Show simple item record

dc.contributor.authorWang, Yonghao
dc.date.accessioned2018-09-18T16:41:19Z
dc.date.available2018-09-18T16:41:19Z
dc.date.issued2018-09-07
dc.date.submitted2018-09-18T15:55:39.698Z
dc.identifier.citationWang, Y. 2018. Low Latency Audio Processing. Queen Mary University of Londonen_US
dc.identifier.urihttp://qmro.qmul.ac.uk/xmlui/handle/123456789/44697
dc.descriptionPhDen_US
dc.description.abstractLatency in the live audio processing chain has become a concern for audio engineers and system designers because significant delays can be perceived and may affect synchronisation of signals, limit interactivity, degrade sound quality and cause acoustic feedback. In recent years, latency problems have become more severe since audio processing has become digitised, high-resolution ADCs and DACs are used, complex processing is performed, and data communication networks are used for audio signal transmission in conjunction with other traffic types. In many live audio applications, latency thresholds are bounded by human perceptions. The applications such as music ensembles and live monitoring require low delay and predictable latency. Current digital audio systems either have difficulties to achieve or have to trade-off latency with other important audio processing functionalities. This thesis investigated the fundamental causes of the latency in a modern digital audio processing system: group delay, buffering delay, and physical propagation delay and their associated system components. By studying the time-critical path of a general audio system, we focus on three main functional blocks that have the significant impact on overall latency; the high-resolution digital filters in sigma-delta based ADC/DAC, the operating system to process low latency audio streams, and the audio networking to transmit audio with flexibility and convergence. In this work, we formed new theory and methods to reduce latency and accurately predict latency for group delay. We proposed new scheduling algorithms for the operating system that is suitable for low latency audio processing. We designed a new system architecture and new protocols to produce deterministic networking components that can contribute the overall timing assurance and predictability of live audio processing. The results are validated by simulations and experimental tests. Also, this bottom-up approach is aligned with the methodology that could solve the timing problem of general cyber-physical systems that require the integration of communication, software and human interactions.en_US
dc.language.isoenen_US
dc.publisherQueen Mary University of Londonen_US
dc.rightsThe copyright of this thesis rests with the author and no quotation from it or information derived from it may be published without the prior written consent of the author
dc.subjectElectronic Engineering and Computer Scienceen_US
dc.subjectlive audio processingen_US
dc.subjectlatency problemsen_US
dc.titleLow Latency Audio Processingen_US
dc.typeThesisen_US


Files in this item

Thumbnail

This item appears in the following Collection(s)

  • Theses [4235]
    Theses Awarded by Queen Mary University of London

Show simple item record