New Technologies in Development For An Approaching Nonbinary Domain Which Theoretically Allow Analogue Audio To Surpass Current Digital Sampling Disadvantages.

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013
NEW TECHNOLOGIES IN DEVELOPMENT FOR AN APPROACHING NONBINARY DOMAIN WHICH THEORETICALLY ALLOW ANALOGUE AUDIO TO SURPASS CURRENT DIGITAL SAMPLING DISADVANTAGES.
Gareth Ramsay
July 2013
Dissertation submitted in partial fulllment for the Degree of Bachelor of Arts in Creative Music Production BA(Hon's)
Creative Music Production The Academy of Contemporary Music

in association with
University of Surrey
There might be future generations who never hear the sound of uncompressed ! ! ! ! ! ! audio. !Thats ! a hateful thought. Lets get going before its too late. Perhaps there is a new !golden age of audio to come. I have faith that it is possible... ~ Phillip Holmes
Abstract
The conversion of integral single analogue files into the digital domain remains to undermine the integrity and the fidelity of the real-world source of recorded capture through signal chopping, even as technology has advanced significantly within these areas. This study shows significance in the investigations of various requirements of such conversion in order to understand the effects (both advantageous and adverse) it has had on the source signal and therefore why technological advancements towards the betterment of fidelity (and reduction in losses thereof) have been undertaken, and presently continue to do so. In order for a means for the sound heard naturally to be heard in the same fashion through recorded means, it was contemplated as necessary to consider the eradication of the binary Pulse Code Modulation system and all its advanced attempts to better it in favour of researching new theories. Current advancements in capture technology as well as environmental sound reconstruction processes have inspired future design considerations using developmental technologies discovered and advancing in a non-binary computing domain. The rise of such a domain is reflected upon for future audio production aspects.
Table of Contents
1. INTRODUCTION 1.1 Overview/Context.................................................................................................................. pg. 1 1.2 Rationale................................................................................................................................. pg. 2 1.3 Hypothesis............................................................................................................................... pg. 3 1.4 Summary of Aims and Objectives.......................................................................................... pg. 5 1.5 Keywords................................................................................................................................. pg. 6 2. LITERATURE REVIEW 2.1 Introduction............................................................................................................................. pg. 7 2.1.1 Sound Travel......................................................................................................................... pg. 7 2.1.2 Principles of Neural Science & Human Hearing................................................................. pg. 8 2.1.3 Sampling............................................................................................................................... pg. 9 2.1.4 Pulse Code Modulation...................................................................................................... pg. 10 2.1.5 Error Correction................................................................................................................. pg. 10 2.2 Non-Binary Algorithms......................................................................................................... pg. 11 2.3 Nyquist-Shannon Sampling Theorem................................................................................... pg. 12 2.4 ADC Advancements.............................................................................................................. pg. 12 2.4.1 Sigma-Delta Modulation and Digital Isolation.................................................................. pg. 13 2.4.2 Pseudorandom Sampling, Photonic ADC, Femtosecond Laser Technologies................... pg. 14 2.5 Organically Inspired Technology and Organic Behavioral Programming............................ pg. 16 2.5.1 Cross-comparisons of Functional Capacity in Other New Non-binary Technologies....... pg. 16 2.6 Challenges.............................................................................................................................. pg. 17 2.7 D-Wave Computer and Two-Colour Ramsey Number Calculation..................................... pg. 18 2.8 Current Cross-Domain Processing Considerations............................................................... pg. 18
2.9 Microphone Technology Advancements............................................................................... pg. 19 2.10 Wave Field Synthesis and Spacial Reconstruction............................................................... pg. 21 2.11 Qubits and Quantum Computing....................................................................................... pg. 22 2.12 Conclusion........................................................................................................................... pg. 23 3. METHODOLOGY................................................................................................................. pg. 24 4. RESULTS 4.1 Introduction........................................................................................................................... pg. 26 4.2 Online Surveys....................................................................................................................... pg. 26 4.3 Online Surveys Compared With Surveys Conducted by S. Persson in 2005........................ pg. 29 4.4 Audiophile Statements........................................................................................................... pg. 29 4.5 Audiophile Arguments........................................................................................................... pg. 31 4.6 Expert Consistencies.............................................................................................................. pg. 31 4.7 Public Opinion Poll................................................................................................................ pg. 31 4.8 Sound & Listening Survey...................................................................................................... pg. 32 4.8 Music Student Audio Illusion Listening Test......................................................................... pg. 38 4.9 Summary & Overall Consistent & Inconsistencies Results.................................................... pg. 38 5. DISCUSSION......................................................................................................................... pg. 39 6. CONCLUSION...................................................................................................................... pg. 41 7. RECOMMENDATIONS....................................................................................................... pg. 43 8. REFERENCES....................................................................................................................... pg. 46 9. BIBLIOGRAPHY................................................................................................................... pg. 55 10. DISCOGRAPHY.................................................................................................................. pg. 62 11. APPENDICES ! ! Appendix A - Advantages and Disadvantages of Binary-Coded Decimal.......................................... pg. 68 Appendix B - A Brief History of Digital Audio............................................................................ pg. 71
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Appendix C - The Nyquist-Shannon Sampling Theorem................................................................ pg. 76 Appendix D - Notes on Discussion With Dr. Tim Brookes............................................................. pg. 78 Appendix E - Online Discussion With Phil Thornton................................................................... pg. 79 Appendix F - Interview With Adi Winman.................................................................................. pg. 81 Appendix G - Weber-Fechner Law............................................................................................... pg. 83 Appendix H - A Neural Science Overview of Human Hearing........................................................ pg. 86 Appendix I - Interview/Discussion Setup Emails........................................................................... pg. 87 Appendix J - Survey 1 Online Questions....................................................................................... pg. 92 Appendix K - Perrson 2005 Survey: Producers Answers............................................................... pg. 101 Appendix L - Sound & Listening Survey.................................................................................... pg. 115
12. GLOSSARY OF TERMS....................................................................................................pg. 145
List of Figures, Tables & Charts
Figure 1: The results of Sampling.................................................................................................. pg. 9 Figure 2: Aliasing of two sine waves.............................................................................................. pg. 9 Figure 3: Pulse Code Modulation................................................................................................ pg. 10 Figure 4: Example of interleaved error correction....................................................................... pg. 11 Figure 5: ADC Types, Resolution, Speed & Cost Comparison.................................................... pg. 13 Figure 6: Pseudorandom sampled photonic ADC architecture.................................................... pg. 15 Figure 7: Spectral analysis, using DASP algorithms, of pseudorandomly sampled two-tone test signals ! ! with wide-band RF signals separated by (a) <1 equivalent Nyquist zone (b)>2 Nyquist ! zones............................................................................................................................... pg. 15
Figure 8: Captured Pseudorandom pulse trace after the T/H circuit of Figure 6 with digital sampling ! oscilloscope...................................................................................................................... pg. 15
Figure 9: Spherical Array, Planar Wheel Array and Foldable Array Beamforming Technology... pg. 19 Figure 10: Frequency & Sampling Quality Comparison............................................................... pg. 20 Figure 11: Spatial Resolution Versus Frequency for Different Beamforming Techniques ............ pg. 20 Chart 1: Survey 1&2 Industry Function Response Percentage Levels........................................... pg. 26 Chart 2: Survey 1&2 Audio Production Medium Preference Response Percentage Levels......... pg. 27 Table 1: Survey 1&2 Audio Production Medium Preference Advantages and Disadvantages ! ! Responses............................................................................................................................... pg. 15
Chart 3: Survey 1&2 Pro Non-Binary Response Percentage Levels.............................................. pg. 28 Chart 4: Survey 1&2 Non-Digital & Analogue Technological advancements Awareness Percentage ! Levels............................................................................................................................... pg. 28 Table 2: Survey 1&2 Non-Digital & Analogue Technological advancements Awareness Percentage ! Levels .................................................................................................................................... pg. 29
Table 3: Famous Audiophile Statements and Opinions Recorded Between 1950 and 1990 ........ pg. 31 Table 4: General & Summarized Pro-Analogue & Pro-Digital Opinions..................................... pg. 31 Table 5: Expert Opinion Consistency Breakdown......................................................................... pg. 31 Figure 12: Digital Music Quality Opinion Poll.............................................................................. pg. 31 Table 6: Music Student Survey Item 1.......................................................................................... pg. 32 Table 7: Music Student Survey Item 2.......................................................................................... pg. 32 Table 8: Music Student Survey Item 3.......................................................................................... pg. 33 Table 9: Music Student Survey Item 4.......................................................................................... pg. 34 Table 10: Music Student Survey Item 5........................................................................................ pg. 35 Table 11: Music Student Survey Item 6........................................................................................ pg. 36 Table 12: Music Student Survey Item 8........................................................................................ pg. 37 Table 13: Music Student Survey Item 7........................................................................................ pg. 38 Table 14: Results Summary Coding............................................................................................... pg. 39
Declaration of Originality
I conrm that the submitted work is my own work and that I have clearly identied and fully acknowledged all material that is entitled to be attributed to others (whether published or unpublished) using the referencing system set out in the programme handbook. I agree that the University may submit my work to means of checking this, such as the plagiarism detection service Turnitin UK. I conrm that I understand that assessed work that has been shown to have been plagiarised will be penalised.
If requested, I agree to submit the draft material that was also completed solely by me prior to this submission
Acknowledgements I would like to express my gratitude to: ! The Head of Research, Dr. Tim Brookes at the Department of Audio Recording, University of Surrey for his
time, for answering my queries, for his advice in researching Sampling Theorem as well as practical explanations, and suggestions for further research, on the subject of Wave Field Synthesis. ! ! Chris Baume of the Audio Engineering Society UK Executive Committee for responding to my email and kindly pointing me in the direction of Dr. Brookes. ! To Earl Vickers, the Principal Audio Algorithm Engineer at the AES for his response to a previous email
requesting information on the Loudness Wars and who, with updated information, thereby unwittingly assisted me by providing fresh leads for research in certain areas of this paper. I am grateful for the time you took to respond. ! ! To my friend and a very great musical inspiration Phil Thornton, who always seems to magically be there on those rare occasions when I need to chat about things like Pro Tools or the task of writing the ensuing dissertation. It is my great pleasure to have been able to include his quotes in this paper and to simply be understood and encouraged by a like-minded thinker. ! ! ! To Christopher McClymont, who gave me some directional pointers in the binary code and Max MSP
research areas when I was still grasping at straws. Thank you for being a fellow geek! ! ! ! My brother Mark, for his ear and his vast binary knowledge as well as the excellence of his frequency Also, for his instinctive way of calculation chart which he very kindly supplied for me to include in the paper. ! " My sister-in-law Angela, whos creative vision and intuition allowed her to instantly recognise my vision and goal, and who shared the excitement at the germination stage which was inspirational and also assisted in my will to move ahead. ! To the participants of my surveys, thank you for your time and adventurousness.
explaining my creative thoughts back to me in technical translations - a very enviable ability.
To my parents, who I am certain I bored to tears on many occasion whilst trying to contain my technical
excitement through expounding dissertation related information during their favourite television shows. Also, for their support of my studies and for understanding when I needed to shut myself away in order to focus a hundred percent on my work. I love you very much. ! ! Lastly but certainly not the least, to my head lecturer and supervisor at ACM, Adi Winman, who has inspired me to attempt my very best in achieving as much as I could through time given to me in listening to my thoughts and by giving me his feedback as well as knowledgeable input . His questions and points led me to attempt my best in the aim of discovering a means for true 3D sound and provided impetus for me to conduct research and creative ideas (in a way that I may not have originally) in the pursuit of considerations of the invention of a sufcient playback system. I would like to thank him sincerely for sharing his passion for aspects of this subject and not least for his simply being an inspiration through his achievements in the industry. His high standards and expectations have been a yard stick that encouraged me to continue attempting to achieve the best I am capable of. I hope more than anything to have done him proud in return for his friendship. Thank you for motivating my journey to the end.
1. INTRODUCTION
1.1 Overview/Context
In recent years, there has been an increasing interest in the means of decreasing storage space to a nanoscale, as well as means of increasing audio sampling rates. This indicates a keen interest in attempting better resolution in the binary domain. As the
nature of digital is finite regardless of any increasing sample and bit rates, there is consideration about the maximum level of improvement possible in the conversion processes. (See Appendix A pg. for Advantages and Disadvantages of Binary-Coded Decimal)
Whilst facts and considerations exist over the human hearing range and perception, audiophile arguments continuously disagree over frequencies not heard by the human ear which, once absent, effect those that are. This study considers the worth in investigating how technology has advanced to such a degree, and yet is still an ultimate interference rather than a means of true transfer.
In short, the natural sound is not the precise sound (even if beneficially enhanced) that we are given back through the processing of the signal. It is considered that instead of (even the most advanced) audio technology being a more translucent semi-conductor or filter, it is merely a net, or puzzle, through which the whole is caged and may only be sensed in bits and pieces.
1.2 Rationale
Whilst it must be established that sound is time-based frequency (Thornton, 2013) and that (as in terms of analogue and digital audio) magnetic tape resolution increases with speed in the same way as higher sample rate in PCM (Thornton, 2013), it must also be established that any new technologies invented and/or compiled must be equally considerate of this fact, and should aim to remove interference with the sound captured and processed by them and rather to sustain the natural signal as faithfully as possible. (See Appendix E, pg. for Online Discussion With Phil Thornton)
Through the considerations of suitable capture means which theoretically shows the ability to enhance and preserve sound in the conversion process better than previously, this examination ultimately attempts to solve remaining issues by drawing on literary records, experiments and discussions within the areas of both existent and foreseen technologies in order to determine where the advantages and disadvantages of current audio production and processing lie, with emphasis on audio conversion and sampling. It also aims to answer challenges by means of contemplating the capabilities and uses of new technology being functional through the binary domain, but also a nonbinary domain.
1.3 Hypothesis
It seems appropriate to draw comparisons between the nature of sound and to strengthen this point with beneficial advancements being achieved in order to highlight issues of sound discrepancy being addressed through constant invention. As this study aims towards investigation into how sound could be more naturally processed by technologies for minimum energy transfer interference and generation loss, additional considerations of the uses of newly developed technologies which could be foreseen to enhance audio production are acknowledged.
The speed of operations suggested by the advanced performance capacity of the new technologies addressed in this paper amplifies the drive to constantly increase speed in the conversion processes with established and continuously advancing technologies. Overall and precise examples of attempts in this area must be studied to solidify the cause for this investigation.
After recognition that current transducer technologies offer us the closest sounds capable of being heard by the human ear, consideration is given to issues of microphone technology being imperfect by issues of energy transfer because: Unlike humans, microphones dont know which sounds are important and which are not, " " " " and so pick up everything. It is important to realise that microphones dont have attentional selectivity. (Watkinson, 2013)
Whilst the progress of sampling is acknowledged as having solved many previous issues with signal chopping there is the concern that, as the nature of digital is finite, there would always be a form of file segmentation in order for storage in the binary
domain. Whilst there may currently be no one conclusive reason as to why this might be a detrimental concern to the future of audio, it does highlight an opportunity to explore what possibilities might be presented if audio were no longer to be subjected to discrete coding and all of its compatible technology.
Whilst this paper does not claim to provide conclusive evidence for the necessity of implementing suggested advancements, it does highlight concerns that may offer reason to explore alternative theories.
1.4 Summary of Aims and Objectives
The main issues/questions addressed in this investigation are:
A) General considerations about the nature of sound and the perception of human hearing.
B) Prohibitive issues with Pulse Code Modulation and the nature of sampling, and reflection on the necessity of reconsidering/re- engineering the computer algorithm.
C) How might a revolution in audio production/processing benefit the audio industry and what possibilities could it present that are not already existent?
D) What design considerations, and the use of which available technologies (both established and still in development) would be most beneficial in achieving the most satisfactory attempt to eradicate the necessity of continual signal sampling through the invention of a new domain based system as well as means of capture to playback?
KEYWORDS
Term Pulse Code Modulation Wave Field Synthesis Principles of Neural Science Analogue/Digital Conversion Sigma Delta Modulation Binary-Coded Decimal Quantum Computing Pseudorandom Sampling Photonic Analogue to Digital Conversion
Acronym PCM WFS PNS ADC SDM BCD QC PS PADC
2. LITERATURE REVIEW !
2.1 Introduction
The published research concerning the main question is problematic. There is no specific database or mass source available to address the unique nature of the topic and comparisons between available knowledge and suggested studies have needed to be established for advancement. The investigation broaches complete reworking and
rethinking of processes and technologies both existent and non-existent and therefore certain parallels are measured as appropriate.
General aspects of the introduction topics are necessary for reflective mention in order to understand the nature of sound before and during current processing operations. This may then be related to concerns taken into account for any suggested progressive processing activities suggested in later sections of this paper.
2.1.1 Sound Travel A large volume of literature has been published and explored over the years with regards to how sound travels and is perceived through the human anatomy as hearing. A basic description states that: Sound is transmitted from a source to the surrounding air particles, which vibrate or collide ! " ! and pass the sound energy along to our ears. (BBC, 2013)
Without any particles to vibrate we wouldn't hear the sound (BBC, 2013), because in the
process of sound travel the tightly compressed molecules under high pressure begin to exert an
outward force on their neighbors in an effort to move toward areas of lower pressure. (Huber, 2010) This is necessary as where there is no medium, no sound can be transmitted (NASA, 2013) because ittravels through liquids, gases and solids (Strickland, 2013) in order to impel itself away from the source over an accordingly varied timespan ...that continues until the molecules have used up their energy in the form of heat. (Huber, 2010) ! ! For further context, this needs to be related to how the sound is eventually
perceived by the human ear.
2.1.2 Principles of Neural Science & Human Hearing In terms of the explanation of human hearing by PNS, it is stated: Hearing commences with the capture of sound energy by the ear. Psychophysical ! ! ! ! ! ! ! ! experiments have established that we perceive an approximately equal increment in loudness for each 10-fold increase in the amplitude of a !s ound stimulus. This type of relation is characteristic of many of our senses and is the basis of the Weber -Fechner law. (Kandel, Schwartz & Jessel, 2000) (See Appendix G, pg. for Weber-Fechner Law) To represent sound intensity in a way that corresponds with perceived loudness, and especially in terms of translating this loudness for technological application, it is useful to employ a logarithmic scale. (See Appendix H, pg. for A Neural Science Overview of Human Hearing) (See Glossary) " In terms of this logarithmic translation being necessary for technological
applications, it is a requirement to know how the process effects the original analogue signal as this lays foundation for understanding in further investigation especially in terms of frequency losses.
2.1.3 Sampling Sampling is best explained as the process by which: Values are obtained from a continuous signal at regular time intervals and converted into a " " " " " " string of numbers. The reverse of this process is called interpolation whereby signal values at arbitrary instants of time and which fall somewhere between the actual samples are guessed in order to re-create a continual signal. (Schmidt, 2010)
What is then known as aliasing occurs.
This is caused by the sample rate of the
signal meeting the rate of the sampler during data acquisition (NI, 2006) It is stated: After sampling there is no signal between samples all the information that existed between " the samples in the original signal is irretrievably lost in the sampling process.
By ignoring anything that goes on between samples, the sampling process throws away information about the original signal. This information loss must be taken into account during system design, but most importantly must be considered in relation to any concerns over sound losses that the ear may or may not hear. quantization is a foundation factor. Knowledge of
2.1.4 Pulse Code Modulation The nature of quantization calls for the division of bit groups for encoding. Data coding, data modulation and data correction are the parts of a necessary process in order to encode the sampled signal into a language that understands further data and processing necessary for storage in the binary domain. PCM is the standard form of sampling for digital audio in computers. (Acroam, 2013)
Figure 3: Pulse Code Modulation (See Glossary)
Through this fundamental process, magnification of root issues with discrepancy and information loss occurs.
2.1.5 Error Correction ! Because PCM storage density is susceptible to irretrievable error by interference of
static or dirt, error correction is necessary in order to significantly enhance digital audio playback quality. One method uses redundant data in the form of parity bits and check codes in order to retrieve and/or reconstruct lost data .(Huber, 2010).
This is clearly not an ideal solution, which is substantiated by the statement: Reed-Solomon error correction works at the theoretical level of performance and there is no ! ! expectation that any future technology will perform any better. (Watkinson, 2013)
"
To support this unideal solution, a preferable second method uses complex
mathematical interleaved patterns to spread the data out over a large surface of the recording media or is deliberately scattered across the digital bitstream. (Huber, 2010)
Figure 4: Example of interleaved error correction
Even though the latter method ensures that the media will bypass a variety of dropouts, it is not completely immune, bringing additional inconsistency to light and therefore the notion that a great deal of consideration may be fundamental over the rethinking of the binary algorithm if sound is to be processed technologically and yet be liberated as best as possible from malformation, because Compression algorithms help only to recover some of the analogue attributes (Watkinson, 2013)
2.2 Non-Binary Algorithms It is significant at this point to note that non-binary based domains are showing early promises of the possibilities of audio being more expediently processed through technology in terms of algorithm re-engineering. It offers further suggestion as to unfulfilled potentials that discrete coding demands. Achievement has already been made to begin ensuring that this format is more beneficial than binary algorithms: The innovation of the new algorithm is the importing of logarithmic operations, which will transform multiplication operations to addition operations in the logarithmic domains. As a
result, the achieved algorithm has the advantages of reduced complexity and more convenient hardware implementation. (Hong & Sun, 2011)
In more recent experimentations, the previous indication of speed and simplicity is reinforced by the introduction of the SMSA algorithm: This hardware efficient decoding algorithm is devised based on significantly reducing the search space of combinatorial optimization. Two practical realizations ensure effective complexity-performance tradeoffs. Simulation results show that SMSA can produce negligible error performance loss. (Prasad & Ananth, 2013)
2.3 Nyquist-Shannon Sampling Theorem These algorithmic advances present cause to reflect back on another major issue that has dominated the field of audio production for many years which concerns the Nyquist-Shannon sampling theorem which has incurred much misuse by engineers through their designs of anti-aliasing filters and establishments of sampling rates due to being based on the assumption that the band limit of the frequency to be sampled is perfect. Unfortunately, a true and perfect band limit cannot be assigned to any real-world signal, as such a signal is dependent on existing within a frequency band in order to have any finite energy. (Wescott, 2010) (See Appendix C, pg. for The Nyquist-Shannon Sampling Theorem)
The attempts to remedy this situation are numerous and ongoing with regards to
types and functions of Analogue To Digital Conversions.
2.4 ADC Advancements Taking into account the varying levels of speed and power consumption which balance the benefits of Flash, Pipeline, Sigma-Delta, Dual Slope and Successive Approximation converters against each other, in depth studies from 2002 concluded that: There is no single ADC architecture that is suitable for high speed, high-resolution applications while maintaining low power and small area requirements. When all
parameters are optimized in a given design, it is often the inevitable mismatch that limits a converters accuracy. (Fung, 2002)
Figure 5: ADC Types, Resolution, Speed & Cost Comparison
2.4.1 Sigma-Delta Modulation and Digital Isolation If an example of the speed issue is studied in the cost-friendly Sigma-Delta Conversion, progress can be seen by how it has overcome the binary-driven nature of digital isolators because: The sigma-delta modulator simply translates an analog signal bounded to a certain range ! ! ! ! " ! ! ! ! into a high speed stream of bits, a density stream, which represents the analog signal - a high speed stream of ones and zeros that have an average weighting proportional to the input signal. The original signal can then be reconstructed on the other end through digital processing or even simple analog ltering. (Fosler, 2013)
The constant increase of digital techniques and the constant drive for integration makes
such an idea far more attractive in certain systems. (Fosler, 2013) Although residual tones are no longer an issue in first-order Sigma-Delta Modulators (Beis, 2005), there are still the issues of speed: The hardware must operate at the over-sampled rate, far exceeding the maximum signal ! ! ! ! bandwidth, and therefore demands great complexity of the digital circuitry. (Analog Devices, 2010).
2.4.2 Pseudorandom Sampling With Photonic ADC and Femtosecond Laser Technologies With any benefits of SDM and any previously mentioned ADCs set aside, little attention has been paid to means of revolutionising sampling issues outside of the binary domain as opposed to ways of advancing production within it - especially in terms of sampling rates and data coding.
Even as Photonic Analogue to Digital Conversion uses Femtosecond Laser Technologies in attempts to over-ride limitations by the timing jitter of the electronic clocking circuits and the jitter performance of electronic oscillators, the address remains stagnant.
As a search for sources of ultrafast optical pulse trains that show timing jitter on the level of a few femtoseconds over the time spans of typical sampling windows explores possibility of further size decrease in the future - any issues of sampling pockets are still not eradicated.
This technology is capable of overcoming many of the limitations in traditional electronic ADCs yet remains also to be the subject of much experimental investigation (Callahan, Dennis & Clark Jr., 2012), such as observations carried out as experimental demonstrations of a photonic analogue-to-digital converter architecture with pseudorandom sampling, using ...nonuniform, sub-Nyquist sampling combined with digital signal processing algorithms to unambiguously identify microwave signals of interest (Airola et al, 2008), which were found to: [Capitalize] on the strengths of "photonicshigh speed, wide bandwidth and precise timing " " " " capabilitieswhile eliminating the need to negotiate between performance and size, weight and power considerations of back-end processing hardware. (Airola et al, 2008)
Figure 6: Pseudorandom sampled photonic ADC architecture. Components are as follows. MLL: mode-locked laser. PPG: pulse pattern generator. EDFA: erbium-doped bre amplier. PD: photodiode detector. PS: phase shifter. ADC: electronic analogue-to-digital converter. PR CLK: pseudorandom clock.
Figure 7: Spectral analysis, using DASP algorithms, of pseudorandomly sampled two-tone test signals with wide-band RF signals separated by (a) <1 equivalent Nyquist zone (b)>2 Nyquist zones. Number of frequency points in each power spectrum is 400
The T/H device proved to be more than sufficient in ultimately including more noise bandwidth than required and as exemplification of the process which is seen here:
Figure 8: Captured Pseudorandom pulse trace after the T/H circuit of Figure 6 with digital sampling oscilloscope. Also shown with black circles are uniformly sampled digitizer data points. (Airola et al, 2008) !
This is described as follows: Large edge drops and rises indicate the onset of pseudorandom T/H times (lower time
! !
! !
units), followed by hold times long enough for the electronic digitizer to uniformly acquire the samples (upper time units). Filled black circles represent uniform time centers (spaced
! !
! !
every 1005 ps for a 995.3-MHz acquisition rate) of the electronic digitizer illustrating sufcient hold and timing to avoid sampling a transition time. (Airola et al, 2008)
"
Because these advancements (which show the greatest progress) are mainly still
experimental and/or outside of the binary domain, this again invites the option of considering less mechanical oppression over the natural signal. As a result, new nonsilicone based technology developments are considered in the areas of having organic properties.
2.5 Organically Inspired Technology and Organic Behavioral Programming In similarity to the human ear drum inspiring the microphone diaphragm, a further observation in organically inspired technology is recently celebrated as being the Harvard achievement of human DNA inspired and replicated storage which, now also having been advanced to store words, pictures, and audio (Wile, 2013), is explained through suggested instruction: In order for DNA stored data to be read, it is simply to be sequenced in the same fashion in which human genome sequencing occurs and then to begin conversion of each of the TGAC (The Genome Analysis Centre) bases back into binary. (Anthony, 2012) 2.5.1 Cross-comparisons of Functional Capacity in Other New Non-binary Technologies DNA storage now has the capacity for cramming 700 terabytes of data into one single gram. It has notoriety for surpassing the data density record in previous DNA storage versions by a thousand times (Anthony, 2012)
It can be paralleled with the advancement of the Graphene-QD Photodetector which (constructed from Graphene and semiconducting quantum dots) is acutely sensitive, is one billion times more sensitive to light than previous graphene-based photodetectors and may be suited to a varied selection of applications including light sensors and solar cells, infrared cameras for night vision and in biomedical imaging. (Dum, 2012) In a facilitating non-silicone based computing platform, the uses of this technology might be
considered for any image to sound mapping processes by way of quantum entanglement (See page 22, para. 4) which: Transcribes or "maps" the entangled beams of light onto two distinct groups of roughly 100,000 laser-cooled cesium atoms, separated from each other by 1 millimeter. (Banegas, 2009) In terms of organically supporting the means for audio to exist in a non-silicone based computing platform, there are also Phonitons which are half sound, half matter objects. They may act as tools for coupling quantum systems with examples including cases of ion traps or solid-state qubits. Their considerations for use in the simulation of many-body dynamics... (Sokyol; et al, 2011) might be self-explanatory in considerations of the integral role they could play where the granularity of capture and metadata need increasing in the nanoscale for improved sonic calibration.
Phonons which are found in the cavities of phonitons have further benefits: Phonons are slow compared to light, a property that is already used, e.g., in signal processing. Their smaller wavelengths are potentially relevant for nanoscale imaging and devices. Phonon analogues of lasers have been proposed and demonstrated. Theoretically these conditions can also be met in a realizable, silicon-based devices. (Sokyol; et al, 2011)
2.6 Challenges It is timely to interject that this study does encounter the opinions of experts and authorities in audio production sectors who generally agree that: Binary based applications [are] sufcient in the reconstruction of analogue and digital ! ! ! ! audio in terms of acceptability in the discrepancies that may not be heard by the human ear (Brookes; Winman, 2013).
(See Appendix F, pg. for Interview With Adi Winman)
It is also a major problem that the previously mentioned types of considered applications currently provide little experimentation or lucrative evidence of their integrations with studio industry or domestic audio systems: In addition, the difficulty in executing the vast improvements in speed and computing in the areas of quantum-based technology has been latently exposed through the fragmented and minimal progressions over the past decade. (Ekert & Ol, 2012)
2.7 D-Wave Computer and Two-Colour Ramsey Number Calculation It is, however, significant to note that there has been advancement beyond at least the speed issues by an experimental breakthrough whereby: What is considered to be the mind-numbingly difficult mathematics of the Two-colour Ramsey Number calculation was solved by a D-Wave quantum computer. Considered to be the largest-ever quantum calculation, 84 qubits were used and the process took a mere 270 milliseconds. (Dillow, 2012) 2.8 Current Cross-Domain Processing Considerations A further challenge exists in that even as the speed of such quantum technology advances, there have been no controlled studies which compare differences in audio processing between the binary and the qubit domains as there have been abundantly throughout the history of analogue and digital audio in terms of any benefits provided for the industry of audio production.
It may be beneficial to compare latest advances in certain areas with possible advances in the future as a means of determining precisely where the ultimate benefits in surpassing digital lie. The following areas consider the main benefits: (See Appendix B, pg. for A Brief History of Digital Audio)
2.9 Microphone Technology Advancements Acoustical Beam Forming is significant concept because of how certain microphone technology has advanced to fulfill the aims of improved audio capture by morphing into an acoustical camera.
Spherical arrays obtained by a complete omnidirectional noise map in any acoustic environment based on one simple measurement [by using] a spherical harmonics-based algorithm (Bksv.com, 2013) and Planar Wheel Arrays mainly used for beamforming measurements and for acoustical holography measurements, provided that the array can be moved close enough to the source (Bksv.com, 2013) are new means of sound capture technology along with Foldable Arrays which are used where reflections from the ground are important, such as measurements alongside a vehicle during a pass-by, or a model high speed train in a wind tunnel. (Bksv.com, 2013)
Figure 9: Spherical Array, Planar Wheel Array and Foldable Array Beamforming Technology
The main disadvantage of beamforming aside from spacial resolution being proportionate to wavelength and that it can not be used to calculate sound power (Debleuwe et al, 2007) is that it does not perform well in the low frequency range. This can however be improved by using a dedicated acoustic beamforming technique called near-field focalization (Debleuwe et al, 2007)
The two following charts show a) sampling measurements in terms of frequency and quality (for contextual reference) and b) spatial resolution versus frequency for different techniques of beamforming:
Figure 10: Frequency & Sampling Quality Comparison
Figure 11: Spatial Resolution Versus Frequency for Different Beamforming Techniques
Although Near-field focalization improves the spatial resolution to 0.44!, sampling remains prohibitive over results gathered by these advanced techniques. This is addressed and emphasized (as an example) by the introduction of CMOS radar transceivers which intend to possibly revolutionize a range of close range imaging
applications such as land mine detection, through wall imaging and even medical imaging, and yet still has unresolved issues: One of the main challenges lies in control over the CMOS process to produce a reproducible delay line for sampling the received signal at a very high frequency which is uniform. Direct calibration corrects only some of the quality degradation. (Bakken et al, 2012)
2.10 Wave Field Synthesis and Spacial Reconstruction Spacial reconstruction is another area attempted through possibilities but hindered by challenges presented by Wave Field Synthesis due to the necessity of working with binary based technology and processing, in terms of wave font rendering. Through a very basic overview of WFS, which defines the generation of a holographic illusion whereall of the localized cues are present no matter where the listener is positioned (Reichel, 2012), the sound source is produced by the positioning of a very large number of speakers. Though it is space which is not digital but a series of isolated sounds [which] in small slices compares with real space up to certain frequency, there are issues of spacial resolution, reflection and obstacles (Brookes, 2013).
During summary of the main challenge of WFS, the issue of missing frequency once again arises: The missing high frequencies affect the way you perceive the lower frequencies. In other words, the ear is a much more complex organ than people thought it was. (Mallinson, 2013) In relation it is also important to recall that the brain plays a significant role in individual perception of sound: It turns out that the way we perceive pitch isn't just to do with the tones we hear but also the pattern of tones. It is the way that you interpret that depends on how your brain works. In other words your brain is hearing the fundamental harmonic that isn't there. (Mould, 2005) This might once again present cause for further investigation into alternative means of spacial reconstruction and therefore, the natural reconstruction of sound in general.
2.11 Qubits and Quantum Computing The previous considerations of concepts and advancements in binary processed audio have been necessary in this study in order to assimilate its current benefits as a point of known reference in establishing a) what to retain as digital advantages and b) how to surpass them through understanding an evolution into a new and predominantly experimental domain such as Quantum Computing. It is assured that: A working, large-scale quantum computer is still a decade away, but researchers are ! ! ! ! currently turning a critical corner from theory to building the rst small quantum systems. (Dillow, 2012)
This is reassuring in the light of the prediction: There has been no real technical progress over the last ten years with regards to hi products ! ! ! ! ! and [as] digital quality is becoming meaningless, in ten years that process will have completed. (Watkinson, 2013)
General questions arising over the benefits of quantum computing are answered
largely in knowledge of the language of qubits, which are probability based.
A basic explanation would be that whereas a classical binary bit can only take the value of 0 or 1, a qubit may take both 0 and 1 as the value of 1. The continuous
analogue nature of this state exhibits quantum entanglement, a nonlocal property which allows superpositions of different binary strings simultaneously: Such "quantum parallelism" is one of the keys to the potential power of quantum computation. In essence, each independent state of the quantum particle used in the computer can follow its own independent computation path to conclusion while its other states are observed and changed. (Quantiki, 2010)
It is also of significance to note: It is not clear which technology will support Quantum Computation in the future. Today,
simple quantum logic gates involving two qubits are being realised in laboratories. Current
! ! ! !
! ! ! !
experiments range from trapped ions via atoms in an array of potential wells created by a pattern of crossed laser ! beams to electrons in semiconductors. The next decade should bring control over several qubits and, without any doubt, we shall already begin to benet from our new way of harnessing nature. (Ekert & Ol, 2012)
With regards to image to sound mapping ( as considering Beamforming and
Graphene QD chips), there have also been welcomed developments in Quantum Mechanics where the process can work vice versa as well and naturally cohere to the ways of acoustic sound: Now that we have a quantum transducer that converts electromagnetic energy into mechanical oscillations, we can start thinking about optical control of solid-state devices or spin systems. This in turn opens up a potential new class of hybrid optical/mechanical quantum devices, with potential applications in quantum computing. (Francis 2012)
Quantum Signal Processing (QSP) which is the same as DSP but with different mathematics, borrows from the mathematical formalism of quantum mechanics to reformulate signal processing [and] is not limited by physical constraints. (Aboy 2013, pp. 3-4) brings further possibilities for future and improved audio transitions from the binary code.
2.12 Conclusion This concludes that digital has prohibitions by its nature as shown by the various and continuing attempts to remedy its disadvantages through advancements. In
addition, the desire for the faithful capture and reconstruction of analogue sound inspires technology design as well as considerations of how to achieve the task as naturally as possible within the domains of binary code. It is generally apparent that
whilst there is no other available means for this to fully occur more accurately yet, the current level of binary development is therefore considered sufficient. A constant
throughout is the concern of sound apportionment, both in terms of time as well as frequency and the importance of speed.
3. METHODOLOGY
It was necessary to determine the level of use and opinion between analogue and digital production and processing in order to strengthen arguments and determine any reasons for this study, with greatest emphasis on how any discrepancies might be addressed through technological advancement and invention. User functionality was
paramount for insight as was any opinion and further research/study pointers gained from industry experts/professionals. Two interviews/discussions were set up (both in person and by email) and undertaken with authorities in the industry (both academic and practical) who were carefully selected for their knowledgeable expertise in key areas of the subject, ranging from spatial relationships, award-winning sound engineering achievements and work with game sound in the industry. Further to that, a couple of instances during online discussions with an award-winning multi-million album selling artist and also current audio production students through social network commenting added to the analysis as well. Survey questions were partially based on feedback from these authorities for
further study determination. The design of surveys distributed online was to cultivate information and insight, mainly for the purposes of maintaining evidence for the importance and necessity of this study, yet also for tempering any progressions in both developmental technologies and research with opinions and reactions from people involved with the processes and general utilization of current audio technology in order to provide - as complete as possible - a cross-study. The first survey was aimed at determining a general level of awareness of future trends and developments in audio technology with emphasis on the computing domain and considerations of cost etc. as well as to gather insight into the preferred use
of current audio formats as a means to further strengthen reasons for delving into what proved to be a very controversial study. The second survey was, once again, set to establish awareness of future computing domain formats but with emphasis on introductory determination of further questions for the participants, being namely concerned with the subject of technological advancement in general and reasons therefore. Because the sample sizes of the two surveys, it was felt that comparisons between whatever results were gathered and other studies, research and experiments undertaken in the field would prove beneficial and enhance insights gained, and to best eradicate any potential sources for error that may have been presented. Secondary research into the opinions and insights of experts within fields of problems discussed in this paper have been tempered with all results gathered in the primary methods and thereby used to supply primary results in relation to the course of discussion in this study. Audiophile comments across five decades were also researched and included in support as attestation of expert opinion of progress on the threshold of technological advancement throughout history. A third survey was conducted in person at a music school with 30 student participants aged between 13 and 19 with questions based on the general study of this paper for further evidence. Their age group was considered to be important for current music format consumption and technology use insights. In addition, a harmonics
listening test was conducted to emphasise the importance of taking individual listening perception and missing frequency into account. (See Appendix I, pg. for Interview/Discussion Setup Emails)
4. RESULTS
4.1 Introduction
The two surveys conducted online have been broken down into summarized charts
in order to show a more coded impression of the overall trends. Similarly, the Persson Surveys have been dissected in order to show the same value, additionally being crossexamined against the previous charts in order to show any discrepancies as well as similarities in trends. Feedback from interviews, discussions and magazine articles
written/stated by industry authorities have been taken into account in comparison with each other and furthermore added as a cross calculation of results. The results of the
live survey have been broken down between items to be considered in relation to other results gathered. A final result shows these preliminary results added and cross
examined against current information gathered from sources expert in the field of this papers discussed problems.
4.2 Online Surveys Ten items in the first survey were measured by the responses of eleven participants and six items in the second survey were measured by the responses of four participants across audio production groups on social networking forums. (See Appendix J, pg. for Survey 1 Online Questions) (See Appendix K, pg. for Survey 2 Online Questions)
Producer Composer/Writer Sound Engineer Technician 0 22.5 45 67.5 90
Chart 1: Survey 1&2 Industry Function Response Percentage Levels
Analogue Digital 0 20 40 60 80
Chart 2: Survey 1&2 Audio Production Medium Preference Response Percentage Levels
Table 1: Survey 1&2 Audio Production Medium Preference Advantages and Disadvantages Responses
Yes No Dont Know 0 12.5 25 37.5 50
Chart 3: Survey 1&2 Pro Non-Binary Response Percentage Levels
0% 20% 40% 60% 80% 100% 0 12.5 25 37.5 50

Chart 4: Survey 1&2 Non-Digital & Analogue Technological advancements Awareness Percentage Levels
4.3 Online Surveys Compared With Producer/Engineer Surveys Conducted by S. Persson in 2005
Table 2: Survey 1&2 Non-Digital & Analogue Technological advancements Awareness Percentage Levels
4.4 Audiophile Statements
Table 3: Famous Audiophile Statements and Opinions Recorded Between 1950 and 1990
4.5 Audiophile Arguments
Table 4: General & Summarized Pro-Analogue & Pro-Digital Opinions (Strickland, 2013)
4.6 Expert Consistencies
Table 5: Expert Opinion Consistency Breakdown
4.7 Public Opinion Poll
Figure 12: Digital Music Quality Opinion Poll (Telegraph, 2013)
4.8 Sound & Listening Survey Eight questions were circulated between 30 participants, including one case of a live listening test.
(See Appendix L, pg. for Sound & Listening Survey Response sheets)
Participant Age
13 y.o. x 4 14 y.o. x 4 15 y.o. x 7 16 y.o. x 11 17 y.o. x 2 18 y.o. x 1 19 y.o. x 1 = 30 participants
Table 6: Music Student Survey Item 1
Favourite Type of Music
Acoustic based x 4
Electric/Electronica based x5
Both x 21
Reasons
- Good instrumental resonance - Spanish & classic - Softer sound and generally more emotional - Requires years of skill and chemistry between band members - Dubstep is good and acoustic is tiring - It is the music of the future - More room to explore with various effects - Can really feel the groove, creates huge tensions and generates energy and excitement
5 x Both have good aspects - 5 x More variety - 7 x Good variation and technique contrast - 2 x Adaptable qualities
Analogue Advantages
Remembering it longer Natural Sound Main melody is audible Brighter tone x 2 Natural x 4 Faithful transmission of band quality and skill More original Exhibits raw talent Better, crisper sound quality x 3 Reliable Original Louder and good on vinyl More reliable x 2 Freer and more raw Traditional Captures and equipment cannot be lost all at once Physical and encumbering Nice antique feel Not altered by conversions Less bugs Better feel Warm true and expressive sound Unprocessed/Unedited
Digital Disadvantages Advantages

Can hear it all around the world Varies the music Interest and depth Easier to record with x 2 Has good beat Easier to make x 2 Portable and clear x 4 Quick A lot more storage space x4 More common More professional Easier alternative Easy editing and altering x 2 Quick & Lasts longer x 2 Clearer long distance transmission USB Cleaner sound Easy access Hi-Fi
Disadvantages
Not as powerful as a live performance Is not original Needs backing vocals to enhance it Less quality More common A threat to music stores Less free and unreliable Overrated Overall sound is sometimes particularly unclear Not traditional and less fresh x 2 Could all be lost at once No weight Data corrupting bugs Compressed and complicated to use in some cases Can sound clinical Quality maybe sacrificed
Will only hear it once Can be a little dull More room for error Special effects can be added to enhance with less accuracy Harder to make Can be unclear x2 Harder to purchase Not perfect and less reliable Takes longer to make Medium gets damaged Inferior quality Rare Background noise
Favourite Form of Music Purchase Vinyl x 5 Compact Disc x 12 Reasons Digital Download x 16
Playback listening preference Already on disc and better quality Better sound Original Recorded on analogue equipment Helps the artist by buying physical copy ( same with CD) Better listening experience Like having the actual format to hold x 3 Reasonable price and traditional Like the feeling of having a collection x 2 Grew up with it and is nice to own Can be ripped Easy access x 5 Easy management x 2 Easy use x 2 Fixable faults Efficient, cheaper and less storage space needed x 5 Can keep it all in one place Portable
Music Production Advantages Quirky Efficiency and good quality Possible to create easily Good genres Time has allowed for more modern/ openminded takes on music People can sell millions of records online easily Has progressed to becoming more available to younger audiences Home recording is good for young bands High quality recording and great possibility The old ways are just as good in some areas as the new ways Available to everyone, quick & cheap Can alter with perception Can get the exact sound you are going for Easy to learn now Live music sounds different now Difficulty in the uses of some production softwares Openly available Anyone can sound great Technology has made it far easier to produce music in a variety of ways
Disadvantages
Open to piracy x 6 Can all sound the same Can sound robotic Recently popular music is auto-tuned and horrible Can make good/raw talent sound fake/ generic Organizations are needed mostly for success Home recording produces too many Justin Biebers When it takes longer than it should The use of mainly digital is making it a lazy predicament Can be expensive x 3 Mostly takes a long time Too many producers decrease the quality and emotion.
Too much auto tune kills the soul Excessive dubbing kills band chemistry Serious musicians are discredited through easy editing of less talented musicians
Need For Non-Binary Technology Advancement Yes x 17 Reasons It needs a better control for cooling systems So that music can sound better x 2 There is still room for improvement x 2 To make illegal downloads more difficult To make sound manipulation easier Auto-tune would be used more Its becoming too lazy Advancement means more development and more capabilities Things are always better when improved Time reduction The field has so much unused potential To incorporate better systems to aid future electronic music Should decrease size and increase speed Better individual processes needed in software Better processing power, rendering time DAWs with better functions Smaller file sizes Computing can only help us
No x 12
Free music Its okay as it is x 5 Artists can already achieve suitable quality music Its advanced far enough and should be left alone x 3
Wouldnt know how it could advance further Not finished with current formats yet
Changes To/The Way That Music Should Be Recorded Nothing x 5 Equipment should cost less x 2 Make it more real and dont lose track of talent
Protect talented artists by scrapping auto-tune
Change back to tapes - no more macs in studio
Increase types of genres to record
Increase ability to change errors and decrease production time x 2
Revert to analogue
More live recordings and presences x 3
Make it more accessible x 2
Needs higher quality and faster rendering
The amount of equipment needed
Serious quality analogue modeling plugins
More realistic sound and quicker midi drum programming
4.9 Music Student Audio Illusion Listening Test Is the Second Sound Higher or Lower Than the First Sound?
Higher x 22
Lower x6
Cant Tell x2
Table 13: Music Student Survey Item 7 (Please refer to the Harmonic Bleeps Audio CD)
4.10 Summary & Overall Consistent & Inconsistencies Results
Consistencies
Binary is sufficient Storage space and speed Analogue mostly favoured for sound quality Listener Perception differs Cost, portability and access
Inconsistencies
Non-binary Technological advancements favoured Quality is preferred Digital favoured for editing processes, ease and capability Majority hear the same thing Desire for more live recordings
Table 14: Results Summary Coding
5. DISCUSSION
Sample sizes between online and Perrson surveys show no discrepancy in the concurrence of points and opinions overall results for either pro or anti formats. This is confirmed by the results of the live survey. The small sample size that participated in the future computing format survey attests to the minuscule level of awareness of computing domains beyond current trends and supports the experimental nature of any studies of technologies initiated within these areas. Once again, results from the live
survey show this, yet add insight into what would be changed if possible.
Audiophile statements gathered between 1950 and 1990 show similar trends to opinions and statements gathered in research during this study to show that there is a consistent skepticism and hesitation on the brink of sonic technological evolution, existential or suggested across history and currently, and defines the main root of this as the need for analogue to be preserved whilst technology advances for improved benefits of processing and storage.
Results show a marked effect that computing technology has on audio production and highlights concerns and reasons for continuous advancements for issues with timesaving, speed, space and budget.
The medium of analogue audio versus the preservation of digital are shown to be the greatest considerations overall as experts in the field agree beyond technological flaws and in the face of sampling progress that it is the nature of the playback experience that should ultimately define the process of the signal, which supports this investigation
into the areas of exploring any means of liberating audio from a mere dual-integer format.
This listening test results conclude that whilst the majority of listeners hear the same thing, there are individual cases which differ and also those which cannot differ. This could suggest that the nature of all feedback from experts and non-experts has a level of bias and opinion in addition to technical/user knowledge.
The poll shows an obvious public opinion that high resolution is key to the experience of satisfying audio. The contradiction suggested by the overall desire for the unaltered nature of analogue audio highlights a cause for the investigation into ways to achieve this balance where current technology apparently may not at this time.
Whilst age is shown as a major factor in opinionated feedback persuasions, it appears to have little effect on the overall agreement of study consistencies.
6. CONCLUSION
This paper has generally explored technical and practical evidence of both the
advantages and disadvantages of binary-based technology and processes concerning the fields of sound recording and audio production. Attention has been paid to awareness of how audio technology and sampling has progressed, as well as to awareness (or lack thereof) with regards to further development - both in design and employment of new application.
Reflection upon the binary systems positive acknowledgements has compared it with the possibilities of a less finite domain in comparison and what opportunities might be presented as a result which are not yet fully achievable - albeit proven feasible through existential progression.
Points of consistency in this study are agreements that current technology is flawed in certain areas and that the continuous analogue signal, without affectation from data loss due to sampling, is universally quested after for various reasons - even inadvertently amongst pro-digital opinions. This has given rise to issues and questions of the time basis of frequency which has in turn been reinforced in aspect to the desirability of time reduction during audio production processes. This study also recalls how technology continues to focus on portability and therefore reiterates data density requirements.
It is the conclusion of this paper that whilst digital sampling is generally accepted to be sufficient for the closest reproduction of recorded analogue sound after conversions, that a new capture system which regards all frequency as important in the
genetic make-up of real-life sound is accounted for in a manner which affords no data losses and permits a means of eventual true real life experience playback integrity.
It may hereby be considered that the finite nature of digital foresees a culmination of progress that might fall short of the capabilities of a new format which, whether or not acknowledged for any awareness thereof, presents possibilities not yet proven inferior.
7. RECOMMENDATIONS
It is suggested that further research and study be conducted within the areas of multi-data medium synthesis and integration as a means for definitive results (sound waves in the instance of this study) being achieved by multifarious levels of response ranging from image and ultrasonic measurement to the employment of meteorologist equipment in the pursuit of the most faithful environmental reproduction of sound capture. A reason for this (as well as a further suggested field of research)
would be the implications that such a progressive means of audio production would have on many levels of interest within both the processing as well as industry concerns, essentially exploring the possibilities of more integral editing as well as the reduction of necessary studio time as a result of the capabilities presented in terms of audio manipulation on a significantly advanced fractal level.
Another avenue of investigation is suggested as being in the pursuit of a means to integrate any qubit based technology possibilities within the binary domain as it presently operates by means of re-engineering the computer algorithm through means of its employment as opposed to its structure. As an example, considerations might be
taken into account of whether a more granular and varied array of source capture would be better served on the intake, thereby eradicating time based frequency signals altogether through markers of distance points which may be integrated and crossed over with captures of area density and temperature activity etc. thereby retaining all necessary time based information. This in turn would theoretically afford a nano-scale segmented capture which may be seamlessly joined together without the data loss caused by aliasing and generation. As an extension of this possibility, a full investigation into the discrepancies between the disadvantages presented by WFS and the proposed
advantages of a new reversible capture system may or may not highlight further reasons for technological advancements within these areas and yet may present insight into design concepts for a sufficient playback system.
More broadly, research and experimentation is also needed to determine the differences in audio perception through studies of Neural Science, outlining the responses between individual listeners through any foreseeable experiences of these suggested technological applications in order for maximum authenticity during the playback experience.
In the areas of organically behaving and compatible technology in advancement (which are suggested inclusions in the applications of the audio industry especially in the domains of quantum computing), it is urged that further research be conducted within the areas of the following materials, elements and mechanics with audio production in mind:
Graphene (with great emphasis on its altering properties by means of various doping treatments) Silicene (analogue version of Graphene) Graphene QD Photodetector chips Quantum Dots (caged) Chromatt and Chromea (employed in CGI and 3D vision technologies) Ferrofluids and oil magnetizations Electrons Phonitons Chemical fluid mechanics
Animatronics Image to sound mapping (both Quantum and Binary) Thermal Imaging Nanoscale magnetic sensing Photoelectronic sensing Solid-state physics X-ray and infrared Quantum applications Fluid and Quantum transducers Integration of Ultrasound Technologies Motion processing Preparata codes Holographic storage Pipeline storage DNA storage Meteorology/Meteorological Equipments Holophony Holography Plasma Wave Technologies
(Please see Nature: The Future of Sound Documentary DVD with Companion book)
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SQUIDs,IEEE Trans. Electron Devices, vol.27, pp. 1887, Oct. 1980. C. A. Hamilton and F. L. Lloyd,100 GHz binary counter based on dc SQUIDs,IEEE Electron Device Lett., vol. 3, pp. 335338, Nov.
DISCOGRAPHY
Phil Thornton Flying (Expandis) Expandisongs Special Limited Edition DVD - A Global Voyage Bexhill Museum Living like this Expandisongs Not Waving, I'm Drowning Expandisongs Tribal Spirit FatChanceBellyDance, Inc. Egypt Unveiled ARC Angelic Harmony New World Music Nexus Tribal Expandisongs Rhythm of the rainforest. New World Music Enchanted Egypt New World Music Zen Beats Smith & Co/Libra Films! 2004 2005 2009 2009 2010 2011 2011 2011 2011 2012 2012
Tibetan Meditation New World Music The Buddha Experience DVD Balance and Harmony Bathtime Bliss Spectrum The Buddha Experience collection Balance and Harmony Native American Chants Balance and Harmony Capoeira Balance and Harmony Dreamscapes New World Music Illusions New World Music Solstice New World Music Immortal Egypt New World Music Eternal Egypt New World Music Alien Encounter New World Music
2003
2003
2002
2002
2002
2002
2001
2001
1999
1998
1996
1996
Sorcerer New World Music Beyond Heaven's River Disky Shaman New World Music Pharaoh New World Music While the Green Man Sleeps Mystic Stones Alchemy New World Music Tibetan Horn New World Music Between Two Worlds New World Music Fire Queen New World Music Initiation New World Music Natural Magic New World Music Forever Dream Disky
1996
1995
1995
1995
1993
1993
1993
1991
1991
1990
1990
1989
Transformation New World Music From Another Sky New World Music Flying Disky Edge of Dreams New World Music Cloud Sculpting New World Music Flying (Expandis Version) Expandisongs Complete Home Expandis Expandisongs Not Waving, I'm Drowning Underground
1989
1988
1987
1986
1986
1983
1982
1981
Adi Winman Production The Only Ones (Part Two) Hybirds, The - Take You Down (CD, EP) Heavenly 1997 Don't Let Them Get You Down (CD, Dig) V2 Records, Inc. 1998 GoesAroundComesAround ! (2 versions) Big Deal, KMG Records 1999 Step Inside Your Love ! (2 versions)
Technical Artful Records 2001 Mrs Lullaby Marc Carroll - Mrs Lullaby (CD, Single) Evangeline Records 2002 London Calling Various - White Riot Vol. One A Tribute To The Clash (CD, Comp) Uncut Magazine 2003 Carter Brown (2) / Oracleboy, The - Say Hello / Clovers (7") Worst Case Scenario Records 2006 Heartbleeder Midnight Juggernauts / Coronation, The Raised By Wolves / Heartbleeder (7") Worst Case Scenario Records 2006 Dust Of Rumour ! (2 versions) High Noon Recordings 2009 Matty Groves (CDr, EP, Promo) High Noon Recordings 2010 The Sisters EP. ! (5 versions) Island Records 1994 The Living Dead Suede - Stay Together ! (6 versions) Nude Records 1994 A Map Of The Universe By Blink ! (2 versions) Parlophone 1994 Chrome ! (6 versions) Fontana 1993 Boaphenia ! (3 versions) Polydor 1993 Jim Diamond (CD, Album) Polygram TV 1993 Crank ! (5 versions) Fontana, Fontana 1993
Happy Days ! (4 versions) Mercury 1995 Babies Pulp - Common People '96 (CD, Single, Comp) Island Records 1996
Skunkworks ! (6 versions) Raw Power 1996 Different Class ! (2 versions) Island Records 1996 This Is The Sound
Juice ! (2 versions) EastWest 1996
(CD, Dig) V2 Records, Inc. 1997 Almost Here
Stoosh ! (12 versions) One Little Indian 1996
! (5 versions) Virgin 1998
APPENDIX A
Advantages and Disadvantages of Binary-Coded Decimal
The BCD representation shows its main advantage in the ease by which it is capable of being converted into decimals and vice versa. A good example of this might be through the necessity for
electronic decoding of binary numbers for the display of a device such as a pocket calculator. A BCD formatted number needs only to be segmented into four bits, and
each of those bit groups are then simply converted directly to the decimal digit that applicably corresponds. A further advantage of BDC representation is evident through fractional number storage. Because of the separate encoding of each decimal digit, the representation of
the complete number is based on as many bits as are required and this results in no number rounding. This is the reason for BCD arithmetic being employed in business applications, for example, where there is the necessity for the retainment of every significant digit in a result.
The main disadvantage of using BCD (in relation to pure Binary) is that more bits are required in number storage. In the same relation, another disadvantage is that BCD based calculations are far more complex. As an example, if the BCD representations of 1 and 19 were in equation, the result would be: 0000 0001 0001 1001 ------------0001 1010
From this we can see that 1, which is the first digit, is incorrect and 1010 is not valid as a code. The problem arising is explained by the first ten of sixteen four digit
combinations which are the only used to encode the decimal symbols 0 to 9.
As a
result, the greater the sum of two binary digits is over 9, an addition of 6 is required for the result by means of bypassing the six unused codes.
The addition of the 6 binary representation to 1010: 0001 1010 0110 ------------0010 0000 This results as 20 in BDC which arrives at the correct equation.
In summation (as well as further consideration) the following tables are set forth between the two aspects:
ADVANTAGES OF BDC OVER PURE BINARY
Many non-integral values, such as decimal 0.2, have an infinite place-value representation in binary (.001100110011...) but have a finite place-value in binarycoded decimal (0.0010). Consequently a system based on binary-coded decimal representations of decimal fractions avoids errors representing and calculating such values. Scaling by a factor of 10 (or a power of 10) is simple; this is useful when a decimal scaling factor is needed to represent a non-integer quantity (e.g., in financial calculations)
Rounding at a decimal digit boundary is simpler. Addition and subtraction in decimal does not require rounding.
ADVANTAGES OF BDC OVER PURE BINARY
Alignment of two decimal numbers (for example 1.3 + 27.08) is a simple, exact, shift.
Conversion to a character form or for display (e.g., to a text-based format such as XML, or to drive signals for a seven-segment display) is a simple per-digit mapping, and can be done in linear (O(n)) time. Conversion from pure binary involves relatively complex logic that spans digits, and for large numbers no linear-time conversion algorithm is known.
DISADVANTAGES OF BDC OVER PURE BINARY
Some operations are more complex to implement. Adders require extra logic to cause them to wrap and generate a carry early. 1520 percent more circuitry is needed for BCD add compared to pure binary. Multiplication requires the use of algorithms that are somewhat more complex than shift-mask-add (a binary multiplication, requiring binary shifts and adds or the equivalent, per-digit or group of digits is required).
Standard BCD requires four bits per digit, roughly 20 percent more space than a binary encoding (the ratio of 4 bits to log210 bits is 1.204). When packed so that three digits are encoded in ten bits, the storage overhead is greatly reduced, at the expense of an encoding that is unaligned with the 8-bit byte boundaries common on existing hardware, resulting in slower implementations on these systems.
Practical existing implementations of BCD are typically slower than operations on binary representations, especially on embedded systems, due to limited processor support for native BCD operations.
APPENDIX B
A Brief History of Digital Audio
T.A. Edisons invention of the phonograph in 1877 marked the birth of the audio signal recording and reproduction evolution. The main driving force since that time has been the effort to research, develop and therefore enhance ways of eliminating any form of distortion and noise in any given audio signal. This achievement began shortly after Edisons invention with P. Berliners introduction of the gramophone (or disc phonograph) in 1893 which heralded an enhanced sound quality with the addition and possibility of easy reproduction.
With the rise of electrical recording systems in 1925, it is noted that an acoustic recording process similar to the predecessor systems was still employed by which a membrane within a horn was used in sound receipt and generation, and as being attached to a responsive needle, a groove was able to be etched into the disc or record through instantaneous correspondence with the sonic vibrations.
Further gradual developments saw this system enhanced with the electric crystal pickup, and at the dawn of the 1930s, the standard playing record (SP) with its 78 rpm was popularized by the broadcast AM radio stations.
In 1948, CBS further increased the popularity of the record with its development of the 33 rpm known as the long play (LP) which held the capacity for 25 minutes playtime on each side.
RCA offered the next installment which was known as the extended play (EP), which boasted a superior quality in record sound and turned at 45 revolutions per minute. Simultaneously, companies such as Pickering and General Electric were developing pick-up cartridges which were light weight and operated with only a few grams of pressure from the stylus.
1956 is generally considered to be the year that saw the beginning of the fundamental change brought about by the quest for faithful audio recording and playback signals. With a stereo reproduction tape recorder intended initially for industrial master use being the goal of a manufacturing race, the contest brought with it a simplification of techniques which made equipment for domestic use an included possibility.
Responsible for the resulting improvements in sound quality, broadcast radio began its revolution from AM to FM with the possibility of FM stereo broadcasting becoming a reality in 1960.
The same period saw the rise of the globe-conquering Philips creation of the compact cassette recorder.
Each of the three audio technology developments circa 1960 were analog. Since that time, turntables have seen developments ranging from a reduction to 1 gram stylus pressure and accompanying precision groove tracing tonearms which eventually graduated to the biotracer (electronically controlled tonearm) by means of the collaborative adoption of the servo, BSL (brushless and slot-less linear) and quartzlocked DC servo motors. The DC Servo and hysteresis motors had been developed for regular and quieter rotation which reduced rumble, and various developments in facets
of the turntable ranging from a variety of insulators and turntable platters, as well as the high quality and heavyweight frames were manufactured. With its humble origins of 2 minutes of 78 rpm play time on each side, a frequency range of 200 Hz-300 kHz and a dynamic range of 18db, the LP record eventually enjoyed a frequency range of 30 Hz -15 kHz and a 65db stereo dynamic range.
Between the 1960s and the 1970s, constant attempts to improve the signal of both record production and broadcast were at the forefront of manufacturer agendas, as the main instrument used in both instances was the open reel tape recorder. The recording tape, reproduction heads and tape path drive mechanism with an ultimate 0.04% wrms at 19 cms and 0.02% wrms at 38 cms of wow and flutter were the most significant developmental aims. In addition to this, the available signal-to-noise ratios enjoyed improvements with the introductions of Dolby, dBx and other such compression/expansion systems.
Gradually in the wake of the release of the previously mentioned compact cassette recorder in1963 (domestic use of which ensured that the recorder had become an essential acquisition for everyone from the music lover to the employer of the Dictaphone), the advancing developments enjoyed by the open reel tape recorder were eventually applicable to the domestic versions as well.
Although many progressive developments were enjoyed in the analog age of invention and advancement, the end of the 1970s saw the industry unable to progress further in the field without dramatically having to increase the resulting cost of equipment through further performances of development. As the analog means of recording and broadcast was the foundation determining the resulting distortion, quality
and dynamic range by the processing equipment, Analog reproduction techniques has just about reached the limits of their characteristics. (Maes & Vercammen. 2001, pg.5)
In the analog chain of reproduction, original dynamic range is still adversely effected. Aspects of this system including distortion, frequency response and signal-tonoise ratio are afflicted by similar limits as a result, simply, of involved analog processes. It was generally for these reasons that manufacturers turned to the possibilities and opportunities presented by Digital Audio Technology.
May 1967 was the year that saw the first presentation of Pulse Code Modulated (PCM) Digital Audio by the Japanese Broadcasting Corporation, NHK who employed a recording medium such as a 1-inch, two head, helical scan VTR, whereby it was generally conceded that the quality of the audio was both superior and incomparable to the previous analog efforts largely due to the fact that the previous limitations were no longer in effect. Explained briefly here, In PCM recording...the original bit value pattern corresponding to the audio signal, and thus the audio signal itself, can be fully recovered, even if the record signal is distorted by tape non-linearities and other causes. ( Maes & Vercammen. 2001, pg. 8)
Following this advancement came the PCM-1 (an analog Hi-Fi system with attached Digital Audio Processor for reproduction) which was initially marketed by Sony in 1977. In the year following that, a professionally applicable Digital Audio Processor called the PCM-1600 was marketed, and in the April of 1978, the AES (Audio Engineering Society) accepted the use of a 44.056 kHz sampling frequency as used in both aforementioned PCM models.
In October 1982, the release of the compact disc into the market was to enjoy a wide-spread and steadily growing popularity and was to become an essential addition to the audio recording and reproduction inventory.
APPENDIX C
The Nyquist-Shannon Sampling Theorem
The process of converting a signal (being a function of continual time and space) into a numeric sequence is called sampling. states: If a function x(t) contains no frequencies higher than B hertz, it is completely ! ! determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart. The Shannon version of the theorem
In essence, the theorem shows that a bandlimited analogue signal that has been sampled can be perfectly reconstructed from an infinite sequence of samples if the sampling rate exceeds 2B samples per second, where B is the highest frequency in the original signal. If a signal contains a component at exactly B hertz, then samples spaced at exactly 1/(2B) seconds do not completely determine the signal, Shannon's statement notwithstanding. This sufficient condition can be weakened. More recent statements of the theorem are sometimes careful to exclude the equality condition; that is, the condition is if x(t) contains no frequencies higher than or equal to B; this condition is equivalent to Shannon's except when the function includes a steady sinusoidal component at exactly frequency B. The theorem assumes an idealization of any real-world situation, as it only applies to signals that are sampled for infinite time; any time-limited x(t) cannot be perfectly bandlimited. Perfect reconstruction is mathematically possible for the idealized model but only an approximation for real-world signals and sampling techniques, albeit in practice often a very good one.
The theorem also leads to a formula for reconstruction of the original signal. The constructive proof of the theorem leads to an understanding of the aliasing that can occur when a sampling system does not satisfy the conditions of the theorem. The sampling theorem provides a sufficient condition, but not a necessary one, for perfect reconstruction. The field of compressed sensing provides a stricter sampling condition when the underlying signal is known to be sparse. Compressed sensing specifically yields a sub-Nyquist sampling criterion.
APPENDIX D
Notes on Discussion With Dr. Tim Brookes

The following notes are mere pointers jotted down during an arranged discussion with Dr. Brookes at the University of Surrey on the 29th of January 2013 during which the following issues were broached:
The range of human hearing - what precisely is not being heard:
- Suggested research into high frequency. - High-low sampling rate. - Sampling theorem, particularly concerning signal chopping. - Note: Dr. Brookes is of the opinion that Binary based applications are sufficient in the reconstruction of analogue and digital audio in terms of acceptability in the discrepancies that may not be heard by the human ear.
Spacial Sound Reconstruction:
- Not digital, but a series of isolated sounds. - Space in small slices compares with real space up to certain frequency.
Suggested research into Wave Field Synthesis:
- Spacial resolution problems. - Issues of reflection and obstacles.
Note: This discussion leads me to the question: Are frequencies not heard by the human ear integral in anatomy to sounds that ARE heard?
APPENDIX E
Online Discussion With Phil Thornton
This conversation took place on Facebook on the 22nd of January, 2013.
Ramsay: Have now spent two whole days just categorizing and filing preparatory dissertation research and have yards more research to do yet on a subject there is no direct information compiled on or written about.....anywhere in the universe. whole encyclopedia of Harvard referencing when I'm finally done. Thornton: Really want to know what its about now ..... Ramsay: Ha ha Phil, the working question is WHAT NEW DEVELOPMENTAL TECHNOLOGIES MIGHT THEORETICALLY ALLOW ANALOGUE AUDIO TO EXPERIENCE THE BENEFITS OF THE BINARY DOMAIN WITHOUT THE FILE SEGMENTATION CAUSED BY SAMPLING? I'm basically having a go at re-designing audio technology and processing for the world of tomorrow - just a little light work here and there LOL Thornton: ...and you choose this question? Ramsay: LOL. Yes, its all my fault, I admit. Thornton: Digital control of analog synths? No file segmentation there but of course Hello
control resolution could be regarded as the same problem. Ramsay: Absolutely - so the answer is, unfortunately for me, to eradicate binary
completely (it's coming anyway) and to re-design the entire system from capture/means to processing to playback. In order to eradicate binary PCM, a reworking of handling of the sound source is necessary by any means available or conceivable. Thornton: Wax Drum?
Ramsay: LOVE IT! Will have to quote you ha ha. Thornton: So are we thinking of some kind of fractal capture? Ramsay: Well funny you mention - it's along those lines but way more complex
although I'm certain that holographic storage (holographic not as in current holographic but more literal) will also come into play. I'm designing a very intricate and advanced microphone system already but I'll just have to send you a copy of the paper once it's written because I couldn't go into it all here even if I tried LOL. Thornton: It has to be time based - magnetic tape resolution increases with speed in
the same way as higher sample rate in PCM - a mike has to mirror sound waves in some way, they have a frequency - so the output must also have a frequency? Ramsay: Indeed yes! The microphone is designed to imitate the environment on a nanoscale in order to faithfully recreate what is captured. In fact the entire system - even 'hard-drive' will be frequency based and primarily designed only to pass on and handle incoming information rather than directly store it. Basically, technology will perform more organically and bow to the ways of sound rather than sound submitting to the machinery. Thornton: I will look forward to reading the finished work! - good luck! Ramsay: So will I LOL I'm also mystified as to where it's all to land up! Thanks for your thoughts and the chat and the good luck - always a pleasure. If you don't mind I will quote you on a couple of thoughts here in the paper to introduce or strengthen points.
APPENDIX F
Interview With Adi Winman
Q 1:
! In your experience/knowledge, what do you consider to be the most
beneficial audio production process from capture to master using both analogue and digital!technology? Winman: Beneficial? Mmm. I think the microphone, as "flawed" as the technology is in terms of energy transfer, the transducer has enabled us to hear sound as "close" to what we hear via current technology. It is also cheap so has prevailed over the years as a "good enough" technology to record and play back audio. Other technologies have been invented for this task many times, but have not prevailed due to 1. Cost 2. The status quo 3. Perception of sound and the listeners experience.! ! ! Q 2: What would you consider to be the benefits of analogue of digital and vice!versa. Winman: Analogue has it limits, but they seem to be moving targets i.e. one can push it a bit more. Digital is finite so has defined limits. Yes you can increase sampling rates & bit rates etc but they are still finite. Analogue however is still based on electrical signals produced by transducer technology which could be much more efficient. That said, we (humans) find it hard (generally) to notice any differences so the status quo prevails.! ! ! Q 3: In the production of game sound, what considerations are present that may not be so in other areas of audio recording. Winman: Mmm? The creation, capture and editing is almost identical, but the
implementation and playback of audio is fundamentally different. It is non linear. It is based on interaction at any one moment in time and so needs to be flexible, interesting and often random to mimic player input and gameplay. This gets very complicated when real world scenarios are emulated in a 3D sense with surround for example. Real world models
are imposed and modeled onto that game. FPS games do this a lot. Each input creates a code triggering a sound event or events that have to be positioned etc...i.e. input + maths and programming + audio = game audio. ! Q 4: In the history of your engineering experience, how has advancement in both technologies changed how you/ the industry approaches music production? Winman: Digital has made things simpler to manipulate sound in many ways. However, some say that it has also meant that too much emphasis is placed on absolute perfection. This has always been the case of course, it's just that the goal posts and expectations have moved slightly often meaning that things sound different, but not necessarily "better". ! Q 5: If audio processing was capable of eradicating PCM and therefore naturalizing the sound! as we perceive it, what would you imagine to be the benefits of this and the possible applications of it? Winman: I think this ultimately depends on playback and perception. Capturing a
"flawless" sound is one thing but playing back so that it sounds "as if you were there" is another. In my honest opinion, this is the greater challenge. It's an interesting concept but I think one CANNOT be considered without the other.! ! Q 6: As an exercise purely for creative thinking, how would you consider technology to explore possible means of making a compact disc behave more like a vinyl? Winman: Metadata on the CD that stores/emulates analogue traits. The user chooses the sound/vibe on their amp?
APPENDIX G
Weber-Fechner Law
Webers law
This general relationship between the initial intensity of something and the smallest detectable increment is exactly what Weber noticed and formalized into Webers Law. The discrimination threshold, or the threshold for detecting an increment in the quantity or intensity of something, changes depending on how much there is before we add the increment. Webers law is a hypothesis about how this threshold change happens. Lets call the initial intensity (the intensity before adding the increment) I, and lets call the amount needed to detect a difference !I. (Often, the specific letter that researchers choose bears some relation to the stimulus dimension being discriminated if we are interested in duration discrimination, we might have D and !D.) In a discrimination experiment, then, we are interested in measuring !I as a function of I. That is, we want to find the discrimination threshold !I such that a stimulus with an intensity I+!I is just discriminable from a stimulus of intensity I. Webers law characterizes how the !I we measure depends on I. It states that: #I = KwI for some constant Kw. The constant Kw is called the Weber " " " " " " " " " " " " " " " Fraction. You may notice the similarity between this equation and the equation for a straight line, y = m * x + b. Webers law expresses the equation for a straight line, with the slope m being the Weber Fraction, Kw and the y -intercept b being zero . Likewise, I plays the part of x and #I plays the part of y. (Landy, 2012)
Thus, if Webers law holds, we expect the data to graph as a straight line through the origin, as shown in Figure 1.
Fechners Law
Fechners law provides an explanation for Webers law. Fechners explanation has two parts. The first part is that two stimuli will be discriminable if they generate a visual response that exceeds some threshold. The second part is that the visual response R to an intensity I is given by the equation R = log(I ). This relation is plotted in the next graph. Suppose we call the change in response necessary for discrimination one. Then by Fechners law the two intensities indicated by arrows will be just discriminable. The discrimination threshold !I is given by the distance on the x-axis between I and I+!I.
If you use Fechners law and the graph to find the discrimination threshold for a stimulus more intense than the I shown, you will you will discover that the discrimination threshold is larger. This is consistent with Webers law, which predicts that the discrimination threshold grows as we increase the base intensity I.
APPENDIX H
A Neural Science Overview of Human Hearing
Hearing, a key sense in human communication, commences with the ear's capture of sound. Mechanical energy flows through the middle ear to the cochlea, where it makes the elastic basilar membrane vibrate. An array of 16,000 hair cells detects each frequency component of a stimulus, transduces it into receptor potentials, and encodes it in the firing pattern of eighth-nerve fibers. The complex auditory pathways of the brain stem mediate certain functions, such as the localization of sound sources, and forward auditory information to the cerebral cortex. Here, several distinct areas analyze sound to detect the complex patterns characteristic of speech.
Figure The central auditory pathways extend from the cochlear nucleus to the auditory cortex
APPENDIX I Interview/Discussion Setup Emails
APPENDIX J
Survey 1 Online Questions
Survey 2 Online Questions
APPENDIX K
Perrson 2005 Survey: Producers Answers
QUESTION 1. How long have you been working? 2. Do you produce major label artists or independent artists or both? 3. What genres or styles of music do you record? 4. Do you believe that the computer technology changed the way we record music today?
Answer: Participant 1 13 years
Both
Mostly independent Powerpop, Singer/ songwriter, Retro Rock
Both
Soul/RnB, Rock, Country
All (Except White Power)
Yes
Yes
Yes
5a. Do you prefer analog or digital for recording instruments?
Analogue, but rarely is there a budget for it.
Analogue
Both. Analogue for warmth, rst in the chain. Digital for ease of use, editing, collaboration, saves time.
QUESTION
Answer: Participant 1
Answer: Participant 3 Both. Analogue for warmth, rst in the chain. Digital for ease of use, editing, collaboration, saves time. Both. Analogue for warmth, rst in the chain. Digital for ease of use, editing, collaboration, saves time. Both. The les are digital but then are put through analog EQs, compressors back to digital
5b. Do you prefer analog or digital for recording vocals?
Analogue, but rarely is there a budget for it.
Digital
5c. Do you prefer analog or digital for mixing?
Prefers Analogue, but rarely is there a budget for it.
Analogue
5d. Do you prefer analog or digital for mastering?
Almost always digital
Both
6. On average, has the total length of time for recording projects (from preproduction through nal product) increased or decreased since the intro of digital technology? 7. Same question as nr 6, but on negotiations with companies/labels? 8. Same question as nr 6 but in preproduction? 9. Same as before but on rehearsing with musicians?
Increased
Increased
No real change
Not applicable
Same
More
More
Same
Same
Not applicable
More
Same
QUESTION 10. Same as before but on talking and planning with artist regarding sound, aesthetic approach, etc. 11. Same as before but on talking and planning with the engineer? 12. Same as before but on tracking/recording vocals and instruments? 13a. Same question as before but on communicating with performers about performance during recording sessions?
Not applicable
More
Same
More
More
Same
Basic tracking: More. Overdubs: Less
More
Same
Same.
Same.
Same.
13b. Has what you talked about changed? If so how?
With performers, no real change. With engineers the new talk is usually more directed towards nding a way of making the recording sound warm
Yes, in some cases. If you work with mouse clicking it can get too much focus on copy/paste. But with analog recording, withouth clicking it is just as it was before.
No, its the same.
14. Same question as nr 6, but on editing?
More
More
Same, but different.
QUESTION
15a. same as above, but on mixing? 15b.Have the people present in sessions changed since the change to digital? If yes, who was there before and after? 15c. Has what you talk about with the engineer during mixing sessions changed? If so, how? 16. Same question as nr 6 but on Mastering 17. Do the choices of music consumers directly affect what types of projects you work on and how you work on in the studio? If yes, how, in what way? 18. Are you able to work with less established artists or develop new artists more or less since the change to digital? 19. On average are your budgets larger or smaller?
Same.
More
Did not answer
Not the way I work
No
No
Yes, in the way of choosing plug-ins rather then using outboards.
Not applicable
Not really
Same
Same
Same
Not really
No
No
Absolutely more. Its cheaper; therefore there are more small sessions.
Same
Yes
Much smaller since 2001 but not necessarily due to digital
Smaller
Smaller
QUESTION 20. Is more or less money budgeted for time to develop a unique sound for artists/projects? 21. Do you spend more or less time developing a unique sound for artists/projects?
Less money
Less
Less money
About the same. I feel you have to give each artist a unique sound. No. Not in the way I see it. I think that the overall climate regarding the entertainment industries has changed. Music has a lot more competition from games, lm, sports.
More
The same amount
22. Has Internet technology (other than P2P le sharing) affected music production?
Yes. I usually mix on distance and send mp3 versions of the mix, and I can get quick replies on possible comments, so that everything is of satisfaction.
No. A little, I can send pre-mixes over the Net instead of mailing a CD, and the ability to chat with video has changed sessions were I collaborate globally
23. Has P2P le sharing impacted budgets?
Yes, absolutely
Did not answer
No. Thats just propaganda from the Major labels!
QUESTION
Answer: Participant 2 Yes. A lot of mixing is to be done when bands records in home or in a small pc based studio. Everything to keep the costs done. Six years ago you could take 5.000 SEK each day on mixing. These days half of that
24. Has the dropping costs of production tools and increased in home recordings impacted your job? Yes or No
Yes
Yes
25. Are there more or fewer music producers today?
A lot more
Probably more. If you have a PC and the interest you can work with fewer things, which makes it possible to get working with music out to a wider scale of people. Its cheap and you dont need a studio on 200 m2 that costs a lot of money
Far too many!
26. Has the number of personnel in the average recording session increased or decreased in studios over the years?
Decreased
In my case, no difference
It depends
QUESTION
Answer: Participant 2 No difference. Ive always taken in the role that each project demands of me. Sometimes only as an engineers, mostly both as an engineer and producer, and make sure that the people involved really get what they want I can only see it from my point of View and digital technology had nothing to do with it For me its always been a shared interest for the technical and the musical, and through the years Ive realized that it is the musical side that always is the most important one. ...Digital curiosity mostly just ruined the music. Good thing one nally woken up
27. What roles have been added or removed?
Tape operator, sound engineer.
Added: Programming, Pro Tools, editing, and system operator
28. Do you see the role of the producer changing since the adoption of digital?
Yes. A lot more songwriters also become producers
Yes
29. Do you see the role of the producer and engineer emerging?
Yes
Its already happened.
Perrson 2005 Survey: Engineers Answers
GLOSSARY OF TERMS
Sampling: Given a continuous signal x(t) to be sampled, and a sample interval T, the sampled version of ! ! x is simply the continuous version of x taken at integer intervals of T:
! Logarithmic Scale
The level, L, of any sound may ! then be ! expressed (in units of decibels sound pressure level, ! ! or dB SPL) as:
" "
" "
where P, the magnitude of the stimulus, is given as the root mean square of the sound pressure (in units of pascals, or Pa). (Kandel, Schwartz & Jessel, 2000)
Quantization Quantization represents the amplitude component of the digital sampling process. It is used ! ! ! ! ! ! to translate the voltage levels of a continuous analogue signal (at discrete sample points over time) into binary digits (bits) for the purpose of manipulating or storing audio data in the digital domain. (Huber, 2010)
Transmission Theory If eight bits are allowed for the PCM sample, this gives a total of 256 possible values. ! ! ! ! ! ! ! ! PCM assigns these 256 possible values as 127 positive and 127 negative encoding levels,plus the zero-amplitude level. (PCM assigns two samples to the zero level.) These levels are divided up into eight bands called chords. Within each chord is sixteen steps. (Poland, 2008)
Figure 3: PCM Quantization levels - Chords and Steps
GLOSSARY Flash ADC ! ! ! ! ! ! ! ! ! ! ! ! A type of analog-to-digital converter that uses a linear voltage ladder with a comparator at each "rung" of the ladder to compare the input voltage to successive reference voltages. Often these reference ladders are constructed of many resistors; ! however modern implementations show that capacitive voltage division is also possible. The output of these comparators is generally fed into a digital encoder which converts the inputs into a binary value (the collected outputs from the comparators can be thought of as a unary value).
Resolution " " " " " " " " " " " " " " " " " " " " The resolution of the converter indicates the number of discrete values it can produce over the range of analog values. The resolution determines the magnitude of the quantization error and therefore determines the maximum possible average signal to noise ratio for an ideal ADC without the use of oversampling. The values are usually stored electronically in binary form, so the resolution is usually expressed in bits. In consequence, the number of discrete values available, or "levels", is assumed to be a power of two. For example, an ADC with a resolution of 8 bits can encode an analog input to one in 256 different levels, since 28# =# 256. The values can represent the ranges from 0 to 255 (i.e. unsigned integer) or from $128 to 127 (i.e. signed integer), depending on the application.
Oversampling " " " " " " " " " " " " " " " " " " Signals are often sampled at the minimum rate required, for economy, with the result that the quantization noise introduced is white noise spread over the whole pass band of the converter. If a signal is sampled at a rate much higher than the Nyquist frequency and then digitally ltered to limit it to the signal bandwidth there are the following advantages: digital lters can have better properties (sharper rolloff, phase) than analogue lters, so a sharper anti-aliasing lter can be realised and then the signal can be downsampled giving a better result a 20-bit ADC can be made to act as a 24-bit ADC with 256% oversampling the signal-to-noise ratio due to quantization noise will be higher than if the whole available band had been used. ""
ADC Types ! ! " " " " " " " " " " " " " " " " " " " " " " " ! ! ! ! ! ! ! ! ! ! " " " " " " " " " " " " " " " " " " " " " " " ! ! ! ! ! ! ! ! A direct-conversion ADC or ash ADC has a bank of comparators sampling the input signal in parallel, each ring for their decoded voltage range. The comparator bank feeds a logic circuit that generates a code for each voltage range. Direct conversion is very fast, capable of gigahertz sampling rates, but usually has only 8 bits of resolution or fewer, since the number of comparators needed, 2N 1, doubles with each additional bit, requiring a large, expensive circuit. ADCs of this type have a large die size, a high input capacitance, high power dissipation, and are prone to produce glitches at the output (by outputting an out of-sequence code). Scaling to newer submicrometre technologies does not help as the device mismatch is the dominant design limitation. They are often used for video, wideband communications or other fast signals in optical storage. A successive-approximation ADC uses a comparator to successively narrow a range that contains the input voltage. At each successive step, the converter compares the input voltage to the output of an internal digital to analog converter which might represent the midpoint of a selected voltage range. At each step in this process, the approximation is stored in a successive approximation register (SAR). For example, consider an input voltage of 6.3 V and the initial range is 0 to 16 V. For the rst step, the input 6.3 V is compared to 8 V (the midpoint of the 016#V range). The comparator reports that the input voltage is less than 8# V, so the SAR is updated to narrow the range to 08# V. For the second step, the input voltage is compared to 4 V (midpoint of 08). The comparator reports the input voltage is above 4# V, so the SAR is updated to reect the input voltage is in the range 48#V. For the third step, the input voltage is compared with 6 V (halfway between 4 V and 8 V); the comparator reports the input voltage is greater than 6 volts, and search range becomes 68#V. The steps are continued until the desired resolution is reached. A ramp-compare ADC produces a saw-tooth signal that ramps up or down then quickly returns to zero. When the ramp starts, a timer starts counting. When the ramp voltage matches the input, a comparator res, and the timer's value is recorded. Timed ramp converters require the least number of transistors. The ramp time is sensitive to temperature because the circuit generating the ramp is often just some simple oscillator. There are two solutions: use a clocked counter driving a DAC and then use the comparator to preserve the counter's value, or calibrate the timed ramp. A special advantage of the ramp-compare system is that comparing a second signal just requires another comparator, and another These are the most common ways of implementing an electronic ADC:
! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! !
! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! !
register to store the voltage value. A very simple (non-linear) ramp-converter can be implemented with a microcontroller and one resistor and capacitor.[11] Vice versa, a lled capacitor can be taken from an integrator, time-to-amplitude converter, phase detector, sample and hold circuit, or peak and hold circuit and discharged. This has the advantage that a slow comparator cannot be disturbed by fast input changes. An integrating ADC (also dual-slope or multi-slope ADC) applies the unknown input voltage to the input of an integrator and allows the voltage to ramp for a xed time period (the run-up period). Then a known reference voltage of opposite polarity is applied to the integrator and is allowed to ramp until the integrator output returns to zero (the run-down period). The input voltage is computed as a function of the reference voltage, the constant run-up time period, and the measured run-down time period. The run-down time measurement is usually made in units of the converter's clock, so longer integration times allow for higher resolutions. Likewise, the speed of the converter can be improved by sacricing resolution. Converters of this type (or variations on the concept) are used in most digital voltmeters for their linearity and exibility. A delta-encoded ADC or counter-ramp has an up-down counter that feeds a digital to analog converter (DAC). The input signal and the DAC both go to a comparator. The comparator controls the counter. The circuit uses negative feedback from the comparator to adjust the counter until the DAC's output is close enough to the input signal. The number is read from the counter. Delta converters have very wide ranges and high resolution, but the conversion time is dependent on the input signal level, though it will always have a guaranteed worst-case. Delta converters are often very good choices to read real-world signals. Most signals from physical systems do not change abruptly. Some converters combine the delta and successive approximation approaches; this works especially well when high frequencies are known to be small in magnitude. A pipeline ADC (also called subranging quantizer) uses two or more steps of subranging. First, a coarse conversion is done. In a second step, the difference to the input signal is determined with a digital to analog converter (DAC). This difference is then converted ner, a ! nd the results are combined in a last step. This can be considered a renement of the successive-approximation ADC wherein the feedback reference signal consists of the interim conversion of a whole range of bits (for ! example, four bits) rather than just the next-most signicant bit. By combining the merits of the successive
! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! !
! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! !
approximation and ash ADCs this type is fast, has a high resolution, and only requires a small die size. A sigma-delta ADC (also known as a delta-sigma ADC) oversamples the desired signal by a large factor and lters the desired signal band. Generally, a smaller number of bits than required are converted using a Flash ADC after the lter. The resulting signal, along with the error generated by the discrete levels of the Flash, is fed back and subtracted from the input to the lter. This negative feedback has the effect of noise shaping the error due to the Flash so that it does not appear in the desired signal frequencies. A digital lter (decimation lter) follows the ADC which reduces the sampling rate, lters off unwanted noise signal and increases the resolution of the output (sigma-delta modulation, also called delta-sigma modulation). A time-interleaved ADC uses M parallel ADCs where each ADC samples data every M:th cycle of the effective sample clock. The result is that the sample rate is increased M times compared to what each individual ADC can manage. In practice, the individual differences between the M ADCs degrade the overall performance reducing the SFDR.[14] However, technologies exist to correct for these time-interleaving mismatch errors. An ADC with intermediate FM stage rst uses a voltage-to-frequency converter to convert the desired signal into an oscillating signal with a frequency proportional to the voltage of the desired signal, and then uses a frequency counter to convert that frequency into a digital count proportional to the desired signal voltage. Longer !integration times allow for higher resolutions. Likewise, the speed of the converter can be improved by sacricing resolution. The two parts of the ADC may be widely separated, with the frequency signal passed through an opto-isolator or transmitted wirelessly. Some such ADCs use sine wave or square wave frequency modulation; others use pulse-frequency !modulation. Such ADCs were once the most popular way to show a digital display of the status of a remote analog sensor.
Pseudorandom ! ! ! ! Of, relating to, or being random numbers generated by a denite, nonrandom computational process.
Femtosecond
! !
One quadrillionth (10^-15) of a second; one thousandth of a picosecond
"

New Technologies in Development For An Approaching Nonbinary Domain Which Theoretically Allow Analogue Audio To Surpass Current Digital Sampling Disadvantages.

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GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Creative Music Production The Academy of Contemporary Music

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

12. GLOSSARY OF TERMS....................................................................................................pg. 145

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

List of Figures, Tables & Charts

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

explaining my creative thoughts back to me in technical translations - a very enviable ability.

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

1.4 Summary of Aims and Objectives

The main issues/questions addressed in this investigation are:

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Acronym PCM WFS PNS ADC SDM BCD QC PS PADC

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

perceived by the human ear.

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

What is then known as aliasing occurs.

This is caused by the sample rate of the

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Figure 3: Pulse Code Modulation (See Glossary)

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

To support this unideal solution, a preferable second method uses complex

Figure 4: Example of interleaved error correction

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

types and functions of Analogue To Digital Conversions.

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Figure 5: ADC Types, Resolution, Speed & Cost Comparison

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

(See Appendix F, pg. for Interview With Adi Winman)

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Figure 10: Frequency & Sampling Quality Comparison

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

largely in knowledge of the language of qubits, which are probability based.

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

With regards to image to sound mapping ( as considering Beamforming and

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Producer Composer/Writer Sound Engineer Technician 0 22.5 45 67.5 90

Chart 1: Survey 1&2 Industry Function Response Percentage Levels

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

GARETH RAMSAY - CMP-501 - RESEARCH METHODS - TRIMESTER 6 - 2013

Yes No Dont Know 0 12.5 25 37.5 50

Chart 3: Survey 1&2 Pro Non-Binary Response Percentage Levels

0% 20% 40% 60% 80% 100% 0 12.5 25 37.5 50