-steering subtractive beamforming achieves efficient speech enhancement, but resultant speech signals include artificial distortion by non-linear signal processing. Each beamformer has both an advantage and a drawback. Directionof- arrival (DOA) is the most important spatial feature for beamforming. The proposed versatile beamforming aims at integrating delay-and-sum beamforming with null-steering beamforming adaptively under time-varying acoustic environments. Confidence for each DOA estimate can be estimated based on a spatial feature, e.g., a cross-correlation function for stereo observations, and is the key for achieving the adaptive integration. Null-steering beamforming can be employed with a reliable DOA estimate, and delay-and-sum beamforming should be used with an unreliable DOA estimate. The DOA confidence varies time by time under non-stationary acoustic environments. The propose versatile beamformer chooses between delay-and-sum and null-steering subtractive beamformers relying on the confidence of each DOA confidence in non-stationary acoustic environments.

Scopus

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Speech enhancement with adaptive integration between spatial and spectral filtering 査読有り

Mizumachi M.

40th International Congress and Exposition on Noise Control Engineering 2011, INTER-NOISE 2011 2 1432 - 1437 2011年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is indispensable for robust speech processing in noisy environments. It is, however, difficult to achieve effective speech enhancement under practical, non-stationary, and unknown noisy environments. Challenge to design an almighty algorithm is not realistic for speech enhancement in real environments. In other words, the right scheme should be prepared for speech enhancement in the right noisy environment. In this paper, a novel approach is introduced to achieve robust speech enhancement. Provided that two or more microphones are available, we can estimate a direction of arrival (DOA) from a spatial feature, which can be also used for spatial filtering as a means of speech enhancement. The author has already proposed to give reliability, which is also estimated from the spatial feature, to each estimate of the DOA. The proposed method achieves speech enhancement by adaptively selecting the suitable approach relying on the reliability of each DOA estimate. Beamforming and spectral subtraction are prepared as candidates for spatial and spectral filtering, respectively. Each output is merged with the time-varying mixing ratio depending on the reliability of each DOA estimate. Then, the proposed method can deal with non-stationary noisy environments by introducing the adaptive integration between spatial and spectral filtering. Feasibility of the proposed idea is confirmed under a less-reverberant, non-stationary noisy condition. Copyright © (2011) by the Institute of Noise Control Engineering.

Scopus

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Relationship between subjective and objective evaluation of distorted speech 査読有り

Mitsunori Mizumachi

Proc. AES 131st Convention 2011年10月

Speech enhancement with adaptive integration between spatial and spectral filtering 招待有り査読有り

Mitsunori Mizumachi

Proc. Inter-noise 2011年09月

Versatile beamforming for speech enhancement in non-stationary acoustic environments 査読有り

Mitsunori Mizumachi

Proc. Forum Acusticum 2011年06月

Statistical confidence measure for direction-of-arrival estimate 査読有り

Mizumachi M.

European Signal Processing Conference 666 - 670 2010年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

This paper proposes to estimate the reliability of each direction-of-arrival (DOA) estimate explicitly. Observed spatial features, which give DOA estimates, are usually distorted by acoustical noise and reverberation. Then, distorted spatial features are filtered out by particle filters with a system model, which describes sound source dynamics on spatial state space. After particle filtering is carried out, set of the weighted particles can be considered as the discrete approximation of the true spatial feature. It is reasonable that particles with large weights concentrate around the true DOA on spatial state space. In this paper, crest factor, skewness, and kurtosis are calculated based on the second, third, and forth order moments from the particle set for representing how particles concentrate simply. It is confirmed that the skewness confidence measure is the most suitable for estimating DOA reliability under both directional and distributed noise conditions. © EURASIP, 2010.

Scopus

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Speech enhancement based on spatial and acoustical features in non-stationary noisy environments 査読有り

Mizumachi M.

17th International Congress on Sound and Vibration 2010, ICSV 2010 5 3603 - 3608 2010年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is essential for many audio applications with speech input. Those devices are supposed to be used under a wide variety of acoustic conditions. Then, speech enhancement should be effective even in unknown, time-varying, and non-stationary noisy environments. The author has proposed an efficient noise reduction method, which aims at making the most use of stochastic spatial information [Mizumachi, Proc. ICSV16, 2009]. The spatial and acoustical features have been prepared based on directions of sound sources and spectral characteristics of a target and acoustical interferences, respectively. In this paper, those stochastic features are extended including much richer spatial information. New spatial features are derived from higher order statistics of the spatial information extracted from the 2-ch observations. The proposed method can deal with non-stationary noisy environments, because it takes the suitable scheme in noise reduction relying on the spatial and acoustical features flexibly. In this paper, the feasibility of the proposed method is investigated under non-stationary noisy environments.

Scopus

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Self-organized approach to noise reduction based on acoustic and spatial features 査読有り

Mizumachi M.

39th International Congress on Noise Control Engineering 2010, INTER-NOISE 2010 7 5645 - 5652 2010年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Noise reduction has been one of important issues in acoustic signal processing. Until now, various kinds of algorithms have been proposed to achieve noise reduction, but we have not established an almighty algorithm, which always works well in the real world. Each algorithm has both advantage and disadvantage in noise reduction. In this paper, a practical approach to noise reduction is proposed by introducing an idea of self-organization for determining the suitable algorithm in time-varying acoustic scenes based on acoustic and spatial features. Time evolution of acoustic and spatial features and reliability of direction-of-arrival estimates are used for selecting the suitable approach out of spectral subtraction with acoustic features and beamforming with spatial features. Feasibility of the self-organized approach has been confirmed under non-stationary noise environments.

Scopus

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Confidence measure based on higher-order statistics for spatial feature in particle filtering 査読有り

Mizumachi M.

2010 World Automation Congress, WAC 2010 2010年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Scopus

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Adaptive source separation based on reliability of spatial feature using multi-channel acoustic observations 査読有り

Mizumachi M.

128th Audio Engineering Society Convention 2010 2 1092 - 1098 2010年12月

記述言語：英語掲載種別：研究論文（国際会議プロシーディングス）

Separation of sound source can be achieved by spatial filtering with multi-channel acoustic observations. However, the right algorithm should be prepared in each condition of acoustic scene. It is difficult to provide the suitable algorithm under real acoustic environments. In this paper, an adaptive source separation scheme is proposed based on the reliability of a spatial feature, which gives an estimate of direction of arrival (DOA). As confidence measures for DOA estimates, the third and fourth moments for spatial features are employed to measure how sharp the main-lobes of spatial features are. This paper proposes to selectively use either spatial filters or frequency-selective filters without spatial filtering depending on the reliability of each DOA estimate.

Scopus

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Confidence measure based on higher-order statistics for spatial feature in particle filtering 査読有り

Mitsunori Mizumachi

Proc. 2010 International Forum on Multimedia and Image Processing (IFMIP 2010) 2010年09月

Statistical confidence measure for direction-of-arrival estimate 査読有り

Mitsunori Mizumachi

Proc. 18th European Signal Processing Conference (EUSIPCO2010) 666 - 670 2010年08月

Speech enhancement based on spatial and acoustical features in non-stationary noisy environments 査読有り

水町　光徳

Proc. 17th International Congress on Sound and Vibration (ICSV17) 2010年07月

Self-organized approach to noise reduction based on acoustic and spatial features 査読有り

水町　光徳

Proc. 39th International Congress and Exposition on Noise Control Engineering (INTER-NOISE 2010) 2010年06月

Adaptive source separation based on reliability of spatial feature using multi-channel acoustic observations 査読有り

水町　光徳

Proc. AES 128th Convention in London 2010年05月

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