Language Modeling II

Home
Full List of Titles
1: Speech Processing
CELP Coding
Large Vocabulary Recognition
Speech Analysis and Enhancement
Acoustic Modeling I
ASR Systems and Applications
Topics in Speech Coding
Speech Analysis
Low Bit Rate Speech Coding I
Robust Speech Recognition in Noisy Environments
Speaker Recognition
Acoustic Modeling II
Speech Production and Synthesis
Feature Extraction
Robust Speech Recognition and Adaptation
Low Bit Rate Speech Coding II
Speech Understanding
Language Modeling I
2: Speech Processing, Audio and Electroacoustics, and Neural Networks
Acoustic Modeling III
Lexical Issues/Search
Speech Understanding and Systems
Speech Analysis and Quantization
Utterance Verification/Acoustic Modeling
Language Modeling II
Adaptation /Normalization
Speech Enhancement
Topics in Speaker and Language Recognition
Echo Cancellation and Noise Control
Coding
Auditory Modeling, Hearing Aids and Applications of Signal Processing to Audio and Acoustics
Spatial Audio
Music Applications
Application - Pattern Recognition & Speech Processing
Theory & Neural Architecture
Signal Separation
Application - Image & Nonlinear Signal Processing
3: Signal Processing Theory & Methods I
Filter Design and Structures
Detection
Wavelets
Adaptive Filtering: Applications and Implementation
Nonlinear Signals and Systems
Time/Frequency and Time/Scale Analysis
Signal Modeling and Representation
Filterbank and Wavelet Applications
Source and Signal Separation
Filterbanks
Emerging Applications and Fast Algorithms
Frequency and Phase Estimation
Spectral Analysis and Higher Order Statistics
Signal Reconstruction
Adaptive Filter Analysis
Transforms and Statistical Estimation
Markov and Bayesian Estimation and Classification
4: Signal Processing Theory & Methods II, Design and Implementation of Signal Processing Systems, Special Sessions, and Industry Technology Tracks
System Identification, Equalization, and Noise Suppression
Parameter Estimation
Adaptive Filters: Algorithms and Performance
DSP Development Tools
VLSI Building Blocks
DSP Architectures
DSP System Design
Education
Recent Advances in Sampling Theory and Applications
Steganography: Information Embedding, Digital Watermarking, and Data Hiding
Speech Under Stress
Physics-Based Signal Processing
DSP Chips, Architectures and Implementations
DSP Tools and Rapid Prototyping
Communication Technologies
Image and Video Technologies
Automotive Applications / Industrial Signal Processing
Speech and Audio Technologies
Defense and Security Applications
Biomedical Applications
Voice and Media Processing
Adaptive Interference Cancellation
5: Communications, Sensor Array and Multichannel
Source Coding and Compression
Compression and Modulation
Channel Estimation and Equalization
Blind Multiuser Communications
Signal Processing for Communications I
CDMA and Space-Time Processing
Time-Varying Channels and Self-Recovering Receivers
Signal Processing for Communications II
Blind CDMA and Multi-Channel Equalization
Multicarrier Communications
Detection, Classification, Localization, and Tracking
Radar and Sonar Signal Processing
Array Processing: Direction Finding
Array Processing Applications I
Blind Identification, Separation, and Equalization
Antenna Arrays for Communications
Array Processing Applications II
6: Multimedia Signal Processing, Image and Multidimensional Signal Processing, Digital Signal Processing Education
Multimedia Analysis and Retrieval
Audio and Video Processing for Multimedia Applications
Advanced Techniques in Multimedia
Video Compression and Processing
Image Coding
Transform Techniques
Restoration and Estimation
Image Analysis
Object Identification and Tracking
Motion Estimation
Medical Imaging
Image and Multidimensional Signal Processing Applications I
Segmentation
Image and Multidimensional Signal Processing Applications II
Facial Recognition and Analysis
Digital Signal Processing Education

Author Index
A B C D E F G H I
J K L M N O P Q R
S T U V W X Y Z

Speech Recognition Experiments Using Multi-Span Statistical Language Models

Authors:

Jerome R Bellegarda,

Page (NA) Paper number 1748

Abstract:

A multi-span framework was recently proposed to integrate the various constraints, both local and global, that are present in the language. In this approach, local constraints are captured via n-gram language modeling, while global constraints are taken into account through the use of latent semantic analysis. The performance of the resulting multi-span language models, as measured by perplexity, has been shown to compare favorably with the corresponding n-gram performance. This paper reports on actual speech recognition experiments, and shows that word error rate is also substantially reduced. On a subset of the Wall Street Journal speaker-independent, 20,000-word vocabulary, continuous speech task, the multi-span framework resulted in a reduction in average word error rate of up to 17%.

IC991748.PDF (From Author) IC991748.PDF (Rasterized)

TOP


Spoken Language Variation over Time and State in a Natural Spoken Dialog System

Authors:

Allen L Gorin,
Giuseppe Riccardi,

Page (NA) Paper number 2045

Abstract:

We are interested in adaptive spoken dialog system for automated services. Peoples' spoken language usage varies over time for a fixed task, and furthermore varies depending on the state of the dialog. We will characterize and quantify this variation based on a database of 20K user-transactions with AT&T's experimental " How May I Help You" spoken dialog system. We then report on a language adaptation algorithm which was used to train state-dependent ASR language models, experimentally evaluating their improved performance with respect to word accuracy and perplexity.

IC992045.PDF (From Author) IC992045.PDF (Rasterized)

TOP


Comparative Evaluation Of Spoken Corpora Acquired By Presentation Of Visual Scenarios And Textual Descriptions

Authors:

Demetrio Aiello,
Cristina Delogu,
Renato De Mori,
Andrea Di Carlo,
Marina Nisi,
Silvia Tummeacciu,

Page (NA) Paper number 1526

Abstract:

The paper describes a system, in JAVA, for written and visual scenario generation used to collect speech corpora in the framework of a Tourism Information System. Experimental evidence shows that the corpus generated with visual scenarios has a higher perplexity and a richer vocabulary than the corpus generated using the same conceptual derivations to produce textual scenarios. Furthermore, there is evidence that textual scenarios influence speakers in the choice of the lexicon used to express the concepts more than visual scenarios.

IC991526.PDF (From Author) IC991526.PDF (Rasterized)

TOP


Using Smoothed K-TSS Language Models In Continuous Speech Recognition

Authors:

Amparo Varona,
Ines Torres,

Page (NA) Paper number 1907

Abstract:

A syntactic approach of the well-known N-grams models, the K-Testable Language in the Strict Sense (K-TSS), is used in this work to be integrated in a Continuous Speech Recognition (CSR) system. The use of smoothed K-TSS regular grammars allowed to obtain a deterministic Stochastic Finite State Automaton (SFSA) integrating K k-TSS models into a self-contained model. An efficient representation of the whole model in a simple array of and adequate size is proposed. This structure can be easily handled at decoding time by a simple search function through the array. This formulation strongly reduced the number of parameters to be managed and thus the computing complexity of the model. An experimental evaluation of the proposed SFSA representation was carried out over an Spanish recognition task. These experiments showed important memory saving to allocate K-TSS Language models, more important for higher values of K. They also showed that the decoding time did not meaningfully increased when K did. The lower word error rates for the Spanish task tested were achieved for K=4 and 5. As a consequence the ability of this syntactic approach of the N-grams to be well integrated in a CSR system, even for high values of K, has been established.

IC991907.PDF (From Author) IC991907.PDF (Rasterized)

TOP


Interfacing a CDG Parser with an HMM Word Recognizer Using Word Graphs

Authors:

Mary P Harper,
Michael T Johnson,
Leah H Jamieson,
Stephen A Hockema,
Christopher M White,

Page (NA) Paper number 2403

Abstract:

In this paper, we describe a prototype spoken language system that loosely integrates a speech recognition component based on hidden Markov models with a constraint dependency grammar (CDG) parser using a word graph to pass sentence candidates between the two modules. This loosely coupled system was able to improve the sentence selection accuracy and concept accuracy over the level achieved by the acoustic module with a stochastic grammar. Timing profiles suggest that a tighter coupling of the modules could reduce parsing times of the system, as could the development of better acoustic models and tighter parsing constraints for conjunctions.

IC992403.PDF (From Author) IC992403.PDF (Rasterized)

TOP


An Automatic Acquisition Method Of Statistic Finite-State Automaton For Sentences

Authors:

Motoyuki Suzuki,
Shozo Makino,
Hirotomo Aso,

Page (NA) Paper number 1925

Abstract:

Statistic language models obtained from a large number of training samples play an important role in speech recognition. In order to obtain higher recognition performance, we should introduce long distance correlations between words. However, traditional statistic language models such as word n-grams and ergodic HMMs are insufficient for expressing long distance correlations between words. In this paper, we propose an acquisition method for a language model based on HMnet taking into consideration long distance correlations and word location.

IC991925.PDF (From Author) IC991925.PDF (Rasterized)

TOP


Robust Dialogue-State Dependent Language Modeling Using Leaving-One-Out

Authors:

Frank Wessel,
Andrea Baader,

Page (NA) Paper number 1385

Abstract:

The use of dialogue-state dependent language models in automatic inquiry systems can improve speech recognition and understanding if a reasonable prediction of the dialogue-state is feasible. In this paper, the dialogue-state is defined as the set of parameters which are contained in the system prompt. For each dialogue-state a separate language model is constructed. In order to obtain robust language models despite the small amount of training data we propose to interpolate all of the dialogue-state dependent language models linearly for each dialogue-state and to train the large number of resulting interpolation weights with the EM-Algorithm in combination with Leaving-One-Out. We present experimental results on a small Dutch corpus which has been recorded in the Netherlands with a train timetable information system and show that the perplexity and the word error rate can be reduced significantly.

IC991385.PDF (From Author) IC991385.PDF (Rasterized)

TOP


On-line Algorithms for Combining Language Models

Authors:

Adam Kalai,
Stanley F. Chen,
Avrim Blum,
Ronald Rosenfeld,

Page (NA) Paper number 2175

Abstract:

Multiple language models are combined for many tasks in language modeling, such as domain and topic adaptation. In this work, we compare on-line algorithms from machine learning to existing algorithms for combining language models. On-line algorithms developed for this problem have parameters that are updated dynamically to adapt to a data set during evaluation. On-line analysis provides guarantees that these algorithms will perform nearly as well as the best model chosen in hindsight from a large class of models, e.g., the set of all static mixtures. We describe several on-line algorithms and present results comparing these techniques with existing language modeling combination methods on the task of domain adaptation. We demonstrate that, in some situations, on-line techniques can significantly outperform static mixtures (by over 10% in terms of perplexity) and are especially effective when the nature of the test data is unknown or changes over time.

IC992175.PDF (From Author) IC992175.PDF (Rasterized)

TOP