Demo Data from VLDB 2026 / reviewers in the wild / expert
Hongmei Ai
dblp:99/5695
· DBLP profile ↗
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
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing
audio coding |
0.0 | 1 | 2003 | High-fidelity multichannel audio coding with Karhunen-Loeve transform · IEEE Trans. Speech Audio Process. 2003 |
Audio and music processing › audio coding
multichannel audio coding |
0.0 | 1 | 2003 | High-fidelity multichannel audio coding with Karhunen-Loeve transform · IEEE Trans. Speech Audio Process. 2003 |
Image and video coding
transform coding |
0.0 | 1 | 2003 | 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
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2003 | High-fidelity multichannel audio coding with Karhunen-Loeve transformabstractA 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 featuresabstractMPEG 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 |
ICASSP | 2 |