The generation of panning laws for irregular speaker arrays using heuristic methods

Currently, the ITU standard surround sound speaker arrangement is based on an irregular 5 speaker array. However, this may change to an irregular 7 speaker array (as is now the standard on computer hardware) or more in the future. The Ambisonic system, pioneered by Micheal Gerzon, among others, in the late 1960’s, is very well suited to situations where the end system speaker configuration is not fixed in terms of number or position while also offering a simple way (via energy and velocity vector analysis) of quantifying the performance of such systems. However, while the derivation of the decoders is well documented for regular speaker arrangements [1], optimising the decoders for irregular layouts is not a simple task, where optimisation requires the solution of a set of non linear simultaneous equations, complicated further by the fact that multiple solutions are possible [2]. Craven [3] extended the system to use higher order circular harmonics and presented a 4th order Ambisonic decoder (9 input channels), although the derivation method used was not presented. In this paper a semi-automated decoder optimisation system using heuristic methods will be presented that will be shown to be robust enough to generate higher order Ambisonic decoders based on the energy and velocity vector parameters. This method is then analytically compared to Craven’s decoder using both energy/velocity vector and head related transfer function based methods.EPSR

Let's mix it up: interviews exploring the practical and technical challenges of interactive mixing in games

Game audio has come a long way since the simple electronic beeps of the early 1970s, when significant technical constraints governed the scope of creative possibilities. Recent years have witnessed technological advancements on an unprecedented scale; no sooner is one technology introduced than it is superseded by another, boasting a range of new refinements and enhanced performance

Loudness measurement of multitrack audio content using modifcations of ITU-R BS.1770

The recent loudness measurement recommendations by the ITU and the EBU have gained widespread recognition in the broadcast community. The material it deals with is usually full-range mastered audio content, and its applicability to multitrack material is not yet clear. In the present work we investigate how well the evaluated perception of single track loudness agrees with the measured value as de ned by ITU-R BS.1770. We analyze the underlying features that may be the cause for this disparity and propose some parameter alterations that might yield better results for multitrack material with minimal modi cation to their rating of broadcast content. The best parameter sets are then evaluated by a panel of experts in terms of how well they produce an equal-loudness multitrack mix, and are shown to be signi cantly more successful

A differential fault attack, key representation and mapping tables for the advanced encryption standard

In 1997, the National Institute of Standards and Technology (NIST) announced an open competition for an Advanced Encryption Standard (AES), a symmetric-key cryptographic algorithm, to replace the ageing DES and to be able to run on 128-bit data blocks, using 128-bit, 192-bit or 256-bit keys. After five years of intense peer review, attacks and discussions, the Rijndael algorithm was declared the AES. Being used by the entire world for most of the encryption, it has to be subjected to every possible analysis to find new vulnerabilities. This thesis discusses three new features of the cipher. A powerful attack is initially formulated, which exploits cipher properties in its final and penultimate rounds under applications of precise bit/byte tweaking mechanisms. Next, the Key Schedule of the cipher is simplistically represented and a few interesting properties are noted. Finally, the AES is represented as a set of simple mapping tables and the relevance of such a representation is addressed

Generalization of a 3-D resonator model for the simulation of spherical enclosures

A rectangular enclosure has such an even distribution of resonances that it can be accurately and efficiently modelled using a feedback delay network. Conversely, a non rectangular shape such as a sphere has a distribution of resonances that challenges the construction of an efficient model. This work proposes an extension of the already known feedback delay network structure to model the resonant properties of a sphere. A specific frequency distribution of resonances can be approximated, up to a certain frequency, by inserting an allpass filter of moderate order after each delay line of a feedback delay network. The structure used for rectangular boxes is therefore augmented with a set of allpass filters allowing parametric control over the enclosure size and the boundary properties. This work was motivated by informal listening tests which have shown that it is possible to identify a basic shape just from the distribution of its audible resonances.Comment: 39 pages, 16 figures, 6 tables. Accepted for publication in Applied Signal Processin