Maximum Likelihood Successive State Splitting Algorithm for Tied-Mixture HMnet
スポンサーリンク
概要
- 論文の詳細を見る
This paper shows how a divisive state clustering algorithm that generates acoustic Hidden Markov models(HMM)can benefit from a tied-mixture representation of the probability density function(pdf)of a state and increase the recognition performance. Popular decision tree based clustering algorthms, like for example the Successive State Splitting algorithm(SSS)make use of a simplification when clustering data. They represent a state using a single Gaussian pdf. We show that this approximation of the true pdf by a single Gaussian is too coarse, for example a single Gaussian cannot represent the differences in the symmetric parts of the pdf's of the new hypothetical states generated when evaluating the state split gain(which will determine the state split). The use of more sophisticated representations would lead to intractable computational problems that we solve by using a tied-mixture pdf representation. Additionally, we constrain the codebook to be immutable during the split. Between state spilits, this constraint is relaxed and the codebook is updated. In this paper, we thus propose an extension to the SSS algorithm, the so-called Tied-mixture Successive State Splitting algorithm(TM-SSS). TM-SSS shows up to about 31% error reduction in comparison with Maximum-Likelihood Successive State Split algorithm(ML-SSS)for a word recognition experiment.
- 社団法人電子情報通信学会の論文
- 2000-10-25
著者
-
Shikano K
Chiba University And National Institute Of Information And Communications Technology
-
Singer Harald
The Author Is With Speech Works International
-
NAKAMURA Satoshi
the ATR Spoken Language Communication Research Labs.
-
Shikano Kiyohiro
The Authors Are With Nara Institute Of Science And Technology
-
GIRARDI Alexandre
The author is with Lernout & Hauspie Speech Products
-
Girardi Alexandre
The Author Is With Lernout & Hauspie Speech Products
-
Nakamura Satoshi
The Atr Spoken Language Communication Research Laboratories
-
Nakamura Satoshi
The Authors Are With Nara Institute Of Science And Technology
関連論文
- EA2010-24 Development of real-time audio localization control system
- Sound reproduction based on multi-channel inverse filtering and WFS
- Building an Effective Speech Corpus by Utilizing Statistical Multidimensional Scaling Method
- Cost Reduction of Acoustic Modeling for Real-Environment Applications Using Unsupervised and Selective Training
- Reducing Computation Time of the Rapid Unsupervised Speaker Adaptation Based on HMM-Sufficient Statistics(Speech and Hearing)
- Improving Rapid Unsupervised Speaker Adaptation Based on HMM-Sufficient Statistics in Noisy Environments Using Multi-Template Models(Speech Recognition, Statistical Modeling for Speech Processing)
- Utterance-Based Selective Training for the Automatic Creation of Task-Dependent Acoustic Models(Speech Recognition, Statistical Modeling for Speech Processing)
- Designing Target Cost Function Based on Prosody of Speech Database(Speech Synthesis and Prosody, Corpus-Based Speech Technologies)
- Cross-language Voice Conversion Evaluation Using Bilingual Databases (特集 音声言語情報処理とその応用)
- 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)
- Effect of Central Limit Theorem non-compliance on blind separation of speech by negentropy maximization
- Blind Separation of Speech by Fixed-Point ICA with Source Adaptive Negentropy Approximation(Blind Source Separation, Multi-channel Acoustic Signal Processing)
- Robots that can hear, understand and talk
- Probability Distribution of Time-Series of Speech Spectral Components(Audio/Speech Coding)(Applications and Implementations of Digital Signal Processing)
- Using Hybrid HMM/BN Acoustic Models : Design and Implementation Issues(Speech Recognition, Statistical Modeling for Speech Processing)
- A Microphone Array-Based 3-D N-Best Search Method for Recognizing Multiple Sound Sources
- 3D N-best 探索法に基づく複数音源の位置推定と音声認識の統合
- 複数話者の音声認識における音源方向経路間距離を用いた3-D N-best探索法の評価
- A Non-stationary Noise Suppression Method Based on Particle Filtering and Polyak Averaging(Speech Recognition, Statistical Modeling for Speech Processing)
- Non-Audible Murmur (NAM) Recognition Exploiting Adaptation Techniques
- Development and evaluation of pocket-size real-time blind source separation microphone
- Objective sound quality comparison based on higher-order statistics for nonlinear noise reduction methods (応用音響)
- Objective sound quality evaluation for combination method of beamforming and spectral subtraction (応用音響)
- Fast Convergence Blind Source Separation Using Frequency Subband Interpolation by Null Beamforming
- Rapid Compensation of Temperature Fluctuation Effect for Multichannel Sound Field Reproduction System
- Development, Long-Term Operation and Portability of a Real-Environment Speech-Oriented Guidance System
- Interface for Barge-in Free Spoken Dialogue System Using Nullspace Based Sound Field Control and Beam forming (Speech/Audio Processing, Multidimensional Signal Processing and Its Application)
- On-Line Relaxation Algorithm Applicable to Acoustic Fluctuation for Inverse Filter in Multichannel Sound Reproduction System(Sound Field Reproduction, Multi-channel Acoustic Signal Processing)
- 複数モデルを用いた十分統計量に基く教師なし話者適応における学習話者のクラス化の検討
- Iterative Inverse Filter Relaxation Algorithm for Adaptation to Acoustic Fluctuation in Sound Reproduction System
- Sound Reproduction System Including Adaptive Compensation of Temperature Fluctuation Effect for Broad-Band Sound Control(Special Section on Digital Signal Processing)
- Maximum Likelihood Successive State Splitting Algorithm for Tied-Mixture HMnet
- A Self-Generator Method for Initial Filters of SIMO-ICA Applied to Blind Separation of Binaural Sound Mixtures(Blind Source Separation, Multi-channel Acoustic Signal Processing)
- Multistage SIMO-Model-Based Blind Source Separation Combining Frequency-Domain ICA and Time-Domain ICA(Adaptive Signal Processing and Its Applications)
- Evaluation of Extremely Small Sound Source Signals Used in Speaking-Aid System with Statistical Voice Conversion
- Improvements of the One-to-Many Eigenvoice Conversion System
- Esophageal Speech Enhancement Based on Statistical Voice Conversion with Gaussian Mixture Models
- Adaptive Training for Voice Conversion Based on Eigenvoices
- Blind Separation and Deconvolution for Convolutive Mixture of Speech Combining SIMO-Model-Based ICA and Multichannel Inverse Filtering(Engineering Acoustics)
- High-Fidelity Blind Separation of Acoustic Signals Using SIMO-Model-Based Independent Component Analysis(Engineering Acoustics)
- Overdetermined Blind Separation for Real Convolutive Mixtures of Speech Based on Multistage ICA Using Subarray Processing(Speech/Acoustic Signal Processing)(Digital Signal Processing)
- Stable Learning Algorithm for Blind Separation of Temporally Correlated Acoustic Signals Combining Multistage ICA and Linear Prediction(Digital Signal Processing)
- Blind Source Separation of Acoustic Signals Based on Multistage ICA Combining Frequency-Domain ICA and Time-Domain ICA
- Fast-Convergence Algorithm for Blind Source Separation Based on Array Signal Processing
- Sound Field Reproduction by Wavefront Synthesis Using Directly Aligned Multi Point Control
- Speech Prior Estimation for Generalized Minimum Mean-Square Error Short-Time Spectral Amplitude Estimator