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Veljko Miljanic

dblp:315/0786 · DBLP profile ↗
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2ranked-venue papers
0as first author
2since 2021 · last 2023
—ORCID · none

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

Artificial intelligence and machine learning · 1 · 1 since 2021Graphics, computer vision, multimedia, augmented reality and games · 1 · 1 since 2021

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 · 67% Language models and text generation · 33%

Topics — the 3 heaviest of 3, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis
automatic speech recognition
0.612022
Towards Contextual Spelling Correction for Customization of End-to-End Speech Recognition Systems · IEEE ACM Trans. Audio Speech Lang. Process. 2022
Natural language and speech › Speech recognition and synthesis › automatic speech recognition
end-to-end speech recognition
0.612022
Towards Contextual Spelling Correction for Customization of End-to-End Speech Recognition Systems · IEEE ACM Trans. Audio Speech Lang. Process. 2022
Natural language and speech › Language models and text generation › text correction
spelling correction
0.612022
Towards Contextual Spelling Correction for Customization of End-to-End Speech Recognition Systems · IEEE ACM Trans. Audio Speech Lang. Process. 2022

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

sequence-to-sequence model · 0.6non-autoregressive decoding · 0.6context encoder · 0.6autoregressive decoding · 0.6
YearPublicationVenuePosition
2023 Fast and Accurate Factorized Neural Transducer for Text Adaption of End-to-End Speech Recognition Models
abstract
Neural transducer is now the most popular end-to-end model for speech recognition, due to its naturally streaming ability. However, it is challenging to adapt it with text-only data. Factorized neural transducer (FNT) model was proposed to mitigate this problem. The improved adaptation ability of FNT on text-only adaptation data came at the cost of lowered accuracy compared to the standard neural transducer model. We propose several methods to improve the performance of the FNT model. They are: adding CTC criterion during training, adding KL divergence loss during adaptation, using a pre-trained language model to seed the vocabulary predictor, and an efficient adaptation approach by interpolating the vocabulary predictor with the n-gram language model. A combination of these approaches results in a relative word-error-rate reduction of 9.48% from the standard FNT model. Furthermore, n-gram interpolation with the vocabulary predictor improves the adaptation speed hugely with satisfactory adaptation performance.
Rui Zhao 0017, Partha Parthasarathy, Veljko Miljanic, Jinyu Li 0001
ICASSP4
2022 Towards Contextual Spelling Correction for Customization of End-to-End Speech Recognition Systems
abstract
Contextual biasing is an important and challenging task for end-to-end automatic speech recognition (ASR) systems, which aims to achieve better recognition performance by biasing the ASR system to particular context phrases such as person names, music list, proper nouns, etc. Existing methods mainly include contextual LM biasing and adding bias encoder into end-to-end ASR models. In this work, we introduce a novel approach to do contextual biasing by adding a contextual spelling correction model on top of the end-to-end ASR system. We incorporate contextual information into a sequence-to-sequence spelling correction model with a shared context encoder. The proposed model includes two different mechanisms: autoregressive (AR) and non-autoregressive (NAR). We also propose filtering algorithms to handle large-size context lists, and performance balancing mechanisms to control the biasing degree of the model. The proposed model is a general biasing solution which is domain-insensitive and can be adopted in different scenarios. Experiments show that the proposed method achieves as much as 51% relative word error rate (WER) reduction over ASR system and outperforms traditional biasing methods. Compared to the AR solution, the NAR model reduces model size by 43.2% and speeds up inference by 2.1 times.
Xiaoqiang Wang 0006, Jinyu Li 0001, Veljko Miljanic, Sheng Zhao 0002, Hosam Khalil
IEEE ACM Trans. Audio Speech Lang. Process.4