AES Paris 2016
Paper Session P11
P11 - Audio Content Management & Applications in Audio
Sunday, June 5, 14:00 — 16:30 (Room 353)
Chair:
Mark Drews, University of Stavanger - Stavanger, Norway; Norwegian Institute of Recorded Sound - Stavanger, Norway
P11-1 Development Tools for Modern Audio Codecs—Jonas Larsen, Dolby Germany GmbH - Nuremberg, Germany; Martin Wolters, Dolby Germany GmbH - Nuremberg, Germany
The Dolby Bitstream Syntax Description Language (BSDL) is a generic, XML-based language for describing the syntactical structure of compressed audio-visual streams. This paper describes how the representation of a bitstream syntax in the BSDL is used to ease the development of serialization, deserialization, and editing tools. Additionally, the formal syntax description allows realizing a range of novel analysis methods including bitstream syntax coverage measurements, detailed bitrate profiles, and the automatic generation of rich specification documentation. The approach is exemplified using the AC-4 codec.
Convention Paper 9537 (Purchase now)
P11-2 Can Bluetooth ever Replace the Wire?—Jonny McClintock, Qualcomm Technology International Ltd. - Belfast, Northern Ireland, UK
Bluetooth is widely used as a wireless connection for audio applications including mobile phones, media players, and wearables, removing the need for cables. The combination of the A2DP protocol and frame based codecs used in many Bluetooth stereo audio implementations have led to excessive latency and acoustic performance significantly below CD quality. This paper will cover the latest developments in Bluetooth audio connectivity that will deliver CD quality audio, or better, and low latency for video and gaming applications. These developments together with the increased battery life delivered by Bluetooth Smart could lead to the elimination of wires for many applications.
[Also a poster—see session P15-11]
Convention Paper 9538 (Purchase now)
P11-3 Deep Neural Networks for Dynamic Range Compression in Mastering Applications—Stylianos Ioannis Mimilakis, Fraunhofer Institute for Digital Media Technology (IDMT) - Ilmenau, Germany; Konstantinos Drossos, Tampere University of Technology - Tampere, Finland; Tuomas Virtanen, Tampere University of Technology - Tampere, Finland; Gerald Schuller, Ilmenau University of Technology - IImenau, Germany; Fraunhofer Institute for Digital Media technology (IDMT) - Ilmenau, Germany
The process of audio mastering often, if not always, includes various audio signal processing techniques such as frequency equalization and dynamic range compression. With respect to the genre and style of the audio content, the parameters of these techniques are controlled by a mastering engineer, in order to process the original audio material. This operation relies on musical and perceptually pleasing facets of the perceived acoustic characteristics, transmitted from the audio material under the mastering process. Modeling such dynamic operations, which involve adaptation regarding the audio content, becomes vital in automated applications since it significantly affects the overall performance. In this work we present a system capable of modelling such behavior focusing on the automatic dynamic range compression. It predicts frequency coefficients that allow the dynamic range compression, via a trained deep neural network, and applies them to unmastered audio signal served as input. Both dynamic range compression and the prediction of the corresponding frequency coefficients take place inside the time-frequency domain, using magnitude spectra acquired from a critical band filter bank, similar to humans’ peripheral auditory system. Results from conducted listening tests, incorporating professional music producers and audio mastering engineers, demonstrate on average an equivalent performance compared to professionally mastered audio content. Improvements were also observed when compared to relevant and commercial software.
Also a poster—see session P15-9]
Convention Paper 9539 (Purchase now)
P11-4 Principles of Control Protocol Design and Implementation—Andrew Eales, Rhodes University - Grahamstown, South Africa; Richard Foss, Rhodes University - Grahamstown, Eastern Cape, South Africa
Control protocols are used within audio networks to manage both audio streams and networked audio devices. A number of control protocols for audio devices have been recently developed, including the AES standards AES64-2012 and AES70-2015. Despite these developments, an ontology of control protocol design and implementation does not exist. This paper proposes design and implementation heuristics for control protocols. Different categories of control protocol design and implementation heuristics are presented and the implications of individual heuristics are discussed. These heuristics allow the features provided by different control protocols to be compared and evaluated and provide guidelines for future control protocol development.
Convention Paper 9540 (Purchase now)
P11-5 Absorption Materials in Reflex Loudspeakers—Juha Backman, Genelec Oy - Iisalmi, Finland; Microsoft Mobile - Espoo, Finland
It is well known that the placement of absorbent material has an effect on the behavior of ported (reflex) enclosures, even if the acoustic solution of the field inside the enclosure would predict that the pressure field is quite homogeneous and that the flow velocities in the acoustic field are small. A CFD model is used to study this phenomenon, and the results indicate that there is strong vortex formation inside an unlined enclosure even at small volume velocities, and that the presence and the distribution of porous material has a strong effect on these vortices.
Convention Paper 9541 (Purchase now)
