AES New York 2013
Broadcast & Streaming Media Track Event Details

Wednesday, October 16, 5:00 pm — 7:00 pm (Room 1E10)

Workshop: W21 - Lies, Damn Lies, and Audio Gear Specs

Chair:
Ethan Winer, RealTraps - New Milford, CT, USA
Panelists:
Scott Dorsey, Williamsburg, VA, USA
David Moran, Boston Audio Society - Wayland, MA, USA
Mike Rivers, Gypsy Studio - Falls Church, VA, USA

Abstract:
The fidelity of audio devices is easily measured, yet vendors and magazine reviewers often omit important details. For example, a loudspeaker review will state the size of the woofer but not the low frequency cut-off. Or the cut-off frequency is given, but without stating how many dB down or the rate at which the response rolls off below that frequency. Or it will state distortion for the power amps in a powered monitor but not the distortion of the speakers themselves, which of course is what really matters. This workshop therefore defines a list of standards that manufacturers and reviewers should follow when describing the fidelity of audio products. It will also explain why measurements are a better way to assess fidelity than listening alone.

Excerpts from this workshop are available on YouTube.

 
 

Thursday, October 17, 9:00 am — 11:00 am (Room 1E12)

Live Sound Seminar: LS1 - AC Power and Grounding

Chair:
Bruce C. Olson, Olson Sound Design - Brooklyn Park, MN, USA; Ahnert Feistel Media Group - Berlin, Germany
Panelist:
Bill Whitlock, Jensen Transformers, Inc. - Chatsworth, CA, USA; Whitlock Consulting - Oxnard, CA, USA

Abstract:
There is a lot of misinformation about what is needed for AC power for events. Much of it has to do with life-threatening advice. This panel will discuss how to provide AC power properly and safely and without causing noise problems. This session will cover power for small to large systems, from a couple boxes on sticks up to multiple stages in ballrooms, road houses, and event centers; large scale installed systems, including multiple transformers and company switches, service types, generator sets, 1ph, 3ph, 240/120 208/120. Get the latest information on grounding and typical configurations by this panel of industry veterans.

 
 

Thursday, October 17, 9:00 am — 10:30 am (Room 1E11)

Workshop: W1 - Applications of 3D Audio in Automotive

Chair:
Alan Trevena, Jaguar Land Rover - Gaydon, UK
Panelists:
Jean-Marc Jot, DTS, Inc. - Los Gatos, CA, USA
Andreas Silzle, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Gilbert Soulodre, Camden Labs - Ottawa, ON, Canada
Bert Van Daele, Auro Technologies NV - Mol, Belgium

Abstract:
While there are a number of technologies aimed at improving the spatial rendering of recorded sounds in automobiles, few are offer the advantages and challenges as 3D surround. This workshop will explore theoretical applications; system configurations as well as limitations of 3D surround applications in automotive. Questions such as what is the reference experience, and how is a system evaluated will be addressed.

This session is presented in association with the AES Technical Committee on Automotive Audio

 
 

Thursday, October 17, 9:00 am — 10:30 am (Room 1E10)

Product Design: PD1 - Best Practices in Audio Software Development

Presenter:
Pascal Brunet, Setem Technologies - Newbury, MA, USA

Abstract:
This presentation reviews best practices accumulated through 25 years of software development experience. We first present the classical development "V" cycle: requirement specifications, prototyping, design (general and specific), coding & tests, validation. We then focus on each individual step: what should be included in good specifications; importance and good usage of prototyping; design methods; coding guidelines; testing methods; independent validation and beta testing. We finish with miscellaneous topics: estimation methods and risk assessment, project management, team work, source code control.

 
 

Thursday, October 17, 9:00 am — 10:30 am (Room 1E13)

Tutorial: T1 - FXpertise: Compression

Presenter:
Alex Case, University of Massachusetts Lowell - Lowell, MA, USA

Abstract:
Compressors were invented to control dynamic range. The next day, engineers started doing so much more—increasing loudness, improving intelligibility, adding distortion, extracting ambience, and, most importantly, reshaping timbre. This diversity of signal processing possibilities is realized only indirectly, by choosing the right compressor for the job and coaxing the parameters of ratio, threshold, attack, and release into place. Learn when to reach for compression, know a good starting place for compressor settings, and advance your understanding of what to listen for and which way to tweak.

 
 

Thursday, October 17, 9:00 am — 10:30 am (Room 1E08)

Broadcast and Streaming Media: B1 - Television Loudness and Metadata

Chair:
Fred Willard, Univision - Washington, DC, USA
Panelists:
J. Todd Baker, DTS, Inc. - Laguna Hills, CA, USA
Arne Borsum, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Florian Camerer, ORF - Austrian TV - Vienna, Austria; EBU - European Broadcasting Union
Tim Carroll, Linear Acoustic Inc. - Lancaster, PA, USA
Michael Kahsnitz, RTW - Cologne, Germany
Robert Orban, Orban - San Leandro, CA, USA

Abstract:
Television broadcasters and Multichannel Video Program Distributors (MVPDs) are required to put in place procedures, software, and hardware to “effectively control program-to-interstitial loudness … and loudness management at the boundaries of programs and interstitial content.” Objective data must be supplied to the FCC to support compliance with the legislation as well as timely resolution of listener complaints. Similar rules have been developed in the UK and other parts of the world. Members of our panel of experts have worked tirelessly to either create loudness control recommendations that have become the law or to bring those recommendations to implementation at the companies they represent. This session will cover the FCC’s Report and Order on the CALM Act, the development of the ATS’s A/85 Recommended Practice that is now part of the U.S. legislation and both domestic and European technical developments by major media distributors and P/LOUD.

 
 

Thursday, October 17, 10:30 am — 12:00 pm (Room 1E08)

Broadcast and Streaming Media: B2 - Audio for 4K TV

Chair:
Jonathan Abrams, Nutmeg Post - New York, NY, USA
Panelists:
Robert Bleidt, Fraunhofer USA Digital Media Technologies - San Jose, CA, USA
Tim Carroll, Linear Acoustic Inc. - Lancaster, PA, USA
Dave Casey, DTS
Poppy Crum, Dolby Laboratories - San Francisco, CA, USA
Robert Orban, Orban - San Leandro, CA, USA
Robert Reams, Psyx Research
Jim Starzynski, NBC Universal - New York, NY, USA

Abstract:
4K Television is the future. Video will be improved but what is happening to the audio? How will audio enhance the video experience? This panel will discuss television’s future sound.

 
 

Thursday, October 17, 10:30 am — 12:30 pm (Room 1E14)

Workshop: W3 - Acoustic Enhancements Systems—Implementations

Chair:
Ben Kok, SCENA acoustic consultants - Uden, The Netherlands
Panelists:
Steve Barbar, Lares Associates - Belmont, MA, USA
Peter Mapp, Peter Mapp Associates - Colchester, Essex, UK
Thomas Sporer, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Ilmenau University of Technology - Ilmenau, Germany
Takayuki Watanabe, Yamaha Corp. - Hamamatsu, Shizuoka, Japan
Wieslaw Woszczyk, McGill University - Montreal, QC, Canada
Diemer de Vries, RWTH Aachen University - Aachen, Germany; TU Delft - Delft, Netherlands

Abstract:
Acoustic enhancement systems offer the possibility to change the acoustics of a venue by electronic means. How this is achieved varies by the working principle and philosophy of the system implemented. In this workshop various researchers, consultants, and suppliers active in the field of enhancement systems will discuss working principles and implementations.

This workshop is in close relation with the tutorial on acoustic enhancement systems; people not yet too familiar with the applications and working principles of these systems are recommended to attend the tutorial before attending the workshop.

This session is presented in association with the AES Technical Committee on Acoustics and Sound Reinforcement

 
 

Thursday, October 17, 11:00 am — 1:00 pm (Room 1E12)

Workshop: W4 - Microphone Specifications—Believe it or Not

Chair:
Eddy B. Brixen, EBB-consult/DPA Microphones - Smorum, Denmark
Panelists:
Juergen Breitlow, Neumann - Berlin, Germany
Jackie Green, Audio-Technica U.S., Inc. - Stow, OH, USA
Bill Whitlock, Jensen Transformers, Inc. - Chatsworth, CA, USA; Whitlock Consulting - Oxnard, CA, USA
Helmut Wittek, SCHOEPS GmbH - Karlsruhe, Germany
Joerg Wuttke, Joerg Wuttke Consultancy - Pfinztal, Germany

Abstract:
There are lots and lots of microphones available to the audio engineer. The final choice is often made on the basis of experience or perhaps just habits. (Sometimes the mic is chosen because of the looks … .) Nevertheless, there is essential and very useful information to be found in the microphone specifications. This workshop will present the most important microphone specs and provide the attendees with up-to-date information on how these are obtained and understood. Each member of the panel—all related to industry top brands—will present one item from the spec sheet. The workshop takes a critical look on how specs are presented to the user, what to look for and what to expect. The workshop is organized by the AES Technical Committee on Microphones and Applications.

This session is presented in association with the AES Technical Committee on Microphones and Applications

 
 

Thursday, October 17, 2:15 pm — 3:45 pm (Room 1E08)

Broadcast and Streaming Media: B3 - Listener Fatigue and Retention

Chair:
Richard Burden, Richard W. Burden Associates - Canoga Park, CA, USA
Panelists:
Frank Foti, Telos - New York, NY, USA
Greg Ogonowski, Orban - San Leandro, CA, USA
Sean Olive, Harman International - Northridge, CA, USA
Robert Reams, Psyx Research
Elliot Scheiner

Abstract:
This panel will discuss listener fatigue and its impact on listener retention. While listener fatigue is an issue of interest to broadcasters, it is also an issue of interest to telecommunications
service providers, consumer electronics manufacturers, music producers, and others. Fatigued listeners to a broadcast program may tune out, while fatigued listeners to a cell phone conversation may switch to another carrier, and fatigued listeners to a portable media player may purchase another company’s product. The experts on this panel will discuss their research and experiences with listener fatigue and its impact on listener retention.

 
 

Thursday, October 17, 2:30 pm — 4:30 pm (Room 1E12)

Live Sound Seminar: LS2 - Audio Network and Transport

Chair:
Jim Risgin, On Stage Audio - Wood Dale, IL, USA
Panelists:
Mark Dittmar, Firehouse Productions
Phil Reynolds, System Tech, The Killers
Robert Silfvast, Avid - Mountain View, CA, USA

Abstract:
As audio and control over network become more predominate in today’s live sound environment managing the network becomes more challenging. This panel will discuss the problems, challenges, and solutions required that are associated with sharing the bandwith between audio and control as well as the unique challenges created by all the different manufacturers and protocols. Our discussion will rely heavily upon questions and comments from the audience as your experiences, pitfalls and questions are central to these common challenges today.

 
 

Thursday, October 17, 3:45 pm — 5:15 pm (Room 1E08)

Broadcast and Streaming Media: B4 - Loudness Control for Radio and Internet Streaming

Chair:
David Bialik, CBS - New York, NY, USA
Panelists:
Robert Bleidt, Fraunhofer USA Digital Media Technologies - San Jose, CA, USA
Florian Camerer, ORF - Austrian TV - Vienna, Austria; EBU - European Broadcasting Union
Frank Foti, Telos - New York, NY, USA
John Kean, NPR
Robert Orban, Orban - San Leandro, CA, USA

Abstract:
Is the “Loudness War” in radio over? Has it moved over to internet streaming? With content being injected from multiple sources, levels are varying. How can we control level without disrupting the audience? Some countries are introducing regulation—is it needed?

 
 

Thursday, October 17, 4:30 pm — 6:00 pm (Room 1E13)

Networked Audio: N1 - One Network to Rule Them All

Chair:
Kevin Gross, AVA Networks - Boulder, CO, USA
Panelists:
Mattias Allevik, Video Corporation of America - New York, NY, USA
Dave Revel, Technical Multimedia Design, Inc. - Burbank, CA, USA

Abstract:
Networked audio distribution is now less frequently accomplished as a separate infrastructure. The promise of running audio on the same network as other facility services and applications is now coming to fruition. This workshop will discuss the motivation for combining services, the challenges in doing so, and requirements this approach puts on audio networking technologies.

This session is presented in association with the AES Technical Committee on Network Audio Systems

 
 

Thursday, October 17, 5:30 pm — 7:00 pm (Room 1E08)

Broadcast and Streaming Media: B5 - Is it Time to Retire the MP3 Protocol for Streaming

Chair:
Ray Archie, CBS - New York, NY, USA; Music is My First Language - New York, NY, USA
Panelists:
Karlheinz Brandenburg, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Ilmenau University of Technology - Ilmenau, Germany
John Kean, NPR
Jan Nordmann, Fraunhofer USA - San Jose, CA, USA
Greg Ogonowski, Orban - San Leandro, CA, USA
Greg Shay, The Telos Alliance - Cleveland, OH, USA

Abstract:
It has been over 25 years since the MP3 codec was introduced to the audio community. With lossy audio encoding, such as an MP3, there is a not so fine balance between audio quality and file size. With the ever increasing availability of bandwidth, file size has diminished as a consideration for audio streaming and codec related loss in audio quality is much more apparent.

This panel will be an in-depth discussion about this phenomenon. We will also discuss challenges related to introducing new codecs into the space.

 
 

Thursday, October 17, 6:00 pm — 7:00 pm (Room 1E13)

Networked Audio: N2 - A Primer on Fundamental Concepts of Media Networking

Presenter:
Landon Gentry, Audinate - Portland, OR, USA; Sydney, Australia

Abstract:
This session will cover the OSI model and how data travels through network layers (a “networking stack”): Layers 1, 2, 3 and 4; Cables, MAC Addresses, IP Addresses, and networking protocols. An overview of some networking standards and standards organizations, including the IEEE and the IETF. An introduction to IP data networking . . . it is how everything is already wired together. Identify some of the advantages and limitations of IP data networks with respect to real-time media. A brief discussion of IP networking standards and protocols that can be leveraged for media networking.

 
 

Friday, October 18, 9:00 am — 10:30 am (Room 1E08)

Broadcast and Streaming Media: B6 - Audio for Mobile TV

Chair:
Joe Giardina, DSI RF Systems
Panelists:
J. Todd Baker, DTS, Inc. - Laguna Hills, CA, USA
Tim Carroll, Linear Acoustic Inc. - Lancaster, PA, USA
Greg Ogonowski, Orban - San Leandro, CA, USA
Robert Reams, Psyx Research
Jim Starzynski, NBC Universal - New York, NY, USA

Abstract:
A panel discussion highlighting the various challenges facing Mobile TV audio transmissions. Focus will be on dialog intelligibility, signal routing and issues and applications unique to Mobile TV audio broadcasts.

 
 

Friday, October 18, 9:00 am — 10:30 am (Room 1E13)

Tutorial: T6 - FXpertise: Equalization

Presenter:
Alex Case, University of Massachusetts Lowell - Lowell, MA, USA

Abstract:
Equalization might be the most intuitive of effects. We’ve had tone controls since we were kids, after all. Advanced applications of equalization are born from a deep understanding of EQ parameters and technologies, plus broad knowledge of the spectral properties and signatures of the most common pop and rock instruments. In this tutorial, Alex Case shares his approach for applying EQ and strategies for its use: fixing frequency problems, fitting the spectral pieces together, enhancing flattering features, and more.

 
 

Friday, October 18, 9:00 am — 12:00 pm (Room 1E07)

Paper Session: P8 - Recording and Production

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

P8-1 Music Consumption Behavior of Generation Y and the Reinvention of the Recording Industry—Barry Marshall, The New England Institute of Art - Brookline, MA, USA
This paper will give an overview of the last 15 years of the recording industry’s problems with piracy and decreasing sales, while reporting on research into the music consumption behavior of a group of audio students in both the United States and in eight European countries. Audio students have a unique perspective on the issues surrounding the recording industry’s problems since the advent of Napster and the later generations of illegal file sharing. Their insights into issues like the importance of access to music, the quality of the listening experience, and the ethical quandary of participating in copyright infringement, may help point to a direction for the future of the recording industry.
Convention Paper 8956 (Purchase now)

P8-2 (Re)releasing the Beatles—Brett Barry, Syracuse University - Syracuse, NY, USA
This paper presents a comparative analysis of various Beatles releases, including original 1960s vinyl, early compact discs, and present-day digital downloads through services like iTunes. I will provide original research using source material and interviews with persons directly involved in recording and releasing Beatles albums, examining variations in dynamic range, spectral distribution, psychoacoustics, and track anomalies. Considerations are given to mastering and remastering a catalog of classics.
Convention Paper 8957 (Purchase now)

P8-3 Maximum Averaged and Peak Levels of Vocal Sound Pressure—Braxton Boren, New York University - New York, NY, USA; Agnieszka Roginska, New York University - New York, NY, USA; Brian Gill, New York University - New York, NY, USA
This work describes research on the maximum sound pressure level achievable by the spoken and sung human voice. Trained actors and singers were measured for peak and averaged SPLs at an on-axis distance of 1 m at three different subjective dynamic levels and also for two different vocal techniques (“back” and “mask” voices). The “back” sung voice was found to achieve a consistently lower SPL than the “mask” voice at a corresponding dynamic level. Some singers were able to achieve high averaged levels with both spoken and sung voice, while others produced much higher levels singing than speaking. A few of the vocalists were able to produce averaged levels above 90 dB_A<, the highest found in the existing literature.
Convention Paper 8958 (Purchase now)

P8-4 Listener Adaptation in the Control Room: The Effect of Varying Acoustics on Listener Familiarization—Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Brett Leonard, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Scott Levine, Skywalker Sound - San Francisco, CA, USA; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Grzegorz Sikora, Bang & Olufsen Deutschland GmbH - Pullach, Germany
The area of auditory adaptation is of central importance to a recording engineer operating in unfamiliar or less-than-ideal acoustic conditions. This study prompts expert listeners to perform a controlled level-balancing task while exposed to three different acoustic conditions. The length of exposure is varied to test the role of adaptation on such a task. Results show that there is a significant difference in the variance of participants’ results when exposed to one condition for a longer period of time. In particular, subjects seem to most easily adapt to reflective acoustic conditions.
Convention Paper 8959 (Purchase now)

P8-5 Spectral Characteristics of Popular Commercial Recordings 1950-2010—Pedro Duarte Pestana, Catholic University of Oporto - CITAR - Oporto, Portugal; Lusíada Universityof Portugal (ILID); Centro de Estatística e Aplicacoes; Zheng Ma, Queen Mary University of London - London, UK; Joshua D. Reiss, Queen Mary University of London - London, UK; Alvaro Barbosa, Catholic University of Oporto - CITAR - Oporto, Portugal; Dawn A. A. Black, Queen Mary University of London - London, UK
In this work the long-term spectral contours of a large dataset of popular commercial recordings were analyzed. The aim was to analyze overall trends, as well as yearly and genre-specific ones. A novel method for averaging spectral distributions is proposed that yields results that are prone to comparison. With it, we found out that there is a consistent leaning toward a target equalization curve that stems from practices in the music industry but also to some extent mimics natural, acoustic spectra of ensembles.
Convention Paper 8960 (Purchase now)

P8-6 A Knowledge-Engineered Autonomous Mixing System—Brecht De Man, Queen Mary University of London - London, UK; Joshua D. Reiss, Queen Mary University of London - London, UK
In this paper a knowledge-engineered mixing engine is introduced that uses semantic mixing rules and bases mixing decisions on instrument tags as well as elementary, low-level signal features. Mixing rules are derived from practical mixing engineering textbooks. The performance of the system is compared to existing automatic mixing tools as well as human engineers by means of a listening test, and future directions are established.
Convention Paper 8961 (Purchase now)

 
 

Friday, October 18, 9:00 am — 11:30 am (Room 1E09)

Paper Session: P9 - Applications in Audio—Part I

Chair:
Sungyoung Kim, Rochester Institute of Technology - Rochester, NY, USA

P9-1 Audio Device Representation, Control, and Monitoring Using SNMP—Andrew Eales, Wellington Institute of Technology - Wellington, New Zealand; Rhodes University - Grahamstown, South Africa; Richard Foss, Rhodes University - Grahamstown, Eastern Cape, South Africa
The Simple Network Management Protocol (SNMP) is widely used to configure and monitor networked devices. The architecture of complex audio devices can be elegantly represented using SNMP tables. Carefully considered table indexing schemes support a logical device model that can be accessed using standard SNMP commands. This paper examines the use of SNMP tables to represent the architecture of audio devices. A representational scheme that uses table indexes to provide direct-access to context-sensitive SNMP data objects is presented. The monitoring of parameter values and the implementation of connection management using SNMP are also discussed.
Convention Paper 8962 (Purchase now)

P9-2 IP Audio in the Real-World; Pitfalls and Practical Solutions Encountered and Implemented when Rolling Out the Redundant Streaming Approach to IP Audio—Kevin Campbell, WorldCast Systems /APT - Belfast, N Ireland; Miami, Florida
This paper will review the development of IP audio links for audio delivery and chiefly look at the possibility of harnessing the flexibility and cost-effectiveness of the public internet for professional audio delivery. We will discuss first the benefits of IP audio when measured against traditional synchronous audio delivery and also the typical problems associated with delivering real-time broadcast audio across packetized networks, specifically in the context of unmanaged IP networks. The paper contains an examination of some techniques employed to overcome these issues with an in-depth look at the redundant packet streaming approach.
Convention Paper 8963 (Purchase now)

P9-3 Implementation of AES-64 Connection Management for Ethernet Audio/Video Bridging Devices—James Dibley, Rhodes University - Grahamstown, South Africa; Richard Foss, Rhodes University - Grahamstown, Eastern Cape, South Africa
AES-64 is a standard for the discovery, enumeration, connection management, and control of multimedia network devices. This paper describes the implementation of an AES-64 protocol stack and control application on devices that support the IEEE Ethernet Audio/Video Bridging standards for streaming multimedia, enabling connection management of network audio streams.
Convention Paper 8964 (Purchase now)

P9-4 Simultaneous Acquisition of a Massive Number of Audio Channels through Optical Means—Gabriel Pablo Nava, NTT Communication Science Laboratories - Kanagawa, Japan; Yutaka Kamamoto, NTT Communication Science Laboratories - Kanagawa, Japan; Takashi G. Sato, NTT Communication Science Laboratories - Kanagawa, Japan; Yoshifumi Shiraki, NTT Communication Science Laboratories - Kanagawa, Japan; Noboru Harada, NTT Communicatin Science Labs - Atsugi-shi, Kanagawa-ken, Japan; Takehiro Moriya, NTT Communicatin Science Labs - Atsugi-shi, Kanagawa-ken, Japan
Sensing sound fields at multiple locations often may become considerably time consuming and expensive when large wired sensor arrays are involved. Although several techniques have been developed to reduce the number of necessary sensors, less work has been reported on efficient techniques to acquire the data from all the sensors. This paper introduces an optical system, based on the concept of visible light communication, which allows the simultaneous acquisition of audio signals from a massive number of channels via arrays of light emitting diodes (LEDs) and a high speed camera. Similar approaches use LEDs to express the sound pressure of steady state fields as a scaled luminous intensity. The proposed sensor units, in contrast, transmit optically the actual digital audio signal sampled by the microphone in real time. Experiments to illustrate two examples of typical applications are presented: a remote acoustic imaging sensor array and a spot beamforming based on the compressive sampling theory. Implementation issues are also addressed to discuss the potential scalability of the system.
Convention Paper 8965 (Purchase now)

P9-5 Blind Microphone Analysis and Stable Tone Phase Analysis for Audio Tampering Detection—Luca Cuccovillo, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Sebastian Mann, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Patrick Aichroth, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Marco Tagliasacchi, Politecnico di Milano - Milan, Italy; Christian Dittmar, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany
In this paper we present an audio tampering detection method based on the combination of blind microphone analysis and phase analysis of stable tones, e.g., the electrical network frequency (ENF). The proposed algorithm uses phase analysis to detect segments that might have been tampered. Afterwards, the segments are further analyzed using a feature vector able to discriminate among different microphone types. Using this combined approach, it is possible to achieve a significantly lower false-positive rate and higher reliability as compared to standalone phase analysis.
Convention Paper 8966 (Purchase now)

 
 

Friday, October 18, 9:00 am — 10:30 am (Room 1E10)

Workshop: W10 - National Recording Preservation Plan: Best Practices for Creating and Preserving Born-Digital Audio Files

Chair:
Konrad Strauss, Indiana University - Bloomington, IN, USA
Panelists:
Chris Lacinak, AVPreserve - New York, NY, USA
George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada
Charles Van Winkle, Adobe - Minneapolis, MN, USA

Abstract:
In December of 2012 the Library of Congress released the National Recording Preservation Plan. The result of nearly 10 years of work by the Library and the National Recording Preservation Board, the Plan outlines a series of recommendations for implementing a national recorded sound preservation plan. This workshop will explore recommendations 2.7: Best Practices for Creating and Preserving Born-Digital Audio Files, and 2.6: Tools to Support Preservation throughout the Content Life Cycle; and will focus on best practices for the creation of born-digital recordings and strategies for short-term backup and long-term preservation.

This session is presented in association with the AES Technical Committee on Archiving Restoration and Digital Libraries

 
 

Friday, October 18, 10:30 am — 12:30 pm (Room 1E08)

Workshop: W11 - Audio Source Separation

Chair:
Gautham Mysore, Adobe Research - San Francisco, CA, USA
Panelists:
Nicholas Bryan, Stanford University - Stanford, CA, USA
Derry FitzGerald, Cork Institute of Technology - Cork, Ireland
Elias Kokkinis, accusonus - Patras, Greece
Pierre Leveau, Audionamix - Paris, France

Abstract:
Audio source separation algorithms aim to take a recording of a mixture of sound sources as an input and provide the separated sources as outputs. Algorithmically, this is a very challenging problem. However, some recent technological advances have made this possible for multiple real world scenarios such as denoising in the presence of complex noises, pitch correcting certain notes while preserving others, processing only the vocals of a song while preserving the background music, extracting dialogue from old films to provide a higher quality soundtrack, removing microphone leakage from multichannel drum recordings, upmixing mono to stereo with panning of sound sources, and more generally, music remixing. Some of these technologies are available in products (Adobe Audition CC, Melodyne, ISSE, ADX Trax, Drumatom). Others are used by specialized sound engineers and are offered as a service (Audionamix, Derry Fitzgerald). This panel is comprised of some of the inventors of these technologies, who will discuss the ideas and their practical use.

This session is presented in association with the AES Technical Committee on Semantic Audio Analysis

 
 

Friday, October 18, 10:30 am — 12:00 pm (Room 1E13)

Workshop: W12 - FX Design Panel: Equalization

Chair:
Francis Rumsey, Logophon Ltd. - Oxfordshire, UK
Panelists:
Nir Averbuch, Sound Radix Ltd. - Israel
George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada
Saul Walker, New York University - New York, NY, USA

Abstract:
Meet the designers whose talents and philosophies are reflected in the products they create, driving sound quality, ease of use, reliability, price, and all the other attributes that motivate us to patch, click and tweak their effects processors.

 
 

Friday, October 18, 11:30 am — 1:00 pm (Room 1E11)

Sound for Picture: SP1 - Creative Dimension of Immersive Sound—Sound in 3D

Chair:
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Panelists:
Marti Humphrey CAS, The Dub Stage - Burbank, CA, USA
Branko Neskov, Loudness Films - Lisbon, Portugal

Abstract:
Audio for Cinema has always struggled to replicate the motion shown on the screen, a fact that became more apparent with 3D films. Several methodologies for "immersive sound" are currently under evaluation by the industry, with theater owners and film companies both advising that they will not tolerate a format war, with a common format a commercial requirement.

The two major methods of creating immersive sound and audio motion are referred to as "object-based" and "channel-based." Each has its strengths and limitations for retrofit into the current cinema market. With few sound mixers experienced in either of these techniques, we're pleased to welcome two of the pioneers, one with experience at Auro3D and the other in Atmos, in a discussion of their experiences and comments on working with the two systems.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Friday, October 18, 12:30 pm — 2:00 pm (Room 1E08)

Broadcast and Streaming Media: B7 - Broadcasting During a Disaster

Chair:
Glynn Walden, CBS Radio - Philadelphia, PA, USA
Panelists:
Rob Bertrand, CBS
Howard Price, ABC/Disney
Tom Ray, Tom Ray Broadcasting Consulting
Richard Ross, WADO/Univision

Abstract:
No power!
Water rising!
Roads out!
No phones!

Broadcasters have always been “First Informers.” Not only do they convey information to the audience, the station has to maintain the broadcast throughout the disaster. Superstorm Sandy challenged many broadcasters. This panel will discuss how they dealt with non-ideal situations to keep the broadcast on during the storm, the aftermath, and the recovery.

 
 

Friday, October 18, 1:00 pm — 2:00 pm (Stage)

Project Studio Expo: Mixing Secrets: Production Tricks to Use with any DAW

Presenter:
Mike Senior, Sound On Sound - Munich, Germany; Cambridge Music Technology

Abstract:
Affordable DAW software now provides all the processing tools you need to create commercially competitive music mixes within a home, college, or project studio. As such, the overriding concern for budget-conscious engineers these days should be to develop effective habits with regard to studio monitoring, mix balancing, and quality control. Important techniques in each of these three areas are often neglected in small-scale productions, leading to mixes that don't stack up against professional releases, or that collapse on some mass-market listening systems. In this seminar Sound On Sound magazine's "Mix Rescue" columnist Mike Senior will draw on his experience of thousands of project-studio mixes to highlight the most frequently overlooked studio tricks. In the process he'll demonstrate how these methods can powerfully upgrade your sonics without breaking the bank, no matter which DAW you're using.

 
 

Friday, October 18, 2:15 pm — 4:45 pm (Room 1E09)

Paper Session: P11 - Perception—Part 1

Chair:
Jason Corey, University of Michigan - Ann Arbor, MI, USA

P11-1 On the Perceptual Advantage of Stereo Subwoofer Systems in Live Sound Reinforcement—Adam J. Hill, University of Derby - Derby, Derbyshire, UK; Malcolm O. J. Hawksford, University of Essex - Colchester, Essex, UK
Recent research into low-frequency sound-source localization confirms the lowest localizable frequency is a function of room dimensions, source/listener location, and reverberant characteristics of the space. Larger spaces therefore facilitate accurate low-frequency localization and should gain benefit from broadband multichannel live-sound reproduction compared to the current trend of deriving an auxiliary mono signal for the subwoofers. This study explores whether the monophonic approach is a significant limit to perceptual quality and if stereo subwoofer systems can create a superior soundscape. The investigation combines binaural measurements and a series of listening tests to compare mono and stereo subwoofer systems when used within a typical left/right configuration.
Convention Paper 8970 (Purchase now)

P11-2 Auditory Adaptation to Loudspeakers and Listening Room Acoustics—Cleopatra Pike, University of Surrey - Guildford, Surrey, UK; Tim Brookes, University of Surrey - Guildford, Surrey, UK; Russell Mason, University of Surrey - Guildford, Surrey, UK
Timbral qualities of loudspeakers and rooms are often compared in listening tests involving short listening periods. Outside the laboratory, listening occurs over a longer time course. In a study by Olive et al. (1995) smaller timbral differences between loudspeakers and between rooms were reported when comparisons were made over shorter versus longer time periods. This is a form of timbral adaptation, a decrease in sensitivity to timbre over time. The current study confirms this adaptation and establishes that it is not due to response bias but may be due to timbral memory, specific mechanisms compensating for transmission channel acoustics, or attentional factors. Modifications to listening tests may be required where tests need to be representative of listening outside of the laboratory.
Convention Paper 8971 (Purchase now)

P11-3 Perception Testing: Spatial Acuity—P. Nigel Brown, Ex'pression College for Digital Arts - Emeryville, CA, USA
There is a lack of readily accessible data in the public domain detailing individual spatial aural acuity. Introducing new tests of aural perception, this document specifies testing methodologies and apparatus, with example test results and analyses. Tests are presented to measure the resolution of a subject's perception and their ability to localize a sound source. The basic tests are designed to measure minimum discernible change across a 180° horizontal soundfield. More complex tests are conducted over two or three axes for pantophonic or periphonic analysis. Example results are shown from tests including unilateral and bilateral hearing aid users and profoundly monaural subjects. Examples are provided of the applicability of the findings to sound art, healthcare, and other disciplines.
Convention Paper 8972 (Purchase now)

P11-4 Evaluation of Loudness Meters Using Parameterization of Fader Movements—Jon Allan, Luleå University of Technology - Piteå, Sweden; Jan Berg, Luleå University of Technology - Piteå, Sweden
The EBU recommendation R 128 regarding loudness normalization is now generally accepted and countries in Europe are adopting the new recommendation. There is now a need to know more about how and when to use the different meter modes, Momentary and Short term, proposed in R 128, as well as to understand how different implementations of R 128 in audio level meters affect the engineers’ actions. A method is tentatively proposed for evaluating the performance of audio level meters in live broadcasts. The method was used to evaluate different meter implementations, three of them conforming to the recommendation from EBU, R 128. In an experiment, engineers adjusted audio levels in a simulated live broadcast show and the resulting fader movements were recorded. The movements were parameterized into “Fader movement,” “Adjustment time,” “Overshoot,” etc. Results show that the proposed parameters produced significant differences caused by the meters and that the experience of the engineer operating the fader is a significant factor.
Convention Paper 8973 (Purchase now)

P11-5 Validation of the Binaural Room Scanning Method for Cinema Audio Research—Linda A. Gedemer, University of Salford - Salford, UK; Harman International - Northridge, CA, USA; Todd Welti, Harman International - Northridge, CA, USA
Binaural Room Scanning (BRS) is a method of capturing a binaural representation of a room using a dummy head with binaural microphones in the ears and later reproducing it over a pair of calibrated headphones. In this method multiple measurements are made at differing head angles that are stored separately as data files. A playback system employing headphones and a headtracker recreates the original environment for the listener, so that as they turn their head, the rendered audio during playback matches the listeners' current head angle. This paper reports the results of a validation test of a custom BRS system that was developed for research and evaluation of different loudspeakers and different listening spaces. To validate the performance of the BRS system, listening evaluations of different in-room equalizations of a 5.1 loudspeaker system were made both in situ and via the BRS system. This was repeated using three different loudspeaker systems in three different sized listening rooms.
Convention Paper 8974 (Purchase now)

 
 

Friday, October 18, 2:30 pm — 4:00 pm (Room 1E11)

Sound for Picture: SP2 - Cinema Sound Standards Collapse Leaving Turmoil—An Overview of the State of the Art

Chair:
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Panelists:
Glenn Leembruggen, Acoustics Directions Pty Ltd. - Summer Hill, NSW, Australia; Sydney University
David Murphy, Krix Loudspeakers - Hackham, South Australia

Abstract:
Dr. Floyd Toole first documented in his book Sound Reproduction: Loudspeakers and Rooms in 2008 the failure of the Standards process in producing quality sound in movie theaters. The work was expanded on in experiments done by a group led by Philip Newell in Europe, and this work was cited by Brian McCarty in order to get both the SMPTE and AES to begin work on scientific, comprehensive new Standards for cinema and eventually home audio reproduction.

This workshop reviews the flawed Standards and presents new experiments that further define the areas of work that will need to be undertaken for new Standards to be written.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Friday, October 18, 2:30 pm — 3:30 pm (Room 1E13)

Networked Audio: N3 - The Role of Standards in Audio Networking

Chair:
Mark Yonge, Blakeney, Gloucestershire, UK
Panelists:
Jeff Berryman, Bosch Communications - Ithaca, NY, USA
Kevin Gross, AVA Networks - Boulder, CO, USA
Andreas Hildebrand, ALC NetworX - Munich, Germany
Lee Minich, Lab X Technologies - Rochester, NY, USA

Abstract:
A number of standards organizations and industry associations have been active in promoting standards relating to audio networks, such as EBU, IEC, and not least AES with recent standards AES64, AES67, and project X-210. Networks themselves are standardized under the auspices of bodies such as the IEEE and IETF. This session will describe the landscape of standards bodies and their areas of interest in audio networking and will examine the questions:

• Are standards important?
• How does all this standard activity impact the real world of audio networks?
• How do these standards benefit the marketplace, end users and the technology suppliers to this market?
• Is development of and adherence to standards better for suppliers and end users than letting the manufacturers’ proprietary solutions compete for market dominance?

 
 

Friday, October 18, 2:30 pm — 4:30 pm (Room 1E12)

Live Sound Seminar: LS6 - Wireless Microphones and Performers: Mic Placement and Handling for Multiple Actors

Panelists:
Mary McGregor, Freelance, Local 1 - New York, NY, USA
Stephanie Vetter, Freelance, Local 1 - New York, NY, USA

Abstract:
Fitting actors with wireless microphone elements and transmitters has become a detailed art form. From ensuring the actor is comfortable and the electronics are safe and secure, to getting the proper sound with minimal detrimental audio effects all while maintaining the visual illusion, one of the most widely recognized artisans in this field provide hands on demonstrations of basic technique along with some time tested “tricks of the trade.”

 
 

Friday, October 18, 3:00 pm — 4:30 pm (1EFoyer)

Poster: P12 - Signal Processing

P12-1 Temporal Synchronization for Audio Watermarking Using Reference Patterns in the Time-Frequency Domain—Tobias Bliem, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Juliane Borsum, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Giovanni Del Galdo, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Stefan Krägeloh, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Temporal synchronization is an important part of any audio watermarking system that involves an analog audio signal transmission. We propose a synchronization method based on the insertion of two-dimensional reference patterns in the time-frequency domain. The synchronization patterns consist of a combination of orthogonal sequences and are continuously embedded along with the transmitted data, so that the information capacity of the watermark is not affected. We investigate the relation between synchronization robustness and payload robustness and show that the length of the synchronization pattern can be used to tune a trade-off between synchronization robustness and the probability of false positive watermark decodings. Interpreting the two-dimensional binary patterns as one-dimensional N-ary sequences, we derive a bond for the autocorrelation properties of these sequences to facilitate an exhaustive search for good patterns.
Convention Paper 8975 (Purchase now)

P12-2 Sound Source Separation Using Interaural Intensity Difference in Real Environments—Chan Jun Chun, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Hong Kook Kim, Gwangju Institute of Science and Tech (GIST) - Gwangju, Korea
In this paper, a sound source separation method is proposed by using the interaural intensity difference (IID) of stereo audio signal recorded in real environments. First, in order to improve the channel separability, a minimum variance distortionless response (MVDR) beamformer is employed to increase the intensity difference between stereo channels. Then, IID between stereo channels processed by the beamformer is computed and applied to sound source separation. The performance of the proposed sound source separation method is evaluated on the stereo audio source separation evaluation campaign (SASSEC) measures. It is shown from the evaluation that the proposed method outperforms a sound source separation method without applying a beamformer.
Convention Paper 8976 (Purchase now)

P12-3 Reverberation and Dereverberation Effect on Byzantine Chants—Alexandros Tsilfidis, accusonus, Patras Innovation Hub - Patras, Greece; Charalampos Papadakos, University of Patras - Patras, Greece; Elias Kokkinis, accusonus - Patras, Greece; Georgios Chryssochoidis, National and Kapodistrian University of Athens - Athens, Greece; Dimitrios Delviniotis, National and Kapodistrian University of Athens - Athens, Greece; Georgios Kouroupetroglou, National and Kapodistrian University of Athens - Athens, Greece; John Mourjopoulos, University of Patras - Patras, Greece
Byzantine music is typically monophonic and is characterized by (i) prolonged music phrases and (ii) Byzantine scales that often contain intervals smaller than the Western semitone. As happens with most religious music genres, reverberation is a key element of Byzantine music. Byzantine churches/cathedrals are usually characterized by particularly diffuse fields and very long Reverberation Time (RT) values. In the first part of this work, the perceptual effect of long reverberation on Byzantine music excerpts is investigated. Then, a case where Byzantine music is recorded in non-ideal acoustic conditions is considered. In such scenarios, a sound engineer might require to add artificial reverb on the recordings. Here it is suggested that the step of adding extra reverberation can be preceded by a dereverberation processing to suppress the originally recorded non ideal reverberation. Therefore, in the second part of the paper a subjective test is presented that evaluates the above sound engineering scenario.
Convention Paper 8977 (Purchase now)

P12-4 Cepstrum-Based Preprocessing for Howling Detection in Speech Applications—Renhua Peng, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Jian Li, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Chengshi Zheng, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Xiaoliang Chen, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Xiaodong Li, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China
Conventional howling detection algorithms exhibit dramatic performance degradations in the presence of harmonic components of speech that have the similar properties with the howling components. To solve this problem, this paper proposes a cepstrum preprocessing-based howling detection algorithm. First, the impact of howling components on cepstral coefficients is studied in both theory and simulation. Second, according to the theoretical results, the cepstrum pre-processing-based howling detection algorithm is proposed. The Receiver Operating Characteristic (ROC) simulation results indicate that the proposed algorithm can increase the detection probability at the same false alarm rate. Objective measurements, such as Speech Distortion (SD) and Maximum Stable Gain (MSG), further confirm the validity of the proposed algorithm.
Convention Paper 8978 (Purchase now)

P12-5 Delayless Method to Suppress Transient Noise Using Speech Properties and Spectral Coherence—Chengshi Zheng, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Xiaoliang Chen, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Shiwei Wang, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Renhua Peng, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China; Xiaodong Li, Chinese Academy of Sciences - Beijing, China; Chinese Academy of Sciences - Shanghai, China
This paper proposes a novel delayless transient noise reduction method that is based on speech properties and spectral coherence. The proposed method has three stages. First, the transient noise components are detected in each subband by using energy-normalized variance. Second, we apply the harmonic property of the voiced speech and the continuity of the speech signal to reduce speech distortion in voiced speech segments. Third, we define a new spectral coherence to distinguish the unvoiced speech from the transient noise to avoid suppressing the unvoiced speech. Compared with those existing methods, the proposed method is computationally efficient and casual. Experimental results show that the proposed algorithm can effectively suppress transient noise up to 30 dB without introducing audible speech distortion.
Convention Paper 8979 (Purchase now)

P12-6 Artificial Stereo Extension Based on Hidden Markov Model for the Incorporation of Non-Stationary Energy Trajectory—Nam In Park, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Kwang Myung Jeon, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Seung Ho Choi, Prof., Seoul National University of Science and Technology - Seoul, Korea; Hong Kook Kim, Gwangju Institute of Science and Tech (GIST) - Gwangju, Korea
In this paper an artificial stereo extension method is proposed to provide stereophonic sound from mono sound. While frame-independent artificial stereo extension methods, such as Gaussian mixture model (GMM)-based extension, do not consider the correlation of energies of previous frames, the proposed stereo extension method employs a minimum mean-squared error estimator based on a hidden Markov model (HMM) for the incorporation of non-stationary energy trajectory. The performance of the proposed stereo extension method is evaluated by a multiple stimuli with a hidden reference and anchor (MUSHRA) test. It is shown from the statistical analysis of the MUSHRA test results that the stereo signals extended by the proposed stereo extension method have significantly better quality than those of a GMM-based stereo extension method.
Convention Paper 8980 (Purchase now)

P12-7 Simulation of an Analog Circuit of a Wah Pedal: A Port-Hamiltonian Approach—Antoine Falaize-Skrzek, IRCAM - Paris, France; Thomas Hélie, IRCAM-CNRS UMR 9912-UPMC - Paris, France
Several methods are available to simulate electronic circuits. However, for nonlinear circuits, the stability guarantee is not straightforward. In this paper the approach of the so-called "Port-Hamiltonian Systems" (PHS) is considered. This framework naturally preserves the energetic behavior of elementary components and the power exchanges between them. This guarantees the passivity of the (source-free part of the) circuit.
Convention Paper 8981 (Purchase now)

P12-8 Improvement in Parametric High-Band Audio Coding by Controlling Temporal Envelope with Phase Parameter—Kijun Kim, Kwangwoon University - Seoul, Korea; Kihyun Choo, Samsung Electronics Co., Ltd. - Suwon, Korea; Eunmi Oh, Samsung Electronics Co., Ltd. - Suwon, Korea; Hochong Park, Kwangwoon University - Seoul, Korea
This study proposes a method to improve temporal envelope control in parametric high-band audio coding. Conventional parametric high-band coders may have difficulties with controlling fine high-band temporal envelope, which can cause the deterioration in sound quality for certain audio signals. In this study a novel method is designed to control temporal envelope using spectral phase as an additional parameter. The objective and the subjective evaluations suggest that the proposed method should improve the quality of sound with severely degraded temporal envelope by the conventional method.
Convention Paper 8982 (Purchase now)

 
 

Friday, October 18, 3:30 pm — 5:00 pm (Room 1E13)

Networked Audio: N4 - Command and Control Protocols, Target Application Use Cases

Chair:
Tim Shuttleworth, Renkus Heinz - Oceanside, CA, USA
Panelists:
Jeff Berryman, Bosch Communications - Ithaca, NY, USA
Andrew Eales, Wellington Institute of Technology - Wellington, New Zealand; Rhodes University - Grahamstown, South Africa
Richard Foss, Rhodes University - Grahamstown, Eastern Cape, South Africa
Jeff Koftinoff, Meyer Sound Canada - Vernon, BC, Canada

Abstract:
With the increasing utilization of data networks for the command and control of audio devices a number of protocols have been defined and promoted. These competing protocol initiatives, while providing methods suited to their target applications, have created confusion among potential adopters as to which protocol best fits their needs. In addition, the question is being asked, Why do we need so many “standard” protocols? At least four different industry organizations have involved themselves in some form of standardized protocol effort. AES is currently pursuing standardization of two such protocols, AES64 and X-210 (aka OCA). IEC has IEC62379, while IEEE is defining AVDECC (IEE1722.1) and ESTA offers ACN and there’s OSC from opensoundcontrol.org. This workshop addresses what differentiates these protocols by examining their target use applications.

This session is presented in association with the AES Technical Committee on Network Audio Systems

 
 

Friday, October 18, 3:45 pm — 5:15 pm (Room 1E14)

Broadcast and Streaming Media: B8 - Content Delivery and the Mobile Initiative

Chair:
Neil Glassman, WhizBangPowWow - Jersey City, NJ, USA
Panelists:
Karlheinz Brandenburg, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Ilmenau University of Technology - Ilmenau, Germany
John Kean, NPR
Raymond Lau, RAMP Holdings, Inc. - Boston, MA, USA
Leigh Newsome, Targetspot - New York, NY, USA
Jan Nordmann, Fraunhofer USA - San Jose, CA, USA
Greg Ogonowski, Orban - San Leandro, CA, USA

Abstract:
Consumer use of mobile devices for entertainment and information is exploding. Smart phones and tablets are used for both primary programming and "second screen" applications. These devices are increasingly being integrated into the "connected car," where legacy receivers are no longer the only built-in listening option. Applying the term "streaming" to a broad range of delivery platforms, this panel will look at the established and nascent technical advancements that have enabled content providers to reach the expanding mobile audience. We'll also explore whether audio and data technologies are changing consumer preferences or merely keeping up with them. Panelists will also pull out their crystal balls to predict the future technologies that will help help some of the platforms grow their listener base and turn other platforms dark.

 
 

Friday, October 18, 5:30 pm — 7:00 pm (Room 1E07)

Product Design: PD2 - High-Order Harmonic Distortion Measurement of Amplifiers and its Impact on Fidelity

Presenters:
Dan Foley, Audio Precision - Worcester, MA, USA
Roger Gibboni, Rogers High Fidelity - Warwick, NY, USA

Abstract:
The electronics side of the audio industry has standardized on THD and THD+N as the main means of characterizing distortion, especially for amplifiers. However in 1942, RCA engineers who wrote the Radiotron Handbook proposed a weighted THD metric that weighted the energy of high-order harmonics to a much greater degree than low-order harmonics. Listening tests back then did show a correlation of amplifiers with very little high-order harmonic distortion being more acceptable compared to other designs with greater high-order distortion even though THD differed slightly. This presentation will focus on current measurement methods that can be used to separate high-order and low-order distortion.

 
 

Friday, October 18, 5:30 pm — 7:00 pm (Room 1E14)

Broadcast and Streaming Media: B9 - Modern Audio Transportation Techniques for Remote Broadcasts

Chair:
Herb Squire, Herb Squire - Martinsville, NJ
Panelists:
Chris Crump, Comrex
Chris Nelson, NPR
Greg Shay, The Telos Alliance - Cleveland, OH, USA
Chris Tobin, CCS-IPcodecs - Newark, NJ USA

Abstract:
Evolving technology has made great strides in audio transport versatility, connectivity, availability, and reliability. Whether wired or wireless, this discussion will provide real-time remote program solution options for broadcasters trying to make ends meet.

 
 

Saturday, October 19, 9:00 am — 10:30 am (Room 1E11)

Sound for Picture: SP3 - Dialog Editing and Mixing for Film (Sound for Pictures Master Class)

Presenters:
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Fred Rosenberg

Abstract:
Film soundtracks contain three elements—dialog, music, and sound effects. Dialog is the heart of the process, with “telling the story” the primary goal of the dialog. With multiple sources of dialog available, the assessment and planning of the dialog and subsequent mixing is a critical element in the process. This Master Class with one of Hollywood's leading professionals puts the process under the microscope.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Saturday, October 19, 9:00 am — 11:00 am (Room 1E09)

Product Design: PD3 - Telephony: An Introduction to the Acoustics of Personal Telecommunications Devices

Presenter:
Christopher Struck, CJS Labs - San Francisco, CA, USA

Abstract:
The basic concepts of telephonometry and electroacoustics of telephones and personal telecommunications devices are introduced. Techniques for assessing analogue, digital, cellular, VoIP, USB, and other telephone devices are presented. Objective evaluation of the performance of handsets, headsets, speakerphones, and hands-free devices is discussed and interfacing to these devices is explained. Selection, calibration, and use of microphones, ear simulators, mouth simulators, and test fixtures are described. The send, receive, sidetone, and echo transmission paths are defined. The use of real speech test signals and pulsed noise for distortion is illustrated using examples from IEEE 269. The concept of Loudness Rating, its history, and standardized methods for its calculation are reviewed. Methods specified in ITU-T, IEEE, TIA, ETSI, and the 3GPP standards are explained.

This session is presented in association with the AES Technical Committee on Audio for Telecommunications

 
 

Saturday, October 19, 9:00 am — 11:00 am (Room 1E12)

Live Sound Seminar: LS8 - Design Meets Reality: The A2’s and Production Sound Mixer’s Challenges, Obstacles, and Responsibilities for Loading in and Implementing the Sound Designer’s Concept

Chair:
Christopher Evans, Benedum Center - Pittsburgh, PA, USA
Panelists:
Colle Bustin, IRES-Partners, LLC - New York, NY, USA
Paul Garrity, Auerbach Pollock Friedlander - New York, NY, USA; Auerbach Pollock Friedlander - San Francisco, CA, USA
Scott Lehrer, Scott Lehrer Sound Design, Ltd. - New York, NY, USA
Augie Propersi, NYC City Center
Dominic Sack, Sound Associates, Inc.
Christopher Sloan, Production Engineer, The Book of Mormon

Abstract:
The best intentions of the sound designer don’t always fit in with the venue’s interior or infrastructure, other departments’ needs, or other changes as a production is loaded in and set up for the first time. How the designer’s designated representative on site addresses these issues is critical to keeping the overall vision of the sound design and production aesthetics intact while keeping an eye on the budget and schedule.

 
 

Saturday, October 19, 9:00 am — 10:30 am (Room 1E13)

Tutorial: T12 - FXpertise: Distortion

Presenter:
Alex Case, University of Massachusetts Lowell - Lowell, MA, USA

Abstract:
Distortion can be good, or bad. With the right touch, it can lift a track up out of a crowded arrangement and add excitement to a performance. Yet too much distortion renders the track too messy, too murky to be enjoyed. Accidental distortion is a certain sign that the production is unprofessional. Amps, stomp boxes, tubes, transformers, tape machines, the plug-ins that emulate them, and the plug-ins that create wholly new forms of distortion all offer a rich palette of possibilities. Audio engineers must choose the right tool for the job and then tailor the distortion to the music. This advanced tutorial takes a close look at distortion, detailing the technical goings-on when things break-up, and defining the production potential of this always-tempting effect.

 
 

Saturday, October 19, 9:00 am — 11:30 am (Room 1E07)

Paper Session: P13 - Applications in Audio—Part 2

Chair:
Hans Riekehof-Boehmer, SCHOEPS Mikrofone - Karlsruhe, Germany

P13-1 Level-Normalization of Feature Films Using Loudness vs Speech—Esben Skovenborg, TC Electronic - Risskov, Denmark; Thomas Lund, TC Electronic A/S - Risskov, Denmark
We present an empirical study of the differences between level-normalization of feature films using the two dominant methods: loudness normalization and speech (“dialog”) normalization. The sound of 35 recent “blockbuster” DVDs were analyzed using both methods. The difference in normalization level was up to 14 dB, on average 5.5 dB. For all films the loudness method provided the lowest normalization level and hence the greatest headroom. Comparison of automatic speech measurement to manual measurement of dialog anchors shows a typical difference of 4.5 dB, with the automatic measurement producing the highest level. Employing the speech-classifier to process rather than measure the films, a listening test suggested that the automatic measure is positively biased because it sometimes fails to distinguish between “normal speech” and speech combined with “action” sounds. Finally, the DialNorm values encoded in the AC-3 streams on DVDs were compared to both the automatically and the manually measured speech levels and found to match neither one well. AES 135th Convention Best Peer-Reviewed Paper Award Cowinner
Convention Paper 8983 (Purchase now)

P13-2 Sound Identification from MPEG-Encoded Audio Files—Joseph G. Studniarz, Montana State University - Bozeman, MT, USA; Robert C. Maher, Montana State University - Bozeman, MT, USA
Numerous methods have been proposed for searching and analyzing long-term audio recordings for specific sound sources. It is increasingly common that audio recordings are archived using perceptual compression, such as MPEG-1 Layer 3 (MP3). Rather than performing sound identification upon the reconstructed time waveform after decoding, we operate on the undecoded MP3 audio data as a way to improve processing speed and efficiency. The compressed audio format is only partially processed using the initial bitstream unpacking of a standard decoder, but then the sound identification is performed directly using the frequency spectrum represented by each MP3 data frame. Practical uses are demonstrated for identifying anthropogenic sounds within a natural soundscape recording.
Convention Paper 8984 (Purchase now)

P13-3 Pilot Workload and Speech Analysis: A Preliminary Investigation—Rachel M. Bittner, New York University - New York, NY, USA; Durand R. Begault, Human Systems Integration Division, NASA Ames Research Center - Moffett Field, CA, USA; Bonny R. Christopher, San Jose State University Research Foundation, NASA Ames Research Center - Moffett Field, CA, USA
Prior research has questioned the effectiveness of speech analysis to measure a talker's stress, workload, truthfulness, or emotional state. However, the question remains regarding the utility of speech analysis for restricted vocabularies such as those used in aviation communications. A part-task experiment was conducted in which participants performed Air Traffic Control read-backs in different workload environments. Participant's subjective workload and the speech qualities of fundamental frequency (F₀) and articulation rate were evaluated. A significant increase in subjective workload rating was found for high workload segments. F₀ was found to be significantly higher during high workload while articulation rates were found to be significantly slower. No correlation was found to exist between subjective workload and F₀ or articulation rate.
Convention Paper 8985 (Purchase now)

P13-4 Gain Stage Management in Classic Guitar Amplifier Circuits—Bryan Martin, McGill University - Montreal, QC, Canada
The guitar amplifier became a common tool in musical creation during the second half of the 20th Century. This paper attempts to detail some of the internal mechanisms by which the tones are created and their dependent interactions. Two early amplifier designs are examined to determine the circuit relationships and design decisions that came to define the sound of the electric guitar.
Convention Paper 8986 (Purchase now)

P13-5 Audio Pre-Equalization Models for Building Structural Sound Transmission Suppression—Cheng Shu, University of Rochester - Rochester, NY, USA; Fangyu Ke, University of Rochester - Rochester, NY, USA; Xiang Zhou, Bose Corporation - Framingham, MA, USA; Gang Ren, University of Rochester - Rochester, NY, USA; Mark F. Bocko, University of Rochester - Rochester, NY, USA
We propose a novel audio pre-equalization model that utilizes the transmission characteristics of building structures to reduce the interference reaching adjacent neighbors while maintaining the audio quality for the target listener. The audio transmission profiles are obtained by field acoustical measurements in several typical types of building structures. We also measure the spectrum of audio to adapt the pre-equalization model to a specific audio segment. We apply a computational auditory model to (1) monitor the perceptual audio quality for the target listener and (2) access the interference caused to adjacent neighbors. The system performance is then evaluated using subjective rating experiments.
Convention Paper 8987 (Purchase now)

 
 

Saturday, October 19, 9:30 am — 11:00 am (Room 1E08)

Broadcast and Streaming Media: B10 - Technology and Storytelling: How Can We Best Use the Tools Available to Tell Our Stories?

Panelists:
Butch D'Ambrosio, Manual SFX
Robert Fass, Voice Talent
Bill Rogers, Voice Talent
David Shinn, SueMedia Productions - Carle Place, NY, USA
Sue Zizza, SueMedia Productions - Carle Place, NY, USA

Abstract:
This session will showcase three examples of how the choices we make around technology and the way we use it effect the storytelling process for all entertainment media. With on-site demonstrations by Sue Zizza and David Shinn of SueMedia Productions.

1) Microphones and the Voice in Storytelling. Whether producing an audiobook or narration for a film or game, you want your talent to sound right for the story. This session will begin by looking at how we select microphones for voice talent. Two voice actors will demonstrate how working with different microphones effect their performance abilities.

2) Sound Effects: Studio vs. On Location Recordings. Sound Effects enhance the storytelling process by helping to create location, specific action, emotion, and more. Do you have to create every sound effect needed for your project, or can you work with a combination of already recorded elements, alongside studio produced sound effects (foley), or on-location effects, and what are some tips and tricks to recording sound design elements?

3) Digital Editing and Mixing. How can you better manage multiple voice, sound effect, and music elements into "stems," or sub-mixes for better control over final mixing as well as integrating plug-ins for mastering.

 
 

Saturday, October 19, 10:30 am — 12:00 pm (Room 1E11)

Sound for Picture: SP4 - Music Production for Film (Sound for Pictures Master Class)

Presenters:
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Simon Franglen, Class1 Media - Los Angeles, CA, USA; London
Chris Hajian

Abstract:
Film soundtracks contain three elements: dialog, music, and sound effects. The creation of a music soundtrack is far more complex than previously, now encompassing “temp music” for preview screenings, synthesizer-enhanced orchestra tracks, and other special techniques. This Master Class with one of Hollywood's leading professionals puts the process under the microscope.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Saturday, October 19, 10:30 am — 12:00 pm (Room 1E13)

Workshop: W16 - FX Design Panel: Distortion

Chair:
Jan Berg, Luleå University of Technology - Piteå, Sweden
Panelists:
Ken Bogdanowicz, SoundToys - Burlington, VT, USA
Marc Gallo, Studio Devil Virtual Tube Amplification - New York, NY, USA
Aaron Wishnick, iZotope - Somerville, MA, USA

Abstract:
Meet the designers whose talents and philosophies are reflected in the products they create, driving sound quality, ease of use, reliability, price, and all the other attributes that motivate us to patch, click, and tweak their effects processors.

 
 

Saturday, October 19, 11:00 am — 12:30 pm (Room 1E09)

Product Design: PD4 - Loudspeaker Nonlinear Identification

Presenter:
Pascal Brunet, Setem Technologies - Newbury, MA, USA

Abstract:
This presentation reviews recent developments in the domain of loudspeaker nonlinear identification and explores new possibilities to improve modeling that is better match to the loudspeaker response. First we present the loudspeaker operation principles and the major causes of distortion, then we explore the successive modeling approaches that have been investigated in the last decades. Finally we provide new directions of research in the frequency domain and propose two techniques based on state-space for modeling of loudspeaker which can effectively be used in identification process.

 
 

Saturday, October 19, 12:00 pm — 1:00 pm (Stage)

Project Studio Expo: It Won't Sound Right If You Don't Hear It Right: Studio Acoustics, Monitoring & Critical Listening

Presenters:
Hugh Robjohns, Technical Editor, Sound on Sound - Cambridge, UK
Paul White

Abstract:
The monitoring environment acoustics and the monitoring loudspeakers are critical links in every music production chain. Any weaknesses impact negatively not only on the overall quality of mixes, but also on the confidence and ability of the user to assess and process audio material efficiently and effectively. This workshop examines the theoretical requirements and practical optimization of high-quality monitoring systems for home and project studios, drawing on the author’s experiences in the “Studio SOS” series published in Sound On Sound magazine. It will also explore choosing new monitoring loudspeakers, optimizing control room acoustics, and honing critical listening skills.

 
 

Saturday, October 19, 1:00 pm — 2:00 pm (Stage)

Project Studio Expo: Take Your Studio On Stage: Live Performance with Laptops, Looping Pedals & Other Studio Tech

Presenter:
Craig Anderton, Harmony Central / Electronic Musician - Santa Fe, NM, USA

Abstract:
For many musicians, as well as DJs and electronic acts, a 21st century live performance requires much more than just a mixer and a bunch of amps. This workshop takes a practical look at how to use technology on stage without being overwhelmed by it, ways to insure a smooth performance, and includes invaluable information on the “care and feeding” of laptops to insure optimum performance—and uninterrupted performances. Other topics include using controllers for a more vibrant live performance, performing with Ableton Live and dedicated control surfaces, improvisation with looping pedals and DAW software, and the evolution of DJ controller/laptop combinations into tools for a musical, complex new art form.

 
 

Saturday, October 19, 1:30 pm — 3:00 pm (Room 1E08)

Broadcast and Streaming Media: B11 - Maintenance, Repair, and Troubleshooting

Chair:
John Bisset, Telos Alliance
Panelists:
Michael Azzarello, CBS
Bill Sacks, Orban / Optimod Refurbishing - Hollywood, MD, USA
Kimberly Sacks, Optimod Refurbishing - Hollywood, MD, USA

Abstract:
Much of today's audio equipment may be categorized as “consumer, throw-away” gear, or so complex that factory assistance is required for a board or module swap. The art of Maintenance, Repair, and Troubleshooting is actually as important as ever, even as the areas of focus may be changing. This session brings together some of the sharpest troubleshooters in the audio business. They'll share their secrets to finding problems, fixing them, and working to ensure they don't happen again. We'll delve into troubleshooting on the systems level, module level, and the component level, and explain some guiding principles that top engineers share.

 
 

Saturday, October 19, 2:00 pm — 3:30 pm (Room 1E11)

Sound for Picture: SP5 - Sound Design for Film (Sound for Pictures Master Class)

Presenters:
Michael Barry
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Eugene Gearty
Skip Lievsay

Abstract:
Film soundtracks contain three elements: dialog, music, and sound effects. Sound effects, which used to be an afterthought, are now constructed by sound designers, often working from the start of production. This Master Class with Hollywood's leading professionals puts the process under the microscope.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Saturday, October 19, 2:30 pm — 4:30 pm (Room 1E12)

Live Sound Seminar: LS10 - Production Wireless Systems: An Examination of Antennas, Coax, Filters, and Other Tips and Tricks from the Experts

Chair:
James Stoffo, Radio Active Designs - Key West, FL, USA
Panelists:
Brooks Schroeder, Frequency Coordination Group - Orlando, FL, USA
Vinnie Siniscal, Firehouse Productions - Red Hook, NY, USA
Ed Weizcerak, Freelance

Abstract:
Beyond the basics of accepted RF practices for wireless microphones, intercoms, IEMs, and IFBs is a plethora of facts about antennas, coax, and other passives not commonly understood by the production community at large. This session is comprised of an expert group of RF practitioners who will discuss the various types and performance characteristics of antennas, coax, filters, isolators/circulators, hybrid combiners, directional couplers, and other devices along with their own tips and tricks for dealing with difficult deployments.

 
 

Saturday, October 19, 3:00 pm — 4:00 pm (Stage)

Project Studio Expo: Loudness, Levels, and Metering

Presenter:
Hugh Robjohns, Technical Editor, Sound on Sound - Cambridge, UK

Abstract:
This seminar will cover the development and history of audio metering and discuss why traditional analog instruments are obsolete in the digital age. It will then cover digital metering and the associated problems, and contrast the concepts and practices of peak and loudness normalization. That will lead on to the aims of the ITU-R BS1770 loudness standard, its practical implementation, and then examples of how it has been implemented by a number of manufacturers and how it works in practice. There will be audio/visual examples throughout.

 
 

Saturday, October 19, 3:00 pm — 4:30 pm (1EFoyer)

Poster: P15 - Applications in Audio—Part I

P15-1 An Audio Game App Using Interactive Movement Sonification for Targeted Posture Control—Daniel Avissar, University of Miami - Coral Gables, FL, USA; Colby N. Leider, University of Miami - Coral Gables, FL, USA; Christopher Bennett, University of Miami - Coral Gables, FL, USA; Oygo Sound LLC - Miami, FL, USA; Robert Gailey, University of Miami - Coral Gables, FL, USA
Interactive movement sonification has been gaining validity as a technique for biofeedback and auditory data mining in research and development for gaming, sports, and physiotherapy. Naturally, the harvesting of kinematic data over recent years has been a function of an increased availability of more portable, high-precision sensory technologies, such as smart phones, and dynamic real time programming environments, such as Max/MSP. Whereas the overlap of motor skill coordination and acoustic events has been a staple to musical pedagogy, musicians and music engineers have been surprisingly less involved than biomechanical, electrical, and computer engineers in research efforts in these fields. Thus, this paper proposes a prototype for an accessible virtual gaming interface that uses music and pitch training as positive reinforcement in the accomplishment of target postures.
Convention Paper 8995 (Purchase now)

P15-2 Evaluation of the SMPTE X-Curve Based on a Survey of Re-Recording Mixers—Linda A. Gedemer, University of Salford - Salford, UK; Harman International - Northridge, CA, USA
Cinema calibration methods, which include targeted equalization curves for both dub stages and cinemas, are currently used to ensure an accurate translation of a film's sound track from dub stage to cinema. In recent years, there has been an effort to reexamine how cinemas and dub-stages are calibrated with respect to preferred or standardized room response curves. Most notable is the work currently underway reviewing the SMPTE standard ST202:2010 "For Motion-Pictures - Dubbing Stages (Mixing Rooms), Screening Rooms and Indoor Theaters -B-Chain Electroacoustic Response." There are both scientific and anecdotal reasons to question the effectiveness of the SMPTE standard in its current form. A survey of re-recording mixers was undertaken in an effort to better understand the efficaciousness of the SMPTE standard from the users' point of view.
Convention Paper 8996 (Purchase now)

P15-3 An Objective Comparison of Stereo Recording Techniques through the Use of Subjective Listener Preference Ratings—Wei Lim, University of Michigan - Ann Arbor, MI, USA
Stereo microphone techniques offer audio engineers the ability to capture a soundscape that approximates how one might hear realistically. To illustrate the differences between six common stereo microphone techniques, namely XY, Blumlein, ORTF, NOS, AB, and Faulkner, I asked 12 study participants to rate recordings of a Yamaha Disklavier piano. I examined the inter-rating correlation between subjects to find a preferential trend toward near-coincidental techniques. Further evaluation showed that there was a preference for clarity over spatial content in a recording. Subjects did not find that wider microphone placements provided for more spacious-sounding recordings. Using this information, this paper also discusses the need to re-evaluate how microphone techniques are typically categorized by distance between microphones.
Convention Paper 8997 (Purchase now)

P15-4 Tampering Detection of Digital Recordings Using Electric Network Frequency and Phase Angle—Jidong Chai, University of Tennessee - Knoxville, TN, USA; Yuming Liu, Electrical Power Research Institute, Chongqing Electric Power Corp. - Chongqing, China; Zhiyong Yuan, China Southern Power Grid - Guangzhou, China; Richard W. Conners, Virginia Polytechnic Institute and State University - Blacksburg, VA, USA; Yilu Liu, University of Tennessee - Knoxville, TN, USA; Oak Ridge National Laboratory
In the field of forensic authentication of digital audio recordings, the ENF (electric network frequency) Criterion is one of the possible tools and has shown promising results. An important task for forensic authentication is to determine whether the recordings are tampered or not. Previous work performs tampering detection by looking for the discontinuity in either the extracted ENF or phase angle from digital recordings. However, using only frequency or phase angle to detect tampering may not be sufficient. In this paper both frequency and phase angle with a corresponding reference database are used to do tampering detection of digital recordings, which result in more reliable detection. This paper briefly introduces the Frequency Monitoring Network (FNET) at UTK and its frequency and phase angle reference database. A Short-Time Fourier transform (STFT) is employed to estimate the ENF and phase angle embedded in audio files. A procedure of using the ENF criterion to detect tampering, ranging from signal preprocessing, ENF and phase angle estimation, frequency database matching to tampering detection, is proposed. Results show that utilizing frequency and phase angle jointly can improve the reliability of tampering detection in authentication of digital recordings.
Convention Paper 8998 (Purchase now)

P15-5 Portable Speech Encryption Based Anti-Tapping Device—C. R. Suthikshn Kumar, Defence Institute of Advanced Technology (DIAT) - Girinagar, Pune, India
Tapping telephones nowadays is a major concern. There is a need for a portable device that can be attached to a mobile phone that can prevent tapping. Users want to encrypt their voice during conversation, mainly for privacy. The encrypted conversation can prevent tapping of the mobile calls as the network operator may tap the calls for various reasons. In this paper we propose a portable device that can be attached to the mobile phone/landline phone that serves as an anti-tapping device. The device encrypts the speech and decrypts the encrypted speech in real time. The main idea is that speech is unintelligible when encrypted.
Convention Paper 8999 (Purchase now)

P15-6 Personalized Audio Systems—A Bayesian Approach—Jens Brehm Nielsen, Technical University of Denmark - Kongens Lyngby, Denmark; Widex A/S - Lynge, Denmark; Bjørn Sand Jensen, Technical University of Denmark - Kongens Lyngby, Denmark; Toke Jansen Hansen, Technical University of Denmark - Kongens Lyngby, Denmark; Jan Larsen, Technical University of Denmark - Kgs. Lyngby, Denmark
Modern audio systems are typically equipped with several user-adjustable parameters unfamiliar to most listeners. To obtain the best possible system setting, the listener is forced into non-trivial multi-parameter optimization with respect to the listener's own objective and preference. To address this, the present paper presents a general interactive framework for robust personalization of such audio systems. The framework builds on Bayesian Gaussian process regression in which the belief about the user's objective function is updated sequentially. The parameter setting to be evaluated in a given trial is carefully selected by sequential experimental design based on the belief. A Gaussian process model is proposed that incorporates assumed correlation among particular parameters, which provides better modeling capabilities compared to a standard model. A five-band constant-Q equalizer is considered for demonstration purposes, in which the equalizer parameters are optimized for each individual using the proposed framework. Twelve test subjects obtain a personalized setting with the framework, and these settings are significantly preferred to those obtained with random experimentation.
Convention Paper 9000 (Purchase now)

 
 

Saturday, October 19, 3:00 pm — 5:00 pm (Room 1E13)

Workshop: W19 - "Help! I Have a Tape Recorder!"—Restoration and Rebuilding Analog Tape Machines

Chair:
Noah Simon, New York University - New York, NY, USA
Panelists:
John French, JRF Magnetic Sciences Inc - Greendell, NJ, USA
Bob Shuster, Shuster Sound - Smithtown, NY USA
Daniel Zellman, Zeltec Service Labs - New York, NY, USA; Zeltec Research & Development

Abstract:
A new generation of engineers, musicians, and audiophiles are discovering how the analog recorders from the “good old days” are helping them get a better sound or get that “analog sound” into their recordings. At the same time at the other end, archivists, preservationists, remastering engineers, and high end audiophiles need to know what’s involved in taking care of these machines. This workshop will discuss the various options for these folks when they look for purchasing, maintaining, restoring, and using these recorders. During the workshop discussion, we hope to show examples of tape recorder repairs and restoration and have a running Q&A session.

This session is presented in association with the AES Technical Committee on Archiving Restoration and Digital Libraries

 
 

Saturday, October 19, 3:15 pm — 4:45 pm (Room 1E09)

Product Design: PD5 - The Power of the Brand

Presenter:
Adrian Weidmann, StoreStream Metrics - USA

Abstract:
This session will define "Brand" and explore its power and importance for the commercial success of your product development and/or service—be it a microphone, audio processing software, or recording studio. Developing, defining, and maintaining your brand and its message may the most important "product" you ever develop. This session will explore the power of Brand and outline seven key components to define your brand. The "Brand as Publisher" concept will be introduced, defined, and examples will be presented that can be used to create a meaningful dialog between your customers and your brand across available customer touchpoints—human, mobile, social media, web, and print. Understanding the power of this customer dialog can provide innovative insights for your product design team as well as propel your Brand.

 
 

Saturday, October 19, 3:15 pm — 4:45 pm (Room 1E08)

Broadcast and Streaming Media: B12 - HTML5 and Streaming

Chair:
Valerie Tyler, College of San Mateo - San Mateo, CA, USA
Panelists:
Jan Linden, Google - Mountain View, CA, USA
Greg Ogonowski, Orban - San Leandro, CA, USA
Charles Van Winkle, Adobe - Minneapolis, MN, USA

Abstract:
HTML5 is a language for structuring and presenting content for the World Wide Web, a core technology of the Internet. It is the fifth revision of the HTML standard. HTML5 has many features built into the code. One feature is the media player and how it handles media being downloaded or streamed. This session will look into the technical considerations for media to be played back as well as the user interfaces.

 
 

Saturday, October 19, 3:30 pm — 5:00 pm (Room 1E11)

Sound for Picture: SP6 - World-Class Cinema Sound Mixers Discuss Their Craft

Chair:
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Panelists:
Marti Humphrey CAS, The Dub Stage - Burbank, CA, USA
Chris M. Jacobson, The Dub Stage - Los Angeles, CA, USA
Branko Neskov, Loudness Films - Lisbon, Portugal

Abstract:
In what is fast becoming one of the most popular events in the "sound for picture" track, we again put together a panel of four of the top sound mixers for film and television.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Saturday, October 19, 4:30 pm — 7:00 pm (Room 1E12)

Live Sound Seminar: LS11 - TVBDs, Geo-Location Databases, and Upcoming Spectrum Auctions: An In-Depth Look and Their Impact on Wireless Microphone Operations

Chair:
Henry Cohen, CP Communications
Panelists:
Joe Ciaudelli, Sennheiser Electronic Corporation - Old Lyme, CT, USA
Ira Keltz, Federal Communications Commission
Michael Marcus, Marcus Spectrum Solutions - Cabin John, MD, USA
David Pawlik, Skadden, Arps, Slate, Meagher & Flom - Washington, DC, USA
Edgar Reihl, Shure, Incorporated - Niles, IL, USA
Peter Stanforth, Specrum Bridge
James Stoffo, Radio Active Designs - Key West, FL, USA

Abstract:
Television band devices (TVBD) and geo-location databases directing TVBD operations are a reality, and the first certified fixed TVBDs are in service. The 600 MHz auction may likely occur in 2014 with a vacate date within the next six to eight years. Operating wireless microphones, IEMs, intercoms, and cueing in this new environment requires understanding how the databases work, the rules governing both licensed and unlicensed wireless production equipment, and what spectrum is currently available and will be available in the future. This panel brings together a diverse group of individuals intimately involved from the beginning with TVBDs, databases, spectrum auctions, and the new FCC rules as well as seasoned veterans of medium- to large-scale wireless microphone deployments to discuss how the databases operate, how to use the database for registering TV channel usage, and best procedures and practices to insure minimal problems.

 
 

Saturday, October 19, 5:00 pm — 6:30 pm (Room 1E08)

Broadcast and Streaming Media: B13 - Facility Design

Chair:
Sergio Molho, Walters Storyk Design Group - Highland, NY, USA
Panelists:
Jim Servies, Jr., ESPN - Bristol, CT, USA
John Storyk, Walters-Storyk Design Group - Highland, NY, USA

Abstract:
Part 1: A Ground Up Design – ESPN, Bridgeport, CT
Part 2: Corrective Measures – QTV Doha, Qatar

The wisest course of action to insure optimal acoustics for broadcast facilities is to begin at the design stage. ESPN’s new production complex in Bridgeport, CT, represents an ideal example of the value of bringing acousticians in at the earliest possible opportunity. The panel will illustrate the critical issues to be addressed and the many advantages of acoustician participation at the design phase of a facility design. In a contrasting scenario, the panel will discuss QTV in Doha, Qatar. Last year after construction was completed on this state-of-the-art broadcast production complex, the three primary permanent sets designed for the new complex required sophisticated (and undetectable) acoustic treatments to alleviate excessive reverberation and related sound reflection/absorption issues. A commitment to a mid-December broadcast première presented acousticians with an inflexible sixty-day window to accomplish and evaluate, critical acoustic measurements and simulation tests, present recommendations, and complete the installation. This panel will provide insights into the evaluation and recommendation process, including a description of programs and tools, supplier outreach, installation issues, and client coordination concerns.

 
 

Saturday, October 19, 5:00 pm — 7:00 pm (Room 1E14)

Workshop: W22 - Loudness Wars: Leave Those Peaks Alone

Chair:
Thomas Lund, TC Electronic A/S - Risskov, Denmark
Panelists:
John Atkinson
Florian Camerer, ORF - Austrian TV - Vienna, Austria; EBU - European Broadcasting Union
Bob Ludwig, Gateway Mastering Studios, Inc. - Portland, ME, USA
George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada
Susan Rogers

Abstract:
Music production, distribution, and consumption has been caught in a vicious spiral rendering two decades of our music heritage damaged. Because of irreversible dynamics processing and data reduction from production onwards, new tracks and remastered ones typically sound worse than what could even be expected from compact cassette. However, with Apple, WiMP, and Spotify now engaged in a competition on quality, and FM radio in Europe adopting EBU R128 loudness normalization, limbo-practice is finally losing its grip on distribution.

The panel uses terms "Peak to Loudness Ratio" (PLR) and "Headroom" to analyze recorded music fidelity over the past 50 years from four different angles: physiological, production, distribution, and consumption. In the new realm, it's futile to master music louder than –16 LKFS.

 
 

Sunday, October 20, 9:00 am — 12:00 pm (Room 1E07)

Paper Session: P16 - Spatial Audio—Part 2

Chair:
Jean-Marc Jot, DTS, Inc. - Los Gatos, CA, USA

P16-1 Defining the Un-Aliased Region for Focused Sources—Robert Oldfield, University of Salford - Salford, Greater Manchester, UK; Ian Drumm, University of Salford - Salford, Greater Manchester, UK
Sound field synthesis reproduction techniques such as wave field synthesis can accurately reproduce wave fronts of arbitrary curvature, including sources with the wave fronts of a source in front of the array. The wave fronts are accurate up until the spatial aliasing frequency, above which there are no longer enough secondary sources (loudspeakers) to reproduce the wave front accurately, resulting in spatial aliasing contribution manifesting as additional wave fronts propagating in directions other than intended. These contributions cause temporal, spectral, and spatial errors in the reproduced wave front. Focused sources (sources in front of the loudspeaker array) have a unique attribute in this sense in that there is a clearly defined region around the virtual source position that exhibits no spatial aliasing contributions even at an extremely high frequency. This paper presents a method for the full characterization of this un-aliased region using both a ray-based propagation model and a time domain approach.
Convention Paper 9001 (Purchase now)

P16-2 Using Ambisonics to Reconstruct Measured Soundfields—Samuel W. Clapp, Rensselaer Polytechnic Institute - Troy, NY, USA; Anne E. Guthrie, Rensselaer Polytechnic Institute - Troy, NY, USA; Arup Acoustics - New York, NY, USA; Jonas Braasch, Rensselaer Polytechnic Institute - Troy, NY, USA; Ning Xiang, Rensselaer Polytechnic Institute - Troy, NY, USA
Spherical microphone arrays can measure a soundfield's spherical harmonic components, subject to certain bandwidth constraints depending on the array radius and the number and placement of the array's sensors. Ambisonics is designed to reconstruct the spherical harmonic components of a soundfield via a loudspeaker array and also faces certain limitations on its accuracy. This paper looks at how to reconcile these sometimes conflicting limitations to produce the optimum solution for decoding. In addition, binaural modeling is used as a method of evaluating the proposed decoding method and the accuracy with which it can reproduce a measured soundfield.
Convention Paper 9002 (Purchase now)

P16-3 Subjective Evaluation of Multichannel Sound with Surround-Height Channels—Sungyoung Kim, Rochester Institute of Technology - Rochester, NY, USA; Doyuen Ko, Belmont University - Nashville, TN, USA; McGill University - Montreal, Quebec, Canada; Aparna Nagendra, Rochester Institute of Technology - Rochester, NY, USA; Wieslaw Woszczyk, McGill University - Montreal, QC, Canada
In this paper we report results from an investigation of listener perception of surround-height channels added to standard multichannel stereophonic reproduction. An ITU-R horizontal loudspeaker configuration was augmented by the addition of surround-height loudspeakers in order to reproduce concert hall ambience from above the listener. Concert hall impulse responses (IRs) were measured at three heights using an innovative microphone array designed to capture surround-height ambience. IRs were then convolved with anechoic music recordings in order to produce seven-channel surround sound stimuli. Listening tests were conducted in order to determine the perceived quality of surround-height channels as affected by three loudspeaker positions and three IR heights. Fifteen trained listeners compared each reproduction condition and ranked them based on their degree of appropriateness. Results indicate that surround-height loudspeaker position has a greater influence on perceived sound quality than IR height. Listeners considered the naturalness, spaciousness, envelopment, immersiveness, and dimension of the reproduced sound field when making judgments of surround-height channel quality.
Convention Paper 9003 (Purchase now)

P16-4 A Perceptual Evaluation of Recording, Rendering, and Reproduction Techniques for Multichannel Spatial Audio—David Romblom, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Catherine Guastavino, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
The objective of this project is to perceptually evaluate the relative merits of two different spatial audio recording and rendering techniques within the context of two different multichannel reproduction systems. The two recordings and rendering techniques are "natural," using main microphone arrays, and "virtual," using spot microphones, panning, and simulated acoustic delay. The two reproduction systems are the 3/2 system (5.1 surround) and a 12/2 system, where the frontal L/C/R triplet is replaced by a 12-loudspeaker linear array. The perceptual attributes of multichannel spatial audio have been established by previous authors. In this study magnitude ratings of selected spatial audio attributes are presented for the above treatments and results are discussed.
Convention Paper 9004 (Purchase now)

P16-5 The Optimization of Wave Field Synthesis for Real-Time Sound Sources Rendered in Non-Anechoic Environments—Ian Drumm, University of Salford - Salford, Greater Manchester, UK; Robert Oldfield, University of Salford - Salford, Greater Manchester, UK
Presented here is a technique that employs audio capture and adaptive recursive filter design to render in real time dynamic, interactive, and content rich soundscapes within non-anechoic environments. Typically implementations of wave field synthesis utilize convolution to mitigate for the amplitude errors associated with the application of linear loudspeaker arrays. Although recursive filtering approaches have been suggested before, this paper aims to build on the work by presenting an approach that exploits Quasi Newton adaptive filter design to construct components of the filtering chain that help compensate for both the particular system configuration and mediating environment. Early results utilizing in-house developed software running on a 112-channel wave field synthesis system show the potential to improve the quality of real-time 3-D sound rendering in less than ideal contexts.
Convention Paper 9005 (Purchase now)

P16-6 A Perceptual Evaluation of Room Effect Methods for Multichannel Spatial Audio—David Romblom, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Catherine Guastavino, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
The room effect is an important aspect of sound recording technique and is typically captured separately from the direct sound. The perceptual attributes of multichannel spatial audio have been established by previous authors, while the psychoacoustic underpinnings of room perception are known to varying degrees. The Hamasaki Square, in combination with a delay plan and an aesthetic disposition to "natural" recordings, is an approach practiced by some sound recording engineers. This study compares the Hamasaki Square to an alternative room effect and to dry approaches in terms of a number of multichannel spatial audio attributes. A concurrent experiment investigated the same spatial audio attributes with regard to the microphone and reproduction approach. As such, the current study uses a 12/2 system based upon 3/2 (5.1 surround) where the frontal L/C/R triplet has been replaced by a linear wavefront reconstruction array. AES 135th Convention Student Technical Papers Award Cowinner
Convention Paper 9006 (Purchase now)

 
 

Sunday, October 20, 9:00 am — 10:30 am (Room 1E13)

Tutorial: T17 - FXpertise: Reverb

Presenter:
Alex Case, University of Massachusetts Lowell - Lowell, MA, USA

Abstract:
Reverberation in the recording studio comes from a variety of technologies and achieves a great range of results. Echo chambers, plates, and springs still have their place in contemporary music production even as digital reverb algorithms dominate. This tutorial reviews the technologies behind studio reverb units, shares a broad range of measurement data, and offers organization and insight into the creative, musical applications of reverb. Audio engineers reach for reverb effects to create space and ambience, to be sure. Reverb is also employed to influence timbre, create textures, invoke scene changes, manipulate masking, and synthesize new sounds entirely.

 
 

Sunday, October 20, 10:30 am — 12:30 pm (Room 1E11)

Sound for Picture: SP7 - Sound for "A Deadliest Catch"—Reality Is Hard Work

Chair:
Brian McCarty, Coral Sea Studios Pty. Ltd - Clifton Beach, QLD, Australia
Panelists:
Bob Bronow, Max Post - Burbank, CA; Audio Cocktail
Josh Earl, Original Productions - Burbank, CA, USA
Sound Crew from "Deadliest Catch"

Abstract:
Television has seen the development of a new category of TV program—the "reality" show. While many of these shows are TV fluff, one of the first was set around the dangerous profession of crab fishing in the Bering Sea. This hit rated show is one of the most difficult and challenging productions for not only the fisherman but for the capture and mixing of the soundtrack. We present two of the key Emmy-winning sound professionals in this workshop.

This session is presented in association with the AES Technical Committee on Audio for Cinema

 
 

Sunday, October 20, 10:30 am — 12:00 pm (Room 1E13)

Workshop: W26 - FX Design Panel: Reverb

Chair:
Joshua D. Reiss, Queen Mary University of London - London, UK
Panelists:
Michael Carnes, Exponential Audio - Cottonwood Heights, UT, USA
Casey Dowdell, Bricasti

Abstract:
Meet the designers whose talents and philosophies are reflected in the products they create, driving sound quality, ease of use, reliability, price, and all the other attributes that motivate us to patch, click, and tweak their effects processors.

 
 

Sunday, October 20, 11:00 am — 12:30 pm (Room 1E08)

Networked Audio: N5 - X192 / AES67: How the New Networked Audio Interoperability Standard Was Designed

Chair:
Greg Shay, The Telos Alliance - Cleveland, OH, USA
Panelists:
Kevin Gross, AVA Networks - Boulder, CO, USA
Stefan Heinzmann, Heinzmann - Konstanz, Germany
Andreas Hildebrand, ALC NetworX - Munich, Germany
Gints Linis, University of Latvia - IMCS - Riga, Latvia

Abstract:
It is said, to really understand a solution, you must clearly understand the problems it is solving. The nature of a technical specification like AES67 is that it is the end result of much discussion and deliberation. However, many of the intentions, the tradeoffs that were made, and an understanding of what problems were being solved, are not fully contained in the resulting document.

This panel will present the background of a number of the
decisions that were made and embodied into AES67. It will
describe the problems that were targeted to be solved, as best as they were understood. What were some of the difficult tradeoffs?

Networked audio will be new for some users, while some of the roots of the networked audio experience of the members of X192 go back 20 years. Given a proverbial clean slate by the AES, come listen to the reasons why the choices in AES67 were made.

 
 

Sunday, October 20, 1:00 pm — 2:15 pm (Stage)

Systems Sound Symposium: SS3 - AV/IT Convergence—The Practicalities of Networked Audio in Permanent Installations

Abstract:
The basics on digital audio networking in applications large and small. How are things shaping up in the real world?

 
 

Sunday, October 20, 2:30 pm — 4:30 pm (Room 1E10)

Product Design: PD7 - Is Your Equipment Design a Noise Problem Waiting to Happen?

Presenter:
Bill Whitlock, Jensen Transformers, Inc. - Chatsworth, CA, USA; Whitlock Consulting - Oxnard, CA, USA

Abstract:
A design goal for all audio equipment is freedom from hum and buzz. But AC power normally creates a system environment of ground voltage differences. While a balanced interface is the first line of defense against this noise source, the balanced interface itself is very poorly understood by most engineers ... and practical aspects of its design are rarely taught in engineering schools. This leads engineers to design balanced input circuits that perform impressively in the lab but exhibit poor noise rejection in real-world systems. To make matters worse, internal equipment grounding schemes are often thoughtlessly designed. Two common results are noise coupled via cable shield connections, known as the "pin 1 problem," and by the AC power cord (so-called "sensitive" equipment). These and other design pitfalls, and how to avoid them, are the focus of this class.

 
 

---

Return to Broadcast & Streaming Media Track Events

REGISTER TODAY