Investigation into digital audio equaliser systems and the effects of arithmetic and transform errors on performance

Merged with duplicate record 10026.1/2685 on 07.20.2017 by CS (TIS)Discrete-time audio equalisers introduce a variety of undesirable artefacts into audio mixing systems, namely, distortions caused by finite wordlength constraints, frequency response distortion due to coefficient calculation and signal disturbances that arise from real-time coefficient update. An understanding of these artefacts is important in the design of computationally affordable, good quality equalisers. A detailed investigation into these artefacts using various forms of arithmetic, filter frequency response, input excitation and sampling frequencies is described in this thesis. Novel coefficient calculation techniques, based on the matched z-transform (MZT) were developed to minimise filter response distortion and computation for on-line implementation. It was found that MZT-based filter responses can approximate more closely to s-plane filters, than BZTbased filters, with an affordable increase in computation load. Frequency response distortions and prewarping/correction schemes at higher sampling frequencies (96 and 192 kHz) were also assessed. An environment for emulating fractional quantisation in fixed and floating point arithmetic was developed. Various key filter topologies were emulated in fixed and floating point arithmetic using various input stimuli and frequency responses. The work provides detailed objective information and an understanding of the behaviour of key topologies in fixed and floating point arithmetic and the effects of input excitation and sampling frequency. Signal disturbance behaviour in key filter topologies during coefficient update was investigated through the implementation of various coefficient update scenarios. Input stimuli and specific frequency response changes that produce worst-case disturbances were identified, providing an analytical understanding of disturbance behaviour in various topologies. Existing parameter and coefficient interpolation algorithms were implemented and assessed under fihite wordlength arithmetic. The disturbance behaviour of various topologies at higher sampling frequencies was examined. The work contributes to the understanding of artefacts in audio equaliser implementation. The study of artefacts at the sampling frequencies of 48,96 and 192 kHz has implications in the assessment of equaliser performance at higher sampling frequencies.Allen & Heath Limite

Hardward and algorithm architectures for real-time additive synthesis

Additive synthesis is a fundamental computer music synthesis paradigm tracing its origins to the work of Fourier and Helmholtz. Rudimentary implementation linearly combines harmonic sinusoids (or partials) to generate tones whose perceived timbral characteristics are a strong function of the partial amplitude spectrum. Having evolved over time, additive synthesis describes a collection of algorithms each characterised by the time-varying linear combination of basis components to generate temporal evolution of timbre. Basis components include exactly harmonic partials, inharmonic partials with time-varying frequency or non-sinusoidal waveforms each with distinct spectral characteristics. Additive synthesis of polyphonic musical instrument tones requires a large number of independently controlled partials incurring a large computational overhead whose investigation and reduction is a key motivator for this work. The thesis begins with a review of prevalent synthesis techniques setting additive synthesis in context and introducing the spectrum modelling paradigm which provides baseline spectral data to the additive synthesis process obtained from the analysis of natural sounds. We proceed to investigate recursive and phase accumulating digital sinusoidal oscillator algorithms, defining specific metrics to quantify relative performance. The concepts of phase accumulation, table lookup phase-amplitude mapping and interpolated fractional addressing are introduced and developed and shown to underpin an additive synthesis subclass - wavetable lookup synthesis (WLS). WLS performance is simulated against specific metrics and parameter conditions peculiar to computer music requirements. We conclude by presenting processing architectures which accelerate computational throughput of specific WLS operations and the sinusoidal additive synthesis model. In particular, we introduce and investigate the concept of phase domain processing and present several “pipeline friendly” arithmetic architectures using this technique which implement the additive synthesis of sinusoidal partials

The design and implementation of a wideband digital radio receiver

Historically radio has been implemented using largely analogue circuitry. Improvements in mixed signal and digital signal processing technology are rapidly leading towards a largely digital approach, with down-conversion and filtering moving to the digital signal processing domain. Advantages of this technology include increased performance and functionality, as well as reduced cost. Wideband receivers place the heaviest demands on both mixed signal and digital signal processing technology, requiring high spurious free dynamic range (SFDR) and signal processing bandwidths. This dissertation investigates the extent to which current digital technology is able to meet these demands and compete with the proven architectures of analogue receivers. A scalable generalised digital radio receiver capable of operating in the HF and VHF bands was designed, implemented and tested, yielding instantaneous bandwidths in excess of 10 MHz with a spurious-free dynamic range exceeding 80 decibels below carrier (dBc). The results achieved reflect favourably on the digital receiver architecture. While the necessity for minimal analogue circuitry will possibly always exist, digital radio architectures are currently able to compete with analogue counterparts. The digital receiver is simple to manufacture, based on the use of largely commercial off-the-shelf (COTS) components, and exhibits extreme flexibility and high performance when compared with comparably priced analogue receivers

Quantisation mechanisms in multi-protoype waveform coding

Prototype Waveform Coding is one of the most promising methods for speech coding at low bit rates over telecommunications networks. This thesis investigates quantisation mechanisms in Multi-Prototype Waveform (MPW) coding, and two prototype waveform quantisation algorithms for speech coding at bit rates of 2.4kb/s are proposed. Speech coders based on these algorithms have been found to be capable of producing coded speech with equivalent perceptual quality to that generated by the US 1016 Federal Standard CELP-4.8kb/s algorithm. The two proposed prototype waveform quantisation algorithms are based on Prototype Waveform Interpolation (PWI). The first algorithm is in an open loop architecture (Open Loop Quantisation). In this algorithm, the speech residual is represented as a series of prototype waveforms (PWs). The PWs are extracted in both voiced and unvoiced speech, time aligned and quantised and, at the receiver, the excitation is reconstructed by smooth interpolation between them. For low bit rate coding, the PW is decomposed into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW). The SEW is coded using vector quantisation on both magnitude and phase spectra. The SEW codebook search is based on the best matching of the SEW and the SEW codebook vector. The REW phase spectra is not quantised, but it is recovered using Gaussian noise. The REW magnitude spectra, on the other hand, can be either quantised with a certain update rate or only derived according to SEW behaviours

New strategies for low noise, agile PLL frequency synthesis

Phase-Locked Loop based frequency synthesis is an essential technique employed in wireless communication systems for local oscillator generation. The ultimate goal in any design of frequency synthesisers is to generate precise and stable output frequencies with fast switching and minimal spurious and phase noise. The conflict between high resolution and fast switching leads to two separate integer synthesisers to satisfy critical system requirements. This thesis concerns a new sigma-delta fractional-N synthesiser design which is able to be directly modulated at high data rates while simultaneously achieving good noise performance. Measured results from a prototype indicate that fast switching, low noise and spurious free spectra are achieved for most covered frequencies. The phase noise of the unmodulated synthesiser was measured −113 dBc/Hz at 100 kHz offset from the carrier. The intermodulation effect in synthesisers is capable of producing a family of spurious components of identical form to fractional spurs caused in quantisation process. This effect directly introduces high spurs on some channels of the synthesiser output. Numerical and analytic results describing this effect are presented and amplitude and distribution of the resulting fractional spurs are predicted and validated against simulated and measured results. Finally an experimental arrangement, based on a phase compensation technique, is presented demonstrating significant suppression of intermodulation-borne spurs. A new technique, pre-distortion noise shaping, is proposed to dramatically reduce the impact of fractional spurs in fractional-N synthesisers. The key innovation is the introduction in the bitstream generation process of carefully-chosen set of components at identical offset frequencies and amplitudes and in anti-phase with the principal fractional spurs. These signals are used to modify the Σ-Δ noise shaping, so that fractional spurs are effectively cancelled. This approach can be highly effective in improving spectral purity and reduction of spurious components caused by the Σ-Δ modulator, quantisation noise, intermodulation effects and any other circuit factors. The spur cancellation is achieved in the digital part of the synthesiser without introducing additional circuitry. This technique has been convincingly demonstrated by simulated and experimental results

On-line health monitoring of passive electronic components using digitally controlled power converter

This thesis presents System Identification based On-Line Health Monitoring to analyse the dynamic behaviour of the Switch-Mode Power Converter (SMPC), detect, and diagnose anomalies in passive electronic components. The anomaly detection in this research is determined by examining the change in passive component values due to degradation. Degradation, which is a long-term process, however, is characterised by inserting different component values in the power converter. The novel health-monitoring capability enables accurate detection of passive electronic components despite component variations and uncertainties and is valid for different topologies of the switch-mode power converter. The need for a novel on-line health-monitoring capability is driven by the need to improve unscheduled in-service, logistics, and engineering costs, including the requirement of Integrated Vehicle Health Management (IVHM) for electronic systems and components. The detection and diagnosis of degradations and failures within power converters is of great importance for aircraft electronic manufacturers, such as Thales, where component failures result in equipment downtime and large maintenance costs. The fact that existing techniques, including built-in-self test, use of dedicated sensors, physics-of-failure, and data-driven based health-monitoring, have yet to deliver extensive application in IVHM, provides the motivation for this research ... [cont.]

Analog to digital conversion in beam instrumentation systems

Analog to digital conversion is a very important part of almost all beam instrumentation systems. Ideally, in a properly designed system, the used analog to digital converter (ADC) should not limit the system performance. However, despite recent improvements in ADC technology, quite often this is not possible and the choice of the ADC influences significantly or even restricts the system performance. It is therefore very important to estimate the requirements for the analog to digital conversion at an early stage of the system design and evaluate whether one can find an adequate ADC fulfilling the system specification. In case of beam instrumentation systems requiring both, high time and amplitude resolution, it often happens that the system specification cannot be met with the available ADCs without applying special processing to the analog signals prior to their digitisation. In such cases the requirements for the ADC even influence the system architecture. This paper aims at helping the designer of a beam instrumentation system in the process of selecting an ADC, which in many cases is iterative, requiring a trade off between system performance, complexity and cost. Analog to digital conversion is widely and well described in the literature, therefore this paper focusses mostly on aspects related to beam instrumentation. The ADC fundamentals are limited to the content presented as an introduction during the CAS one-hour lecture corresponding to this paper.Comment: 36 pages, contribution to the CAS - CERN Accelerator School: Beam Instrumentation, 2-15 June 2018, Tuusula, Finlan

Evanescent single-molecule biosensing with quantum limited precision

Sensors that are able to detect and track single unlabelled biomolecules are an important tool both to understand biomolecular dynamics and interactions at nanoscale, and for medical diagnostics operating at their ultimate detection limits. Recently, exceptional sensitivity has been achieved using the strongly enhanced evanescent fields provided by optical microcavities and nano-sized plasmonic resonators. However, at high field intensities photodamage to the biological specimen becomes increasingly problematic. Here, we introduce an optical nanofibre based evanescent biosensor that operates at the fundamental precision limit introduced by quantisation of light. This allows a four order-of-magnitude reduction in optical intensity whilst maintaining state-of-the-art sensitivity. It enable quantum noise limited tracking of single biomolecules as small as 3.5 nm, and surface-molecule interactions to be monitored over extended periods. By achieving quantum noise limited precision, our approach provides a pathway towards quantum-enhanced single-molecule biosensors.Comment: 17 pages, 4 figures, supplementary informatio

Design and implementation of a modified fourier analysis harmonic current computation technique for power active filters using DSPs

The design and implementation of a harmonic current computation technique based on a modified Fourier analysis, suitable for active power filters incorporating DSPs is presented. The proposed technique is suitable for the monitoring and control of load current harmonics for real-time applications. The derivation of the basic equations based on the proposed technique and the system implementation using the Analogue Devices SHARC processor are presented. The steady state and dynamic performance of the system are evaluated for a range of loading conditions