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Hongmei Ai

dblp:99/5695 · DBLP profile ↗
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2ranked-venue papers
0as first author
0since 2021 · last 2003
—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

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 · 87% Image and video coding · 13%

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

TopicWeightPapersLastEvidence papers
Audio and music processing
audio coding
0.012003
High-fidelity multichannel audio coding with Karhunen-Loeve transform · IEEE Trans. Speech Audio Process. 2003
Audio and music processing › audio coding
multichannel audio coding
0.012003
High-fidelity multichannel audio coding with Karhunen-Loeve transform · IEEE Trans. Speech Audio Process. 2003
Image and video coding
transform coding
0.012003
High-fidelity multichannel audio coding with Karhunen-Loeve transform · IEEE Trans. Speech Audio Process. 2003

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

karhunen-loeve transform · 0.0channel transmission control · 0.0MPEG-2 AAC · 0.0
YearPublicationVenuePosition
2003 High-fidelity multichannel audio coding with Karhunen-Loeve transform
abstract
A new quality-scalable high-fidelity multichannel audio compression algorithm based on MPEG-2 advanced audio coding (AAC) is presented. The Karhunen-Loeve transform (KLT) is applied to multichannel audio signals in the preprocessing stage to remove interchannel redundancy. Then, signals in decorrelated channels are compressed by a modified AAC main profile encoder. Finally, a channel transmission control mechanism is used to re-organize the bitstream so that the multichannel audio bitstream has a quality scalable property when it is transmitted over a heterogeneous network. Experimental results show that, compared with AAC, the proposed algorithm achieves a better performance while maintaining a similar computational complexity at the regular bit rate of 64 kbit/sec/ch. When the bitstream is transmitted to narrowband end users at a lower bit rate, packets in some channels can be dropped, and slightly degraded, yet full-channel, audio can still be reconstructed in a reasonable fashion without any additional computational cost.
Dai Yang, Hongmei Ai, Chris Kyriakakis, C.-C. Jay Kuo
IEEE Trans. Speech Audio Process.2
2002 Progressive multichannel audio codec (PMAC) with rich features
abstract
MPEG Advanced Audio Coding (AAC) is one of the most distinguished multichannel digital audio compression systems. Based on AAC, we develop a progressive syntax-rich multichannel audio codec in this work. It not only supports fine grain bit rate scalability for multichannel audio bitstreams but also provides several desirable functionalities. MPEG-4 version-2 audio coding supports fine grain bit rate scalablility in its Generic Audio Coder with the Bit-Sliced Arithmetic Coding (BSAC) tool that provides scalability for mono or stereo audio, but not for multichannel audio. A formal subjective listening test shows that, the proposed algorithm achieves a better performance at several different bit rates when compared with MPEG-4 BSAC for mono audio sources.
Dai Yang, Hongmei Ai, C.-C. Jay Kuo
ICASSP2