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Neethu Mariam Joy

dblp:160/8908 · DBLP profile ↗
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9ranked-venue papers
5as first author
0since 2021 · last 2018
0000-0003-2460-5189ORCID · corroborated

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

Graphics, computer vision, multimedia, augmented reality and games · 8 · 4 first-authorArtificial intelligence and machine learning · 7 · 4 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.

Computer graphics and multimedia
1 paper
Audio and music processing · 100%

Topics — the 4 heaviest of 4, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing › speech recognition
speaker normalization
0.312018
FMLLR Speaker Normalization With i-Vector: In Pseudo-FMLLR and Distillation Framework · IEEE ACM Trans. Audio Speech Lang. Process. 2018
Audio and music processing
speech processing
0.312018
FMLLR Speaker Normalization With i-Vector: In Pseudo-FMLLR and Distillation Framework · IEEE ACM Trans. Audio Speech Lang. Process. 2018
Audio and music processing
speech recognition
0.312018
FMLLR Speaker Normalization With i-Vector: In Pseudo-FMLLR and Distillation Framework · IEEE ACM Trans. Audio Speech Lang. Process. 2018
Audio and music processing › speaker recognition › speaker verification
i-vector
0.112018
FMLLR Speaker Normalization With i-Vector: In Pseudo-FMLLR and Distillation Framework · IEEE ACM Trans. Audio Speech Lang. Process. 2018

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

knowledge distillation · 0.3i-vector · 0.3deep neural network · 0.3FMLLR · 0.3
YearPublicationVenuePosition
2018 FMLLR Speaker Normalization With i-Vector: In Pseudo-FMLLR and Distillation Framework
abstract
When an automatic speech recognition (ASR) system is deployed for real-world applications, it often receives only one utterance at a time for decoding. This single utterance could be of short duration depending on the ASR task. In these cases, robust estimation of speaker normalizing methods like feature-space maximum likelihood linear regression (FMLLR) and i-vectors may not be feasible. In this paper, we propose two unsupervised speaker normalization techniques-one at feature level and other at model level of acoustic modeling-to overcome the drawbacks of FMLLR and i-vectors in real-time scenarios. At feature level, we propose the use of deep neural networks (DNN) to generate pseudo-FMLLR features from time-synchronous pair of filterbank and FMLLR features. These pseudo-FMLLR features can then be used for DNN acoustic model training and decoding. At model level, we propose a generalized distillation framework, where a teacher DNN trained on FMLLR features guides the training and optimization of a student DNN trained on filterbank features. In both the proposed methods, the ambiguity in choosing the speaker-specific FMLLR transform can be reduced by augmenting i-vectors to the input filterbank features. Experiments conducted on 33-h and 110-h subsets of Switchboard corpus show that the proposed methods provide significant gains over DNNs trained on FMLLR, i-vector appended FMLLR, filterbank and i -vector appended filterbank features, in real-time scenario.
Neethu Mariam Joy, Sandeep Kothinti, Srinivasan Umesh
IEEE ACM Trans. Audio Speech Lang. Process.1
2017 Joint Estimation of Articulatory Features and Acoustic Models for Low-Resource Languages
Basil Abraham, Srinivasan Umesh, Neethu Mariam Joy
INTERSPEECH3
2017 Generalized Distillation Framework for Speaker Normalization
Neethu Mariam Joy, Sandeep Kothinti, Srinivasan Umesh, Basil Abraham
INTERSPEECH1
2017 On Improving Acoustic Models for TORGO Dysarthric Speech Database
Neethu Mariam Joy, Srinivasan Umesh, Basil Abraham
INTERSPEECH1
2017 DNNs for unsupervised extraction of pseudo speaker-normalized features without explicit adaptation data
Neethu Mariam Joy, Murali Karthick Baskar, Srinivasan Umesh
Speech Commun.1
2016 Articulatory Feature Extraction Using CTC to Build Articulatory Classifiers Without Forced Frame Alignments for Speech Recognition
Basil Abraham, Srinivasan Umesh, Neethu Mariam Joy
INTERSPEECH3
2016 Overcoming Data Sparsity in Acoustic Modeling of Low-Resource Language by Borrowing Data and Model Parameters from High-Resource Languages
Basil Abraham, Srinivasan Umesh, Neethu Mariam Joy
INTERSPEECH3
2016 DNNs for Unsupervised Extraction of Pseudo FMLLR Features Without Explicit Adaptation Data
Neethu Mariam Joy, Murali Karthick Baskar, Srinivasan Umesh, Basil Abraham
INTERSPEECH1
2014 A data-driven phoneme mapping technique using interpolation vectors of phone-cluster adaptive training
abstract
One of the major problems in acoustic modeling for a low-resource language is data sparsity. In recent years, cross-lingual acoustic modeling techniques have been employed to overcome this problem. In this paper we propose multiple cross-lingual techniques to address the problem of data insufficiency. The first method, which we call as the cross-lingual phone-CAT, uses the principles of phone-cluster adaptive training (phone-CAT), where the parameters of context-dependent states are obtained by linear interpolation of monophone cluster models. The second method uses the interpolation vectors of phone-CAT, which is known to capture the phonetic context information, to map phonemes between two languages. Finally, the data-driven phoneme-mapping technique is incorporated into the cross-lingual phone-CAT, to obtain what we call as the phoneme-mapped cross-lingual phone-CAT. The proposed techniques are employed in acoustic modeling of three Indian languages namely Bengali, Hindi and Tamil. The phoneme-mapped cross-lingual phone-CAT gave relative improvements of 15.14% for Bengali, 16.4% for Hindi and 11.3% for Tamil over the conventional cross-lingual subspace Gaussian mixture model (SGMM) in low-resource scenario.
Basil Abraham, Neethu Mariam Joy, Navneeth K. S. Umesh
SLT2