The present status, progress, and usage of speech databases in Japan

スポンサーリンク

概要

• 論文の詳細を見る

• The present status, progress and usage of Japanese speech database has been described. The database project in Japan started in the early 1980s. The first was by the Japan Electronic Industry Development Association (JEIDA), which aimed at creating a speech database to evaluate performance of the existing speech input/output machines and systems. Several database projects have been undertaken since then, including the one initiated by the Advanced Telecommunication Research Institute (ATR), and now we have reached a point where an enormous amount of spontaneous speech data is available. A survey was conducted recently on usage of the presently existing speech databases among industry and university institutions in Japan where speech research is now actively going on. It was revealed that the ATR's continuous speech database is the most frequently used, followed by the equivalent version from the Acoustical Society of Japan.

• 社団法人日本音響学会の論文

著者

• NAKAMURA Satoshi

  ATR Spoken Language Translation Research Labs.

• Takeda Kazuya

  Center for Integrated Acoustic Information Research, Graduate School of Information Science, Nagoya

• Yamamoto M

  University Of Tsukuba

• Kuwabara Hisao

  Teikyo University of Science and Technology

• Itahashi Shuich

  University of Tsukuba

• Yamamoto Mikio

  University of Tsukuba

• Takezawa Toshiyuki

  ATR Spoken Language Translation Research Laboratories

• Takezawa T

  Atr Spoken Language Translation Research Laboratories

• Itahashi Shuichi

  University Of Tsukuba Japan

• Takeda Kazuya

  Center For Integrated Acoustic Information Research Nagoya University

• Nakamura Satoshi

  Atr Spoken Language Translation Res. Lab. Kyoto Jpn

• Nakamura Satoshi

  Atr Spoken Language Communication Res. Lab. Kyoto‐fu Jpn

• Nakamura Satoshi

  Atr Spoken Language Translation Research Laboratories

• Itahashi Shuichi

  University of Tsukuba

関連論文

• Combination Therapy with Vascular Endothelial Growth Factor Neutralizing Antibody and Mitomycin C on Human Gastric Cancer Xenograft
• Noise and Channel Distortion Robust ASR System for DARPA SPINE2 Task (Special Issue on Speech Information Processing)
• A Study on Acoustic Modeling of Pauses for Recognizing Noisy Conversational Speech (Special Issue on Speech Information Processing)
• A-16-24 3D Sound Wave Field Representation Based on Ray-Space Method(A-16. マルチメディア・仮想環境基礎, 基礎・境界)
• AURORA-2J: An Evaluation Framework for Japanese Noisy Speech Recognition(Speech Corpora and Related Topics, Corpus-Based Speech Technologies)
• Missing Feature Theory Applied to Robust Speech Recognition over IP Network(Speech Dynamics by Ear, Eye, Mouth and Machine)
• CENSREC-3: An Evaluation Framework for Japanese Speech Recognition in Real Car-Driving Environments(Speech and Hearing)
• A Design for a Collaborative Steering System of Microphone Array and Video Camera Toward Multi-Lingual Tele-Conference (特集 インタラクション技術の革新と実用化)
• A design of adaptive beamformer based on average speech spectrum for noisy speech recognition
• A Microphone Array-Based 3-D N-Best Search Method for Recognizing Multiple Sound Sources
• The present status, progress, and usage of speech databases in Japan
• An Objective Method for Evaluating Speech Translation System : Using a Second Language Learner's Corpus(Speech Corpora and Related Topics, Corpus-Based Speech Technologies)
• IMPROVING ACCURACY IN PARAMETER ESTIMATION IN AN EXTENDED KALMAN PARTICLE FILTERS FOR NOISY SPEECH RECOGNITION
• ATR Parallel Decoding Based Speech Recognition System Robust to Noise and Speaking Styles(Speech Recognition, Statistical Modeling for Speech Processing)
• Construction of Audio-Visual Speech Corpus Using Motion-Capture System and Corpus Based Facial Animation(Life-like Agent and its Communication)
• Monophthongal and diphthongal evidence of isomorphism between formant and cepstral spaces
• The effect of group delay spectrum on timbre
• Passive hybrid subtractive beamformer for near-field sound sources
• An Acoustic Modeling Method Robustagainst Changes of Speaking Stylein Error Recovery
• A Hybrid HMM/BN Acoustic Model Utilizing Pentaphone-Context Dependency(Speech Recognition, Statistical Modeling for Speech Processing)
• Improving Acoustic Model Precision by Incorporating a Wide Phonetic Context Based on a Bayesian Framework(Speech Recognition, Statistical Modeling for Speech Processing)
• A Hybrid HMM/BN Acoustic Model for Automatic Speech Recognition (Special Issue on Speech Information Processing)
• MIXTURE OF FACTOR ANALYZED HMM
• Iterative Estimation and Compensation of Signal Direction for Moving Sound Source by Mobile Microphone Array(Engineering Acoustics)
• TIME-VARYING NOISE COMPENSATION BY SEQUENTIAL MONTE CARLO METHOD
• Burst Error Recovery for Huffman Coding(Algorithm Theory)
• Audio-Visual Speech Recognition Based on Optimized Product HMMs and GMM Based-MCE-GPD Stream Weight Estimation (Special Issue on Speech Information Processing)

もっと見る 閉じる

スポンサーリンク