Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Implementation of a MPEG 1 layer I audio decoder with variable bit lengths

One of the most popular forms of audio compression is MPEG (Moving Picture Experts Group). By using a VHDL (Very high-speed integrated circuit Hardware Description Language) implementation of a MPEG audio decoder and varying the word length of the constants and the multiplications used in the decoding process, and comparing the error, the minimum word length required can be determined. In general, the smaller the word length, the smaller the hardware resources required. This thesis is an investigation to find the minimum bit lengths required for each of the four multiplication sections used in a MPEG Audio decoder, that will still meet the quality levels specified in the MPEG standard. The use of the minimum bit lengths allows the minimum area resources of a FPGA (Field Programmable Gate Array) to be used. A FPGA model was designed that allowed the number of bits used to represent four constants and the results of the multiplications using these constants to vary. In order to limit the amount of data generated, testing was restricted to a single channel of audio data sampled at a frequency of 32kHz. This was then compared to the supplied C model distributed with the MPEG Audio Standard. It was found that for the MPEG audio coder to be fully compliant with the standard the bit lengths of the constants and the multiplications could be reduced by 75% and to be partial compliant with the standard, the bit lengths of the constants and the multiplications could be reduced by up to 82%. An implementation of a MPEG audio decoder in VHDL has the advantage of specific hardware, optimised, for all the different complex mathematical operations thereby reducing the repetitive operations and therefore power consumption and the time required performing these complex operations

Movements in Binaural Space: Issues in HRTF Interpolation and Reverberation, with applications to Computer Music

This thesis deals broadly with the topic of Binaural Audio. After reviewing the literature, a reappraisal of the minimum-phase plus linear delay model for HRTF representation and interpolation is offered. A rigorous analysis of threshold based phase unwrapping is also performed. The results and conclusions drawn from these analyses motivate the development of two novel methods for HRTF representation and interpolation. Empirical data is used directly in a Phase Truncation method. A Functional Model for phase is used in the second method based on the psychoacoustical nature of Interaural Time Differences. Both methods are validated; most significantly, both perform better than a minimum-phase method in subjective testing. The accurate, artefact-free dynamic source processing afforded by the above methods is harnessed in a binaural reverberation model, based on an early reflection image model and Feedback Delay Network diffuse field, with accurate interaural coherence. In turn, these flexible environmental processing algorithms are used in the development of a multi-channel binaural application, which allows the audition of multi-channel setups in headphones. Both source and listener are dynamic in this paradigm. A GUI is offered for intuitive use of the application. HRTF processing is thus re-evaluated and updated after a review of accepted practice. Novel solutions are presented and validated. Binaural reverberation is recognised as a crucial tool for convincing artificial spatialisation, and is developed on similar principles. Emphasis is placed on transparency of development practices, with the aim of wider dissemination and uptake of binaural technology

An investigation into the real-time manipulation and control of three-dimensional sound fields

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.EPRS

Analysis and resynthesis of polyphonic music

This thesis examines applications of Digital Signal Processing to the analysis, transformation, and resynthesis of musical audio. First I give an overview of the human perception of music. I then examine in detail the requirements for a system that can analyse, transcribe, process, and resynthesise monaural polyphonic music. I then describe and compare the possible hardware and software platforms. After this I describe a prototype hybrid system that attempts to carry out these tasks using a method based on additive synthesis. Next I present results from its application to a variety of musical examples, and critically assess its performance and limitations. I then address these issues in the design of a second system based on Gabor wavelets. I conclude by summarising the research and outlining suggestions for future developments

Evaluation of audio source separation in the context of 3D audio

The emergence and broader availability of 3D audio systems allows for new possibilities in mixing, post-production and playback of audio content. Used in movie post-production for cinemas, as special effect by disk jockeys for example and even for live concerts, 3D rendering immerses the listener more than ever before. When existing audio material is to be employed, Audio Source Separation (ASS) techniques enable the extraction of single sources from a mixture. Modern mixing approaches for 3D audio do not assign individual gains and delays for each source in every channel. A sound scene is rather designed, with individual sources treated as objects to be placed within a scene. The hardware layer is mostly irrelevant for mixing in such a setting. ASS is therefore a valuable tool to ¿disassemble¿ amore traditional monophonic, stereophonic, or multichannel mix. However, due to the complexity of the ASS problem, extracted sources are subject to degradations. While state-of-the-art objective measures for ASS quality build on monaural auditory models, they don¿t take into account binaural listening and the psychoacoustic phenomena that are involved, such as binaural unmasking. In this thesis, an extension to Perceptive Evaluation Methods for Audio Source Separation (PEASS) [41] is proposed with spatial rendering in mind. Additionally a new binaural model for ASS evaluation in the context of 3D audio is presented. The performance of the basic and extended versions of PEASS, as well as the proposed binaural model is evaluated in two subjective studies. The first study is conducted with binaural spatialisation presented over headphones, while the second experiment uses a 3DWave Field Synthesis (WFS) system. A set of artificial ASS degradation algorithms is proposed and used for the stimuli of the subjective studies. Results of the studies indicate monotonic decrease of the perceived quality as a function of the amounts of degradations introduced. The most important degradation is found to be target distortion, followed by onset misallocation and musical noise-type artifacts. Additionally, spatialising the extracted target source away from the residue or having it louder than the residue negatively affects the results, indicating a perceived quality degradation. In 3D WFS conditions, results show evidence for monaural and binaural unmasking. The performance of the proposed binauralmodel is consistently superior to that of the basic or extended PEASS versions. In the binaural spatialisation experiment, a correlation coefficient of 0.60 between subjective and objective results is achieved, versus 0.57 and 0.53 with the extended and basic PEASS version respectively. For the 3D WFS study, the binaural model achieves 0.67 prediction accuracy whereas both PEASS versions get 0.57. The perceptual validity of the WFS formulation is also verified in a localisation experiment. Vertical localisation is found to be nearly as good as physical source localisation for an extended listening area with localisation precision of 6± - 9±. The response time is also used as an indicator of localisation performance