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Ling Kok Ng

dblp:90/11353 · DBLP profile ↗
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2ranked-venue papers
1as first author
0since 2021 · last 2000
—ORCID · none

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

Artificial intelligence and machine learning · 1Graphics, 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.

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

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

TopicWeightPapersLastEvidence papers
Audio and music processing › speech processing
speech analysis and synthesis
0.012000
Signal representation based on instantaneous amplitude models with application to speech synthesis · IEEE Trans. Speech Audio Process. 2000
Audio and music processing
speech synthesis
0.012000
Signal representation based on instantaneous amplitude models with application to speech synthesis · IEEE Trans. Speech Audio Process. 2000

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

optimization · 0.0
YearPublicationVenuePosition
2000 Signal representation based on instantaneous amplitude models with application to speech synthesis
abstract
In this correspondence, a model for multicomponent signal representation is proposed. This model can avoid the difficulty in dealing with the time-varying phases and allow us to carry out an optimization procedure easily such that the synthetic signal can be made as close to the original one as possible. This model is successfully applied to speech analysis/synthesis.
Lunji Qiu, Ling Kok Ng
IEEE Trans. Speech Audio Process.3
1999 Wideband speech coding with toll quality based on IA-model
abstract
We propose an instantaneous amplitude (IA) based model for speech signal representation. This can avoid the difficulty in dealing with the time-varying phases and allows us to perform an optimization procedure easily such that the synthetic signal can be made as close to the original one as possible. A simplified frequency picking algorithm is derived to shorten the processing time while still maintaining the quality of the synthetic speech. Experiments show that the synthetic speech with the developed technique is of toll quality and almost perceptually indistinguishable from the original speech. Initial work on the coding of the parameters, for a 16 kHz sampled speech, for the IA model is done and a toll quality synthesized speech at a bit rate of 40 kbps is achieved.
Ling Kok Ng, Gang Li 0010, Guoan Bi
ICASSP1