H. Saruwatari

H. Saruwatari, Shinnosuke Takamichi, Yuki Saito, 2022, ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Multi-Task Adversarial Training Algorithm for Multi-Speaker Neural Text-to-Speech

H. Saruwatari, Yuki Saito, K. Udagawa, 2022, 2022 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Empirical Study Incorporating Linguistic Knowledge on Filled Pauses for Personalized Spontaneous Speech Synthesis

H. Saruwatari, Shinnosuke Takamichi, Takaaki Saeki, 2022, ArXiv.

Spontaneous speech synthesis with linguistic-speech consistency training using pseudo-filled pauses

H. Saruwatari, Shinnosuke Takamichi, Takaaki Saeki, 2022, ArXiv.

Deficient-basis-complementary rank-constrained spatial covariance matrix estimation based on multivariate generalized Gaussian distribution for blind speech extraction

H. Saruwatari, Daichi Kitamura, Norihiro Takamune, 2022, EURASIP Journal on Advances in Signal Processing.

Fast-convergence blind separation of more than two sources combining ICA and beamforming

K. Shikano, T. Nishikawa, H. Saruwatari, 2005 .

Predicting VQVAE-based Character Acting Style from Quotation-Annotated Text for Audiobook Speech Synthesis

H. Saruwatari, Tomoki Koriyama, Shinnosuke Takamichi, 2022, INTERSPEECH.

Exploring the Effectiveness of Self-supervised Learning and Classifier Chains in Emotion Recognition of Nonverbal Vocalizations

H. Saruwatari, Shinnosuke Takamichi, Detai Xin, 2022, ArXiv.

Interface for Barge-in Free Spoken Dialogue System Using Sound Field Control and Microphone Array

Kiyohiro Shikano, Hiroshi Saruwatari, Yoichi Hinamoto, 2002 .

Objective sound quality comparison based on higher-order statistics for nonlinear noise reduction methods (応用音響)

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2009 .

Visual onoma-to-wave: environmental sound synthesis from visual onomatopoeias and sound-source images

H. Saruwatari, Shinnosuke Takamichi, Kazuki Fujii, 2022, 2210.09173.

Text-to-speech synthesis from dark data with evaluation-in-the-loop data selection

H. Saruwatari, Shinnosuke Takamichi, Takaaki Saeki, 2022, ArXiv.

Unsupervised Speaker Adaptation Based on HMM Sufficient Statistics Using Spectral Subtraction and Noise Superimposition

Kiyohiro Shikano, Hiroshi Saruwatari, Shingo Yamade, 2003 .

Spatial Active Noise Control Based on Individual Kernel Interpolation of Primary and Secondary Sound Fields

H. Saruwatari, S. Koyama, Kazuyuki Arikawa, 2022, ICASSP 2022 - 2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Amplitude Matching: Majorization–Minimization Algorithm for Sound Field Control Only with Amplitude Constraint

H. Saruwatari, S. Koyama, Natsuki Ueno, 2021, ICASSP 2021 - 2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

jaCappella Corpus: A Japanese a Cappella Vocal Ensemble Corpus

H. Saruwatari, Shinnosuke Takamichi, Satoru Fukayama, 2022, 2211.16028.

jaCappella Corpus: A Japanese a Cappella Vocal Ensemble Corpus

H. Saruwatari, Shinnosuke Takamichi, Satoru Fukayama, 2022, ArXiv.

Perception analysis of inter-singer similarity in Japanese song

H. Saruwatari, Shinnosuke Takamichi, H. Tamaru, 2020, Acoustical Science and Technology.

Adaptive End-to-End Text-to-Speech Synthesis Based on Error Correction Feedback from Humans

H. Saruwatari, Yuki Saito, Kazuki Fujii, 2022, 2022 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

BLIND SOURCE SEPARATION CO九1BINING SIMO-恥IODEL-BASED ICA AND ADAP TIVE BEAMFORMING

T. Nishikawa, H. Saruwatari, Sαtoshi UKAI、TomoyαTAKATANI, 2012 .

Foreword: Special section on multi-channel acoustic signal processing

Yoshifumi Chisaki, Futoshi Asano, Hiroshi Saruwatari, 2005 .

Unsupervised speaker and noise adaptation based on sufficient HMM statistics

Akinobu Lee, Akira Baba, Shinichi Yoshizawa, 2002 .

Time-Domain Audio Source Separation With Neural Networks Based on Multiresolution Analysis

Hiroshi Saruwatari, Tomohiko Nakamura, Shihori Kozuka, 2021, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Time-Domain Audio Source Separation Based on Wave-U-Net Combined with Discrete Wavelet Transform

Hiroshi Saruwatari, Tomohiko Nakamura, H. Saruwatari, 2020, ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Sampling-Based Speech Parameter Generation Using Moment-Matching Networks

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2017, INTERSPEECH.

Music signal separation based on Bayesian spectral amplitude estimator with automatic target prior adaptation

Satoshi Nakamura, Hiroshi Saruwatari, Daichi Kitamura, 2014, 2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Statistical Analysis and Evaluation of Blind Speech Extraction Algorithms

Hiroshi Saruwatari, Ryoichi Miyazaki, H. Saruwatari, 2014 .

Theoretical analysis of musical noise in nonlinear noise reduction based on higher-order statistics

Hiroshi Saruwatari, Yu Takahashi, Kazunobu Kondo, 2012, Proceedings of The 2012 Asia Pacific Signal and Information Processing Association Annual Summit and Conference.

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

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2012, IEEE Transactions on Audio, Speech, and Language Processing.

Blind Speech Prior Estimation for Generalized Minimum Mean-Square Error Short-Time Spectral Amplitude Estimator

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2012, INTERSPEECH.

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

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2011, IEEE Transactions on Audio, Speech, and Language Processing.

Blind Speech Extraction Combining Generalized MMSE STSA Estimator and ICA-Based Noise and Speech Probability Density Function Estimations

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2010, LVA/ICA.

Musical-Noise Analysis in Methods of Integrating Microphone Array and Spectral Subtraction Based on Higher-Order Statistics

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2010, EURASIP J. Adv. Signal Process..

Convex and Differentiable Formulation for Inverse Problems in Hilbert Spaces with Nonlinear Clipping Effects

Hiroshi SARUWATARI, Natsuki UENO, Shoichi KOYAMA, 2021, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

Empirical Bayesian Independent Deeply Learned Matrix Analysis For Multichannel Audio Source Separation

Takuya Hasumi, Hiroshi Saruwatari, Daichi Kitamura, 2021, 2021 29th European Signal Processing Conference (EUSIPCO).

Independent deeply learned matrix analysis with automatic selection of stable microphone-wise update and fast sourcewise update of demixing matrix

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2021, Signal Process..

Independent Low-Rank Matrix Analysis Based on Time-Variant Sub-Gaussian Source Model for Determined Blind Source Separation

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2020, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Independent Deeply Learned Matrix Analysis for Determined Audio Source Separation

Shinnosuke Takamichi, Hiroshi Saruwatari, Daichi Kitamura, 2019, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

A review of blind source separation methods: two converging routes to ILRMA originating from ICA and NMF

Hirokazu Kameoka, Hiroshi Saruwatari, Daichi Kitamura, 2019, APSIPA Transactions on Signal and Information Processing.

Independent Low-Rank Matrix Analysis Based on Time-Variant Sub-Gaussian Source Model

Hiroshi Saruwatari, Hiroaki Nakajima, Daichi Kitamura, 2018, 2018 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Independent Deeply Learned Matrix Analysis for Multichannel Audio Source Separation

Shinnosuke Takamichi, Hiroshi Saruwatari, Daichi Kitamura, 2018, 2018 26th European Signal Processing Conference (EUSIPCO).

Generalized independent low-rank matrix analysis using heavy-tailed distributions for blind source separation

Nobutaka Ono, H. Saruwatari, Daichi Kitamura, 2018, EURASIP J. Adv. Signal Process..

Vectorwise Coordinate Descent Algorithm for Spatially Regularized Independent Low-Rank Matrix Analysis

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2018, 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Musical-Noise-Free Blind Speech Extraction Based on Higher-Order Statistics Analysis

Hiroshi Saruwatari, Ryoichi Miyazaki, H. Saruwatari, 2018 .

Independent low-rank matrix analysis based on complex student's t-distribution for blind audio source separation

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2017, 2017 IEEE 27th International Workshop on Machine Learning for Signal Processing (MLSP).

Experimental analysis of optimal window length for independent low-rank matrix analysis

Hiroshi Saruwatari, Daichi Kitamura, Nobutaka Ono, 2017, 2017 25th European Signal Processing Conference (EUSIPCO).

Determined Blind Source Separation Unifying Independent Vector Analysis and Nonnegative Matrix Factorization

Hirokazu Kameoka, Hiroshi Sawada, Hiroshi Saruwatari, 2016, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Relaxation of rank-1 spatial constraint in overdetermined blind source separation

Hirokazu Kameoka, Hiroshi Sawada, Hiroshi Saruwatari, 2015, 2015 23rd European Signal Processing Conference (EUSIPCO).

Efficient multichannel nonnegative matrix factorization exploiting rank-1 spatial model

Hirokazu Kameoka, Hiroshi Sawada, Hiroshi Saruwatari, 2015, 2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

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

Satoshi Nakamura, Kiyohiro Shikano, Hiroshi Saruwatari, 2014, Signal Process..

Optimized joint noise suppression and dereverberation based on blind signal extraction for hands-free speech recognition system

Satoshi Nakamura, Hiroshi Saruwatari, Tomoya Takatani, 2014, 2014 4th Joint Workshop on Hands-free Speech Communication and Microphone Arrays (HSCMA).

Toward musical-noise-free blind speech extraction: Concept and its applications

Satoshi Nakamura, Kiyohiro Shikano, Hiroshi Saruwatari, 2013, 2013 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference.

Real-time semi-blind speech extraction with speaker direction tracking on Kinect

Kiyohiro Shikano, Hiroshi Saruwatari, Yuji Onuma, 2012, Proceedings of The 2012 Asia Pacific Signal and Information Processing Association Annual Summit and Conference.

Sound-localization-preserved binaural MMSE STSA estimator with explicit and implicit binaural cues

Kiyohiro Shikano, Hiroshi Saruwatari, Martin Bouchard, 2012, 2012 Proceedings of the 20th European Signal Processing Conference (EUSIPCO).

Blind Source Separation for Speech Application Under Real Acoustic Environment

Hiroshi Saruwatari, Yu Takahashi, H. Saruwatari, 2012 .

Musical-noise-free blind speech extraction using ICA-based noise estimation and iterative spectral subtraction

Kiyohiro Shikano, Hiroshi Saruwatari, Kazunobu Kondo, 2012, 2012 11th International Conference on Information Science, Signal Processing and their Applications (ISSPA).

Semi-blind speech extraction for robot using visual information and noise statistics

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2011, 2011 IEEE International Symposium on Signal Processing and Information Technology (ISSPIT).

Improvement of speech recognition performance for spoken-oriented robot dialog system using end-fire array

Kiyohiro Shikano, Hiroshi Sawada, Hiroshi Saruwatari, 2010, 2010 IEEE/RSJ International Conference on Intelligent Robots and Systems.

Binaural Hearing Aid Using Sound-Localization-Preserved MMSE STSA Estimator with ICA-based Noise Estimation

Kiyohiro Shikano, Hiroshi Saruwatari, Masanobu Go, 2010 .

MMSE STSA estimator with nonstationary noise estimation based on ICA for high-quality speech enhancement

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2010, 2010 IEEE International Conference on Acoustics, Speech and Signal Processing.

Complex Newton algorithm for blind signal extraction of speech in diffuse noise

Kiyohiro Shikano, Hiroshi Saruwatari, Jani Even, 2010, 2010 IEEE International Conference on Acoustics, Speech and Signal Processing.

Fast and versatile blind separation of diverse sounds using closed-form estimation of probability density functions of sources

Akira Tanaka, Kiyohiro Shikano, Hiroshi Saruwatari, 2009, 2009 3rd IEEE International Workshop on Computational Advances in Multi-Sensor Adaptive Processing (CAMSAP).

Semi-blind suppression of internal noise for hands-free robot spoken dialog system

Kiyohiro Shikano, Hiroshi Sawada, Hiroshi Saruwatari, 2009, 2009 IEEE/RSJ International Conference on Intelligent Robots and Systems.

Blind signal extraction based speech enhancement in presence of diffuse background noise

Kiyohiro Shikano, Hiroshi Saruwatari, Jani Even, 2009, 2009 IEEE/SP 15th Workshop on Statistical Signal Processing.

Development and evaluation of pocket-size real-time blind source separation microphone

Kiyohiro Shikano, Hiroshi Saruwatari, Takashi Morita, 2009 .

Blind Spatial Subtraction Array for Speech Enhancement in Noisy Environment

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2009, IEEE Transactions on Audio, Speech, and Language Processing.

Target Speech Enhancement in Presence of Jammer and Diffuse Background Noise

Kiyohiro Shikano, Hiroshi Saruwatari, Jani Even, 2009, ICA.

Enhancement of speech signals separated from their convolutive mixture by FDICA algorithm

Kiyohiro Shikano, Hiroshi Saruwatari, Rajkishore Prasad, 2009, Digit. Signal Process..

An ICA Algorithm for Separation of Convolutive Mixture of Speech Signals

Kiyohiro Shikano, Hiroshi Saruwatari, Rajkishore Prasad, 2008 .

Real-time implementation of blind spatial subtraction array for hands-free robot spoken dialogue system

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2008, 2008 IEEE/RSJ International Conference on Intelligent Robots and Systems.

Development and evaluation of hands-free spoken dialogue system for railway station guidance

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2008, INTERSPEECH.

Extension of score function difference for frequency domain blind source separation

Kiyohiro Shikano, Hiroshi Saruwatari, Jani Even, 2008, 2008 16th European Signal Processing Conference.

Fast Convergence Blind Source Separation Using Frequency Subband Interpolation by Null Beamforming

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2008, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

Frequency domain semi-blind signal separation: application to the rejection of internal noises

Kiyohiro Shikano, Hiroshi Saruwatari, Jani Even, 2008, 2008 IEEE International Conference on Acoustics, Speech and Signal Processing.

Source-oriented localization control of stereo audio signals based on blind source separation

Kiyohiro Shikano, Hiroshi Saruwatari, Toshiyuki Nomura, 2008, 2008 IEEE International Conference on Acoustics, Speech and Signal Processing.

Blind Source Extraction for Hands-Free Speech Recognition Based on Wiener Filtering and ICA-Based Noise Estimation

K. Shikano, H. Saruwatari, Yu. Takahashi, 2008, 2008 Hands-Free Speech Communication and Microphone Arrays.

Permutation-Robust Structure for ICA-Based Blind Source Extraction

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2007, 2007 IEEE International Conference on Acoustics, Speech and Signal Processing - ICASSP '07.

Robust spatial subtraction array with independent component analysis for speech enhancement

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2007, 2007 9th International Symposium on Signal Processing and Its Applications.

SIMO-Model-Based Blind Source Separation - Principle and its Applications

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2007, Blind Speech Separation.

Acoustic Source Separation Technology Based on Independent Component Analysis

Hiroshi Saruwatari, H. Saruwatari, 2007 .

BLIND SPATIAL SUBTRACTION ARRAY WITH INDEPENDENT COMPONENT ANALYSIS FOR HANDS-FREE SPEECH RECOGNITION

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2006 .

Blind Separation of Acoustic Signals Combining SIMO-Model-Based Independent Component Analysis and Binary Masking

Hiroshi Hashimoto, Kiyohiro Shikano, Hiroshi Saruwatari, 2006, EURASIP J. Adv. Signal Process..

Two-stage blind source separation based on ICA and binary masking for real-time robot audition system

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2005, 2005 IEEE/RSJ International Conference on Intelligent Robots and Systems.

Blind sound scene decomposition for robot audition using SIMO-model-based ICA

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2005, 2005 IEEE/RSJ International Conference on Intelligent Robots and Systems.

Approximating Negentropy of Time-Frequency Series of Speech for Fixed Point ICA by Negentropy Maximization

Kiyohiro Shikano, Hiroshi Saruwatari, Rajkishore Prasad, 2005 .

Blind separation of more than two sources based on high-convergence algorithm combining ICA and beamforming

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2005, 2005 13th European Signal Processing Conference.

Blind separation of binaural sound mixtures using SIMO-ICA with self-generator for initial filter

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2005, 2005 13th European Signal Processing Conference.

Blind Separation of Speech by Fixed-Point ICA with Source Adaptive Negentropy Approximation

Kiyohiro Shikano, Hiroshi Saruwatari, Rajkishore Prasad, 2005, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

Blind decomposition of binaural mixed signals using high-convergence algorithm combining SIMO-ICA and DOA estimation

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2005 .

Speech Enhancement Based on Blind Source Separation in Car Environments

K. Shikano, T. Nishikawa, H. Saruwatari, 2005, 21st International Conference on Data Engineering Workshops (ICDEW'05).

Blind Separation and Deconvolution for Convolutive Mixture of Speech Combining SIMO-Model-Based ICA and Multichannel Inverse Filtering

Kiyohiro Shikano, Hiroshi Saruwatari, Tsuyoki Nishikawa, 2003, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

Stable and Low-Distortion Algorithm Based on Overdetermined Blind Separation for Convolutive Mixtures of Speech

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2004, ICA.

Multistage SIMO-model-based blind source separation combining frequency-domain ICA and time-domain ICA

Kiyohiro Shikano, Hiroshi Sawada, Ryo Mukai, 2004, 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing.

Overdetermined blind separation for convolutive mixtures of speech based on multistage ICA using subarray processing

Kiyohiro Shikano, Hiroshi Abe, Tsuyoki Nishikawa, 2004, 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing.

Blind separation of binaural sound mixtures using SIMO-model-based independent component analysis

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2004, 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing.

Overdetermined Blind Separation of Acoustic Signals Based on MISO-Constrained Frequency-Domain ICA

Kiyohiro Shikano, Hiroshi Abe, Tsuyoki Nishikawa, 2004 .

Blind separation and deconvolution of MIMO system driven by colored inputs using SIMO-model-based ICA with information-geometric learning

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003, 2003 IEEE XIII Workshop on Neural Networks for Signal Processing (IEEE Cat. No.03TH8718).

High-Fidelity Blind Separation of Acoustic Signals using SIMO-Model-based ICA with Information-Geometric Learning

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003 .

Stable Learning Algorithm for Blind Separation of Temporally Correlated Acoustic Signals Combining Multistage ICA and Linear Prediction

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

High-fidelity blind separation for convolutive mixture of acoustic signals using SIMO-model-based independent component analysis

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003, Seventh International Symposium on Signal Processing and Its Applications, 2003. Proceedings..

Blind source separation based on binaural ICA

Tsuyoki Nishikawa, Hiroshi Saruwatari, Tomoya Takatani, 2003, 2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03)..

MULTISTAGE ICA FOR BLIND SOURCE SEPARATION OF REAL ACOUSTIC CONVOLUTIVE MIXTURE

Kiyohiro Shikano, Shoko Araki, Shoji Makino, 2003 .

SIMO-Model-Based Independent Component Analysis for High-Fidelity Blind Separation of Acoustic Signals

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003 .

SPEECH ENHANCEMENT AND RECOGNITION IN CAR ENVIRONMENT USING BLIND SOURCE SEPARATION AND SUBBAND ELIMINATION PROCESSING

Kiyohiro Shikano, Hiroshi Saruwatari, Masao Sakata, 2003 .

Blind Source Separation Combining Independent Component Analysis and Beamforming

Kiyohiro Shikano, Kazuya Takeda, Fumitada Itakura, 2003, EURASIP J. Adv. Signal Process..

Stable learning algorithm for low-distortion blind separation of real speech mixture combining multistage ICA and linear prediction

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003, NOLISP.

Speech enhancement in car environment using blind source separation

Kiyohiro Shikano, Hiroshi Saruwatari, Akinobu Lee, 2002, INTERSPEECH.

Blind source separation based on fast-convergence algorithm using ICA and beamforming for real convolutive mixture

Hiroshi Saruwatari, Masao Sakata, Toshiya Kawamura, 2002, 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing.

Bund source separation based on Multi-Stage ICA combining frequency-domain ICA and time-domain ICA

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2002, 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing.

Blind Source Separation Based on Fast-Convergence Algorithm Using ICA and Array Signal Processing

Kiyohiro Shikano, Hiroshi Saruwatari, Toshiya Kawamura, 2001 .

Blind Source Separation Based on Fast-Convergence Algorithm Using ICA and Beamforming

Kiyohiro Shikano, Hiroshi Saruwatari, Toshiya Kawamura, 2001 .

Blind source separation for speech based on fast-convergence algorithm with ICA and beamforming

Kiyohiro Shikano, Hiroshi Saruwatari, Toshiya Kawamura, 2001, INTERSPEECH.

Development of a Toolkit Handling Multiple Speech-Oriented Guidance Agents for Mobile Applications

Kiyohiro Shikano, Sunao Hara, Hiroshi Saruwatari, 2012, Natural Interaction with Robots, Knowbots and Smartphones, Putting Spoken Dialog Systems into Practice.

Improving Speech Prosody of Audiobook Text-to-Speech Synthesis with Acoustic and Textual Contexts

H. Saruwatari, Shinnosuke Takamichi, Sharath Adavanne, 2022, ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

J-MAC: Japanese multi-speaker audiobook corpus for speech synthesis

Shinnosuke Takamichi, Hiroshi Saruwatari, Naoko Tanji, 2022, INTERSPEECH.

Audiobook Speech Synthesis Conditioned by Cross-Sentence Context-Aware Word Embeddings

Tomoki Koriyama, Shinnosuke Takamichi, Yusuke Ijima, 2021, 11th ISCA Speech Synthesis Workshop (SSW 11).

Kernel interpolation of acoustic transfer functions with adaptive kernel for directed and residual reverberations

Juliano G. C. Ribeiro, H. Saruwatari, S. Koyama, 2023, 2303.03869.

Robots that can hear, understand and talk

Kiyohiro Shikano, Hiroshi Saruwatari, Rajkishore Prasad, 2004, Adv. Robotics.

Blind Source Separation for Convolutive Mixtures of Speech using Subband processing

Shoko Araki, Shoji Makino, Tsuyoki Nishikawa, 2002 .

Design and collection of acoustic sound data for hands-free speech recognition and sound scene understanding

Satoshi Nakamura, Tetsunori Kobayashi, Takanobu Nishiura, 2002, Proceedings. IEEE International Conference on Multimedia and Expo.

Barge-in- and noise-free spoken dialogue interface based on sound field control and semi-blind source separation

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2007, 2007 15th European Signal Processing Conference.

Interface for Barge-in Free Spoken Dialogue System Based on Sound Field Reproduction and Microphone Array

Kiyohiro Shikano, Hiroshi Saruwatari, Yosuke Tatekura, 2007, EURASIP J. Adv. Signal Process..

Noise-robust hands-free speech recognition based on spatial subtraction array and known noise superimposition

Kiyohiro Shikano, Yasuaki Ohashi, Tsuyoki Nishikawa, 2005, 2005 IEEE/RSJ International Conference on Intelligent Robots and Systems.

Barge-in free spoken dialogue interface using nullspace-based sound field control and beamforming

Kiyohiro Shikano, Hiroshi Saruwatari, Yosuke Tatekura, 2005, 2005 13th European Signal Processing Conference.

Interface for Barge-in Free Spoken Dialogue System Combining Adaptive Sound Field Control and Microphone Array

Kiyohiro Shikano, Hiroshi Saruwatari, Tatsunori Asai, 2005, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

Interface for barge-in free spoken dialogue system using adaptive sound field control

Kiyohiro Shikano, Hiroshi Saruwatari, Shigeki Miyabe, 2004, INTERSPEECH.

Interface for barge-in free spoken dialogue system based on sound field control and microphone array

Kiyohiro Shikano, Hiroshi Saruwatari, Yoichi Hinamoto, 2003, 2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03)..

Mutual-Information-Based Sensor Placement for Spatial Sound Field Recording

Hiroshi Saruwatari, Shoichi Koyama, Natsuki Ueno, 2020, ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Multichannel Blind Source Separation Based on Evanescent-Region-Aware Non-Negative Tensor Factorization in Spherical Harmonic Domain

Hiroshi Saruwatari, Shoichi Koyama, Norihiro Takamune, 2021, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Compensating of room acoustic transfer functions affected by change of room temperature

Kazuya Takeda, Fumitada Itakura, Michiaki Omura, 1999, 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258).

Unsupervised acoustic model adaptation algorithm using MLLR in a noisy environment

Kiyohiro Shikano, Hiroshi Saruwatari, Akinobu Lee, 2006 .

Spectral subtraction in noisy environments applied to speaker adaptation based on HMM sufficient statistics

Kiyohiro Shikano, Hiroshi Saruwatari, Akinobu Lee, 2002, INTERSPEECH.

Unsupervised noisy environment adaptation algorithm using MLLR and speaker selection

Kiyohiro Shikano, Hiroshi Saruwatari, Akinobu Lee, 2001, INTERSPEECH.

JSUT corpus: free large-scale Japanese speech corpus for end-to-end speech synthesis pdf

Shinnosuke Takamichi, Hiroshi Saruwatari, Ryosuke Sonobe, 2017, ArXiv.

Selective multi-path acoustic model based on database likelihoods

Kiyohiro Shikano, Hiroshi Saruwatari, Akinobu Lee, 2002, INTERSPEECH.

Topic Classification of Spoken Inquiries Using Transductive Support Vector Machine

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoko Matsui, 2012, Natural Interaction with Robots, Knowbots and Smartphones, Putting Spoken Dialog Systems into Practice.

A MAP Estimator for the Enhancement of Speech Signal Separated by ICA Algorithm (国際ワークショップ Frontiers in Speech and Hearing Research)

Hiroshi Saruwatari, Fumitoshi Matsuno, Rajkishore Prasad, 2006 .

TransVoice: Real-Time Voice Conversion for Augmenting Near-Field Speech Communication

Shinnosuke Takamichi, Hiroshi Saruwatari, Riku Arakawa, 2019, UIST.

Learning to Speak from Text: Zero-Shot Multilingual Text-to-Speech with Unsupervised Text Pretraining

H. Saruwatari, Shinji Watanabe, Shinnosuke Takamichi, 2023, ArXiv.

Cross-Lingual Speaker Adaptation Using Domain Adaptation and Speaker Consistency Loss for Text-To-Speech Synthesis

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2021, Interspeech.

Disentangled Speaker and Language Representations Using Mutual Information Minimization and Domain Adaptation for Cross-Lingual TTS

Shinnosuke Takamichi, Hiroshi Saruwatari, Tatsuya Komatsu, 2021, ICASSP 2021 - 2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

JVS corpus: free Japanese multi-speaker voice corpus pdf

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2019, ArXiv.

A tissue-conductive acoustic sensor applied in speech recognition for privacy

Kiyohiro Shikano, Yoshitaka Nakajima, Hiroshi Saruwatari, 2005, sOc-EUSAI '05.

Cross-Lingual Text-To-Speech Synthesis via Domain Adaptation and Perceptual Similarity Regression in Speaker Space

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2020, INTERSPEECH.

GMM-based voice conversion applied to emotional speech synthesis

Tomoki Toda, Kiyohiro Shikano, Hiroshi Saruwatari, 2003, INTERSPEECH.

jaCappella Corpus: A Japanese a Cappella Vocal Ensemble Corpus

H. Saruwatari, Shinnosuke Takamichi, Satoru Fukayama, 2022, ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

JVS-MuSiC: Japanese multispeaker singing-voice corpus pdf

Shinnosuke Takamichi, Hiroshi Saruwatari, Naoko Tanji, 2020, ArXiv.

JSUT and JVS: Free Japanese voice corpora for accelerating speech synthesis research

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2020, Acoustical Science and Technology.

JNV Corpus: A Corpus of Japanese Nonverbal Vocalizations with Diverse Phrases and Emotions

H. Saruwatari, Shinnosuke Takamichi, Detai Xin, 2023, ArXiv.

Prior Distribution Design for Music Bleeding-Sound Reduction Based on Nonnegative Matrix Factorization pdf

Hiroshi Saruwatari, Daichi Kitamura, Kazunobu Kondo, 2021, 2021 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Generative Moment Matching Network-Based Neural Double-Tracking for Synthesized and Natural Singing Voices

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2020, IEICE Trans. Inf. Syst..

BLIND SIGNAL EXTRACTION VIA DlRECT おIfUTUAL INFORおIfATION MINIMIZATION

K. Shikano, H. Saruwatari, J. Even, 2012 .

Real-Time Implementation of Two-Stage Blind Source Separation Combining SIMO-ICA and Binary Masking

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2005 .

Equivalence between Frequency-Domain Blind Source Separation and Frequency-Domain Adaptive Beamforming for Convolutive Mixtures

Shoko Araki, Shoji Makino, Ryo Mukai, 2002, 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing.

V2S attack: building DNN-based voice conversion from automatic speaker verification pdf

Shinnosuke Takamichi, Yusuke Ijima, Hiroshi Saruwatari, 2019, 10th ISCA Workshop on Speech Synthesis (SSW 10).

Statistical Parametric Speech Synthesis Incorporating Generative Adversarial Networks

Shinnosuke Takamichi, Hiroshi Saruwatari, Yuki Saito, 2017, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Independent Deeply Learned Tensor Analysis for Determined Audio Source Separation

Tomohiro Nakatani, Hiroshi Saruwatari, Daichi Kitamura, 2021, 2021 29th European Signal Processing Conference (EUSIPCO).

Spatio-temporal sparse sound field decomposition considering acoustic source signal characteristics

Hiroshi Saruwatari, Shoichi Koyama, Norihiro Takamune, 2017, 2017 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Sampling-Frequency-Independent Convolutional Layer and its Application to Audio Source Separation

H. Saruwatari, K. Yatabe, Tomohiko Nakamura, 2022, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

ChatGPT-EDSS: Empathetic Dialogue Speech Synthesis Trained from ChatGPT-derived Context Word Embeddings

H. Saruwatari, Shinnosuke Takamichi, Kentaro Tachibana, 2023, INTERSPEECH 2023.

VTTS: Visual-Text To Speech

H. Saruwatari, Shinnosuke Takamichi, Katsuhito Sudoh, 2022, 2022 IEEE Spoken Language Technology Workshop (SLT).

Shinnosuke Takamichi, Hiroshi Saruwatari, Yuki Saito, 2021, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Multichannel Active Noise Control with Exterior Radiation Suppression Based on Riemannian Optimization

H. Saruwatari, Kazuyuki Arikawa, Shoichi Koyama, 2023, 2306.08855.

How Generative Spoken Language Modeling Encodes Noisy Speech: Investigation from Phonetics to Syntactics

H. Saruwatari, Shinnosuke Takamichi, Tomohiko Nakamura, 2023, INTERSPEECH 2023.

Region-Restricted Sensor Placement Based on Gaussian Process for Sound Field Estimation

H. Saruwatari, Tomoya Nishida, Natsuki Ueno, 2022, IEEE Transactions on Signal Processing.

An evaluation of alaryngeal speech enhancement methods based on voice conversion techniques

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2011, 2011 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Voice Conversion Using Sequence-to-Sequence Learning of Context Posterior Probabilities

H. Saruwatari, Shinnosuke Takamichi, Yuki Saito, 2017, INTERSPEECH.

JSSS: free Japanese speech corpus for summarization and simplification pdf

Shinnosuke Takamichi, Hiroshi Saruwatari, Mamoru Komachi, 2020, ArXiv.

Ego Noise Reduction for Hose-Shaped Rescue Robot Combining Independent Low-Rank Matrix Analysis and Multichannel Noise Cancellation

Takeshi Yamada, Shoji Makino, Hiroshi Saruwatari, 2016, LVA/ICA.

Implementation of DNN-based real-time voice conversion and its improvements by audio data augmentation and mask-shaped device

Shinnosuke Takamichi, Hiroshi Saruwatari, Riku Arakawa, 2019, 10th ISCA Workshop on Speech Synthesis (SSW 10).

Alaryngeal Speech Enhancement Based on One-to-Many Eigenvoice Conversion

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2014, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

CPJD Corpus: Crowdsourced Parallel Speech Corpus of Japanese Dialects

Shinnosuke Takamichi, Hiroshi Saruwatari, H. Saruwatari, 2018, LREC.

Automatic optimization scheme of spectral subtraction based on musical noise assessment via higher-order statistics

Kiyohiro Shikano, Hiroshi Saruwatari, Yu Takahashi, 2008 .

Speech enhancement in presence of diffuse background noise: Why using blind signal extraction?

Kiyohiro Shikano, Hiroshi Saruwatari, Jani Even, 2010, 2010 IEEE International Conference on Acoustics, Speech and Signal Processing.

Training Data Size Requirements for Topic Classification in a Speech-Oriented Guidance System

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoko Matsui, 2010 .

Comparison of methods for topic classification in a speech-oriented guidance system

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoko Matsui, 2010, INTERSPEECH.

Unknown example detection for example-based spoken dialog system

Kiyohiro Shikano, Hiroshi Saruwatari, Hiromichi Kawanami, 2009, 2009 Oriental COCOSDA International Conference on Speech Database and Assessments.

Question and answer database optimization using speech recognition results

Kiyohiro Shikano, Hiroshi Saruwatari, Hiromichi Kawanami, 2008, INTERSPEECH.

Development, Long-Term Operation and Portability of a Real-Environment Speech-Oriented Guidance System

Kiyohiro Shikano, Ryuichi Nisimura, Hiroshi Saruwatari, 2008, IEICE Trans. Inf. Syst..

Development and portability of ASR and Q&A modules for real-environment speech-oriented guidance systems

Kiyohiro Shikano, Hiroshi Saruwatari, Hiromichi Kawanami, 2007, 2007 IEEE Workshop on Automatic Speech Recognition & Understanding (ASRU).

Deploying a speech‐based information system as a research platform for speech recognition research in real environments

Kiyohiro Shikano, Hiroshi Saruwatari, Akinobu Lee, 2005 .

Sparse Representation Using Multidimensional Mixed-Norm Penalty With Application to Sound Field Decomposition

Hiroshi Saruwatari, Shoichi Koyama, Norihiro Takamune, 2018, IEEE Transactions on Signal Processing.

Sound field decomposition in reverberant environment using sparse and low-rank signal models

Hiroshi Saruwatari, Shoichi Koyama, H. Saruwatari, 2016, 2016 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Sparse sound field decomposition with parametric dictionary learning for super-resolution recording and reproduction

Hiroshi Saruwatari, Shoichi Koyama, Norihiro Takamune, 2015, 2015 IEEE 6th International Workshop on Computational Advances in Multi-Sensor Adaptive Processing (CAMSAP).

Sparse sound field decomposition using group sparse Bayesian learning

Hiroshi Saruwatari, Shoichi Koyama, Naoki Murata, 2015, 2015 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA).

Sparse sound field decomposition for super-resolution in recording and reproduction.

Hiroshi Saruwatari, Shoichi Koyama, Naoki Murata, 2018, The Journal of the Acoustical Society of America.

Structured sparse signal models and decomposition algorithm for super-resolution in sound field recording and reproduction

Hiroshi Saruwatari, Shoichi Koyama, Naoki Murata, 2015, 2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Utterance-Level Sequential Modeling for Deep Gaussian Process Based Speech Synthesis Using Simple Recurrent Unit

Tomoki Koriyama, Hiroshi Saruwatari, H. Saruwatari, 2020, ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Phase Reconstruction from Amplitude Spectrograms Based on Von-Mises-Distribution Deep Neural Network

Shinnosuke Takamichi, Hiroshi Saruwatari, Daichi Kitamura, 2018, 2018 16th International Workshop on Acoustic Signal Enhancement (IWAENC).

Ego Noise Reduction and Sound Localization Adapted to Human Ears Using Hose-shaped Rescue Robot

Nobutaka Ono, H. Saruwatari, Daichi Kitamura, 2018 .

Real-time Cepstrum Mean Normalization Using Codebook

Tomoki Toda, Kiyohiro Shikano, Hiroshi Saruwatari, 2006 .

Differentiable Digital Signal Processing Mixture Model for Synthesis Parameter Extraction from Mixture of Harmonic Sounds

H. Saruwatari, Daichi Kitamura, Yu Takahashi, 2022, ICASSP 2022 - 2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Federated Learning for Human-in-the-Loop Many-to-Many Voice Conversion

H. Saruwatari, Yuki Saito, Ryunosuke Hirai, 2023, 12th ISCA Speech Synthesis Workshop (SSW2023).

Text-to-Speech Synthesis Using STFT Spectra Based on Low-/Multi-Resolution Generative Adversarial Networks

Shinnosuke Takamichi, Hiroshi Saruwatari, Yuki Saito, 2018, 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Acoustic Modeling for End-to-End Empathetic Dialogue Speech Synthesis Using Linguistic and Prosodic Contexts of Dialogue History

H. Saruwatari, Shinnosuke Takamichi, Kentaro Tachibana, 2022, Interspeech.

Sequence-to-Sequence Learning for Deep Gaussian Process Based Speech Synthesis Using Self-Attention GP Layer

Tomoki Koriyama, Hiroshi Saruwatari, Taiki Nakamura, 2021, Interspeech.

Multi-speaker Text-to-speech Synthesis Using Deep Gaussian Processes

Tomoki Koriyama, Hiroshi Saruwatari, Kentaro Mitsui, 2020, INTERSPEECH.

CALLS: Japanese Empathetic Dialogue Speech Corpus of Complaint Handling and Attentive Listening in Customer Center

H. Saruwatari, Shinnosuke Takamichi, Kentaro Tachibana, 2023, INTERSPEECH 2023.

Discriminative and reconstructive basis training for audio source separation with semi-supervised nonnegative matrix factorization

Hiroshi Saruwatari, Daichi Kitamura, Yu Takahashi, 2016, 2016 IEEE International Workshop on Acoustic Signal Enhancement (IWAENC).

Multichannel blind source separation based on non-negative tensor factorization in wavenumber domain

Hiroshi Saruwatari, Shoichi Koyama, Yuki Mitsufuji, 2016, 2016 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Multichannel Non-Negative Matrix Factorization Using Banded Spatial Covariance Matrices in Wavenumber Domain

Hiroshi Saruwatari, Shoichi Koyama, Daichi Kitamura, 2020, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Blind Detection of CLT Disobeying Frequency Bind for Audio Source Separation by Fixed-Point ICA

Kiyohiro Shikano, Hiroshi Saruwatari, Raj Kishore Prasad, 2004 .

Algorithms of Sampling-Frequency-Independent Layers for Non-integer Strides

H. Saruwatari, K. Yatabe, Norihiro Takamune, 2023, ArXiv.

Improving robustness of spontaneous speech synthesis with linguistic speech regularization and pseudo-filled-pause insertion

H. Saruwatari, Shinnosuke Takamichi, Takaaki Saeki, 2022, 12th ISCA Speech Synthesis Workshop (SSW2023).

Coco-Nut: Corpus of Japanese Utterance and Voice Characteristics Description for Prompt-based Control

H. Saruwatari, Shinnosuke Takamichi, Yuki Saito, 2023, ArXiv.

Personalized Filled-pause Generation with Group-wise Prediction Models

H. Saruwatari, Shinnosuke Takamichi, Takaaki Saeki, 2022, LREC.

Noise Robust Acoustic Anomaly Detection System with Nonnegative Matrix Factorization Based on Generalized Gaussian Distribution

Shinnosuke Takamichi, Hiroshi Saruwatari, Daichi Kitamura, 2021, IEICE Trans. Inf. Syst..

Duration-Aware Pause Insertion Using Pre-Trained Language Model for Multi-Speaker Text-To-Speech

H. Saruwatari, Tomoki Koriyama, D. Yang, 2023, ICASSP 2023 - 2023 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Hyperbolic Timbre Embedding for Musical Instrument Sound Synthesis Based on Variational Autoencoders

H. Saruwatari, Norihiro Takamune, Satoru Fukayama, 2022, 2022 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Learning to Speak from Text: Zero-Shot Multilingual Text-to-Speech with Unsupervised Text Pretraining

H. Saruwatari, Shinji Watanabe, Shinnosuke Takamichi, 2023, IJCAI.

JVNV: A Corpus of Japanese Emotional Speech with Verbal Content and Nonverbal Expressions

H. Saruwatari, Shinnosuke Takamichi, Yuki Saito, 2023, ArXiv.

Speech Enhancement Based on Blind Source Separation in Car Environments

K. Shikano, T. Nishikawa, H. Saruwatari, 2005, 21st International Conference on Data Engineering Workshops (ICDEW'05).

Independent low-rank matrix analysis based on parametric majorization-equalization algorithm

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2017, 2017 IEEE 7th International Workshop on Computational Advances in Multi-Sensor Adaptive Processing (CAMSAP).

Joint-diagonalizability-constrained multichannel nonnegative matrix factorization based on time-variant multivariate complex sub-Gaussian distribution

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2021, Signal Process..

Fundamental limitation of frequency domain blind source separation for convolutive mixture of speech

Shoko Araki, Shoji Makino, Tsuyoki Nishikawa, 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).

Development and Evaluation of Pocket-Size Blind Source Separation Microphone

Kiyohiro Shikano, Hiroshi Saruwatari, Takashi Morita, 2008 .

Comparison of time-domain ICA, frequency-domain ICA and multistage ICA for blind source separation

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2002, 2002 11th European Signal Processing Conference.

Regularized Fast Multichannel Nonnegative Matrix Factorization with ILRMA-Based Prior Distribution of Joint-Diagonalization Process

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2020, ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Blind source separation based on independent low-rank matrix analysis with sparse regularization for time-series activity

Nobutaka Ono, H. Saruwatari, Daichi Kitamura, 2017, 2017 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Blind source separation combining frequency-domain ICA and beamforming

Kazuya Takeda, Hiroshi Saruwatari, Satoshi Kurita, 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).

Music signal separation using supervised NMF with all-pole-model-based discriminative basis deformation

Hiroshi Saruwatari, Shoichi Koyama, Hiroaki Nakajima, 2016, 2016 24th European Signal Processing Conference (EUSIPCO).

NoisyILRMA: Diffuse-Noise-Aware Independent Low-Rank Matrix Analysis for Fast Blind Source Extraction

H. Saruwatari, T. Nakatani, Daichi Kitamura, 2023, 2023 31st European Signal Processing Conference (EUSIPCO).

Speech Enhancement by Noise Self-Supervised Rank-Constrained Spatial Covariance Matrix Estimation via Independent Deeply Learned Matrix Analysis pdf

Shoji Makino, Hiroshi Saruwatari, Daichi Kitamura, 2021, 2021 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Blind Speech Extraction Based on Rank-Constrained Spatial Covariance Matrix Estimation With Multivariate Generalized Gaussian Distribution

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2020, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Acceleration of rank-constrained spatial covariance matrix estimation for blind speech extraction

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2019, 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Efficient Full-Rank Spatial Covariance Estimation Using Independent Low-Rank Matrix Analysis for Blind Source Separation

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2019, 2019 27th European Signal Processing Conference (EUSIPCO).

Subband-Based Blind Separation for Convolutive Mixtures of Speech

Shoko Araki, Shoji Makino, Tsuyoki Nishikawa, 2005, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

Subband based blind source separation for convolutive mixtures of speech

Shoko Araki, Shoji Makino, Tsuyoki Nishikawa, 2003, 2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03)..

Region-to-Region Kernel Interpolation of Acoustic Transfer Functions Constrained by Physical Properties

Juliano G. C. Ribeiro, H. Saruwatari, Natsuki Ueno, 2022, IEEE/ACM Transactions on Audio, Speech, and Language Processing.

Speech kurtosis estimation from observed noisy signal based on generalized Gaussian distribution prior and additivity of cumulants

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2012, 2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Improved Method of Blind Speech Separation with Low Computational Complexity

Kazuya Takeda, Hiroshi Saruwatari, Kazunobu Kondo, 2011 .

Efficient blind speech separation suitable for embedded devices

Kazuya Takeda, Hiroshi Saruwatari, Yu Takahashi, 2011, 2011 19th European Signal Processing Conference.

Temporal quantization of spatial information using directional clustering for multichannel audio coding

Kiyohiro Shikano, Hiroshi Saruwatari, Toshiyuki Nomura, 2009, 2009 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics.

Compressive Coding of Stereo Audio Signals Extracting Sparseness among Sound Sources with Independent Component Analysis

Kiyohiro Shikano, Hiroshi Saruwatari, Tomoya Takatani, 2007, 2007 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics.

Audio object individual operation and its application to earphone leakage noise reduction

Kiyohiro Shikano, Hiroshi Saruwatari, Shigeki Miyabe, 2010, 2010 4th International Symposium on Communications, Control and Signal Processing (ISCCSP).

Evaluation of SIMO Separation Methods for Blind Decomposition of Binaural Mixed Signals

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2005 .

Two-stage blind source separation combining SIMO-model-based ICA and adaptive beamforming

K. Shikano, T. Takatani, T. Nishikawa, 2005, 2005 13th European Signal Processing Conference.

Blind Separation and Deconvolution for Convolutive Mixture of Speech Combining SIMO-Model-Based ICA and Multichannel Inverse Filtering

K. Shikano, T. Takatani, T. Nishikawa, 2005, IEICE Trans. Fundam. Electron. Commun. Comput. Sci..

BLIND SEPARATION AND DECONVOLUTION FOR REAL CONVOLUTIVE MIXTURE OF TEMPORALLY CORRELATED ACOUSTIC SIGNALS USING SIMO-MODEL-BASED ICA

Kiyohiro Shikano, Tsuyoki Nishikawa, Hiroshi Saruwatari, 2003 .

Equivalence between frequency domain blind source separation and frequency domain adaptive beamforming

T. Nishikawa, H. Saruwatari, S. Araki, 2002, 2002 IEEE International Conference on Acoustics, Speech, and Signal Processing.

Humangan: Generative Adversarial Network With Human-Based Discriminator And Its Evaluation In Speech Perception Modeling

Yukino Baba, Shinnosuke Takamichi, Hiroshi Saruwatari, 2019, ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Robust Demixing Filter Update Algorithm Based on Microphone-wise Coordinate Descent for Independent Deeply Learned Matrix Analysis

Hiroshi Saruwatari, Daichi Kitamura, Norihiro Takamune, 2019, 2019 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC).

Human-in-the-loop Speaker Adaptation for DNN-based Multi-speaker TTS

H. Saruwatari, Yuki Saito, K. Udagawa, 2022, INTERSPEECH.

Source-Location-Informed Sound Field Recording and Reproduction

Hiroshi Saruwatari, Shoichi Koyama, Koichiro Ito, 2015, IEEE Journal of Selected Topics in Signal Processing.

Directionally Weighted Wave Field Estimation Exploiting Prior Information on Source Direction

Hiroshi Saruwatari, Shoichi Koyama, Natsuki Ueno, 2021, IEEE Transactions on Signal Processing.

Voice Conversion Using Input-to-Output Highway Networks

Shinnosuke Takamichi, Hiroshi Saruwatari, Yuki Saito, 2017, IEICE Trans. Inf. Syst..

Low-delay voice conversion based on maximum likelihood estimation of spectral parameter trajectory

K. Shikano, H. Saruwatari, T. Toda, 2008, INTERSPEECH.

Speaking-aid systems using GMM-based voice conversion for electrolaryngeal speech

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2012, Speech Commun..

Real-Time Full-Band Voice Conversion with Sub-Band Modeling and Data-Driven Phase Estimation of Spectral Differentials

Shinnosuke Takamichi, Hiroshi Saruwatari, Takaaki Saeki, 2021, IEICE Trans. Inf. Syst..

Vocoder-free text-to-speech synthesis incorporating generative adversarial networks using low-/multi-frequency STFT amplitude spectra

Shinnosuke Takamichi, Hiroshi Saruwatari, Yuki Saito, 2019, Comput. Speech Lang..

Prosody Correction Preserving Speaker Individuality for Chinese-Accented Japanese HMM-Based Text-to-Speech Synthesis

Daiki SEKIZAWA, Shinnosuke TAKAMICHI, Hiroshi SARUWATARI, 2019, IEICE Trans. Inf. Syst..

Training algorithm to deceive Anti-Spoofing Verification for DNN-based speech synthesis

Shinnosuke Takamichi, Hiroshi Saruwatari, Yuki Saito, 2017, 2017 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Statistical approach to voice quality control in esophageal speech enhancement

Tomoki Toda, Kiyohiro Shikano, Hiroshi Saruwatari, 2012, 2012 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Computationally efficient body-conducted voice conversion with original excitation signals

Tomoki Toda, Kiyohiro Shikano, Daisuke Deguchi, 2011 .

Speaking-Aid Systems Based on One-to-Many Eigenvoice Conversion for Total Laryngectomees

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2010 .

Linear transformation approaches to many-to-one voice conversion

Tomoki Toda, Yamato Ohtani, Kiyohiro Shikano, 2010, SSW.

Esophageal Speech Enhancement Based on Statistical Voice Conversion with Gaussian Mixture Models

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2010, IEICE Trans. Inf. Syst..

Adaptive voice-quality control based on one-to-many eigenvoice conversion

K. Shikano, H. Saruwatari, T. Toda, 2010, INTERSPEECH.

Acoustic Compensation Method for Accepting Different Recording Devices in Body-Conducted Voice Conversion

Tomoki Toda, Kiyohiro Shikano, Daisuke Deguchi, 2010 .

Adaptive Training for Voice Conversion Based on Eigenvoices

Tomoki Toda, Yamato Ohtani, Kiyohiro Shikano, 2010, IEICE Trans. Inf. Syst..

Non-parallel training for many-to-many eigenvoice conversion

Tomoki Toda, Yamato Ohtani, Kiyohiro Shikano, 2010, 2010 IEEE International Conference on Acoustics, Speech and Signal Processing.

Statistical approach to enhancing esophageal speech based on Gaussian mixture models

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2010, 2010 IEEE International Conference on Acoustics, Speech and Signal Processing.

Enhancement of Esophageal Speech Using Statistical Voice Conversion

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2009 .

Electrolaryngeal speech enhancement based on statistical voice conversion

K. Shikano, H. Saruwatari, T. Toda, 2009, INTERSPEECH.

Many-to-many eigenvoice conversion with reference voice

Tomoki Toda, Yamato Ohtani, Kiyohiro Shikano, 2009, INTERSPEECH.

Acoustic compensation methods for body transmitted speech conversion

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2009, 2009 IEEE International Conference on Acoustics, Speech and Signal Processing.

An improved one-to-many eigenvoice conversion system

Tomoki Toda, Yamato Ohtani, Kiyohiro Shikano, 2008, INTERSPEECH.

Evaluation of speaking-aid system with voice conversion for laryngectomees toward its use in practical environments

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2008, INTERSPEECH.

Maximum a posteriori adaptation for many-to-one eigenvoice conversion

Tomoki Toda, Yamato Ohtani, Kiyohiro Shikano, 2008, INTERSPEECH.

BANDWIDTH EXTENSION OF CELLULAR PHONE SPEECH BASED ON MAXIMUM LIKELIHOOD ESTIMATION WITH GMM

Tomoki Toda, Kiyohiro Shikano, Hiroshi Saruwatari, 2008 .

High-Quality Statistical Parametric Speech Synthesis Using Generative Adversarial Networks

H. Saruwatari, Yuki Saito, 2018 .

Speaker-Adaptive Speech Synthesis Based on Eigenvoice Conversion and Language-Dependent Prosodic Conversion in Speech-to-Speech Translation

Tomoki Toda, Kiyohiro Shikano, Hisashi Kawai, 2011, INTERSPEECH.

PAPER Special Section on Processing Natural Speech Variability for Improved Verbal Human-Computer Interaction Improvements of the One-to-Many Eigenvoice Conversion System

K. Shikano, H. Saruwatari, T. Toda, 2010 .

SelfRemaster: Self-Supervised Speech Restoration for Historical Audio Resources

H. Saruwatari, Shinnosuke Takamichi, Takaaki Saeki, 2023, IEEE Access.

Do learned speech symbols follow Zipf's law?

H. Saruwatari, Shinnosuke Takamichi, Joonyong Park, 2023, ICASSP 2024 - 2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Generative Moment Matching Network-based Random Modulation Post-filter for DNN-based Singing Voice Synthesis and Neural Double-tracking

Tomoki Koriyama, Shinnosuke Takamichi, Hiroshi Saruwatari, 2019, ICASSP 2019 - 2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Kernel interpolation of acoustic transfer function between regions considering reciprocity

Juliano G. C. Ribeiro, H. Saruwatari, Natsuki Ueno, 2020, 2020 IEEE 11th Sensor Array and Multichannel Signal Processing Workshop (SAM).

Evaluation of Extremely Small Sound Source Signals Used in Speaking-Aid System with Statistical Voice Conversion

Tomoki Toda, Kiyohiro Shikano, Keigo Nakamura, 2010, IEICE Trans. Inf. Syst..

Voice conversion algorithm based on Gaussian mixture model with dynamic frequency warping of STRAIGHT spectrum

K. Shikano, T. Toda, H. Saruwatari, 2001, 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221).

Humanacgan: Conditional Generative Adversarial Network with Human-Based Auxiliary Classifier and its Evaluation in Phoneme Perception

Yukino Baba, Shinnosuke Takamichi, Hiroshi Saruwatari, 2021, ICASSP 2021 - 2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).

Multi-speaker Text-to-speech Synthesis Using Deep Gaussian Processes

H. Saruwatari, Tomoki Koriyama, Kentaro Mitsui, 2020, INTERSPEECH.