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Lunji Qiu

dblp:77/11354 · DBLP profile ↗
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4ranked-venue papers
2as first author
0since 2021 · last 2002
—ORCID · none

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

Graphics, computer vision, multimedia, augmented reality and games · 3 · 2 first-authorArtificial intelligence and machine learning · 1

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
2002 A new parameterization for designing stable adaptive IIR filters
abstract
Stability is one of the most important issues in designing recursive digital adaptive filters. It is well known that the conventional transfer coefficient (or direct-form) Parameterization yields a simple implementation but suffers from potential instability problems. In this paper, based on an input-balanced realization a new adaptive filter structure is developed. It is shown that this new Parameterization has some nice numerical properties and possesses very good convergence behavior. More importantly, with this Parameterization the stability can be ensured easily and no stability monitoring is needed.
Lunji Qiu
ICASSP2
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.2
1995 Fundamental frequency determination based on instantaneous frequency estimation
Lunji Qiu, Haiyun Yang, Soo-Ngee Koh
Signal Process.1
1993 The relationship between a time-varying model of non-stationary signals and its Wigner distribution
Lunji Qiu, Ah Chung Tsoi
Signal Process.1