Browsing by Author "Tseng, KK"

Now showing 1 - 3 of 3

A parallel automaton string matching with pre-hashing and root-indexing techniques for content filtering coprocessor
(2005)
Tseng, KK
;
Lin, YD
;
Lee, TH
;
Lai, YC
;
交大名義發表
We propose a new Parallel Automaton string matching approach and its hardware architecture for content filtering coprocessor. This new approach can improve the average matching time of the Parallel Automaton with Pre-Hashing and Root-Indexing techniques. The Pre-Hashing technique uses a hashing function to verify quickly the text against the partial patterns in the Automaton, and the Root-Indexing technique matches multiple bytes for the root state in one single matching. A popular Automaton algorithm, Aho-Corasick (AC) is chosen to be implemented by adding the two techniques; we employ these two techniques in a memory efficient version of AC namely Bilmap AC For the average-case time, our approach improves Bitmap AC by 494% and 224% speedup for URL and Virus patterns, respectively. Since Pre-Hashing and Root-Indexing techniques can be concurrently executed with Bitmap AC in the hardware, our proposed approach has the same worst-case time as Bitmap AC.
Perceptual codec and interaction aware playout algorithms and quality measurement for VoIP systems
(2004-02-01)
Tseng, KK
;
Lai, YC
;
Lin, YD
;
資訊工程學系
;
Department of Computer Science
To reduce the effect of network jitter, the playout algorithm for voice streams should correctly adjust the playout delay. Conventional playout algorithms were based on network delay only; they did not consider the perceptual quality, and were not aware the codec and interactive mode. Therefore, we present two novel approaches: codec aware adaptive playout (CAAP) algorithm and interactive aware adaptive playout (IAAP) algorithm, which intent to optimize the voice quality based on codec and interactive mode respectively. The performance of CAAP and IAAP are superior to the prior algorithms in our substantial evaluation. Since no objective mechanisms for measuring the speech quality of two-way communication exist, we also propose a new quality measurement for it.
User perceived codec and duplex aware playout algorithms and LMOS-DMOS measurement for real time streams
(2003)
Tseng, KK
;
Lin, YD
;
交大名義發表
;
National Chiao Tung University
In the network devices for real time streaming, e.g. VoIP phone, video phone and online game, the playout algorithm controls the playout delay to eliminate the jitter and to minimize the overflow packet loss. Conventional algorithms did this based on network delay only; they did not consider the user perceived quality, and aware the codec and communication duplex mode. Therefore, we present two novel approaches: codec aware adaptive playout (CAAP) and duplex aware adaptive playout (DAAP), they intent to optimize the user perceived quality based on codec and communication duplex mode respectively. Because of their different characteristics, CAAP and DAAP can work either alone or together; and their improvements are accumulative. The out-performance of CAAP and DAAP are superior the prior algorithms in our substantial but conservative evaluation. Since no objective mechanisms for measuring the speech quality of two-way communication exist; a new LMOS-DMOS measurement mechanism is also proposed. Although VoIP is an exemplary embodiment in this paper, we formally model our algorithms into a generalized user perceived QoS control as well.