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Duc Hoang Ha Nguyen

dblp:125/2824 · DBLP profile ↗
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2ranked-venue papers
2as first author
0since 2021 · last 2016
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 1 · 1 first-authorGraphics, computer vision, multimedia, augmented reality and games · 1 · 1 first-author

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Artificial intelligence
1 paper
Speech recognition and synthesis · 50% Transfer learning and domain adaptation · 50%

Topics — the 2 heaviest of 2, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Machine learning › Transfer learning and domain adaptation › domain adaptation
feature adaptation
0.212016
Feature Adaptation Using Linear Spectro-Temporal Transform for Robust Speech Recognition · IEEE ACM Trans. Audio Speech Lang. Process. 2016
Natural language and speech › Speech recognition and synthesis › automatic speech recognition
robust speech recognition
0.212016
Feature Adaptation Using Linear Spectro-Temporal Transform for Robust Speech Recognition · IEEE ACM Trans. Audio Speech Lang. Process. 2016

Methods — techniques the papers use, named apart from their topics

kullback-leibler divergence · 0.2fMLLR · 0.2
YearPublicationVenuePosition
2016 Feature Adaptation Using Linear Spectro-Temporal Transform for Robust Speech Recognition
abstract
Spectral information represents short-term speech information within a frame of a few tens of milliseconds, while temporal information captures the evolution of speech statistics over consecutive frames. Motivated by the findings that human speech comprehension relies on the integrity of both the spectral content and temporal envelope of speech signal, we study a spectro-temporal transform framework that adapts run-time speech features to minimize the mismatch between run-time and training data, and its implementation that includes cross transform and cascaded transform. A Kullback-Leibler divergence based cost function is proposed to estimate the transform parameters. We conducted experiments on the REVERB Challenge 2014 task, where clean and multi-condition trained acoustic models are tested with real reverberant and noisy speech. We found that temporal information is important for reverberant speech recognition and the simultaneous use of spectral and temporal information for feature adaptation is effective. We also investigate the combination of the cross transform with fMLLR, the combination of batch, utterance and speaker mode adaptation, and multicondition adaptive training using proposed transforms. All experiments consistently report significant word error rate reductions.
Duc Hoang Ha Nguyen, Chng Eng Siong, Haizhou Li 0001
IEEE ACM Trans. Audio Speech Lang. Process.1
2014 Generalization of temporal filter and linear transformation for robust speech recognition
abstract
Temporal filtering of feature trajectories and linear transformation of feature vectors are two effective ways to compensate the speech features to achieve robust speech recognition in noisy and reverberant environments. In the previous studies, as the two methods are usually applied in sequence, the interaction between the two methods is not optimized. In this paper, we propose a generalized transform which integrates temporal filter and linear transformation into a single process. The new transform parameters are optimized to minimize an approximated Kullback-Leibler divergence between the distribution of the compensated features and the distribution represented by a clean reference model. The proposed method is evaluated on the Aurora-5 clean condition training task. The experiments show that the generalized transform significantly outperforms the simple cascade of temporal filtering and linear transformation. For example, the word accuracy is improved from 81.55% (cascade) to 83.99% (generalized) and from 72.09% (cascade) to 76.04% (generalized) for office and living room environments, respectively, in speaker based feature adaptation scheme.
Duc Hoang Ha Nguyen, Chng Eng Siong, Haizhou Li 0001
ICASSP1