dc.contributor.advisor | Gonzalez, Ruben | |
dc.contributor.author | Marks, Stuart Keith | |
dc.date.accessioned | 2018-01-23T02:54:25Z | |
dc.date.available | 2018-01-23T02:54:25Z | |
dc.date.issued | 2006 | |
dc.identifier.doi | 10.25904/1912/1743 | |
dc.identifier.uri | http://hdl.handle.net/10072/367550 | |
dc.description.abstract | This thesis investigates mechanisms for providing audio streaming on portable devices over the Wireless Internet. This is no trivial task, as Wireless Internet channels are highly erratic and experience highly variable bandwidth and large bursty error rates; and portable devices have limited storage and computational power. To provide an audio stream of suitable performance highly adaptive mechanisms must be utilised within session layer protocols. The core of this thesis investigates the use of a joint/source channel coder which streams the audio as a set of autonomous audio objects. This is a considerable shift away from the traditional frame-based streaming paradigm, but is warranted as such a coder has the flexibility to accommodate large error bursts and bandwidth variations while producing decoded audio with respectable perceived quality. The goal is to provide an audio coder that is able to stream audio over a channel with low bandwidth and high-error rates. Such an audio coder does not currently exist.
The majority of the material presented in this thesis is devoted to the encoding of audio into a set of autonomous objects. The approach taken is to build high-level objects from mid-level items generated from sinusoidal, transient and noise modelling. The high-level objects are autonomous, flexible, have a low bitrate and a high perceptual-relevance. The penultimate chapter of this thesis demonstrates that the streaming of these autonomous audio objects is superior to the streaming of audio in the traditional frame-based manner. As object-based audio streaming can both automatically mask the effects of packet-loss without the need for an expensive error-concealment scheme at the decoder, and scale bitrate, quality, complexity and memory in a graceful and natural manner. | |
dc.language | English | |
dc.publisher | Griffith University | |
dc.publisher.place | Brisbane | |
dc.rights.copyright | The author owns the copyright in this thesis, unless stated otherwise. | |
dc.subject.keywords | Audio streaming | |
dc.subject.keywords | wireless internet channels | |
dc.subject.keywords | joint/source channel coder | |
dc.subject.keywords | frame-based streaming paradigm | |
dc.title | Joint Source/Channel Coding for Mobile Audio Streaming | |
dc.type | Griffith thesis | |
gro.rights.copyright | The author owns the copyright in this thesis, unless stated otherwise. | |
gro.hasfulltext | Full Text | |
dc.contributor.otheradvisor | Melih, Kathy | |
dc.rights.accessRights | Public | |
gro.identifier.gurtID | gu1315971544868 | |
gro.identifier.ADTnumber | adt-QGU20070228.161413 | |
gro.source.ADTshelfno | ADT0488 | |
gro.source.GURTshelfno | GURT | |
gro.thesis.degreelevel | Thesis (PhD Doctorate) | |
gro.thesis.degreeprogram | Doctor of Philosophy (PhD) | |
gro.department | School of Information and Communication Technology | |
gro.griffith.author | Marks, Stuart K. | |