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Papers by Kiyohiro Shikano

Research paper thumbnail of Maximum likelihood successive state splitting algorithm for tied-mixture HMNET

Interspeech, 1997

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Research paper thumbnail of How to judge reusability of existing speech corpora for target task by utilizing statistical multidimensional scaling

Interspeech 2007

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Research paper thumbnail of Dictation Machine Based on Japanese Character Source Modeling

International Journal of Pattern Recognition and Artificial Intelligence

This paper describes a phonetic typewriter and a dictation machine that utilize the underlying st... more This paper describes a phonetic typewriter and a dictation machine that utilize the underlying statistical structure of phoneme or character sequences. The approach of using syllable or character trigrams is applied to language source modeling. The language source models are obtained by calculating trigram probabilities from a large text database. These models are combined with the HMM-LR continuous speech recognition system.3,6 The phonetic typewriter is tested using 274 phrases uttered by one male speaker. The syllable source model achieves a 94.9% phoneme recognition rate with the test-set phoneme perplexity of 3.9. Without the syllable source model, the phoneme recognition rate is only 73.2%. A trigram model based on characters is also evaluated. This character source model can reduce the syllable perplexity significantly to 7.7, compared with 10.5 of the syllable source model. The character source model achieves a 78.5% character transcription rate for the 274 phrase utterances...

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Research paper thumbnail of Sound reproduction system with adaptive compensation of temperature fluctuation effect

2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628), 2002

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Research paper thumbnail of A broad-band signal control in sound reproduction system by temperature compensation of room impulse responses

6th International Conference on Signal Processing, 2002., 2000

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Research paper thumbnail of Evaluation of a Self-Generator Method for Initial Filters of Simo-Ica Applied to Blind Separation of Binaural Sound Mixtures

Ieee Workshop on Applications of Signal Processing to Audio and Acoustics 2005, Oct 1, 2005

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Research paper thumbnail of Multiple beamforming with source localization based on CSP analysis

Systems and Computers in Japan, May 1, 2003

ABSTRACT In this paper, a method is investigated for using an array of microphones to capture a h... more ABSTRACT In this paper, a method is investigated for using an array of microphones to capture a high-quality recording of voice under reverberation. In the past, multiple beamforming has been proposed by Flanagan's group as a method for efficiently suppressing reverberation. However, because this multiple beamforming method computes the arrival positions of the sound and its reflections based on the prior knowledge of the shape of the room and the position of the target sound source, it is not suited to practical applications. In this paper, a method of multiple beamforming is proposed in which the source positions of the sound and its reflections are localized without using prior knowledge of the shape of the room or the position of the target sound source. The effectiveness of the proposed method was evaluated in a simulation experiment using the image method and in an experiment in an actual hallway environment. First, an experiment was run to localize the positions of the sources of the sound and its reflections under a condition in which both the shape of the room and the source position of the sound were unknown. As the result, it was found that the localization error for the source position of the sound was small. Furthermore, although the localization errors for the source positions of the reflections were large, the localized reflection paths were found to be approximately equal to the true reflection paths. Next, the output signal from the proposed method was evaluated based on SNR. The result demonstrated that multiple beamforming is more effective than single beamforming. © 2003 Wiley Periodicals, Inc. Syst Comp Jpn, 34(5): 69–80, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/scj.1204

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Research paper thumbnail of Problems in Blind Separation of Convolutive Speech Mixtures by Negentropy Maximization

ABSTRACT: This paper aims to examine suitability of the marginal statistics based contrast functi... more ABSTRACT: This paper aims to examine suitability of the marginal statistics based contrast function eg negentropy for the separation of convolutive speech mixtures picked up by a linear microphone array. For this study we choose our frequency domain fixed-point ICA algorithm, based on ...

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Research paper thumbnail of Fast-Convergence Algorithm for Blind Source Separation Based on Array Signal Processing

Ieice Transactions on Fundamentals of Electronics Communications and Computer Sciences, Mar 1, 2003

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Research paper thumbnail of Automatic n-gram language model creation from web resources

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Research paper thumbnail of One-to-Many and Many-to-One Voice Conversion Based on Eigenvoices

Acoustics Speech and Signal Processing 1988 Icassp 88 1988 International Conference on, Apr 15, 2007

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Research paper thumbnail of Musical-noise-free speech enhancement based on iterative Wiener filtering

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Research paper thumbnail of A Fixed-Point Ica Algorithm for Convoluted Speech Signal Separation

ABSTRACT This paper describes a fixed-point independent component analysis (ICA) algorithm in com... more ABSTRACT This paper describes a fixed-point independent component analysis (ICA) algorithm in combination with the null beamforming technique to sieve out speech signals from their convoluted mixture observed using a linear microphone array. The fixed-point algorithm shows fast convergence to the solution, however it is highly sensitive to the initial value from which iteration starts. A good initial value leads to faster convergence and yields better results. We propose the use of a null beamformer-based initial value for iteration and explore its effects on separation performance under different acoustic conditions by examining the noise reduction rate (NRR) and convergence speed. The result of the simulation confirms the efficacy and accuracy of the proposed algorithm.

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Research paper thumbnail of Speaker adaptive training for one-to-many eigenvoice conversion based on Gaussian mixture model

Interspeech, 2007

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Research paper thumbnail of Comparative Study on Directly-Aligned Multi-Point Controlled Wavefront Synthesis and Wave Field Synthesis

Proceedings Apsipa Asc 2009 Asia Pacific Signal and Information Processing Association 2009 Annual Summit and Conference, Oct 4, 2009

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Research paper thumbnail of Simple designing methods of corpus-based visual speech synthesis

Interspeech, 2003

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Research paper thumbnail of Simultaneous recognition of multiple sound sources based on 3-d n-best search using microphone array

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Research paper thumbnail of Recognition of Distant-Talking Speech based on 3-D Trellis Search using a Microphone Array and Adaptive Beamforming

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Research paper thumbnail of Noice Robust Real World Spoken Dialogue System using GMM Based Rejection of Unintended Inputs

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Research paper thumbnail of Fast back-propagation learning methods for large phonemic neural networks

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Research paper thumbnail of Maximum likelihood successive state splitting algorithm for tied-mixture HMNET

Interspeech, 1997

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Research paper thumbnail of How to judge reusability of existing speech corpora for target task by utilizing statistical multidimensional scaling

Interspeech 2007

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Research paper thumbnail of Dictation Machine Based on Japanese Character Source Modeling

International Journal of Pattern Recognition and Artificial Intelligence

This paper describes a phonetic typewriter and a dictation machine that utilize the underlying st... more This paper describes a phonetic typewriter and a dictation machine that utilize the underlying statistical structure of phoneme or character sequences. The approach of using syllable or character trigrams is applied to language source modeling. The language source models are obtained by calculating trigram probabilities from a large text database. These models are combined with the HMM-LR continuous speech recognition system.3,6 The phonetic typewriter is tested using 274 phrases uttered by one male speaker. The syllable source model achieves a 94.9% phoneme recognition rate with the test-set phoneme perplexity of 3.9. Without the syllable source model, the phoneme recognition rate is only 73.2%. A trigram model based on characters is also evaluated. This character source model can reduce the syllable perplexity significantly to 7.7, compared with 10.5 of the syllable source model. The character source model achieves a 78.5% character transcription rate for the 274 phrase utterances...

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Research paper thumbnail of Sound reproduction system with adaptive compensation of temperature fluctuation effect

2002 14th International Conference on Digital Signal Processing Proceedings. DSP 2002 (Cat. No.02TH8628), 2002

Bookmarks Related papers MentionsView impact

Research paper thumbnail of A broad-band signal control in sound reproduction system by temperature compensation of room impulse responses

6th International Conference on Signal Processing, 2002., 2000

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Research paper thumbnail of Evaluation of a Self-Generator Method for Initial Filters of Simo-Ica Applied to Blind Separation of Binaural Sound Mixtures

Ieee Workshop on Applications of Signal Processing to Audio and Acoustics 2005, Oct 1, 2005

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Multiple beamforming with source localization based on CSP analysis

Systems and Computers in Japan, May 1, 2003

ABSTRACT In this paper, a method is investigated for using an array of microphones to capture a h... more ABSTRACT In this paper, a method is investigated for using an array of microphones to capture a high-quality recording of voice under reverberation. In the past, multiple beamforming has been proposed by Flanagan's group as a method for efficiently suppressing reverberation. However, because this multiple beamforming method computes the arrival positions of the sound and its reflections based on the prior knowledge of the shape of the room and the position of the target sound source, it is not suited to practical applications. In this paper, a method of multiple beamforming is proposed in which the source positions of the sound and its reflections are localized without using prior knowledge of the shape of the room or the position of the target sound source. The effectiveness of the proposed method was evaluated in a simulation experiment using the image method and in an experiment in an actual hallway environment. First, an experiment was run to localize the positions of the sources of the sound and its reflections under a condition in which both the shape of the room and the source position of the sound were unknown. As the result, it was found that the localization error for the source position of the sound was small. Furthermore, although the localization errors for the source positions of the reflections were large, the localized reflection paths were found to be approximately equal to the true reflection paths. Next, the output signal from the proposed method was evaluated based on SNR. The result demonstrated that multiple beamforming is more effective than single beamforming. © 2003 Wiley Periodicals, Inc. Syst Comp Jpn, 34(5): 69–80, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/scj.1204

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Research paper thumbnail of Problems in Blind Separation of Convolutive Speech Mixtures by Negentropy Maximization

ABSTRACT: This paper aims to examine suitability of the marginal statistics based contrast functi... more ABSTRACT: This paper aims to examine suitability of the marginal statistics based contrast function eg negentropy for the separation of convolutive speech mixtures picked up by a linear microphone array. For this study we choose our frequency domain fixed-point ICA algorithm, based on ...

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Fast-Convergence Algorithm for Blind Source Separation Based on Array Signal Processing

Ieice Transactions on Fundamentals of Electronics Communications and Computer Sciences, Mar 1, 2003

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Automatic n-gram language model creation from web resources

Bookmarks Related papers MentionsView impact

Research paper thumbnail of One-to-Many and Many-to-One Voice Conversion Based on Eigenvoices

Acoustics Speech and Signal Processing 1988 Icassp 88 1988 International Conference on, Apr 15, 2007

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Musical-noise-free speech enhancement based on iterative Wiener filtering

Bookmarks Related papers MentionsView impact

Research paper thumbnail of A Fixed-Point Ica Algorithm for Convoluted Speech Signal Separation

ABSTRACT This paper describes a fixed-point independent component analysis (ICA) algorithm in com... more ABSTRACT This paper describes a fixed-point independent component analysis (ICA) algorithm in combination with the null beamforming technique to sieve out speech signals from their convoluted mixture observed using a linear microphone array. The fixed-point algorithm shows fast convergence to the solution, however it is highly sensitive to the initial value from which iteration starts. A good initial value leads to faster convergence and yields better results. We propose the use of a null beamformer-based initial value for iteration and explore its effects on separation performance under different acoustic conditions by examining the noise reduction rate (NRR) and convergence speed. The result of the simulation confirms the efficacy and accuracy of the proposed algorithm.

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Speaker adaptive training for one-to-many eigenvoice conversion based on Gaussian mixture model

Interspeech, 2007

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Comparative Study on Directly-Aligned Multi-Point Controlled Wavefront Synthesis and Wave Field Synthesis

Proceedings Apsipa Asc 2009 Asia Pacific Signal and Information Processing Association 2009 Annual Summit and Conference, Oct 4, 2009

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Simple designing methods of corpus-based visual speech synthesis

Interspeech, 2003

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Simultaneous recognition of multiple sound sources based on 3-d n-best search using microphone array

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Recognition of Distant-Talking Speech based on 3-D Trellis Search using a Microphone Array and Adaptive Beamforming

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Noice Robust Real World Spoken Dialogue System using GMM Based Rejection of Unintended Inputs

Bookmarks Related papers MentionsView impact

Research paper thumbnail of Fast back-propagation learning methods for large phonemic neural networks

Bookmarks Related papers MentionsView impact