AES New York 2011
Engineering Brief Details


EB1 - Recording/Production

Thursday, October 20, 11:00 am — 12:00 pm (Room: 1E09)

Richard King, McGill University - Montreal, Quebec, Canada, Centre for Interdisciplinary Research in Music, Media and Technology, Montreal, Quebec, Canada

EB1-1 A New Golden Age of RecordingThomas T. Chen, Custom Recording/Studio C
The “Golden Age” of recording is said to be in the early 1960s and consists of recordings that were made that have great natural sound that are a treat to the ear. I believe that the room is one of the major contributors to the sound of the “Golden Age” recordings. I have developed a technique for live recording producing good sound and also a means of adding quality room to multitrack recordings.
Engineering Brief 16 (Download now)

EB1-2 What Constitutes Innovation in Music Production?Justin Paterson, London College of Music, University of West London
Innovation has often been at the core of record production, yet as production has advanced from Fred Gaisberg through the techniques of Musique Concrète to the plethora of possibilities afforded by the present digital age, the opportunities for genuine innovation might now seem limited. This notion is explored by considering the ontology of production with reference to audio examples, forming a chronological thread that highlights pieces commonly perceived as landmark innovations, their technological backdrops, and the recurrence/evolution of effect and aesthetic through successive generations of technology, and ultimately a nexus. The perception, attribution and value of “quality” is another factor, and while this is a separate subject in its own right, some discussion of this better contextualizes the topic.
Engineering Brief 17 (Download now)

EB1-3 Achieving Great Sound in the Age of Loudness WarsBryan Martin, Sonosphere Mastering - Montreal, Quebec, Canada; Adrian Carr, AC Mastering - Montreal, Quebec, Canada
There has been much concern and discussion about the ever increasing demand for higher and higher mix and mastering levels. Veteran mastering engineers Adrian Carr and Bryan Martin will take a new perspective to discuss techniques and processes to improve and maintain fidelity in the current “Maximum-Volume-Level” marketplace. Though this presentation is geared toward the working engineers of today, it is of particular importance to the up and coming generation. Since the level of CD’s is not decreasing, the young engineer can still be aware of the advantages he has and the pitfalls he faces.
Engineering Brief 18 (Download now)

EB1-4 Phantom Powering the Modern Condenser Microphone Part II: The Effects of Load Impedance on Microphone PerformanceMark Zaim, Audio-Technica U.S. Inc. - Stow, OH, USA
This paper builds on topics discussed in the previous AES paper “Phantom Powering the Modern Condenser Microphone: A Practical Look at Conditions for Optimized Performance.” It is not uncommon for microphone manufacturers to measure performance specifications using open load conditions. However, in application microphones are connected to mixing consoles that have much lower input impedances. These operating conditions can affect microphone performance and cause measurements to deviate from published open load conditions. During the previous investigation, changes were seen in several relevant microphone measurements as load impedance was changed. These observations prompted a more in-depth investigation into the effects of load impedance on microphone performance. The specific performance parameters investigated were: sensitivity, self-noise, current consumption, dynamic range, maximum sound pressure level (max SPL), signal to noise ratio (SNR) and frequency response.
Engineering Brief 19 (Download now)

EB2 - Posters

Thursday, October 20, 5:00 pm — 6:30 pm (Room: 1E Foyer)

EB2-1 Digital Control of an Analog Parametric EqualizerBlair Ryan Conner, Purdue University - West Lafayette, IN, USA
This project focuses on creating a digitally controllable analog band-pass filter with an adjustable resonant frequency for a middle frequency adjuster in an audio equalization stage. The design of the band-pass filter is a standard series resistor, inductor, and capacitor filter network. An adjustable gyrator circuit simulates an inductor to change the resonant frequency of the filter. Inside the gyrator circuit, a voltage-controlled amplifier is configured to simulate a resistor to change the gyrators simulated inductance. A digital to analog converter controls the gain of the voltage-controlled amplifier to make the analog filter digitally controlled. This circuit successfully acts as a bandpass filter with a digitally controllable resonant frequency.
Engineering Brief 22 (Download now)

EB2-2 Mash-Up Stephen Partridge, Buckinghamshire New University - High Wycombe, UK
The project will constitute an exploration of the choices made by users of audio post-production/editing applications in the selection and re-use of digital media files. The intended outcome would be to inform a better understanding of the ways in which contemporary users engage with digital media artifacts. The production context upon which this study will be based concerns an ongoing enterprise project, some further details of which are referred to below.
Engineering Brief 23 (Download now)

EB2-3 Statistical Analysis of Electro-Acoustic Measurements Sets Using ScilabDaniele Ponteggia, Audiomatica Srl - Firenze, Italy
The production management of electro-acoustic systems require the statistical analysis of measurements data. The analysis process should be sufficiently flexible to match the needs of the production process and the number of measured samples should be large enough to ensure the accuracy in statistical terms. Using an open source numerical computation software (Scilab) is possible to create statistical analysis procedures in a simple and cost effective way. Scilab syntax is simple enough to be acquired within a fairly short time, while data analysis capabilities are very advanced. In this work some sample applications are shown, with minimal code edit the provided examples can be adapted to several real world cases.
Engineering Brief 24 (Download now)

EB2-4 Latency Measurements of Audio Sigma Delta Analog to Digital and Digital to Analog ConvertsYonghao Wang, Queen Mary University of London - London, UK, Birmingham City University, Birmingham, UK
Latency is a well recognized issue when using digital audio workstations for live music processing. Previous research has reported measurements of the latency of the whole audio processing chain based on a “blackbox” approach. This report presents the results of latency measurement of typical compact analog to digital and digital to analog converters (ADC/DACs) in isolation from computer system processing overheads by using a high-speed data acquisition device. The report discusses the testing methods and pitfalls. It confirms that the latency is almost exclusively accounted for by the expected group delay of the digital decimation filters and interpolation filters used in the Sigma-Delta convertor.
Engineering Brief 25 (Download now)

EB2-5 The Effect of Reverberation on Music PerformanceElisa Sato, Toru Kamekawa, Atushi Marui, Tokyo University of the Arts - Tokyo, Japan
The structure of playing musical instruments consists of 3 basic steps: performing on the instrument to make musical sounds, recognizing spatial information and music in the space through listening to the sounds themselves, and finally returning the information to the performance to adapt their musical experience and consciousness. These steps go on as a loop during the performance of musical instruments, and it is widely known that reverberation effects recognition of the space, i.e. the second step. Adjusting the whole acoustic environment inside the room with sampling reverberation should help musicians to play just as they want to. In this paper, the relation between multiple parameters of reverberation and the features of solo violin performance is investigated.
Engineering Brief 26 (Download now)

EB2-6 Analysis-Synthesis Techniques for Additive Granular SynthesisJames O’Neill, University of Miami - Coral Gables, FL, USA
This project explores granular synthesis techniques that utilize various basis functions inspired by existing matching pursuit algorithms. The first algorithm performs a STFT on an input signal and synthesizes a new, granular signal using one-dimensional Gabor atoms. These atoms can be made to virtually reproduce the input signal, but a wide variety of granular effects can be achieved by altering the distribution of the atoms in the time and frequency domains, such as granular time stretching and pitch shifting, along with statistical distribution techniques introduced by the author. The second algorithm utilizes a basis set of generated noise bursts, which can be over-complete or an orthonormal basis for the Hilbert space that corresponds to the analysis window by applying the Gram-Schmidt process to the burst library. The noise functions are then used as grain contents in the synthesis stage, where a variety of effects are created with redistribution methods. Audio examples are provided over headphones.
Engineering Brief 27 (Download now)

EB2-7 Harmonic Distortion Analysis in a Class-AB Tube Amplifier: The McIntosh MC-240Phillip Minnick, University of Miami - Coral Gables, FL, USA
Some Class-AB tube amplifiers remain in demand for audiophiles due to their linear gain stages, low feedback, and minimal high-order harmonic distortion. The McIntosh MC240 Class-AB tube amplifier is a benchmark, high-quality stereo amplifier from the 1960s. This presentation gives an overview of the restoration of the amplifier to achieve a satisfactory level of performance, examination of the electrical topology in relation to the output signal’s distortion characteristics, and detailed analysis of the distortions produced by the amplifier using psychoacoustic-based listening tests as well as standard benchmark tests.
Engineering Brief 28 (Download now)

EB2-8 The “Williams Star” Microphone Array Support SystemMichael Williams, Freelance Sound Recording Engineer and Lecturer, Sounds of Scotland - Le Perreux sur Marne, France
With the ever increasing interest in 5-channel recording for home cinema, television, and pure 5-channel audio, the search for an operationally simple, reliable, and high quality surround sound recording system is becoming ever more important. The equal segment 4-channel and 5-channel arrays described in two AES papers by Michael Williams (AES preprints 3157 and 7480) are attracting more and more interest within the audio industry. However the research for a satisfactory operational microphone configuration cannot be dissociated from the purely mechanical problem of finding a suitable microphone array support system. The “Williams Star” Microphone Support System can provide a simple, flexible, and reliable microphone array support system for any equal segment array design. The visual impact of this system has also been reduced to an absolute minimum. Currently only the 4-channel and 5-channel seem to meet operational broadcasting requirements, but a 7-channel format is also provided for future developments with respect to the new Blu-ray format.
Engineering Brief 29 (Download now)

EB3 - Applications of Audio Engineering

Friday, October 21, 12:00 pm — 1:00 pm (Room: 1E09)

John Vanderkooy, University of Waterloo - Waterloo, Ontario, Canada

EB3-1 Foreign-Language Dubbing PracticesTheodor Stojanov, Independent Writer and Post Production Sound Editor - Montreal, Quebec, Canada
The interdisciplinary nature of foreign language dubbing involves the collaboration of actors, directors, producers, translators, sound and video departments, and often develops into an elaborate production project in its own right. Increasingly, particularly with motion pictures, the dubbing industry faces new challenges as studios are required to work at the pace of the original production and follow any and all modifications, leading ultimately to the global release of a film everywhere simultaneously, an economic model that has been gathering momentum in recent years. The technology for foreign dialogue dubbing has undergone a great optimization of the production work-flow to accommodate such new requirements, and this presentation will show various practices currently in use throughout Europe and North America. I will discuss the idealized work-flow versus technology-specific solutions, cost optimization, media security, and the advantages of certain types of technologies on foreign-language synchronization. Areas of development in foreign-language dubbing and internationalization will be discussed.
Engineering Brief 30 (Download now)

EB3-2 Application of Wave Field Synthesis and Analysis on Automotive AudioMichael J. Strauß, Fraunhofer IDMT - Ilmenau, Germany; Peter Gleim, Audi AG - Ingolstadt, Germany
High-quality audio equipment in cars enjoys rising popularity. For a remarkable number of people the car compartment represents the primary listening environment for enjoying music playback. Besides excellent sound quality spatial audio capabilities can also be expected from today’s top systems. In this talk we will give insight into the implementation of a spatial audio system based on Wave Field Synthesis that was realized inside an SUV. Audi AG and Fraunhofer IDMT together present the outcome of their research collaboration named "Audi Sound Concept." The Audi Q7 prototype has, on the one hand, remained a series production vehicle on the exterior, while on the other hand been rebuilt into a HiFi-Studio in the interior. The talk will include a systematic overview, a description of the playback-system, and some remarks about sound field analysis based on array measurements.
Engineering Brief 31 (Download now)

EB3-3 Spooky Sounds: Interactive Audio Systems and Design for a Themed Attraction in an Academic EnvironmentBruce Ellman, NYC College of Technology/CUNY - Sunnyside, NY, USA; John Huntington, NYC College of Technology/CUNY - Brooklyn, NY, USA
The high-tech, interactive Gravesend Inn haunted hotel attraction features a large, distributed, audience-triggered sound system to implement a design both startling and evocative. Sound Designer Bruce Ellman and Systems Engineer John Huntington will discuss the challenges faced in developing this system and using it to teach Entertainment Technology Students.
Engineering Brief 32 (Download now)

EB3-4 High Performance Architectural and Electro Acoustic Isolation SolutionsDave Kotch, John Storyk, Walters-Storyk Design Group
Increasing demands for community noise abatement, most specifically for architectural spaces such as night clubs and performance venue, have resulted in a variety of interesting design solutions to insure high performance sound attenuation, with FSTC results in excess of 80dB. This has often been accomplished with a combination of architectural construction design as well as electro-acoustic systems design such as directional subs, low frequency harmonic processing, and other systems integration devices. This paper will explore these designs and recent field results.
Engineering Brief 33 (Download now)

EB4 - Signal Processing

Saturday, October 22, 12:00 pm — 12:45 pm (Room: 1E09)

Robert Maher, Montana State University - Bozeman, MT, USA

EB4-1 A New Method for Evaluating Loudspeaker Efficiency in the Frequency DomainJoe Jensen, Technical University of Denmark - Lyngby, Denmark
The Constant Input Power (CIP) frequency response is proposed as a new method to evaluate loudspeaker efficiency in the frequency domain. Through a simulation study it is demonstrated how the CIP response can be a valuable tool when designing loudspeakers for which high efficiency is a priority.
Engineering Brief 34 (Download now)

EB4-2 Wave Field Synthesis by Multiple Line ArraysMatthew Montag, Colby Leider, University of Miami - Coral Gables, FL, USA
Wave field synthesis (WFS) is a spatial audio rendering technique that produces a physical approximation of wavefronts for virtual sources. Large loudspeaker arrays can simulate a virtual source that exists outside of the listening room. The technique is traditionally limited to the horizontal plane due to the prohibitive cost of planar loudspeaker arrays. Multiple-line-array wave field synthesis is proposed as an extension to linear WFS. This method extends the virtual source space in the vertical direction using a fraction of the number of loudspeakers required for plane arrays. This paper describes a listening test and software environment capable of driving a loudspeaker array according to the proposed extension, as well as the construction of a modular loudspeaker array that can be adapted to multiple-line configurations.
Engineering Brief 35 (Download now)

EB4-3 Playback Disappointment in Linear PCM Recording SystemsJohn “Beetle” Bailey, The Drive Shed - Recording Studios - Toronto, Ontario, Canada
As an “in-the-trenches” music recording engineer, my workflow has evolved to essentially an all in-the-box approach, with the exception of some external DSP. After spending many hours on a mix, I always get a strong feeling that when I finally print my mix to a stereo track in my workstation, or to a hardware-based digital recorder and play back the resulting 24-Bit-96-kHz WAV file, that it’s just not the same. A sense of disappointment. It seems to lack depth, reverb tails fall off, the transient response seems dulled, and an overall “graininess” to the mix. This paper and presentation will demonstrate the differences by way of 5.6-MHz DSD null tests and explore the difference between a live digital stream, and the disappointing playback of that digital stream that has been captured by a recorder in WAV file format. Upon demonstrating the problem, I will discuss possible solutions and workarounds I have used.
Engineering Brief 36 (Download now)

EB5 - Perception

Sunday, October 23, 2:15 pm — 3:00 pm (Room: 1E09)

EB5-1 Consumer Attitudes Toward Digital Audio QualityAinslie Harris, Robert Gordon University - Aberdeen, Scotland, UK
This paper builds upon an engineering brief submitted to the 130th AES Convention (Harris 2011). Where the May 2011 brief outlined initial findings from focus groups that were conducted, considering questions about preferred audio quality from the point of view of attitudes and consumer behavior, this brief focuses on an outline for future research, discussing important questions for consideration, and proposed methodology.
Engineering Brief 37 (Download now)

EB5-2 The Effect of Downmixing on Measured LoudnessScott G. Norcross, Michel C. Lavoie, Communications Research Center - Ottawa, Ontario, Canada
ITU-R BS.1770 has become the standard for loudness measurement in broadcasting. The measurement algorithm is equally adapted to 5.1 channel audio signals as it to a 2-channel downmix. Due to the manner by which the channels are summed, loudness differences can occur between the 5.1 channel signal and that of the stereo downmix. These differences are dependent on the inter-channel characteristics of the 5-channel mix. This engineering brief will outline the differences that can occur with different signals and provide data using real-world broadcast signals.
Engineering Brief 38 (Download now)

EB5-3 Prediction of Valence and Arousal from Music FeaturesAlbertus den Brinker, Ralph van Dinther, Philips Research Laboratories Eindhoven - Eindhoven, The Netherlands; J. Skowronek, Technical University Berlin - Berlin, Germany
Mood is an important attribute of music, and knowledge on mood can be used as a basic ingredient in music recommender and retrieval systems. Moods are assumed to be dominantly determined by two dimensions: valence and arousal. An experiment was conducted to attain data for song-based ratings of valence and arousal. It is shown that subject-averaged valence and arousal can be predicted from music features by a linear model.
Engineering Brief 39 (Download now)

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