the 'Programme at a Glance'
giving the complete schedule of conference events and venues.
|Monday, June 25
|In the Listening Room, throughout the day, will be demonstrations
of high resolution audio by Linn Audio
|9:00 – 10:30 a.m.
|Paper Session 1 — High Resolution Recording Issues
Introductory Remarks - Mark Sandler, Head of the Centre for Digital
Music at Queen Mary, University of London, Josh Reiss, General Chair
1-1 Creating and Delivering High-Resolution Multiple 5.1
Surround Music Mixes—Mark Waldrep
The future of music reproduction is multichannel and high-resolution.
However, the limited commercial success of recent high-resolution,
surround audio optical formats demands that producers, engineers,
and recording companies define exactly what is meant by high-resolution,
surround music and that consumers experience content that demonstrates
its full potential. Engineer and educator Mark Waldrep has built
a unique library of award-winning, high-resolution, surround music
titles that include multiple surround mixes created from dozens
of stereo microphone pairs. His hybrid recording methodology brings
studio techniques and “live” concert approaches together
in a unique way.
1-3 Precision Measurement of ADC Effective Number of Bits
Using Multitones—Raymond Belcher, Jonathon Chambers, Cardiff
University, Cardiff, UK
A pure sine wave is the conventional test signal used for measuring
the effective number of bits (ENOB) of an analog to digital converter
(ADC) integrated circuit. Sine wave testing of ADCs weights the
result so that peak level distortion is highlighted, and this is
known to be less appropriate for audio. This paper demonstrates
that a multitone test signal can be used to give most weighting
to the central region and therefore produce an ENOB more relevant
to audio, using less samples and without the difficulty of generating
pure test signals.
|Monday, June 25 11:00
|Paper Session 2 — Perception
2-1 Which of the Two Digital Audio Systems Meets Best
with the Analog System?— Wieslaw Woszczyk,1 Jan Engel,2 John
Usher,1 Ronald Aarts,3 Derk Reefman3
1McGill University, Montreal, Quebec, Canada
2Centre for Quantitative Methods CQM BV
3Philips Research, Eindhoven, The Netherlands
In this listening test, two digital audio systems (B and C), and
one analog system (A) were tested by 10 test persons who listened
to a surround sound scene “live” (without recording).
The main question to be answered was: “Which of the two digital
systems meets best with the analog system?” Both digital versions
had 24-bit dynamic resolution but differed in sampling rate with
which the analog signal was sampled. One version (C) was sampled
with a CD rate of 44.1 kHz, the other (B) 8 times faster. There
were also two test conditions, where in one condition there was
a bandwidth cut off at 20 kHz instead of the 100 kHz that was possible
with special 100 kHz microphones and added super-tweeters. For each
subject, the experiment was replicated six times, in each of the
two conditions. The outcome of each experiment was a 0 or 1, where
the 1 means that the, technically best, digital system B has been
chosen as meeting the analog quality. The paper describes the test
and the outcome.
2-2 A Comparative Study of the Performance of Spatialization
Techniques for a Distributed Audience in a Concert Hall Environment—Gavin
Kearney, Enda Bates, Dermot Furlong, Frank Boland, Trinity College,
The performance of various spatialization techniques is evaluated
for a distributed audience using the non-ideal speaker arrangements
found in medium-sized concert halls. The spatialization methods
assessed in terms of their localization accuracy and listener envelopment.
The data is presented by comparison of empirical binaural measurements
and perceptual listening tests to simulations of the speaker arrangements
in an equivalent acoustically modeled environment.
2-3 Audio Quality on the Air in DAB Digital Radio in Norway—Sverre
Holm, University of Oslo, Oslo, Norway
We have analyzed the audio quality as recorded on the air for stations
in the Norwegian DAB network. When the capacity is fully utilized,
most stations with music transmit at a capacity of 128 kbps using
MPEG1 layer II audio coding. By analyzing the mono component and
the stereo component we have found that the audio is characterized.
by a smeared stereo image due to the use of the heaviest from of
joint stereo coding. There is also a lack of treble as the upper
limit is usually 14 kHz. The result is a loss of brightness and
a veiled sound stage which is particularly noticeable to young people.
|Monday, June 25 12:30
|Poster Session 1
Posters will be presented during the
lunch breaks. Posters may also be available on multiple days. See
on-site conference program for details.
Design of Circular Microphone Arrays: The Analysis of Spatial Aliasing
Error and Microphone Placement Error in Circular Microphone Arrays—Abhaya
Parthy, Craig Jin, Andre van Schaik, The University of Sydney, Sydney,
We present a methodology for analyzing the spatial aliasing error,
and the microphone placement error of both an open, or unbaffled,
circular microphone array, and a circular microphone array that
is baffled, ideally, by an infinite-length rigid cylinder. The methodology
describes the practical design of a circular microphone array, with
a specified number of microphones, which satisfies a specified threshold
noise-to-signal ratio, beamforming order, and frequency range. Two
practical designs have been constructed, and results of their analysis
P1-2 The Localization of a Sound Source in a Reverberant
Room Using Arrays of Microphones —Simon Roper,
Timothy Collins, The University of Birmingham, UK
This work is aimed at the problem of continuously measuring and
predicting the impulse response within a reverberant room using
sources-of-opportunity. The target application is that of compensating
a sound reproduction system for variations in room acoustics and
the non-ideal placement of the loudspeakers. The prediction of the
impulse responses corresponding to the locations of an arbitrary
number of listeners requires knowledge of the room geometry and
the acoustic impedance of the boundaries. The room geometry may
be obtained by estimating the image locations of a source. This
has been addressed by the use of small arrays, with novel geometry,
and processing both angular and temporal measurements. Preliminary
practical results are presented and compared to theoretical predictions.
P1-3 Modeling and Control of Class-D Power Amplifiers for
Vented-Box Loudspeaker Systems—Fran Gonzalez-Espin,1
Emilio Figueres,2 Gabriel Garcera, 2 Jesus Sandia2
1VMB, Valencia, Spain
2University Politécnica de Valencia, Valencia, Spain
The purpose of this paper is to analyze the stability of closed
loop switching power amplifiers using an accurate model of a vented-box
loudspeaker system instead of the commonly used resistive model.
The model takes into account the electrical-mechanical-acoustical
parameters of the transducer as well as the voice-coil loudspeaker’s
nonlinear behavior, avoiding stability problems when trying to compensate
for the output filter stage, thus obtaining better THD figures.
This model has been experimentally tested and then used to design
a regulator for Voltage-Mode Control method by means of MATLAB software.
The regulator has been tested using a full-bridge power converter
along with the proposed model of the transducer.
P1-4 Improved Psychoacoustic Noise Shaping for Requantization
of High-Resolution Digital Audio—Christian R.
Helmrich, Martin Holters, Udo Zölzer, Helmut-Schmidt University
The popularity of high-resolution digital audio systems has renewed
the interest in psychoacoustically optimized wordlength reduction.
In this paper we examine recent approaches in fixed (time invariant)
and signal-adaptive (time variant) psychoacoustic noise shaping.
For the fixed case, we identify problems occurring when equal-loudness
contours such as the threshold of hearing defined in ISO standard
226 are used as the basis for psychoacoustic noise shaping. For
the signal adaptive case, we propose a noise shaping solution based
on work by Verhelst and De Koning with improvements in the computation
of the time-variant noise shaping filter. The paper concludes with
a comparative evaluation of fixed and adaptive noise shapers based
on listening tests in different environments.
P1-5 The Phase Amplitude Control Bit Stream Adder: A One-Bit
Processing Structure for Phasor Manipulation of Oversampled Sinusoids—Enrique
Perez Gonzalez, Joshua Reiss, Queen Mary, University of London,
This paper introduces a one-bit signal processing structure called
the phase amplitude control bit stream adder. The proposed method
is a digital filter structure, which is capable of controlling amplitude
and phase over a one-bit sine wave without the need of directly
multiplying the one-bit stream with a floating-point constant. It
has applications in precision variable oscillator control and in
oscillator bank additive synthesis reconstruction models. The research
also explores a method for one-bit oscillator bank synthesis, without
using intermediate multi-bit stages.
|Monday, June 25 14:30 – 15:30
|Paper Session 3 — Processing, Manipulation,
and Preparation of High-Resolution Signals
3-1 Segmented Dynamic Element Matching using Delta-Sigma
Modulation—Ivar Løkken, Anders Vinje, Trond
Sæther, Norwegian University of Science and Technology, Trondheim,
In multibit delta-sigma digital-to-analog converters (DACs), the
distortion from physical element mismatch can be spectrally shaped
using dynamic element matching (DEM). A problem with all DEM schemes
is that the complexity increases very rapidly with the number of
levels in the DAC. To reduce DEM complexity for DACs with many bits,
DEM segmentation using a dedicated segmentation delta-sigma modulator
(DSM) has previously been suggested. Published segmentation DSMs
have usually been first-order error feedback designs, to maximize
the DEM complexity reduction and to minimize the analog overhead.
In this paper high-order segmentation DSMs will be investigated
and improved solutions proposed.
3-2 Energy Balance Decision Threshold in SDM Systems
Malcolm Hawksford, University of Essex, Essex, UK
Conventional SDM employs an amplitude comparator to convert sampled
multilevel signals to a binary level bit stream where negative feedback
is used to shape the output distortion spectrum. A modified threshold
decision process is proposed that takes a holistic view of the state
of the loop. Here a measure related to stored energy within the
loop is calculated over a short look-ahead period and then used
to select the polarity of the output code at each sampling instant.
The motivation is to improve loop stability for high order coders
especially for high amplitude input signals as encountered in switching
power amplifier applications.
|Monday, June 25 15:30-17:30
|Panel Discussion: Preparation, Archiving,
and Distribution of Hi-Res Audio
George Massenburg, GMLLLC, will chair this panel discussion. As of
press time, Jeff Levison, DTS, and Ronald Prent, Galaxy Studios, will
be part of the panel. Additional panelists to be announced.
In the Listening Room, throughout the day, will be demonstrations
of high resolution audio by Meridian Audio
|9:00 – 10:30
|Paper Session 4 — Synthesis and
4-1 Perceptual Investigation into Envelopment, Spatial
Clarity, and Engulfment in Reproduced Multichannel Audio—
Robert Sazdov, University Western Sydney - MARCS Auditory
Laboratories, Penrith South DC, NSW, Australia
Composers of electroacoustic music have engaged with 3-D sound
since the first performances of these works in the 1950s. Currently,
the majority of electroacoustic compositions continue to be presented
in 2-D. Human auditory perception is 3-D, however music composition
has not adequately exploited the creative possibilities of this
dimension. It is argued that ecologically valid perceptual experiments
are required when attempting to formulate compositional techniques
for electroacoustic music composition. Further, the paper presents
a novel research method for the perceptual evaluation of 3-D multichannel
electroacoustic music. The spatial attributes of envelopment, spatial
clarity, and engulfment, are employed to evaluate composed multichannel
3-D sound executed within a concert hall environment. Results support
various findings within the related disciplines of concert hall
acoustics and multichannel reproduced audio.
4-2 Musical Attractors: A New Method for Audio Synthesis—Eric
Nichols, Ian Knopke, Indiana University, Bloomington, IN, USA
In this paper we use mathematical tools developed for chaos theory
and time series analysis and apply them to the analysis and resynthesis
of musical instruments. In particular, we can embed a basic one-dimensional
audio signal time series within a higher-dimensional space to uncover
the underlying generative attractor. Röbel (1999, 2001) described
a neural-net model for audio sound synthesis based on attractor
reconstruction. We present a different methodology inspired by Kaplan
and Glass (1995) to resynthesize the signal based on time-lag embedding
in different numbers of dimensions, and suggest techniques for choosing
the approximate embedding dimension to optimize the quality of the
4-3 Wavelet Based High Resolution Audio Texture Synthesis—Deirdre
O’Regan, Anil Kokaram, Trinity College Dublin, Ireland
Audio (or Sound) Texture Synthesis is performed by application
of a well-known 2-D image texture synthesis algorithm to one dimension.
This nonparametric, statistical approach is used to synthesize long,
acoustically similar, high resolution audio textures from much shorter
examples of both stochastic and quasi-periodic audio samples, which
include a variety of ambient sounds (e.g., crowd noise, a baby crying).
The process employs the Dual-Tree Complex Wavelet Transform (DT-CWT)
to minimize computational load and maintain both temporal and spectral
coherency in the synthesized audio. The results of this method are
compared and contrasted with other state-of-the-art algorithms of
a similar nature.
|Tuesday, June 26 11:00 – 12:30
|Paper Session 5 — Processing, Manipulation,
and Preparation of High-Resolution Signals
5-1 Horizontal Plane HRTF Reproduction Using Continuous
Fourier-Bessel Functions—Wen Zhang, Thushara
Abhayapala, Rodney Kennedy, Australian National University, Canberra,
This paper proposes a method to reproduce the Head-Related-Transfer-Function
(HRTF) in the horizontal plane. The method is based on a functional
representation for HRTFs being a conventional Fourier series expansion
for spatial dependence and a Fourier Bessel series expansion for
the frequency components. The proposed representation can be used
to predict HRTFs at any azimuth source position and at any frequency
point from a finite number of parameters. Measured HRTFs from a
KEMAR are used to validate the fidelity and predictive capabilities
of the method. Errors between measured and modeled HRTFs are generally
less than 2 percent.
5-2 HRIR Customization in the Median Plane via Principal
Components Analysis of Head-Related Impulse Responses—Sungmok
Hwang, Korea Advanced Institute of Science and Technology; Youngjin
Park, Korea Advanced Institute of Science and Technology, Daejeon,
A principal components analysis of the entire median HRIRs in the
CIPIC HRTF database reveals that the individual HRIRs can be approximated
as a linear combination of several orthonormal basis functions.
The basis functions cover the inter-individual and inter-elevation
variations in HRIRs. There are elevation-dependent tendencies in
the weights of basis functions, and the basis functions can be ordered
according to the magnitude of standard deviation of the weights
at each elevation. We propose a HRIR customization method via tuning
of the weights of three dominant basis functions at each elevation.
Subjective evaluation results show that all subjects perceive the
elevation angles more accurately with the customized HRIRs than
the non-individualized and individual HRIRs.
5-3 The Generation of Panning Laws for Irregular Speaker
Arrays Using Heuristic Methods—Bruce Wiggins,
University of Derby, Derby, UK
In this paper an automated decoder optimization system using heuristic
methods will be presented that will be shown to be robust enough
to generate higher order Ambisonic decoders based on the energy
and velocity vector parameters as proposed by Gerzon (Gerzon, 1985).
This method is then analytically compared to Craven’s decoder
(Craven, 2003) using both energy/velocity vector and head related
transfer function based methods including analysis of head turning.
|Tuesday, June 26
12:30 – 14:00
| Poster Session 2
Posters will be presented during the lunch breaks. Posters may also
be available on multiple days. See on-site conference program for
P2-1 A New Approach for CD 16-Bit Audio to High Resolution
24-Bit Audio—Prathibha Dhanushkodi
This paper analyzes the improvement of CD audio quality by using
the differential evolution (DE) algorithm, which is a simple efficient
adaptive scheme based on vector differences. It is used for high
quality resampling of audio with the improvement of bit depth. A
small amount of colored dither has been added to improve the value
of SNR. Matlab results have been shown for resolution using this
interpolation and the increase in the dynamic range of the signal.
P2-2 A Discussion about Subjective Methods for Evaluating
Blind Upmix Algorithms—Nicolas Chétry, Grégory
Pallone, Marc Emerit, David Virette, France Télécom
R&D, Lannion, France
In this paper we discuss the problems that arise when one wishes
to evaluate the performance of blind upmix algorithms. Based on
the characteristics and spatial attributes that an ideal upmix should
exhibit, we first discuss several evaluation methodologies published
in literature. Second, we report internal listening test results
during which the performance of four upmix algorithms has been evaluated.
The dependence upon test material and listener expertise is highlighted.
In order to complement already published research works, we present
our experimental results and thoughts on this evaluation process.
P2-3 Visual enhancement using multiple audio streams in
live music performance—Rozenn Dahyot, Conor Kelly, Gavin Kearney,
Trinity College Dublin, Ireland
The use of multiple audio streams from digital mixing consoles
is presented for application to real-time enhancement of synchronized
visual effects in live music performances. The audio streams are
processed simultaneously and their temporal and spectral characteristics
are used to control the intensity, duration, and color of the lights.
The efficiency of the approach at various audio resolutions is tested
on rock and jazz pieces. The result of the analysis is illustrated
by a visual OpenGL 3-D animation illustrating the synchronous audio-visual
events occurring in the musical piece.
P2-4 Object-Coding for Resolution-Free Musical Audio—Steve
Welburn, 1 Mark Plumbley, 1 Emmanuel Vincent2
1Queen Mary, University of London, London, UK
2IRISA-INRIA, Rennes, France
Object-based coding of audio represents the signal as a parameter
stream for a set of sound-producing objects. Encoding in this manner
can provide a resolution free representation of an audio signal.
Given a robust estimation of the object-parameters and a multi-resolution
synthesis engine, we can “intelligently” upsample a
signal, extending the bandwidth and getting best use out of a high-resolution
signal-chain. We will present some initial findings on extending
bandwidth using harmonic models.
P2-5 St. Kliment (2006/7) 05.40—High Resolution Music
Composition—Robert Sazdov, University Western Sydney, MARCS
Auditory Laboratories, Sydney, NSW, Australia
The proposed high resolution music composition is a 3-D multichannel
work that demonstrates novel compositional techniques based on audio
perceptual research into spatial attributes.
|Tuesday, June 26 14:00 – 15:00
|Paper Session 6 — High Resolution
Recording Issues II—Microphones
6-1 System Configuration for High Quality Audio Capturing
in a Large Microphone Array—Ines Hafizovic,1,2
Morgan Kjølerbakken,1 Vibeke Jahr1
1SquareHead Systems AS, Oslo, Norway
2University of Oslo, Oslo, Norway
In this paper we describe a speech acquisition system developed
and manufactured for directive audio recording in outdoor arenas.
Core technology, initially developed for audio production of sport
meets the high requirements of the broadcasting industry. The system
differs from the present audio recording solutions in its ability
for user controlled focusing and steering of the sound recorded
with a large, highly directive microphone array. Multi-channel real-time
(RT) audio output presented together with video allows for a new
multimedia experience in broadcast applications and other areas
where remote speech acquisition is desired.
6-2 Digital Microphones for High Resolution Audio—Martin
Schneider, Georg Neumann GmbH, Berlin, Germany
Microphones with a digital output format have appeared on the market
in the last few years. They integrate the functions of a microphone,
preamplifier, and analog-to-digital converter in one device. Properly
designed, the microphone dynamic range can thus be optimally adapted
to the intended application. The need to adjust gain settings and
trim levels is reduced to a minimum. Dynamic range issues inside
and outside the microphone are discussed. Advantages of microphones
with a wide dynamic range and 24-bit resolution, according to AES
42 are shown and compared to simpler realizations with 16-bit resolution
|Tuesday, June 26 15:30– 17:30
Panel Discussion — Design Issues
in High Quality Integrated Audio Systems
Panelists will include John Dawson, Arcam, Philip Hobbs, Linn,
and John Atkinson, Stereophile. Additional panelists to be announced.
|Given the many changes occurring in formats and distribution of
audio, high quality playback systems must increasingly support a range
of disc types, from CD to Bluray/HD DVD, along with streamed or downloaded
data, and to integrate with hard disc servers, home/studio networks,
and potentially media center PCs. This panel will discuss whether
an integrated approach is the correct answer for future system design.
What technical problems occur in the design of integrated systems
that must support HDMI, Ethernet, USB, and wireless connectivity as
well as traditional audio interfaces, e.g., jitter, chipset availability
and adequacy, changing protocols, and the like. Are there workable
strategies to keep up with changing software and hardware as formats
and digital rights management evolve? Do PCs and servers belong as
direct components in a high quality audio chain?
|In the Listening Room, throughout the morning, will be demonstrations
of high resolution audio by Jeff Levison, DTS.
|Wednesday, June 27 9:00 – 10:00
Peter Craven will give the Keynote Speech.
|Wednesday, June 27 10:00 – 10:30
|Paper Session 7 — Ambisonics
7-1 The Design of Improved First Order Ambisonic Decoders
by the Application of Range Removal and Importance in a Heuristic
Search Algorithm—David Moore, Jonathan Wakefield,
University of Huddersfield, Queensgate, Huddersfield, UK
This paper presents improvements to previous work on deriving first
order Ambisonic decoders for ITU 5.1. The decoders are derived using
a heuristic search method with an objective function based upon
Gerzon’s metatheory of auditory localization. An analysis
of previously derived decoders shows that they are biased toward
particular design objectives due to the nature of the multiobjective
function guiding the search. This paper applies a technique called
range removal to systematically and logically remove this bias that
leads to improved decoder coefficients that better meet all of the
objectives. A further technique known as importance is introduced
that enables the logical biasing of range-removed objectives. A
case study to develop a “max RE” decoder demonstrates
this technique in action.
|Wednesday, June 27 11:00 – 12:30
|Paper Session 8 — Maintaining Quality
8-1 Achieving Real Bandwidth Beyond 20 kHz with a Loudspeaker
System—Neil Harris, NXT, Huntingdon, UK
There are now quite a number of so-called “super tweeter”
products on the market, all claiming bandwidth beyond 20 kHz. Frequently,
however, this bandwidth is achieved only within a few degrees of
the loudspeaker axis, and the read power bandwidth is much lower.
The limiting factor in all piston-based designs is the “ka
= 2” relationship between wave-number and radius. At 20 kHz,
this results in the requirement that a is about 5 mm. By removing
the requirement for the diaphragm to move as a piston, the “Balanced
Mode Radiator” allows the use of larger diaphragms, making
the possibility of real power bandwidth beyond 20 kHz a practical
8-2 All Digital High Resolution Class D Amplifier Designs
Using Power Supply Feed-Forward and Signal Feedback—Steven
Harris, Jack Andersen, Daniel Chieng, Jeff Klaas, Michael Kost,
Skip Taylor, D2Audio Corporation, Austin, Texas USA
This paper describes a digital input Class D amplifier that uses
an integrated circuit controller. Sophisticated digital pulse width
modulation, combined with digital feed-forward and feedback paths,
yields high resolution amplifier designs. A powerful DSP is included
in the controller to support amplifier control and allows comprehensive
audio signal processing, including loudspeaker load compensation,
EQ, time alignment, room acoustics compensation, bass enhancement,
loudspeaker driver protection, virtual surround, and other audio
signal processing tasks. Power supply feed-forward and closed-loop
feedback technology correct for nonlinearity and other distortion-inducing
8-3 A Digital Amplification Technology to Optimize Performance
with High-Resolution Audio—Craig Bell
This paper describes a performance-orientated closed-loop digital
amplifier architecture that solves the traditional problems facing
digital-input open-loop amplifiers and is shown to have performance
advantages over equivalent systems implemented using conventional
analog amplifier technology. Through the use of a high-resolution
data-path and the global feedback structure, significant improvements
in retained resolution at the amplifier output are evident.
|Wednesday, June 27 14:00 – 15:00
|Paper Session 9 — Storage and Restoration
9-1 Processing Techniques for the Recovery of Audio from
Edison Cylinder Recordings, via Noncontact Surface Measurement—Antony
Nascè, John McBride, Martyn Hill, Peter Boltryk, University
of Southampton, Southampton, UK
A noncontact method for the recovery of sound from an Edison cylinder
record is presented. The cylinder surface is scanned via white light
sensor, capable of submicron axial resolution. Sound recovery is
achieved by estimating the trajectory of a playback stylus over
the measured surface, by tracking the central axis of the grooves.
The processing methods required to extract audio from a discrete
height map are described. We examine the signal to noise ratio as
a function of position across the groove cross-section for different
data sets, and compare spectra from the non-contact and stylus transfer
9-2 MPEG-A Professional Archival Multimedia Application
Format (MAF) Under Development—Noboru Harada,
Yutaka Kamamoto, Takehiro Moriya, NTT Communication Science Labs.,
Atsugi, Kanagawa, Japan
This paper describes MPEG’s latest specifications of the
Professional Archival Multimedia Application Format (MAF) that is
currently under development, and describes proposed extensions to
MPEG-4 Audio Lossless Coding (ALS) in order to support the Sony
Wave64 Format and the Broadcast Wave File Format (BWF) with RF64,
which handle large data size
exceeding 4 GB. The Professional Archival MAF compresses and archives
files and folder structures into a single archive file so that recorded
digital audio projects including meta-information and digitized
traditional documentation (e.g., tracking sheets, lyrics, and engineer
notes) are archived losslessly in the standardized manner. The format
is sufficient for the future-proof archiving tool.
|Wednesday, June 27 15:30-17:30
|Panel Discussion — Achievements, Challenges,
and the Future in High Resolution Audio
to be announced
|High resolution audio has emerged continuously over a period beginning
around the 1980s and culminating in the existing LPCM and DSD high
res formats. The supporting technologies and practice have emerged
across a wide variety of professional, consumer, and research areas
related to high res. Yet for many reasons, the advancements in these
areas have often not been sufficiently exploited and consumers have
moved instead toward portable listening devices at even lower resolution
than CD. The new HD disc formats together with the larger move toward
electronic distribution present new opportunities and major new challenges
for high res audio. There is a strong need to evolve consumer awareness
of high quality sound. In this panel discussion, we highlight some
of the most interesting directions in the field of high resolution
audio. We explore the opportunities, many of them already mentioned
at the conference, for improving the listening experience. We will
reflect on the demonstrations, exhibits, sound installations and recordings
showcased throughout the conference. The discussion in this session
is intended to be stimulating and an open forum for new ideas and
debate over the exciting developments in this field.
in this session, awards will be given for best paper, best student
paper, and best hi-res recording.
The following demonstrations will be held throughout the
Conference: sound installation by The Illustrious Company;
loudspeaker demonstration by Dyer Audio; original high resolution
audio recordings; various demonstrations and exhibitions.
Craven - Keynote speaker
Peter Craven attended Oxford University from 1966-74, studying mathematics
as an undergraduate and astrophysics as a postgraduate. Much of this time
was devoted to the design of recording equipment and making "purist"
uncompressed recordings of groups such as the Schola Cantorum of Oxford.
He met Michael Gerzon in 1967, starting a collaboration that was to last
for 29 years.
A career in academic computing followed, including much work on compilers
for the programming language Algol68. This work was later extended in
a project by N.A. Software Ltd. and is now the basis for their commercial
Fortran90 compiler. In 1982 Dr. Craven left university life to become
an independent consultant specializing in audio digital signal processing
(DSP) software and in high-level methods of generating efficient DSP code.
In the late 1980s, an extensive collaboration with B&W Loud- Speakers
on room equalization resulted in patents relating to high-resolution D/A
conversion and to digital PWM power amplifiers. Current consultancy projects
include Motorola DSP56000 audio software for use in consumer audio-visual
systems, and the audio DSP for the Jubilee Line Extension's public address
system (London Underground).
The many activities jointly with Michael Gerzon include the invention
of the Ambisonic Soundfield Microphone in 1973, a seminal paper on noise
shaping and dither published in 1989, and the inventions of Autodither
and Buried Data. The work on lossless data cornpression was the last major
collaboration before Michael Gerzon's death, and is continuing. In 1999,
Peter received the AES Publications Award.
Despite involvement with state-of-the-art reproduction technology, for
relaxation Peter Craven turns either to live music or to prewar 78s. In
his view, 1927 was a particularly good year.
George Y. Massenburg was born in Baltimore, Maryland and raised between
there and Macon, Georgia. Keenly interested in music, electronics and
sound recording at an early age, he was working part-time both in the
recording studio and in an electronics laboratory at 15 years of age.
As a sophomore majoring in electrical engineering at Johns Hopkins University,
he left and never returned. He designed, authored and presented the 1972
AES paper on the Parametric Equalizer and is regularly published in professional
journals and trade magazines worldwide. He was chief engineer of Europa
Sonar Studios in Paris, France in 1973 and 1974, and also did freelance
engineering and equipment design in Europe during those years.
He chartered an electronics company, GML, Inc., in 1982 to produce equipment
as needed for specific recording applications. Some early ideas' time
had come - notably that of 'Parametric Equalization' but also seminal
features of third and fourth generation automation systems for recording
studios. More recently introduced devices, such as the GML 2032 Mic Pre
and Parametric EQ, have been in development, on and off, for 20 years.
Currently the company manufacturers this, as well as the GML Automation
System, the High Resolution Topology line-level mixing console, and the
GML Microphone Preamplifier. GML also consults and provides independent
design for several major audio electronics manufacturers.
Individually or collaboratively, he has participated in over two hundred
record albums during the past 30 years. He has designed, built and managed
several recording studios, notably "ITI" Studios in Huntsville,
Maryland and "The Complex" in Los Angeles. He has in addition,
contributed acoustical and architectural designs to many others, including
"Skywalker Sound" and "The Site" in Marin County.
He is currently Adjunct Professor of Recording Arts and Sciences at McGill
University in Montreal, Quebec, Canada and visiting lecturer at UCLA and
USC in Los Angeles, California and MTSU in Murfreesboro Tennessee.
He has been working to qualify extended resolution and bandwidth as a
goal of modern professional digital recording standards work, and has
worked unceasingly to improve analog-digital-analog analysis and conversion
methods. He and GML, Inc. are currently researching extended automated
work-surfaces, high resolution graphical interfaces, extensible network
automation for audio production environments, and automation data interchange
George Massenburg's engineering and producing credits include Billy Joel,
Kenny Loggins, Journey, Madeleine Peyroux, James Taylor, Randy Newman,
Lyle Lovett, Aaron Neville, Little Feat, Michael Ruff, Toto, The Dixie
Chicks, Mary Chapin Carpenter, and Linda Ronstadt, among others (see the
discography page for a more complete list). He has been nominated many
times for the non-classical engineering Grammy (including a nomination
in 2001 for the Mary Chapin Carpenter's "Time*Sex*Love"), for
Record Of The Year in several years, and has won Grammys as producer for
Linda Ronstadt's 1996 "Dedicated To The One I Love" and another
for Best Engineered Non-Classical Record in 1990, for Linda Ronstadt's,
"Cry Like A Rainstorm, Howl Like the Wind." In 1998 he received
the Grammy for Technical Achievement, one of only four such awards presented
in the history of NARAS. He also won the Academy of Country Music award
for Record Of The Year in 1988 (for "The Trio"). In 1989, he
received the Mix Magazine TEC Awards for Producer and Engineer Of The
Year (for Little Feat), as well as Engineer Of The Year Award (for Linda
Ronstadt) in 1991, and 1992 (for Lyle Lovett). He currently resides in
Williamson County, Tennessee.
at a Glance -2 page schedule of conference events and
of High Resolution Sound
- "Content preparation, archiving and distribution
of hi-res audio media" chaired by George Massenburg,
- "Design Issues for High Quality Integrated Audio
- "Achievements, Challenges, and the Future in High