Algorithms and VLSI architectures for parametric additive synthesis

A parametric additive synthesis approach to sound synthesis is advantageous as it can model sounds in a large scale manner, unlike the classical sinusoidal additive based synthesis paradigms. It is known that a large body of naturally occurring sounds are resonant in character and thus fit the concept well. This thesis is concerned with the computational optimisation of a super class of form ant synthesis which extends the sinusoidal parameters with a spread parameter known as band width. Here a modified formant algorithm is introduced which can be traced back to work done at IRCAM, Paris. When impulse driven, a filter based approach to modelling a formant limits the computational work-load. It is assumed that the filter's coefficients are fixed at initialisation, thus avoiding interpolation which can cause the filter to become chaotic. A filter which is more complex than a second order section is required. Temporal resolution of an impulse generator is achieved by using a two stage polyphase decimator which drives many filterbanks. Each filterbank describes one formant and is composed of sub-elements which allow variation of the formant’s parameters. A resource manager is discussed to overcome the possibility of all sub- banks operating in unison. All filterbanks for one voice are connected in series to the impulse generator and their outputs are summed and scaled accordingly. An explorative study of number systems for DSP algorithms and their architectures is investigated. I invented a new theoretical mechanism for multi-level logic based DSP. Its aims are to reduce the number of transistors and to increase their functionality. A review of synthesis algorithms and VLSI architectures are discussed in a case study between a filter based bit-serial and a CORDIC based sinusoidal generator. They are both of similar size, but the latter is always guaranteed to be stable

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

Real-time sound synthesis on a multi-processor platform

Real-time sound synthesis means that the calculation and output of each sound sample for a channel of audio information must be completed within a sample period. At a broadcasting standard, a sampling rate of 32,000 Hz, the maximum period available is 31.25 μsec. Such requirements demand a large amount of data processing power. An effective solution for this problem is a multi-processor platform; a parallel and distributed processing system. The suitability of the MIDI [Music Instrument Digital Interface] standard, published in 1983, as a controller for real-time applications is examined. Many musicians have expressed doubts on the decade old standard's ability for real-time performance. These have been investigated by measuring timing in various musical gestures, and by comparing these with the subjective characteristics of human perception. An implementation and its optimisation of real-time additive synthesis programs on a multi-transputer network are described. A prototype 81-polyphonic-note- organ configuration was implemented. By devising and deploying monitoring processes, the network's performance was measured and enhanced, leading to an efficient usage; the 88-note configuration. Since 88 simultaneous notes are rarely necessary in most performances, a scheduling program for dynamic note allocation was then introduced to achieve further efficiency gains. Considering calculation redundancies still further, a multi-sampling rate approach was applied as a further step to achieve an optimal performance. The theories underlining sound granulation, as a means of constructing complex sounds from grains, and the real-time implementation of this technique are outlined. The idea of sound granulation is quite similar to the quantum-wave theory, "acoustic quanta". Despite the conceptual simplicity, the signal processing requirements set tough demands, providing a challenge for this audio synthesis engine. Three issues arising from the results of the implementations above are discussed; the efficiency of the applications implemented, provisions for new processors and an optimal network architecture for sound synthesis

Sound Synthesis Using Programmable System-On-Chip Devices

The last 20 years has witnessed a resurgence of interest in analogue synthesisers 1 . Manufacturers, such as Moog and Sequential Circuits, that had disappeared from the commercial marketplace by the end of the 1980’s, have reappeared with an impressive line of products. Other established companies such as Korg and Roland, as well as entrants that had made their name with digital technology, such as Novation and Arturia, have released analogue instruments. Although the feature set of digital synthesisers is extensive and with a falling comparative cost, the analogue market has continued to grow with more and more devices coming available. They are perceived to be of superior sound quality by users, but their primary drawback is price, as numerous discrete components or specialist integrated circuits are required. This thesis introduces two novel low-cost approaches to building analogue-type synthesisers. Such a low-cost instrument could have applications in an educational laboratory environment for synthesisers. The first approach is to exploit a new mixed-signal technology called the Programmable System-on-Chip (PSoC), which includes a CPU core and mixed-signal arrays of configurable integrated analogue and digital peripherals. The second exploits a System on Chip (SoC) comprising an ARM-based (Acorn RISC Machine) processor and a Field-Programmable Gate Array (FPGA). Two synthesisers were built and were evaluated for difficulty of implementation and assessed for their sound quality. The design and testing process was recorded and documented in detail. The mixed-signal approach was found to be cheaper than the FPGA-approach both in terms of component costs and development time compared to the FPGA-based approach. Actually, the FPGA-approach was determined to be prohibitively expensive in terms of the development time incurred. The sound quality analysis demonstrated that both instruments were perceived by users to be of high quality, achieving a noticeable analogue sound. Future work would be to repackage the PSoC system and modules into rack-mounted form for use in an educational synthesiser laboratory environment

Sound Synthesis Using Programmable System-On-Chip Devices

The last 20 years has witnessed a resurgence of interest in analogue synthesisers 1 . Manufacturers, such as Moog and Sequential Circuits, that had disappeared from the commercial marketplace by the end of the 1980’s, have reappeared with an impressive line of products. Other established companies such as Korg and Roland, as well as entrants that had made their name with digital technology, such as Novation and Arturia, have released analogue instruments. Although the feature set of digital synthesisers is extensive and with a falling comparative cost, the analogue market has continued to grow with more and more devices coming available. They are perceived to be of superior sound quality by users, but their primary drawback is price, as numerous discrete components or specialist integrated circuits are required. This thesis introduces two novel low-cost approaches to building analogue-type synthesisers. Such a low-cost instrument could have applications in an educational laboratory environment for synthesisers. The first approach is to exploit a new mixed-signal technology called the Programmable System-on-Chip (PSoC), which includes a CPU core and mixed-signal arrays of configurable integrated analogue and digital peripherals. The second exploits a System on Chip (SoC) comprising an ARM-based (Acorn RISC Machine) processor and a Field-Programmable Gate Array (FPGA). Two synthesisers were built and were evaluated for difficulty of implementation and assessed for their sound quality. The design and testing process was recorded and documented in detail. The mixed-signal approach was found to be cheaper than the FPGA-approach both in terms of component costs and development time compared to the FPGA-based approach. Actually, the FPGA-approach was determined to be prohibitively expensive in terms of the development time incurred. The sound quality analysis demonstrated that both instruments were perceived by users to be of high quality, achieving a noticeable analogue sound. Future work would be to repackage the PSoC system and modules into rack-mounted form for use in an educational synthesiser laboratory environment

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

A voice operated musical instrument.

Many mathematical formulas and algorithms exist to identify pitches formed by human voices, and this has continued to be popular in the fields of music and signal pro-cessing. Other systems and research perform real time pitch identification implemented by using PCs with system clocks faster than 400MHz. This thesis explores developing an embedded RPTI system using the average magnitude difference function (AMDF), which will also use MIDI commands to control a synthesizer to track the pitch in near real time. The AMDF algorithm was simulated and its performance analyzed in MATLAB with pre-recorded sound files from a PC. Errors inherent to the AMDF and the hardware constraints led to noticeable pitch errors. The MATLAB code was optimized and its performance verified for the Motorola 68000 assembly language. This stage of development led to realization that the original design would have to change for the processing time required for the AMDF implementation. Hardware was constructed to support an 8MHz Motorola 68000, analog input, and MIDI communications. The various modules were constructed using Vectorbord© prototyping board with soldered tracks, wires and sockets. Modules were tested individually and as a whole unit. A design flaw was noticed with the final design, which caused the unit to fail during program execution while operating in a stand-alone mode. This design is a proof of concept for a product that can be improved upon with newer components, more advanced algorithms and hardware construction, and a more aesthetically pleasing package. Ultimately, hardware limitations imposed by the available equipment in addition to a hidden design flaw contributed to the failure of this stand-alone prototype