A History of Audio Effects

Audio effects are an essential tool that the field of music production relies upon. The ability to intentionally manipulate and modify a piece of sound has opened up considerable opportunities for music making. The evolution of technology has often driven new audio tools and effects, from early architectural acoustics through electromechanical and electronic devices to the digitisation of music production studios. Throughout time, music has constantly borrowed ideas and technological advancements from all other fields and contributed back to the innovative technology. This is defined as transsectorial innovation and fundamentally underpins the technological developments of audio effects. The development and evolution of audio effect technology is discussed, highlighting major technical breakthroughs and the impact of available audio effects

Baseband analog circuits in deep-submicron cmos technologies targeted for mobile multimedia

Three main analog circuit building blocks that are important for a mixed-signal system are investigated in this work. New building blocks with emphasis on power efficiency and compatibility with deep-submicron technology are proposed and experimental results from prototype integrated circuits are presented. Firstly, a 1.1GHz, 5th order, active-LC, Butterworth wideband equalizer that controls inter-symbol interference and provides anti-alias filtering for the subsequent analog to digital converter is presented. The equalizer design is based on a new series LC resonator biquad whose power efficiency is analytically shown to be better than a conventional Gm-C biquad. A prototype equalizer is fabricated in a standard 0.18μm CMOS technology. It is experimentally verified to achieve an equalization gain programmable over a 0-23dB range, 47dB SNR and -48dB IM3 while consuming 72mW of power. This corresponds to more than 7 times improvement in power efficiency over conventional Gm-C equalizers. Secondly, a load capacitance aware compensation for 3-stage amplifiers is presented. A class-AB 16W headphone driver designed using this scheme in 130nm technology is experimentally shown to handle 1pF to 22nF capacitive load while consuming as low as 1.2mW of quiescent power. It can deliver a maximum RMS power of 20mW to the load with -84.8dB THD and 92dB peak SNR, and it occupies a small area of 0.1mm2. The power consumption is reduced by about 10 times compared to drivers that can support such a wide range of capacitive loads. Thirdly, a novel approach to design of ADC in deep-submicron technology is described. The presented technique enables the usage of time-to-digital converter (TDC) in a delta-sigma modulator in a manner that takes advantage of its high timing precision while noise-shaping the error due to its limited time resolution. A prototype ADC designed based on this deep-submicron technology friendly architecture was fabricated in a 65nm digital CMOS technology. The ADC is experimentally shown to achieve 68dB dynamic range in 20MHz signal bandwidth while consuming 10.5mW of power. It is projected to reduce power and improve speed with technology scaling

High gain and bandwidth current-mode amplifiers : study and implementation

Doutoramento em Engenharia ElectrotécnicaEsta tese aborda o problema do projecto de amplificadores com grandes produtos de ganho por largura de banda. A aplicação final considerada consistiu no projecto de amplificadores adequados à recepção de sinais ópticos em sistemas de transmissão ópticos usando o espaço livre. Neste tipo de sistemas as maiores limitações de ganho e largura de banda surgem nos circuitos de entrada. O uso de detectores ópticos com grande área fotosensível é uma necessidade comum neste tipo de sistemas. Estes detectores apresentam grandes capacidades intrínsecas, o que em conjunto com a impedância de entrada apresentada pelo amplificador estabelece sérias restrições no produto do ganho pela largura de banda. As técnicas mais tradicionais para combater este problema recorrem ao uso de amplificadores com retroacção baseados em configurações de transimpedância. Estes amplificadores apresentam baixas impedâncias de entrada devido à acção da retroacção. Contudo, os amplificadores de transimpedância também apresentam uma relação directa entre o ganho e a impedância de entrada. Logo, diminuir a impedância de entrada implica diminuir o ganho. Esta tese propõe duas técnicas novas para combater os problemas referidos. A primeira técnica tem por base uma propriedade fundamental dos amplificadores com retroacção. Em geral, todos os circuitos electrónicos têm tempos de atraso associados, os amplificadores com retroacção não são uma excepção a esta regra. Os tempos de atraso são em geral reconhecidos como elementos instabilizadores neste tipos da amplificadores. Contudo, se usados judiciosamente, este tempos de atraso podem ser explorados como uma forma da aumentar a largura de banda em amplificadores com retroacção. Com base nestas ideias, esta tese apresenta o conceito geral de reatroacção com atraso, como um método de optimização de largura de banda em amplificadores com retroacção. O segundo método baseia-se na destruição da dualidade entre ganho e impedância de entrada existente nos amplificadores de transimpedância. O conceito de adaptação activa em modo de corrente é neste sentido uma forma adequada para separar o detector óptico da entrada do amplificador. De acordo com este conceito, emprega-se um elemento de adaptação em modo de corrente para isolar o detector óptico da entrada do amplificador. Desta forma as tradicionais limitações de ganho e largura de banda podem ser tratadas em separado. Esta tese defende o uso destas técnicas no desenho de amplificadores de transimpedância para sistemas de recepção de sinais ópticos em espaço livre.This thesis addresses the problem of achieving high gain-bandwidth products in amplifiers. The adopted framework consisted on the design of a free-space optical (FSO) front end amplifier able to amplify very small optical signals over large frequency bandwidths. The major gain-bandwidth limitations in FSO front end amplifiers arise due to the input circuitry. Usually, it is necessary to have large area optical detectors in order to maximize signal reception. These detectors have large intrinsic capacitances, which together with the amplifier input impedance poses a severe restriction on the gain-bandwidth product. Traditional techniques to combat this gain-bandwidth limitation resort to feedback amplifiers consisting on transimpedance configurations. These amplifiers have small input impedances due to the feedback action. Nevertheless, transimpedance amplifiers have a direct relation between gain and input impedance. Thus reducing the input impedance usually implies reducing the gain. This thesis advances two new methods suitable to combat the above mentioned problems. The first method is based on a fundamental property of feedback amplifiers. In general, all electronic circuits have associated time delays, and feedback amplifiers are not an exception to this rule. Time delays in feedback amplifiers have been recognized as destabilizing elements. Nevertheless, when used with appropriate care, these delays can be exploited as bandwidth enhancement elements. Based on these ideas, this thesis presents the general concept of delayed feedback, as a bandwidth optimization method suitable for feedback amplifiers. The second method is based on the idea of destroying the impedance-gain duality in transimpedance amplifiers. The concept of active current matching is in this sense a suitable method to detach the optical detector from the transimpedance amplifier input. According to this concept, a current matching device (CMD) is used to convey the signal current sensed by the optical detector, to the amplifier’s input. Using this concept the traditional gainbandwidth limitations can be treated in a separate fashion. This thesis advocates the usage of these techniques for the design of transimpedance amplifiers suited for FSO receiving systems

Digital Filters

The new technology advances provide that a great number of system signals can be easily measured with a low cost. The main problem is that usually only a fraction of the signal is useful for different purposes, for example maintenance, DVD-recorders, computers, electric/electronic circuits, econometric, optimization, etc. Digital filters are the most versatile, practical and effective methods for extracting the information necessary from the signal. They can be dynamic, so they can be automatically or manually adjusted to the external and internal conditions. Presented in this book are the most advanced digital filters including different case studies and the most relevant literature

Design and Control of Power Converters 2019

In this book, 20 papers focused on different fields of power electronics are gathered. Approximately half of the papers are focused on different control issues and techniques, ranging from the computer-aided design of digital compensators to more specific approaches such as fuzzy or sliding control techniques. The rest of the papers are focused on the design of novel topologies. The fields in which these controls and topologies are applied are varied: MMCs, photovoltaic systems, supercapacitors and traction systems, LEDs, wireless power transfer, etc

An Exploratory Study of the Distortion Characteristics of Valve Signal Processors

Thesis (MMus)--Stellenbosch University, 2021ENGLISH ABSTRACT: Valve signal processors, a technology ostensibly out of place in the modern world, are still widely used in professional audio practice. It is believed that valves can impart a euphonic timbre on audio signals, in a phenomenon known colloquially as signal colouration. The inherent electronic limitations of valve signal processing result in a variety of distortions to be generated. Subsequently, the generation of distortion corresponds to the emergence of signal colouration. The standard distortion testing methodologies ordinarily employed by engineers have been limited as an analysis tool. These tests were originally designed to analyse the total amount of distortion present in a signal, not the intrinsic distortion character of a signal processor. Therefore, the objective of this thesis was to explore the key characteristics of valve signal processing distortion. The approach to empirical research adopted for this thesis involved the design and implementation of a series of exploratory experiments intended to analyse distortion character. Experiments were conducted on a selection of devices. Data was acquired from three software emulators, and a hardware valve microphone pre-amplifier. The microphone pre-amplifier was able to use two different types of valve for its gain stages, a 12AY7 and 12AX7; data was gathered from both. To make inferences into the processes that lead to the emergence of distortion, MATLAB was used to plot the electronic characteristics of a valve gain stage for analysis, thus permitting the juxtaposition of electronic phenomena to distortion measurements. The research was definitive in finding that different signal processors generate idiosyncratic profiles o f d istortion; t he e xtent t o which t his o ccurred s urpassed p rior a ssumptions. Each device produced distortion profiles that were not static, but dynamic, and evolved with varying input levels. These results suggest that different signal processors can cause distinct timbral changes to audio material under certain conditions.AFRIKAANSE OPSOMMING: Alhoewel vakuumbuise nie meer algemeen gebruik word in in moderne elektronika nie, word dit steeds wyd aangewend in klankverwerking waar die seinverkleuring wat deur die vakuumbuis se nie-liniere gedrag, gewensd is. Die verskeie tipes klankvervorming wat vakuumbuise skep veroorsaak kombineer tot ’n ontluikende effek in die vorm van klankverkleuring. Standaardtoesprosedures wat gebruik word om seinvervorming te meet is gerig daarop om die totale vervorming te meet en nie die spesifieke eienskappe van vervorming wat hoog aangeprys word vanuit ’n toonkleurperspektief nie. Die doelstelling van hierdie tesis was om die kerneienskappe van vakuumbuisvervorming op klankseine te verken. Die empiriese navorsing in hierdie tesis is die ontwerp en implementering van ’n reeks verkennende eksperimente om vakuumbuisvervorming te analiseer. Eksperimente is uitgevoer op ’n verskeidenheid van toestelle. Data is gemyn van drie sagteware-emulators en ’n vakuumbuis mikrofoonvoorversterker. Die mikrofoonvoorversterker was in staat om twee verskillende tipes vakuumbuise in die aanwinsvlak te gebruik, ’n 12AY7 en 12AX7 buis. Om afleidings te maak omtrent die prosesse wat lei tot die ontluiking van ververvorming, is MATLAB gebruik om die elektroniese eienskappe van ’n vakuumbuisbuisaanwinsvlak te stip vir ontledingsdoeleindes. Dit stel ’n vergelyking van elektroniese eienskappe met vervormingsmeting in staat. Die navorsing het bevind dat verskillende seinverwerkers unieke vervormingsprofiele genereer en die mate waartoe dit plaasvind het voormalige aannames oortref. Elke toestel het vervormseienskappe vertoon wat dinamies is en wat verander op grond van die intreeseinvlakke. Die resultate vertoon die invloed van verskillende seinverwerkers op toonkleureienskappe onder sekere toestande.Master

High Speed Integrated Circuits for High Speed Coherent Optical Communications

With the development of (sub) THz transistor technologies, high speed integrated circuits up to sub-THz frequencies are now feasible. These high speed and wide bandwidth ICs can improve the performance of optical components, coherent optical fiber communication, and imaging systems. In current optical systems, electrical ICs are used primarily as driving amplifiers for optical modulators, and in receiver chains including TIAs, AGCs, LPFs, ADCs and DSPs. However, there are numerous potential applications in optics using high speed ICs, and different approaches may be required for more efficient, compact and flexible optical systems.This dissertation will discuss three different approaches for optical components and communication systems using high speed ICs: a homodyne optical phase locked loop (OPLL), a heterodyne OPLL, and a new WDM receiver architecture.The homodyne OPLL receiver is designed for short-link optical communication systems using coherent modulation for high spectral efficiency. The phase-locked coherent receiver can recover the transmitted data without requiring complex back-end digital signal processing to recover the phase of the received optical carrier. The main components of the homodyne OPLL are a photonic IC (PIC), an electrical IC (EIC), and a loop filter. One major challenge in OPLL development is loop bandwidth; this must be of order 1 GHz in order for the loop to adequately track and suppress the phase fluctuations of the locked laser, yet a 1 GHz loop bandwidth demands small (<100 ps) propagation delays if the loop is to be stable. Monolithic integration of the high-speed loop components into one electrical and one photonic IC decreases the total loop delay. We have designed and demonstrated an OPLL with a compact size of 10 × 10 mm2, stably operating with a loop bandwidth of 1.1 GHz, a loop delay of 120 ps, a pull-in time of 0.55 μs and lock time of <10 ns. The coherent receiver can receive 40 Gb/s BPSK data with a bit error rate (BER) of <10-7, and operates up to 35 Gb/s with BER 10-12.The thesis also describes heterodyne OPLLs. These can be used to synthesize optical wavelengths of a broad bandwidth (optical wavelength synthesis) with narrow linewidth and with fast frequency switching. There are many applications of such narrow linewidth optical signal sources, including low phase noise mm-wave and THz-signal sources, wavelength-division-multiplexed optical transmitters, and coherent imaging and sensor systems. The heterodyne OPLL also has the same stability issues (loop delay and sensitivity) as the homodyne OPLL. In the EIC, a single sideband mixer operating using digital design principles (DSSBM) enables precisely controlled sweeping of the frequency of the locked laser, with control of the sign of the frequency offset. The loop's phase and frequency difference detector (PFD) uses digital design techniques to make the OPLL loop parameters only weakly sensitive to optical signal levels or optical or electrical component gains. The heterodyne OPLL operates stably with a loop bandwidth of 550 MHz and loop delay of <200 ps. An initial OPLL design exhibited optical frequency (wavelength) synthesis from -6 GHz to -2 GHz and from 2 GHz to 9 GHz. An improved OPLL reached frequency tuning up to 25 GHz. The homodyne OPLL exhibits -110 dBc/Hz phase noise at 10 MHz offset and -80 dBc/Hz at 5 kHz offset.Finally, the thesis describes a new WDM receiver architecture using broadband electrical ICs. In the proposed WDM receiver, a set of received signals at different optical wavelengths are mixed against a single optical local oscillator. This mixing converts the WDM channels to electrical signals in the receiver photocurrent, with each WDM signal being converted to an RF sub-carrier of different frequency. An electrical IC then separately converts each sub-carrier signal to baseband using single-sideband mixers and quadrature local oscillators. The proposed receiver needs less complex hardware than the arrays of wavelength-sensitive receivers now used for WDM, and can readily adjust to changes in the WDM channel frequencies. The proposed WDM receiver concept was demonstrated through several system experiments. Image rejection of greater than 25 dB, adjacent channel suppression of greater than 20 dB, operation with gridless channels, and six-channel data reception at a total 15 Gb/s (2.5 Gb/s BPSK × 6-channels) were demonstrated

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

"Datum for its own annihilation" : feedback, control, and computing, 1916-1945

Thesis (Ph. D.)--Massachusetts Institute of Technology, Program in Science, Technology, and Society, 1996.Includes bibliographical references.by David A. Mindell.Ph.D

Engineering handbook

1998 handbook for the faculty of Engineerin