• AmbiFreeVerb 2—Development of a 3D ambisonic reverb with spatial warping and variable scattering

      Wiggins, Bruce; Dring, Mark; University of Derby (Audio Engineering Society, 2016-07-14)
      In this paper the development of a three dimensional Ambisonic reverb based on the open source FreeVerb algorithm will be presented and discussed. This model is then extended to include processing in over-specified A-format, rather than B-format, variable scattering between channels along with controls for warping the distribution of the reflections to implement a reverb that is able to react to the source position in a spatially coherent way with an acoustical analysis of its performance.
    • Distance coding and performance of the mark 5 and st350 SoundField microphones and their suitability for Ambisonic reproduction.

      Wiggins, Bruce; Spenceley, Thomas; University of Derby (Institute of Acoustics, 2009-11-19)
      Capturing and replaying distance cues for multi-channel audio is currently an under-explored and under-exploited area. Panners that successfully give control of distance do not, currently exist. However, recordings made with 1st order ambisonic, Soundfield microphones replayed over an ambisonic rig can give realistic results with respect to distance perception (particularly when bringing sound sources inside the speaker array). Near-field effect, resulting from the wave front curvature of near-field sources, is one cue recorded by the microphone, but not reproduced by software or hardware panners. Papers by Daniel (2003, 2004) discuss the encoding and decoding of ambisonic material with particular reference to higher-order ambisonics, and describe ‘near-field coding’ filters which encode near-field effect while pre-compensating for finite loudspeaker reproduction distance. While existing research concentrates on its simulation, this report documents an investigation into near-field effect in Soundfield ST350 and MK V tetrahedral microphones. It is found that, as a result of calibration for a flat frequency response at a practical source distance, the Soundfield microphone responses bear strong similarity to various near-field coding filters, suggesting the existence of an optimum loudspeaker array radius for positional localisation. On determination of this distance, recordings may be adapted for proper reproduction at any chosen reference distance using the WigWare ambisonic plug-ins created at the University of Derby.
    • The generation of panning laws for irregular speaker arrays using heuristic methods

      Wiggins, Bruce; University of Derby (Audio Engineering Society, 2007-06-25)
      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.
    • Has Ambisonics come of age?

      Wiggins, Bruce; University of Derby (Institute of Acoustics, 2008-11)
      Ambisonics was developed in the 1970’s as a flexible, psycho-acoustically aware system1. Developed at the same time as Quadraphonics2, Ambisonics is an often mis-understood system that was far ahead of it’s time. Due to the ubiquity of surround sound equipment in modern computers and interest in live surround events becoming more widespread, is the time, finally, right for Ambisonics to come into its’ own? In this paper, the definition of what makes a system Ambisonic is clarified with reference made to the traditional energy and velocity vector theory, higher order systems and use in both the live and domestic environment. More recent developments by the author are discussed with respect to irregular Ambisonic decoder design (such as for the ITU 5.1 speaker array) and analysis using Head Related Transfer Function data showing the extra insight this can give into the performance of one, seemingly similar, decoder design over another. The freely available suite of VST plug-ins (comprising of decoders, panners and an Ambisonic reverb) created using this technology are also presented, with case studies of their use in student projects at the University of Derby.
    • An investigation into the real-time manipulation and control of three-dimensional sound fields

      Wiggins, Bruce; University of Derby (University of Derby, 2004)
      This thesis describes a system that can be used for the decoding of a three dimensional audio recording over headphones or two, or more, speakers. A literature review of psychoacoustics and a review (both historical and current) of surround sound systems is carried out. The need for a system which is platform independent is discussed, and the proposal for a system based on an amalgamation of Ambisonics, binaural and transaural reproduction schemes is given. In order for this system to function optimally, each of the three systems rely on providing the listener with the relevant psychoacoustic cues. The conversion from a five speaker ITU array to binaural decode is well documented but pair-wise panning algorithms will not produce the correct lateralisation parameters at the ears of a centrally seated listener. Although Ambisonics has been well researched, no one has, as yet, produced a psychoacoustically optimised decoder for the standard irregular five speaker array as specified by the ITU as the original theory, as proposed by Gerzon and Barton (1992) was produced (known as a Vienna decoder), and example solutions given, before the standard had been decided on. In this work, the original work by Gerzon and Barton (1992) is analysed, and shown to be suboptimal, showing a high/low frequency decoder mismatch due to the method of solving the set of non-linear simultaneous equations. A method, based on the Tabu search algorithm, is applied to the Vienna decoder problem and is shown to provide superior results to those shown by Gerzon and Barton (1992) and is capable of producing multiple solutions to the Vienna decoder problem. During the write up of this report Craven (2003) has shown how 4th order circular harmonics (as used in Ambisonics) can be used to create a frequency independent panning law for the five speaker ITU array, and this report also shows how the Tabu search algorithm can be used to optimise these decoders further. A new method is then demonstrated using the Tabu search algorithm coupled with lateralisation parameters extracted from a binaural simulation of the Ambisonic system to be optimised (as these are the parameters that the Vienna system is approximating). This method can then be altered to take into account head rotations directly which have been shown as an important psychoacoustic parameter in the localisation of a sound source (Spikofski et al., 2001) and is also shown to be useful in differentiating between decoders optimised using the Tabu search form of the Vienna optimisations as no objective measure had been suggested. Optimisations for both Binaural and Transaural reproductions are then discussed so as to maximise the performance of generic HRTF data (i.e. not individualised) using inverse filtering methods, and a technique is shown that minimises the amount of frequency dependant regularisation needed when calculating cross-talk cancellation filters.