AES New York 2015
Broadcast & Streaming Media Track Event Details

Thursday, October 29, 9:00 am — 12:30 pm (Room 1A08)

Paper Session: P1 - Signal Processing

Chair:
Scott Norcross, Dolby Laboratories - San Francisco, CA, USA

P1-1 Time-Frequency Analysis of Loudspeaker Sound Power Impulse Response—Pascal Brunet, Samsung Research America - Valencia, CA USA; Audio Group - Digital Media Solutions; Allan Devantier, Samsung Research America - Valencia, CA, USA; Adrian Celestinos, Samsung Research America - Valencia, CA, USA
In normal conditions (e.g., a living room) the total sound power emitted by the loudspeaker plays an important role in the listening experience. Along with the direct sound and first reflections, the sound power defines the loudspeaker performance in the room. The acoustic resonances of the loudspeaker system are especially important, and thanks to spatial averaging, are more easily revealed in the sound power response. In this paper we use time-frequency analysis to study the spatially averaged impulse response and reveal the structure of its resonances. We also show that the net effect of loudspeaker equalization is not only the attenuation of the resonances but also the shortening of their duration.
Convention Paper 9354 (Purchase now)

P1-2 Low-Delay Transform Coding Using the MPEG-H 3D Audio Codec—Christian R. Helmrich, International Audio Laboratories - Erlangen, Germany; Michael Fischer, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Recently the ISO/IEC MPEG-H 3D Audio standard for perceptual coding of one or more audio channels has been finalized. It is a little-known fact that, particularly for communication applications, the 3D Audio core-codec can be operated in a low-latency configuration in order to reduce the algorithmic coding/decoding delay to 44, 33, 24, or 18 ms at a sampling rate of 48 kHz. This paper introduces the essential coding tools required for high-quality low-delay coding–transform splitting, intelligent gap filling, and stereo filling–and demonstrates by means of blind listening tests that the achievable subjective performance compares favorably with, e.g., that of HE-AAC even at low bit-rates.
Convention Paper 9355 (Purchase now)

P1-3 Dialog Control and Enhancement in Object-Based Audio Systems—Jean-Marc Jot, DTS, Inc. - Los Gatos, CA, USA; Brandon Smith, DTS, Inc. - Bellevue, WA, USA; Jeff Thompson, DTS, Inc. - Bellevue, WA, USA
Dialog is often considered the most important audio element in a movie or television program. The potential for artifact-free dialog salience personalization is one of the advantages of new object-based multichannel digital audio formats, along with the ability to ensure that dialog remains comfortably audible in the presence of concurrent sound effects or music. In this paper we review some of the challenges and requirements of dialog control and enhancement methods in consumer audio systems, and their implications in the specification of object-based digital audio formats. We propose a solution incorporating audio object loudness metadata, including a simple and intuitive consumer personalization interface and a practical head-end encoder extension.
Convention Paper 9356 (Purchase now)

P1-4 Frequency-Domain Parametric Coding of Wideband Speech–A First Validation Model—Aníbal Ferreira, University of Porto - Porto, Portugal; Deepen Sinha, ATC Labs - Newark, NJ, USA
Narrow band parametric speech coding and wideband audio coding represent opposite coding paradigms involving audible information, namely in terms of the specificity of the audio material, target bit rates, audio quality, and application scenarios. In this paper we explore a new avenue addressing parametric coding of wideband speech using the potential and accuracy provided by frequency-domain signal analysis and modeling techniques that typically belong to the realm of high-quality audio coding. A first analysis-synthesis validation framework is described that illustrates the decomposition, parametric representation, and synthesis of perceptually and linguistically relevant speech components while preserving naturalness and speaker specific information.
Convention Paper 9357 (Purchase now)

P1-5 Proportional Parametric Equalizers—Application to Digital Reverberation and Environmental Audio Processing—Jean-Marc Jot, DTS, Inc. - Los Gatos, CA, USA
Single-band shelving or presence boost/cut filters are useful building blocks for a wide range of audio signal processing functions. Digital filter coefficient formulas for elementary first- or second-order IIR parametric equalizers are reviewed and discussed. A simple modification of the classic Regalia-Mitra design yields efficient solutions for tunable digital equalizers whose dB magnitude frequency response is proportional to the value of their gain control parameter. Practical applications to the design of tone correctors, artificial reverberators and environmental audio signal processors are described.
Convention Paper 9358 (Purchase now)

P1-6 Comparison of Parallel Computing Approaches of a Finite-Difference Implementation of the Acoustic Diffusion Equation Model—Juan M. Navarro, UCAM - Universidad Católica San Antonio - Guadalupe (Murcia), Spain; Baldomero Imbernón, UCAM Catholic University of San Antonio - Murcia, Spain; José J. López, Universitat Politcnica de Valencia - Valencia, Spain; José M. Cecilia, UCAM Catholic University of San Antonio - Murcia, Spain
The diffusion equation model has been intensively researched as a room-acoustics simulation algorithm during last years. A 3-D finite-difference implementation of this model was proposed to evaluate the propagation over time of sound field within rooms. Despite the computational saving of this model to calculate the room energy impulse response, elapsed times are still long when high spatial resolutions and/or simulations in several frequency bands are needed. In this work several data-parallel approaches of this finite-difference solution on Graphics Processing Units are proposed using a compute unified device architecture programming model. A comparison of their performance running on different models of Nvidia GPUs is carried out. In general, 2D vertical block approach running in a Tesla K20C shows the best speed-up of more than 15 times versus CPU version.
Convention Paper 9359 (Purchase now)

P1-7 An Improved and Generalized Diode Clipper Model for Wave Digital Filters—Kurt James Werner, Center for Computer Research in Music and Acoustics (CCRMA) - Stanford, CA, USA; Stanford University; Vaibhav Nangia, Stanford University - Stanford, CA, USA; Alberto Bernardini, Politecnico di Milano - Milan, Italy; Julius O. Smith, III, Stanford University - Stanford, CA, USA; Augusto Sarti, Politecnico di Milano - Milan, Italy
We derive a novel explicit wave-domain model for “diode clipper" circuits with an arbitrary number of diodes in each orientation, applicable, e.g., to wave digital filter emulation of guitar distortion pedals. Improving upon and generalizing the model of Paiva et al. (2012), which approximates reverse-biased diodes as open circuits, we derive a model with an approximated correction term using two Lambert W functions. We study the energetic properties of each model and clarify aspects of the original derivation. We demonstrate the model's validity by comparing a modded Tube Screamer clipping stage emulation to SPICE simulation.
Convention Paper 9360 (Purchase now)

 
 

Thursday, October 29, 9:00 am — 11:00 am (Room 1A10)

Tutorial: T2 - Microphones—Can You Hear the Specs?

Chair:
Eddy B. Brixen, EBB-consult - Smørum, Denmark; DPA Microphones
Panelists:
Jürgen Breitlow, Georg Neumann Berlin - Berlin, Germany; Sennheiser Electronic - Wedemark, Germany
David Josephson, Josephson Engineering, Inc. - Santa Cruz, CA, USA
Helmut Wittek, SCHOEPS GmbH - Karlsruhe, 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 valuable information in the microphone specifications. This tutorial demystify the most important microphone specs and provide the attendee with up-to-date information on how these specs are obtained and understood and how the numbers relate to the perceived sound. It takes a critical look on how specs are presented to the user, what to look for, and what to expect.

 
 

Thursday, October 29, 10:45 am — 12:45 pm (Room 1A21)

Workshop: W2 - Mixing Music

Chair:
Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
Panelists:
Buford Jones, Meyer Sound - Berekley, CA
George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada
Shawn Murphy, Recording and Live Sound Engineer - Berkeley, CA, USA

Abstract:
Panel discussion and presentations from award-winning expert practitioners in the industry, describing the process of mixing, actual techniques used, and proven methodologies that have yielded successful results over the years. Focus will remain on real information as opposed to anecdotes, such as different ways to approach a mix, how to improve an existing mix, how to best interpret and address mix comments from an artist or a client, or the record label (it happens). A large portion of time will be left open for questions, so that the audience will have the chance to solicit specific and meaningful information from the panelists.

 
 

Thursday, October 29, 11:00 am — 12:30 pm (S-Foyer 1)

Poster: P3 - Transducers/Perception

P3-1 Predicting the Acoustic Power Radiation from Loudspeaker Cabinets: A Numerically Efficient Approach—Mattia Cobianchi, B&W Group Ltd. - West Sussex, UK; Martial Rousseau, B&W Group Ltd. - West Sussex, UK
Loudspeaker cabinets should not contribute at all to the total sound radiation but aim instead to be a perfectly rigid box that encloses the drive units. To achieve this goal, state of the art FEM software packages and Doppler vibro-meters are the tools at our disposal. The modeling steps covered in the paper are: measuring and fitting orthotropic material properties, including damping; 3D mechanical modeling with a curvilinear coordinates system and thin elastic layers to represent glue joints; scanning laser Doppler measurements and single point vibration measurements with an accelerometer. Additionally a numerically efficient post-processing approach used to extract the total radiated acoustic power and an example of what kind of improvement can be expected from a typical design optimization are presented.
Convention Paper 9367 (Purchase now)

P3-2 New Method to Detect Rub and Buzz of Loudspeakers Based on Psychoacoustic Sharpness—Tingting Zhou, Nanjing Normal University - Nanjing, Jiangsu, China; Ming Zhang, Nanjing Normal University - Nanjing, Jiangsu, China; Chen Li, Nanjing Normal University - Nanjing, Jiangsu, China
The distortion detection of loudspeakers has been researched for a very long time. Researchers are committed to finding an objective way to detect Rub and Buzz (R&B) in loudspeakers that is in line with human ear feelings. This paper applies the psychoacoustics to distortion detection of loudspeakers and describes a new method to detect the R&B based on the psychoacoustic sharpness. Experiments show, comparing with existing objective detection methods of R&B, detection results based on the proposed method are more consistent with subjective judgments.
Convention Paper 9368 (Purchase now)

P3-3 Modal Impedances and the Boundary Element Method: An Application to Horns and Ducts—Bjørn Kolbrek, Norwegian University of Science and Technology - Trondheim, Norway
Loudspeaker horns, waveguides, and other ducts can be simulated by general numerical methods, like the Finite Element or Boundary Element Methods (FEM or BEM), or by a method using a modal description of the sound field, called the Mode Matching Method (MMM). BEM and FEM can describe a general geometry but are often computationally expensive. MMM, on the other hand, is fast, easily scalable, requires no mesh generation and little memory but can only be applied to a limited set of geometries. This paper shows how BEM and MMM can be combined in order to efficiently simulate horns where part of the horn must be described by a general meshed geometry. Both BEM-MMM and MMM-BEM couplings are described, and examples given.
Convention Paper 9369 (Purchase now)

P3-4 Audibility Threshold of Auditory-Adapted Exponential Transfer-Function Smoothing (AAS) Applied to Loudspeaker Impulse Responses—Florian Völk, Technische Universität München - München, Germany; WindAcoustics UG (haftungsbeschränkt) - Windach, Germany; Yuliya Fedchenko, Technische Universität München - Munich, Germany; Hugo Fastl, Technical University of Munich - Munich, Germany
A reverberant acoustical system’s transfer function may show deep notches or pronounced peaks, requiring large linear amplification in the play-back system when used, for example, in auralization or for convolution reverb. It is common practice to apply spectral smoothing, with the aim of reducing spectral fluctuation without degrading auditory-relevant information. A procedure referred to as auditory-adapted exponential smoothing (AAS) was proposed earlier, adapted to the spectral properties of the hearing system by implementing frequency-dependent smoothing bandwidths. This contribution presents listening experiments aimed at determining the audibility threshold of auditory-adapted exponential smoothing, which is the maximum amount of spectral smoothing allowed without being audible. As the results depend on the specific acoustic system, parametrization guidelines are proposed.
Convention Paper 9371 (Purchase now)

P3-5 Developing a Timbrometer: Perceptually-Motivated Audio Signal Metering—Duncan Williams, University of Plymouth - Devon, UK
Early experiments suggest that a universally agreed upon timbral lexicon is not possible, and nor would such a tool be intrinsically useful to musicians, composers, or audio engineers. Therefore the goal of this work is to develop perceptually-calibrated metering tools, with a similar interface and usability to that of existing loudness meters, by making use of a linear regression model to match large numbers of acoustic features to listener reported timbral descriptors. This paper presents work towards a proof-of-concept combination of acoustic measurement and human listening tests in order to explore connections between 135 acoustic features and 3 timbral descriptors, brightness, warmth, and roughness.
Convention Paper 9372 (Purchase now)

P3-6 A Method of Equal Loudness Compensation for Uncalibrated Listening Systems—Oliver Hawker, Birmingham City University - Birmingham, UK; Yonghao Wang, Birmingham City University - Birmingham, UK
Equal-loudness contours represent the sound-pressure-level-dependent frequency response of the auditory system, which implies an arbitrary change in the perceived spectral balance of a sound when the sound-pressure-level is modified. The present paper postulates an approximate proportional relationship between loudness and sound-pressure-level, permitting relative loudness modification of an audio signal while maintaining a constant spectral balance without an absolute sound-pressure-level reference. A prototype implementation is presented and accessible at [1]. Preliminary listening tests are performed to demonstrate the benefits of the described method.
Convention Paper 9373 (Purchase now)

 
 

Thursday, October 29, 11:15 am — 12:45 pm (Room 1A10)

Broadcast and Streaming Media: B1 - Streaming Facilities—Broadcast Scaled to Internet Feeds

Chair:
John Storyk, Architect, Studio Designer and Principal, Walters-Storyk Design Group - Highland, NY, USA
Panelists:
Renato Cipriano, Walters Storyk Design Group - Belo Horizonte, Brazil
David Pentecost, Lower Eastside Girls Club of New York - New York, NY
Nick Squire, Boston Symphony Orchestra - Brookline, MA, USA
Jonathan Talley, (le) poisson rouge - New York, NY, USA

Abstract:
Streaming’s emergence as the de facto 21st Century programming distribution medium was confirmed earlier this year when Apple Inc. announced its exclusive "HBO Now" streaming service. Wireless device-viewers may have already eclipsed the numbers of traditional cable TV audiences. As with any new media format, the need for cutting edge production/post-production skills and facilities remains a critical element. The scale of these studios may be smaller than conventional facilities, but their technology and acoustics must be up to professional broadcast standards. This panel will explore four highly diverse and rapidly expanding streaming content producers. It will survey the similarities and disparities between broadcast/cable and streaming facility designs, acoustic requirements, and issues. And, it will provide unparalleled insights into artist (and audience) preferences and technical requirements for high quality live streaming performances.

 
 

Thursday, October 29, 11:15 am — 12:45 pm (Room 1A06)

Tutorial: T5 - Mic It & Record It!

Presenter:
Ian Corbett, Kansas City Kansas Community College - Kansas City, KS, USA; off-beat-open-hats recording & sound reinforcement

Abstract:
Too many resources emphasize “instant” miking solutions, and tell the aspiring recording engineer to simply “mic it this way.” This often results in sounds that have to be significantly electronically processed to force them into place during mixing, degrading the integrity of the sound and making the mix process longer and more difficult than it otherwise might be. Topics discussed in this presentation will include the effects microphone technologies, mic techniques, and the recording room have on the recorded sound, and how they can be explored and exploited to capture the sound you actually need for the mix, improving your mix, and making mixing an easier and quicker process.

 
 

Thursday, October 29, 2:15 pm — 3:15 pm (Room 1A14)

Networked Audio: N1 - Basic Networking and Layer 3 - Protocols: Layers, Models? A Disambiguation in the Context of Audio over IP

Presenter:
Kieran Walsh, Audinate Pty. Ltd. - Ultimo, NSW, Australia

Abstract:
The OSI model is a great starting point to understand a structure for integrating network protocols and creating software. Topics for discussion include: • Examining the positives of a layered approach and fill in the “missing gaps” that are required to create a real implementation. • An implementation from a “solution provider” (manufacturer) is different to creating a real “on the ground” working full system – the Model and layered approach however can be valuable in converging these two challenges. • “Protocols, Standards, implementations” These terms are used interchangeably—they, however, have distinct meanings; we will examine the differences and distinctions of these terms. • Deploying core AoIP services in the context of other technologies that can be leveraged to make a fully working system function in an effective production environment. • Distinguishing between standards, implementations, transports, protocols, layers and have a better insight into what each means and how to define requirements for systems. • Understanding the IT centric approach to a network, and identify challenges and workarounds when deploying an AoIP system. • Understanding some techniques that “come for free” in an enterprise IT network environment

 
 

Thursday, October 29, 2:15 pm — 3:45 pm (Room 1A10)

Broadcast and Streaming Media: B2 - Streaming Audio from the Cloud

Moderator:
John Kean, Consultant - Washington DC, USA
Panelists:
Michael Dube, NPR - Washington D.C.
Dan Jeselsohn, New York Public Radio - New York, NY, USA
Kyle Wesloh, American Public Media
Adrian Wisbey, BBC FM Media Services - London, UK

Abstract:
Delivery of audio content via the online "cloud" has become a significant consumer media enjoyed by millions of listeners each day. This panel will discuss their own form of delivery, such as streaming, podcasts, or progressive file transfer, and their system architecture. The panelists are encouraged to talk about the audio codec(s) and bit rates they use, and why. All are invited to address the forthcoming guidelines for audio loudness being developed by the AES: their likes, concerns, and suggestions for implementing the guidelines in their own system.

 
 

Thursday, October 29, 2:30 pm — 5:30 pm (Room 1A08)

Paper Session: P4 - Transducers—Part 1: Headphones, Amplifiers, and Microphones

Chair:
Christopher Struck, CJS Labs - San Francisco, CA, USA; Acoustical Society of America

P4-1 Headphone Response: Target Equalization Trade-offs and Limitations—Christopher Struck, CJS Labs - San Francisco, CA, USA; Acoustical Society of America; Steve Temme, Listen, Inc. - Boston, MA, USA
The effects of headphone response and equalization are examined with respect to the influence on perceived sound quality. Free field, diffuse field, and hybrid real sound field targets are shown and objective response data for a number of commercially available headphones are studied and compared. Irregular responses are examined to determine the source of response anomalies, whether these can successfully be equalized and what the limitations are. The goal is to develop a robust process for evaluating and appropriately equalizing headphone responses to a psychoacoustically valid target and to understand the constraints.
Convention Paper 9374 (Purchase now)

P4-2 A Headphone Measurement System Covers both Audible Frequency and beyond 20 kHz—Naotaka Tsunoda, Sony Corporation - Shinagawa-ku, Tokyo, Japan; Takeshi Hara, Sony Corporation - Tokyo, Japan; Koji Nageno, Sony Corporation - Tokyo, Japan
New headphone measurement system consisting of a 1/8” microphone and newly developed HATS (Head And Torso Simulator) with a coupler that have realistic ear canal shape is proposed to enable entire frequency response measurement from audible frequency and higher frequency area up to 140 kHz. At the same time a new frequency response evaluation scheme based on HRTF correction is proposed. Measurement results obtained by this scheme enables much better understanding by enabling direct comparison with free field loudspeaker frequency response.
Convention Paper 9375 (Purchase now)

P4-3 Measurements of Acoustical Speaker Loading Impedance in Headphones and Loudspeakers—Jason McIntosh, McIntosh Applied Engineering - Eden Prairie, MN, USA
The acoustical design of two circumaural headphones and a desktop computer speaker have been studied by measuring the acoustical impedance of the various components in their design. The impedances were then used to build an equivalent circuit model for the devices that then predicted their pressure response. There was seen to be good correlation between the model and measurements. The impedance provides unique insight into the acoustic design that is not observed though electrical impedance or pressure response measurements that are commonly relied upon when designing such devices. By building models for each impedance structure, it is possible to obtain an accurate model of the whole system where the effects of each component upon the device's overall performance can be seen.
Convention Paper 9376 (Purchase now)

P4-4 Efficiency Investigation of Switch-Mode Power Audio Amplifiers Driving Low Impedance Transducers—Niels Elkjær Iversen, Technical University of Denmark - Lyngby, Denmark; Henrik Schneider, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
The typical nominal resistance span of an electro dynamic transducer is 4 Ohms to 8 Ohms. This work examines the possibility of driving a transducer with a much lower impedance to enable the amplifier and loudspeaker to be directly driven by a low voltage source such as a battery. A method for estimating the amplifier rail voltage requirement as a function of the voice coil nominal resistance is presented. The method is based on a crest factor analysis of music signals and estimation of the electrical power requirement from a specific target of the sound pressure level. Experimental measurements confirm a huge performance leap in terms of efficiency compared to a conventional battery-driven sound system. Future optimization of low voltage, high current amplifiers for low impedance loudspeaker drivers are discussed.
Convention Paper 9377 (Purchase now)

P4-5 Self-Oscillating 150 W Switch-Mode Amplifier Equipped with eGaN-FETs—Martijn Duraij, Technical University of Denmark - Lyngby, Denmark; Niels Elkjær Iversen, Technical University of Denmark - Lyngby, Denmark; Lars Press Petersen, Technical University of Denmark - Kgs. Lyngby, Denmark; Patrik Boström, Bolecano Holding AB - Helsingborg, Sweden
Where high-frequency clocked system switch-mode audio power amplifiers equipped with eGaN-FETs have been introduced in the past years, a novel self-oscillating eGaN-FET equipped amplifier is presented. A 150 Wrms amplifier has been built and tested with regard to performance and efficiency with an idle switching frequency of 2 MHz. The amplifier consists of a power-stage module with a self-oscillating loop and an error-reducing global loop. It was found that an eGaN-FET based amplifier shows promising potential for building high power density audio amplifiers with excellent audio performance. However care must be taken of the effects caused by a higher switching frequency.
Convention Paper 9378 (Purchase now)

P4-6 Wind Noise Measurements and Characterization Around Small Microphone Ports—Jason McIntosh, Starkey Hearing Technologies - Eden Prairie, MN, USA; Sourav Bhunia, Starkey Hearing Technologies - Eden Prairie, MN, USA
The physical origins of microphone wind noise is discussed and measured. The measured noise levels are shown to correlate well to theoretical estimates of non-propagating local fluid dynamic turbulence pressure variations called “convective pressure.” The free stream convective pressure fluctuations may already be present in a flow independent of its interactions with a device housing a microphone. Consequently, wind noise testing should be made in turbulent air flows rather than laminar. A metric based on the Speech Intelligibility Index (SII) is proposed for characterizing wind noise effects for devices primarily designed to work with speech signals, making it possible to evaluate nonlinear processing effects on reducing wind noise on microphones.
Convention Paper 9379 (Purchase now)

 
 

Thursday, October 29, 3:30 pm — 4:30 pm (Room 1A14)

Networked Audio: N2 - AVB/TSN Ethernet Is Built-In Everywhere Now; How Do You Make the Most of It? A System Implementation Primer for Consultants and Tech Managers

Chair:
Tim Shuttleworth, Renkus Heinz - Oceanside, CA, USA
Panelists:
Richard Bugg, Meyer Sound - North HIlls, CA, USA
Jim Cooper, MOTU - Cambridge, MA, USA
Tom Knesel, Pivitec - Emmaus, PA, USA
Nathan Phillips, Coveloz Technologies - Ottawa, ON, Canada
Curtis Rex Reed, Harman - South Jordan, Utah

Abstract:
This presentation will introduce technology managers, integrators, and specifiers to the basics of distributing audio, video, and control signals over an Ethernet network in ready-to-play fashion. The presentation will also focus on system implementation with Time Sensitive Networking (TSN) standards—the evolution of Audio Video Bridging (AVB).

Attendees will be provided with a system-level understanding on how to achieve networked AV success; discuss the advantages of using a network; and overview challenges and approaches and provide tips and troubleshooting for networking with AVB/TSN. Discover how easy it is to scale and upgrade TSN systems.

An overview of the methods of time synchronization will also be outlined. AVnu Alliance will start by reviewing system requirements for demanding applications such as performance venue installs, house of worship, large convention systems, conference rooms and broadcast and discuss the Ethernet capabilities needed for the network including characteristics and definitions of TSN for these applications.

The presentation will highlight the importance of certification for interoperability. Finally, AVnu Alliance will present the existing tools and resources that designers need for successful TSN system operation.

Learning Objectives: • Gain a basic understanding of distributing audio, video and control systems over an Ethernet network and the advantages to doing so. • Understand the existing tools and resources that designers need to successfully operate TSN systems. • Understand what is required from a network for applications such as performance venue installs, houses of worship, conference rooms etc.

 
 

Thursday, October 29, 4:00 pm — 5:30 pm (Room 1A10)

Broadcast and Streaming Media: B3 - Loudness for Streaming

Moderator:
Bob Katz, Digital Domain Mastering - Orlando, FL, USA
Panelists:
Rob Byers, American Public Media - St. Paul, MN, USA
John Kean, Consultant - Washington DC, USA
Thomas Lund, Genelec Oy - Iisalmi, Finland
Scott Norcross, Dolby Laboratories - San Francisco, CA, USA
Adrian Wisbey, BBC FM Media Services - London, UK

Abstract:
The advent of Internet streaming services has shaken the entire audio industry. Every sector has been quickly affected: Broadcast, radio, TV, music production. One of the prime problems is that of regulating audio levels. There is already a de facto loudness war among streamers, with some using a high target level that requires them to use extreme amounts of compression and limiting that alter producer's intent and/or cause distortion, dilute the impact, etc. As a result, producers may fall into the trap of loudness envy and create their recordings at the lowest common denominator of sound quality by trying to match the level of the loudest streaming service. Enter the AES Subcommittee on Loudness in Streaming and Network Playback, which has produced a new set of Recommendations for streaming entities. Let us hope that these recommendations help civilize the wild west of streaming. Learn about the issues and the new recommendations.

 
 

Thursday, October 29, 5:00 pm — 6:30 pm (Room 1A06)

Tutorial: T7 - Keep it "Reel": The Ultimate Ultra-Portable Production/Recording Studio—From Idea to Final Master: How to Write, Sequence, Record, and Produce Your Music Using Only Your iPad

Presenter:
Andrea Pejrolo, Berklee College of Music - Boston, MA, USA

Abstract:
In this highly interactive and hands on presentation you will learn the tools, techniques, tips, and tricks required to write, produce, and mix a song using only your iPad.

Through practical examples and scenarios you will learn how to: • Pick the best software for sequencing, producing and mixing your music • Pick the best iPad-compatible hardware tools (microphones, audio interface, MIDI interfaces, controller etc.) • Setup your mobile production/recording studio • Sketch your musical ideas • Use your iPad as a creative inspirational tool for music composition and sound design • Sequence and arrange your music ideas on your iPad • Use your iPad as a powerful virtual mixing console • Add audio plug-ins • Master your final mix.
Who should attend? Anyone who wants to create some great music with their iPads, from beginners to advanced. Attendees will learn: • How to assemble the ultimate ultra-portable multi-track recording rig • How to create a complete final production of a song using only the iPad • How to pick the right hardware and software available for mobile music production on the iPad • How to take advantage of the highly interactive interface of the iPad to streamline and enhance your creative process • How to share and interact with other music creators using solely the iPad and the cloud.

 
 

Thursday, October 29, 7:30 pm — 9:30 pm (Off-Site 1)

Special Event: 50th Anniversary of the Master Antenna on the Empire State Building

Co-moderators:
David Bialik, CBS - New York, NY, USA
Scott Fybush, Northeast Radio Watch
Panelists:
Andy Lanset, Historian, WNYC/WQXR - New York, NY, USA
Shane O'Donoghue, Director of Broadcasting, Empire State Building - New York, NY, USA
Tom Silliman, Electronics Research Inc.
Herb Squire, Herb Squire - Martinsville, NJ
Robert Tarsio, Broadcast Devices Inc.

Abstract:
Limited amount of tickets—first come first served! To attend you must have registered for the Convention and go to the Tech Tours desk on Wednesday Oct. 28th (3pm - 7pm) or Thursday Oct. 29th (8am - 1pm) to obtain your ticket. Do not call AES beforehand—you can ONLY register on-site during the above hours!

On December 9, the Alford master FM antenna that rings the 102nd floor observation gallery at the Empire State Building will mark its 50th anniversary. When the antenna went into service in 1965, it marked a revolution in FM broadcast technology: for the first time, most of a market's FM signals could share a single antenna, sharing costs and reducing the amount of space needed for FM transmission at the market's tallest broadcast site.

The Alford antenna at Empire was the model for master antenna sites in places such as Toronto, St. Louis, Houston, Minneapolis-St. Paul, and eventually back at Empire, where a new master antenna system was commissioned in the 1980s to supplant the original Alford. The original 1965 Alford antenna continues to serve as a backup at Empire, being pressed into service after 9/11 to provide emergency replacement antenna capacity for stations that were displaced from their World Trade Center sites.

This special commemorative event will be held in the 67th floor conference room at the Empire State Building.

Produced jointly with AES and SBE

Watch a video of the "switch throwing" presentation here:
https://www.facebook.com/AES.org/videos/10153364038858585/

 
 

Friday, October 30, 9:00 am — 10:30 am (Room 1A10)

Broadcast and Streaming Media: B4 - Audio and IP: Are We There Yet?

Moderator:
Steve Lampen, Belden - San Francisco, CA, USA
Panelists:
Kevin Gross, AVA Networks - Boulder, CO, USA
David Josephson, Josephson Engineering, Inc. - Santa Cruz, CA, USA
Dan Mortensen, Dansound Inc. - Seattle, WA, USA
Tony Peterle, Worldcast Systems - Miami, FL, USA
Tim Pozar, Fandor - San Francisco, CA, USA

Abstract:
In 2010, Reed Hundt, former head of the Federal Communications Commission, said in a speech at Columbia Business School, [We] “decided in 1994 that the Internet should be the common medium in the United States and broadcast should not be.” This was twenty-one years ago. So, are we there yet? I tried to invite Mr. Hundt to participate on this panel, but he is too well protected, I couldn’t even get an invitation to him.

This panel of esteemed experts will look at the “big picture” of audio in networked formats and internet delivery systems. Do we have the hardware and software we need? If not, what is missing? Can we expect the same quality, consistency, and reliability as we had in the old analog audio days? There are dozens, maybe hundreds, of companies using proprietary Layer 2/Layer 3 Ethernet for audio, and there is much work on combining or cross-fertilizing these systems, such as Dante and Ravenna. There are also new standards such as IEEE 802.1BA-2011 AVB (Audio-Video Bridging), and IEEE 802.1ASbt TSN (Time-Sensitive Networks) that use specialized Ethernet switches in a network architecture. But these do not address anything outside of the Ethernet network itself. Then we have AES IP67, specifically looking at “high performance” IP-based audio.

Mixed in with this is the question “What is a broadcaster? Do you have to have a transmitter to be a broadcaster?” Consider that next year (2016) one company claims they will be the largest broadcaster in the world, and that company is Netflix.

 
 

Friday, October 30, 9:00 am — 10:45 am (Room 1A21)

Tutorial: T11 - Main Microphone Techniques for 2.0 and 5.1

Presenter:
Helmut Wittek, SCHOEPS GmbH - Karlsruhe, Germany

Abstract:
This tutorial will explain step-by-step, using many practical examples, what a suitable stereophonic microphone array can look like. With 2.0 stereo setups as the starting point, multichannel setups will also be introduced.

Many factors influence the choice of a stereophonic microphone setup, but the relevance of these factors can vary greatly depending on the application, such that there is never one single “correct” setup. Knowledge of various options gives a Tonmeister the ability to make optimal choices.

In this session the free iPhone App "Image Assistant" will be presented. It calculates the spatial characteristics of arbitrary stereophonic microphone arrays and auralizes the result. Moreover, the educational website "hauptmikrofon.de" is presented offering various comparative sound samples on the subject.

 
 

Friday, October 30, 10:45 am — 12:15 pm (Room 1A10)

Broadcast and Streaming Media: B5 - Audience Measurement for Stream and Broadcast

Moderator:
David Layer, National Association of Broadcasters - Washington, DC, USA
Panelists:
Frank Foti, Telos Systems/Omnia Audio - New York, NY, USA
Rob Green, Vice President WO Streaming, WideOrbit, Inc. - Seattle, WA, USA
John Rosso, President, Market Development, Triton Digital - New York, NY, USA

Abstract:
Radio broadcasting in the 21st Century is not just about over-the-air signals. The Internet and mobile broadband have made audio streaming a viable option to over-the-air delivery. One thing that has not changed, audience measurement is still fundamental to the business of radio no matter how the signal is being delivered. This 90-minute session will focus on some of the latest audience measurement techniques for both over-the-air and streaming and help you to understand some of the nuances of the methods and technologies employed.

 
 

Friday, October 30, 10:45 am — 12:15 pm (Room 1A12)

Workshop: W8 - ISO/MPEG-H Audio - The New Standard for Universal Spatial / 3D Audio Coding

Chair:
Jürgen Herre, International Audio Laboratories Erlangen - Erlangen, Germany; Fraunhofer IIS - Erlangen, Germany
Panelists:
Alexander Krüger, Technicolor
Nils Peters, Qualcomm - San Diego, CA, USA
Jan Plogsties, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany

Abstract:
Recently, the ISO/MPEG standardization group created the MPEG-H 3D Audio specification to go along with Ultra High Definition Television (UHDT) video. The specification features several unique elements, such as handling of channel-based content, object-based content and higher order ambisonics (HOA) content or the capability of rendering encoded high-quality content on a wide range of loudspeaker setups (22.2 ... 5.1 ... stereo / headphones). This workshop provides an overview of the MPEG-H 3D Audio standard regarding its underlying architecture, technology, performance and how to produce immersive content for it.

 
 

Friday, October 30, 11:00 am — 1:00 pm (Room 1A21)

Workshop: W9 - Give Peaks a Chance

Chair:
Thomas Lund, Genelec Oy - Iisalmi, Finland
Panelists:
Florian Camerer, ORF - Austrian TV - Vienna, Austria; EBU - European Broadcasting Union
Bob Katz, Digital Domain Mastering - Orlando, FL, USA
Bob Ludwig, Gateway Mastering Studios, Inc. - Portland, ME, USA
George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada
Susan Rogers, Berklee College of Music - Boston, MA, USA

Abstract:
The Tribunal is ready to present an update on the Loudness Wars. Important 2015 developments include Apple Music and WHO’s sudden attention to sound exposure from Personal Media Players (PMPs). Both will have an impact on our industry at large, good or bad.

The panel will cover production, mastering, and distribution of quality audio, and we will discuss how the prevention of hearing loss fits in as a key element. While our most social sense must be preserved, cultural aspects should not be forgotten. We need something worth listening to also :-).

 
 

Friday, October 30, 1:15 pm — 4:45 pm (Off-Site 3)

Technical Tour: TT5 - WNYC Radio and WQXR

Abstract:
WNYC 93.9 FM and AM 820 are New York's flagship public radio stations, broadcasting the finest programs from NPR, American Public Media, Public Radio International, and the BBC World Service, as well as a wide range of award-winning local programming. The station now occupies two and a half floors of a 12-story former printing building. WNYC reaches more than one million listeners each week and has the largest public radio audience in the United States.

Location: WNYC Radio
160 Varick St., New York, NY
Bus transportation is not provided. We'd suggest taking the 7 Train to the 1 Train to Houston.

Limited to 25 people. A ticket is required - anyone showing up without a ticket will be turned away.

 
 

Friday, October 30, 1:45 pm — 3:15 pm (Room 1A10)

Broadcast and Streaming Media: B6 - Production of A Prairie Home Companion

Moderator:
John Holt, Retired
Panelists:
Samuel Hudson, Producer/Technical Director - St. Paul, Minnesota USA
Nick Kereakos
Thomas Scheuzger, Broadcast/Transmission Engineer - Saint Paul, MN, USA

Abstract:
“From the control board at the Orpheum, PHC travels via underground phone lines to the tiny Satellite Control Room on the fourth floor of Minnesota Public Radio, from there by cable to MPR’s transmitting dish in a junkyard on the East Side of Saint Paul, and from there 22,300 miles to Western Union’s Westar IV satellite…”

A lot has changed technically since that was written over 30 years ago for the 10th anniversary of "A Prairie Home Companion." You’ll hear a little history and a lot about how the technology has changed and will change the production and distribution of this iconic radio program.

 
 

Friday, October 30, 2:00 pm — 4:30 pm (Room 1A07)

Paper Session: P10 - Recording & Production

Chair:
Grzegorz Sikora, Harman - Pullach, Germany

P10-1 Lossless Audio Checker: A Software for the Detection of Upscaling, Upsampling, and Transcoding in Lossless Musical Tracks—Julien Lacroix, Independent Developer - Aix-en-Provence, France; Yann Prime, Independent Developer - Aix-en-Provence, France; Alexandre Remy, Independent Developer - Aix-en-Provence, France; Olivier Derrien, University of Toulon / CNRS-LMA - Toulon, France
Internet music dealers currently sell “CD quality” tracks, or even better (“Studio Master”), thanks to lossless audio coding formats (FLAC, ALAC). However, a lossless format does not guarantee that the audio content is what it seems to be. The audio signal may have been upscaled (increasing the resolution), upsampled (increasing the sample rate), or even transcoded from a lossy to a lossless format. In this paper we describe a new software that analyzes lossless audio tracks and detects upsampling, upscaling, and transcoding (only for AAC in this early version). Validation tests over a large music database (with groundtruth available) show that this method is fast and accurate: 100% of success for upscaling and transcoding, 91.3% for upsampling.
Convention Paper 9416 (Purchase now)

P10-2 Comparison of Audio Signals Obtained with Source Overlay (OAS) and Other Conventional Recording Methods—Juliette Olivella, Universidad de San Buenaventura - Bogotá, Colombia; K2 INGENIERIA; William Romo, Universidad de San Buenaventura - Bogotá, Colombia; Dario Páez, Universidad de San Buenaventura - Bogotá, Colombia
Overlay Model of Acoustic Sources (OAS) is an unconventional recording method with a stereo microphone array. This model was proposed as a methodological alternative that allows emulating a recording single-take of a musical group. It is based on the presumption of a linear behavior in a recording system and involves doing partial captures of musical instruments that integrate the entire assembly. Experimental tests were done to corroborate the system's linearity; two speakers are used instead of musicians and audio is recorded with conventional techniques and model of Overlay of Acoustic Sources. The audios were discretized using MATLAB in order to evaluate their physical parameters and the correlation coefficients between energy, maximum values, minimum values, frequency response, the zero crossings rate, and spatiality of recordings. All the research sought to answer the question if it is possible to get an audio signal able to imitate the signal characteristics captured in real time in a recording by takes. The results showed that it is possible when the recording is performed with the method of overlay of acoustic sources (OAS).
Convention Paper 9417 (Purchase now)

P10-3 Process Improvement in Audio Production from a Sociotechnical Systems Perspective—Gerhard Roux, Stellenbosch University - Stellenbosch, Western Cape, South Africa
Audio professionals involved in live sound reinforcement, record production, and broadcasting are continuously solving complex problems in creative ways. It is wasteful if the pragmatic methodologies used in solving these problems do not contribute towards a reusable model of process improvement. This paper suggests a systems-level engagement with audio production that strikes a balance between human creativity and technological infrastructure. A conceptual model of process improvement is developed through analysis of audio production as a complex system and subsequently implemented through an action research methodology in multiple case studies. The study found that significant quality improvements in audio production could be attained through a sociotechnical systems approach. The results imply that the application of process improvement methodologies can coexist with creative social practice, resulting in improved technical performance of production systems.
Convention Paper 9418 (Purchase now)

P10-4 Listener Preference for Height Channel Microphone Polar Patterns in Three-Dimensional Recording—Will Howie, McGill University - Montreal, QC, 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; Matthew Boerum, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT); David Benson, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Alan Joosoo Han, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada
A listening experiment was conducted to determine if a preference exists among three microphone polar patterns when recording height channels for three-dimensional music production. Seven-channel 3D recordings of four different musical instruments were made using five-channel surround microphone arrays, augmented with two Sennheiser MKH 800 Twin microphones as height channels. In a double-blind listening test, subjects were asked to rate different mixes of the same recordings based on preference. The independent variable element in these mixes was the polar pattern of the height channel microphones. Analysis of the results found that the vast majority of subjects showed no statistically significant preference for any one polar pattern.
Convention Paper 9419 (Purchase now)

P10-5 Listener Discrimination of High-Speed Digitization from Analog Tape Masters with Spectral Matching—Nick Lobel, Belmont University - Nashville, TN, USA; Eric Tarr, Belmont University - Nashville, TN, USA; Wesley Bulla, Belmont University - Nashville, TN, USA
This study investigated whether listeners could discriminate between real-time (RT) and double-speed (DS) digital transfers from analog tape recordings. Signals were recorded to tape at 15 inches per second (ips), then digitized at two copy rates: 15 ips (RT) and 30 ips (DS). The DS transfers were digitally time-stretched and spectrally processed to match the duration and frequency response of the RT transfers. Thirty-one listeners participated in an ABX experiment to discriminate between the RT and DS transfers. Results show discrimination between RT and DS transfers was not statistically significant. Additionally, discrimination did not vary significantly across different types of source signals.
Convention Paper 9420 (Purchase now)

 
 

Friday, October 30, 2:30 pm — 4:00 pm (Room 1A13)

Networked Audio: N5 - Dante Case Studies

Presenters:
John Huntington, NYC College of Technology - Brooklyn, NY, USA
Sam Kusnetz, Team Sound - Brooklyn, NY, USA
Joe Patten, Communications Design Associates - Canton, MA, USA

Abstract:
Part 1: Using an Audio Network for a Themed Attraction in an Academic Environment Sound Designer Sam Kusnetz and Network Engineer John Huntington give an overview of the Dante network that is the backbone of the audio system for the Gravesend Inn haunted attraction at Citytech in downtown Brooklyn. Here are two learning points: • The benefits of networked audio in themed attractions • Using networked audio over managed networks.

Part 2: Cost Saving and Digital Audio Networking The use of digitized audio networks has changed the flow of information and cost associated with it for the better. More channels of audio are available in more location with the installation cost greatly reduced. Benefits: • Savings with infrastructure such as cabling and conduit • Flexibility with routes or multiple routes/distribution of audio • Density of audio paths, 128 channels over single link via CAT6

 
 

Friday, October 30, 3:30 pm — 5:00 pm (Room 1A10)

Broadcast and Streaming Media: B7 - Audio for Adaptive Streaming—Understanding HLS-DASH, HTML5

Moderator:
Ray Archie, MixLuv - New York, NY, USA; Music is My First Language - New York, NY, USA
Panelists:
Richard Doherty, Director, Connected Technology Strategy at Dolby Laboratories - San Francisco, CA, USA
Ronny Katz, DTS - Calabasas, CA, USA
John Kean, Consultant - Washington DC, USA
Jan Nordmann, Fraunhofer USA - San Jose, CA, USA
Charles Van Winkle, Adobe Systems Incorporated - Minneapolis, MN, USA

Abstract:
Adaptive streaming is the process of encoding a single source stream at multiple bit rates - allowing players to switch adaptively to deliver the optimal experience to each viewer based on available bandwidth and CPU capacity. So how do we encode for this? This panel will look at HTTP Live Streaming (HLS), MPEG-DASH, Adobe Dynamic Streaming, and more!

 
 

Friday, October 30, 5:15 pm — 6:45 pm (Room 1A10)

Broadcast and Streaming Media: B8 - Mixing for Telemedia in 21st

Moderator:
Ed Greene
Presenter:
Bob Katz, Digital Domain Mastering - Orlando, FL, USA

Abstract:
A frank discussion from the mixers chair of the present challenges in crafting program audio for broadcast and the www. There will be a discussion of realistic monitoring situations important for mixers who work in different CR’s. The presentation will also include a review of the circumstances leading us to where we are today and the possible effect of pending new broadcast guidelines from ATSC 3.0 and pending recommended streaming practices for the www. from the AES.

 
 

Friday, October 30, 5:30 pm — 6:45 pm (Room 1A07)

Engineering Brief: EB4 - Listening, Hearing, & Production

Chair:
Bruno Fazenda, University of Salford - Salford, Greater Manchester, UK

EB4-1 Why Do My Ears Hurt after a Show (And What Can I Do to Prevent It)—Dennis Rauschmayer, REVx Technologies/REV33 - Austin, TX, USA
In this brief we review the traditional methods of preventing ear fatigue, short-term ear damage, and long term ear damage. A new method to prevent ear fatigue, focused on performing musicians is then presented. This method, which reduces noise and distortion in the artist’s mix, is discussed. Qualitative and quantitative results from a series of trials and experiments is presented. Qualitative results from artist feedback indicate less ear fatigue, less ringing in the ears, and a better ability to have normal conversations after a performance when noise and distortion in their mix is reduced. Quantitative results are consistent with the qualitative results and show a reduction in the change in otoacoustic emissions measured for a set of musicians when noise and distortion are reduced. The result of the study suggests that there is an important new tool for musicians to use to combat ear fatigue and short term hearing loss.
Engineering Brief 213 (Download now)

EB4-2 Classical Recording with Custom Equipment in South Brazil—Marcelo Johann, UFRGS - Porto Alegre, RS, Brazil; Andrei Yefinczuk, UFRGS - Porto Alegre, Brazil; Marcio Chiaramonte, Meber Metais - Bento Gonçalves, Brazil; Hique Gomez, Instituto Marcello Sfoggia - Porto Alegre, Brasil
This paper describes the process developed by Marcello Sfoggia for recording acoustic and classical music in south of Brazil, making intensive use of custom equipment. Sfoggia spent most of his lifetime building dedicated circuits to optimize sound reproduction and recording. He took the task of registering major performances in the city of Porto Alegre, using his home-developed equipment, what became a reference process. We describe the system employed for both sound capture and mixdown. Key components of the signal flow include preamplifiers with precision op-amps, short signal paths, modified A/D/A converters and the mixing desk with pure vacuum tube circuitry. Finally, we address our current efforts to continue his activities and improve upon his system with updated circuits and techniques.
Engineering Brief 214 (Download now)

EB4-3 Techniques For Mixing Sample-Based Music—Paul "Willie Green" Womack, Willie Green Music - Brooklyn, NY, USA
Samples are a great way to add impact, vibe, and texture to a song and can often be the primary component of a new work. From a production standpoint, audio that is already mixed and mastered can add to a producer’s sonic palette. From a mixing perspective, however, these same bonuses also provide a number of challenges. Looking more closely at each of the common issues an engineer often faces with sample-based music, I will illustrate techniques that can enable an engineer to better manipulate a sample, allowing it to sit more naturally inside the mix as a whole.
Engineering Brief 215 (Download now)

EB4-4 Case Studies of Inflatable Low- and Mid-Frequency Sound Absorption Technology—Niels Adelman-Larsen, Flex Acoustics - Copenhagen, Denmark
Surveys among professional musicians and sound engineers reveal that a long reverberation time at low frequencies in halls during concerts of reinforced music is a common cause for an unacceptable sounding event. Mid- and high-frequency sound is seldom a reason for lack of clarity and definition due to a 6 times higher absorption by audience compared to low frequencies, and a higher directivity of speakers at these frequencies. Lower frequency sounds are, within the genre of popular music however, rhythmically very active and loud, and a long reverberation leads to a situation where the various notes and sounds cannot be clearly distinguished. This reverberant bass sound rumble often partially masks even the direct higher pitched sounds. A new technology of inflated, thin plastic membranes seems to solve this challenge of needed low-frequency control. It is equally suitable for multipurpose halls that need to adjust their acoustics by the push of a button and for halls and arenas that only occasionally present amplified music and need to be treated just for the event. This paper presents the authors’ research as well as the technology showing applications in dissimilarly sized venues, including before and after measurements of reverberation time versus frequency.
Engineering Brief 216 (Download now)

EB4-5 Advanced Technical Ear Training: Development of an Innovative Set of Exercises for Audio Engineers—Denis Martin, McGill University - Montreal, QC, Canada; CIRMMT - Montreal, QC, Canada; George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada
There are currently many automated software solutions to tackle the issue of timbral/EQ training for audio engineers but only limited offerings for developing other skills needed in the production process. We have developed and implemented a set of matching exercises in Pro Tools that fill this need. Presented with a reference track, users are trained in matching inter-instrument levels/gain, lead instrument volume automation, instrument spatial positioning/panning, reverberation level, and compression settings on a lead element within a full mix. The goal of these exercises is to refine the listener’s degree of perception along these production parameters and to train the listener to associate these perceived variations to objective parameters they can control. We also discuss possible future directions for exercises.
Engineering Brief 217 (Download now)

 
 

Friday, October 30, 5:30 pm — 7:00 pm (Room 1A14)

Workshop: W13 - Automotive Audio—The Making of the Sound System in a Car

Chair:
Grzegorz Sikora, Harman - Pullach, Germany
Panelists:
Thomas Bachmann, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Peter John Chapman, Harman - Denmark; Bang & Olufsen Automotive

Abstract:
In this workshop we look behind the scenes of developing an OEM sound system, "the making of." It’s a unique opportunity to understand the key fundamentals of the Automotive Audio perspective, from the project team behind many successful OEM sound systems. This workshop will cover a range of topics from the initial idea to the finished product. The experts from the respective fields will discuss car cabin acoustics, loudspeaker selection and placement, signal flow and amplifier characteristics, sound tuning, audio design philosophy, and creation of the 3D sound algorithm. Each topic will be discussed in general and in the context of actual projects.

 
 

Saturday, October 31, 9:00 am — 10:30 am (Room 1A13)

Product Development: PD7 - Modern Digital Processing of Microphone Signals

Presenter:
Paul Beckmann, DSP Concepts, LLC - Sunnyvale, CA, USA

Abstract:
Microphones have been in use for decades in professional audio applications. Recently they are also being incorporated into consumer and automotive products and their use is exploding. And although they are ubiquitous they are usually the weakest link in the audio signal chain. Common problems include dynamic range issues (too loud or too soft) and noise (electrical noise, background noise, wind, and plosives and sybillants). This session covers modern digital approaches to microphone processing. We use an interactive approach and build up the signal chain using graphical tools. We design single and multiband automatic gain controls, noise gates, and dynamics processors for reducing plosives and handling noise (“de-poppers”). We show how these algorithms are designed and tuned in practice.

 
 

Saturday, October 31, 9:00 am — 10:30 am (Room 1A10)

Broadcast and Streaming Media: B9 - Audio for Over the Top Television (OTT)

Moderator:
Skip Pizzi, NAB - Washington DC, USA
Panelists:
Richard Galvan, Senior Technical Marketing Manager for OTT, Dolby Labs - San Francisco, CA, USA
Tom McCarthy, Owner, AM-DVD - Centreville, VA, USA
Sean Richardson, Starz Entertainment - Parker, CO, USA

Abstract:
Like many other industries, television has been disrupted by the Internet, with a growing amount of content delivered to audiences via streaming. The mechanisms involved are substantially different from traditional television delivery, yet the two forms will continue to coexist for some time. Meanwhile, OTT has greater agility to adopt new formats than traditional delivery schemes, and therefore content suppliers are being asked to deliver new components to OTT providers at a rapid pace. This session will address how audio is changing for OTT delivery, presented from the perspectives of technology providers, content producers and service operators.

 
 

Saturday, October 31, 10:45 am — 12:15 pm (Room 1A10)

Broadcast and Streaming Media: B10 - Listener Fatigue and Retention

Moderator:
Marvin Caesar, Founder and Former President Aphex Systems - Sherman Oaks, CA, USA
Panelists:
Robert Arbittier, Noisy Neighbors Productions - Los Angeles, CA, USA
Robert Reams, Consultant - Santa Clara, CA, USA
Bill Sacks, Orban / Optimod Refurbishing - Hollywood, MD, USA
Andrew Scheps, Tonequake Records - Van Nuys, CA, USA
Larry Zinn, Monitor Engineer

Abstract:
Listener fatigue is present in every audio professional's and music consumer's life. It affects the quality of recorded, live, and broadcast audio. It affects the ability of listeners to enjoy and/or understand the program. And, ultimately, can impact long term health.

The session will seek to define the psychological and physiological aspects of listener fatigue, the causes and possible solutions.
The panel is made up of top audio professionals with varied backgrounds and decades of experience.

 
 

Saturday, October 31, 1:30 pm — 2:30 pm (Room 1A13)

Networked Audio: N7 - Benefits of AES67 to the End User

Presenter:
Rich Zwiebel, QSC - Boulder, CO, USA; K2

Abstract:
Presenter Rich Zwiebel has a long history in audio networking. He was a founder of Peak Audio, the company that developed CobraNet, the first widely used audio network for Professional applications. As a VP at QSC he continues to be very active in the field and is currently the Chairman of the Media Networking Alliance.

This presentation reviews the history of professional audio networking, where we are today, and what the future may hold. A clear explanation of what AES67 is, as well as what it is not, along with how it will benefit those who choose to use it will be included. Attendees will understand it’s relationship to existing audio network technologies in the market.

Additionally, an explanation of who the Media Networking
Alliance is, who it’s members are, and what it’s goals are will be presented.

A discussion of the advantages of a single facility network will close out the session.

 
 

Saturday, October 31, 1:30 pm — 3:00 pm (Room 1A10)

Broadcast and Streaming Media: B11 - Audio for Broadcast Video—Immersive, Personalized, 4K, and 8K

Moderator:
Fred Willard, Univision - Washington, DC, USA
Panelists:
Robert Bleidt, Fraunhofer USA Digital Media Technologies - San Jose, CA, USA
Tim Carroll, Telos Alliance - Lancaster, PA, USA
James Moore, DTS, Inc. - Calabasas, CA, USA
Kazuho Ono, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan
Skip Pizzi, NAB - Washington DC, USA
Robert Reams, Consultant - Santa Clara, CA, USA
Jeff Riedmiller, Dolby Laboratories - San Francisco, CA USA

Abstract:
Each of our panelists is at the forefront of harnessing technology to augment the human experience of immersive and personalized audio in 4K, 8K, and ATSC 3.0 broadcast and streaming. Come to share the latest and greatest discoveries and standards proposals forged in the past year. Work in this sphere progresses at a feverish pace as we close in on final standardization and the next magnitude of consumer psycho-acoustic involvement. Don’t miss this chance to present your questions to our experts after they share their most recent stories and accomplishments.

 
 

Saturday, October 31, 2:15 pm — 3:45 pm (Room 1A07)

Paper Session: P16 - Room Acoustics

Chair:
Rémi Audfray, Dolby Laboratories, Inc. - San Francisco, CA, USA

P16-1 Environments for Evaluation: The Development of Two New Rooms for Subjective Evaluation—Elisabeth McMullin, Samsung Research America - Valencia, CA USA; Adrian Celestinos, Samsung Research America - Valencia, CA, USA; Allan Devantier, Samsung Research America - Valencia, CA, USA
An overview of the optimization, features, and design of two new critical listening rooms developed for subjective evaluation of a wide-array of audio products. Features include a rotating wall for comparing flat-panel televisions, an all-digital audio switching system, custom tablet-based testing software for running a variety of listening experiments, and modular acoustic paneling for customizing room acoustics. Using simulations and acoustic measurements, a study of each of the rooms was performed to analyze the acoustics and optimize the listening environment for different listening situations.
Convention Paper 9460 (Purchase now)

P16-2 Low Frequency Behavior of Small Rooms—Renato Cipriano, Walters Storyk Design Group - Belo Horizonte, Brazil; Robi Hersberger, Walters Storyk Design Group - New York, USA; Gabriel Hauser, Walters Storyk Design Group - Basel, Switzerland; Dirk Noy, WSDG - Basel, Switzerland; John Storyk, Architect, Studio Designer and Principal, Walters-Storyk Design Group - Highland, NY, USA
Modeling of sound reinforcement systems and room acoustics in large- and medium-size venues has become a standard in the audio industry. However, acoustic modeling of small rooms has not yet evolved into a widely accepted concept, mainly because of the unavailable tool set. This work introduces a practical and accurate software-based approach for simulating the acoustic properties of studio rooms based on BEM. A detailed case study is presented and modeling results are compared with measurements. It is shown that results match within given uncertainties. Also, it is indicated how the simulation software can be enhanced to optimize loudspeaker locations, room geometry, and place absorbers in order to improve the acoustic quality of the space and thus the listening experience.
Convention Paper 9461 (Purchase now)

P16-3 Measuring Sound Field Diffusion: SFDC—Alejandro Bidondo, Universidad Nacional de Tres de Febrero - UNTREF - Caseros, Buenos Aires, Argentina; Mariano Arouxet, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina; Sergio Vazquez, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina; Javier Vazquez, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina; Germán Heinze, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina
This research addresses the usefulness of an absolute descriptor to quantify the degree of diffusion in a third octave band basis of a sound field. The degree of sound field diffuseness in one point is related with the reflection’s energy control multiplied by the temporal distribution uniformity of reflections. All this information is extracted from a monaural, broadband, omnidirectional, high S/N impulse response. The coefficient range varies between 0 and 1, evaluates the early, late, and total sound field for frequencies above Schroeder’s and in the far field from diffusive surfaces, zero being “no diffuseness” at all. This coefficient allows the comparison of different rooms, different places inside rooms, measurement of the effects of different sound diffusers coatings, and the resulting spatial uniformity variation, among other applications.
Convention Paper 9462 (Purchase now)

 
 

Saturday, October 31, 3:15 pm — 4:45 pm (Room 1A14)

Networked Audio: N8 - How to Get AES67 into Your Systems/Products

Chair:
Andreas Hildebrand, ALC NetworX GmbH - Munich, Germany
Panelists:
Michael Dosch, Lawo AG - Rastatt Germany
Nathan Phillips, Coveloz Technologies - Ottawa, ON, Canada
Greg Shay, The Telos Alliance - Cleveland, OH, USA
Nicolas Sturmel, Digigram S.A. - Montbonnot, France
Arie van den Broek, Archwave Technologies - Schwerzenbach, Switzerland
Kieran Walsh, Audinate Pty. Ltd. - Ultimo, NSW, Australia

Abstract:
This workshop introduces several options to implement AES67 networking capabilities into existing or newly designed products. The session starts with a quick recap on the technical ingredients of AES67 and points out the principal options on implementing AES67 into new or existing products. After providing an overview on commercially available building blocks (modules, software libraries and reference designs), the workshop commences in a discussion on the value of providing AES67 compatibility from the perspective of providers of existing AoIP networking solutions. The workshop is targeted towards product manufacturers seeking ways to implement AES67 into their products, but should also provide valuable insight to those with general technical interest in AES67.

 
 

Saturday, October 31, 3:15 pm — 4:45 pm (Room 1A10)

Broadcast and Streaming Media: B12 - Integrating Mobile Telephony and IP in Broadcast

Moderator:
Kirk Harnack, Telos Alliance - Nashville, TN, USA; South Seas Broadcasting Corp. - Pago Pago, American Samoa
Panelists:
Mitch Glider, iHeart Media
Dave Immer, DIGIFON/Walnut Studio - Fairfield, CT, USA
Tony Peterle, Worldcast Systems - Miami, FL, USA
Paul Shulins, Greater Media - Boston, MA, USA
Christopher Tobin, Newark Public radio - Newark, NJ USA
Andrew Zarian, GFQ Network - Queens, NY

Abstract:
Engineers are tasked to get broadcasts on-air from some very out-of-the-way places. From car dealers to tire shops, from park concerts to canoe races, and even the occasional balloon glow miles from anywhere, broadcast engineers strive to bring audio and video back to the station. The once-reliable telephone companies have already reduced their connection offerings and service footprint, leaving engineers to be creative in tying the studio to the off-site event. Radio engineers need workable alternatives to ISDN, and many television engineers are restricted from using the satellite truck unless absolutely necessary, leaving them to seek high-bandwidth connectivity, too. Today’s connectivity solutions typically involve IP—both wired and wireless. Often times a clever mix of IP connection services and technologies are required to bring a broadcast in from the field.

Leading broadcast engineers and vendor technical representatives will discuss the latest practices and workable solutions to “get there from anywhere.”

 
 

Saturday, October 31, 3:45 pm — 4:15 pm (Room 1A07)

Engineering Brief: EB5 - Acoustics

Chair:
Jung Wook (Jonathan) Hong, McGill University - Montreal, QC, Canada; GKL Audio Inc. - Montreal, QC, Canada

EB5-1 Visualization of Compact Microphone Array Room Impulse Responses—Luca Remaggi, University of Surrey - Guildford, Surrey, UK; Philip Jackson, University of Surrey - Guildford, Surrey, UK; Philip Coleman, University of Surrey - Guildford, Surrey, UK; Jon Francombe, University of Surrey - Guildford, Surrey, UK
For many audio applications, availability of recorded multichannel room impulse responses (MC-RIRs) is fundamental. They enable development and testing of acoustic systems for reflective rooms. We present multiple MC-RIR datasets recorded in diverse rooms, using up to 60 loudspeaker positions and various uniform compact microphone arrays. These datasets complement existing RIR libraries and have dense spatial sampling of a listening position. To reveal the encapsulated spatial information, several state of the art room visualization methods are presented. Results confirm the measurement fidelity and graphically depict the geometry of the recorded rooms. Further investigation of these recordings and visualization methods will facilitate object-based RIR encoding, integration of audio with other forms of spatial information, and meaningful extrapolation and manipulation of recorded compact microphone array RIRs.
Engineering Brief 218 (Download now)

EB5-2 Sensible 21st Century Saxophone Selection —Thomas Mitchell, University of Miami - Coral Gables, FL, USA
This paper presents a method for selecting a saxophone, using data mining techniques with both subjective and objective data as criteria. Immediate, subjective personal impressions are given equal weight with more-objective observations made after the fact, and with hard data distilled from audio data using MIR Toolbox. Offshoots and directions for future research are considered.
Engineering Brief 219 (Download now)

 
 

Saturday, October 31, 4:30 pm — 6:00 pm (Room 1A06)

Workshop: W21 - Fiber Optic Connector Choices for Audio

Chair:
Ronald Ajemian, Owl Fiber Optics - Flushing, NY, USA; Institute of Audio Research - New York, NY, USA
Panelists:
Marc Brunke, Optocore GmbH - Grafelfing, Germany
Fred Morgenstern, Neutrik USA
Harry (Buddy) Oliver, FiberPlex Technologies, LLC - Elkridge, MD, USA
Warren Osse, Applications/Senior Design Engineer, Vistacom, Inc. - Allentown, PA, USA

Abstract:
This AES workshop is designed to educate the user, engineer, technician, and student on the most popular Fiber Optic Connector types that are currently available in the audio/video market place. A brief review will be presented about these connectors. The panel will then discuss user preferences for various audio/video applications. A demo will also be shown on how to terminate (put together) a typical LC fiber optic connector to a glass optical fiber.

 
 

Saturday, October 31, 5:00 pm — 6:30 pm (Room 1A14)

Networked Audio: N9 - How Will AES67 Affect the Industry?

Chair:
Rich Zwiebel, QSC - Boulder, CO, USA; K2
Panelists:
Claude Cellier, Merging Technologies - Puidoux, Switzerland
Andreas Hildebrand, ALC NetworX GmbH - Munich, Germany
Patrick Killianey, Yamaha Professional Audio - Buena Park, CA, USA
Phil Wagner, Focusrite Novation Inc. - El Segundo, CA
Ethan Wetzell, Bosch Communications Systems - Burnsville, MN USA; OCA Alliance

Abstract:
There are many audio networking standards available today. Unfortunately, equipment designers and facility engineers have been forced to choose between them to adopt a single platform for an entire operation, or link disparate network pools by traditional cabling (analog, AES/EBU or MADI). AES67 solves this dilemma, providing a common interchange format for various network platforms to exchange audio without sacrificing proprietary advantages. Published in 2013, manufacturers are already showing products with AES67 connectivity this year. Join our panel of six industry experts for an open discussion on how AES67 will impact our industry.

 
 

Saturday, October 31, 8:00 pm — 10:00 pm (Off-Site 3)

Special Event: Stories for the Ears: Live Audio Drama and Narration

Abstract:
Dolby Laboratories NY Screening Room
1350 Ave of the Americas Main Floor
(Doors open at 7:30 pm – show begins at 8:00 pm)
Limited seating, tickets required (at Tours Desk).

Fantasy, Fiction, and Fun!
The HEAR Now Festival and SueMedia Productions in conjunction with the Audio Engineering Society (AES) presents an evening of live audio/radio drama along with narrative readings celebrating the art of sonic storytelling.

Hosted by Simon Jones (Hitchhiker’s Guide to the Galaxy) featuring performances by Audie Award winning and Golden Voice narrators Robin Miles and Barbara Rosenblat, and the award winning NY-based audio drama troupe VoiceScapes Audio Theater.

Sponsored by Hear Now Festival, Walters Storyk Design Group, Dolby Labs, and the AES

 
 

Sunday, November 1, 9:00 am — 10:30 am (Room 1A21)

Workshop: W25 - Loudness Regulation: New Tools to Keep the Spirit of Dynamics

Chair:
Florian Camerer, ORF - Austrian TV - Vienna, Austria; EBU - European Broadcasting Union
Panelists:
Antoine Hurtado, Isostem - Paris, France
Thomas Lund, Genelec Oy - Iisalmi, Finland
Michael Kashnitz, RTW
Matthieu Parmentier, francetélévisions - Paris, France

Abstract:
Thanks to various laws and recommendations, the loudness regulation is now well spread among TV broadcasters. Now sound mixers and broadcast engineers need for new tools to raise the quality while keeping the dynamics of good programs and gently process others. This workshop will offer an overview of the latest developments concerning loudness measures and process, such as modifying the loudness range, compensating the loudness shift of upmix/downmix, measuring the loudness through an IP network and integrating a smart loudness fader within a web player.

 
 

Sunday, November 1, 10:15 am — 1:30 pm (Off-Site 1)

Technical Tour: TT10 - NBC Universal

Abstract:
The tour will include a visit to Studio 6B, home of The Tonight Show starring Jimmy Fallon, and move on to Studio 8G, home of Late Night with Seth Meyer. We will review the audio technology used to produce both shows.

Location: NBC Universal
30 Rockefeller Pl, New York, NY
Bus transportation is not provided. We'd suggest taking the 7 Train to Times Square and then walking up to Rockefeller Center.

Limited to 40 people. A ticket is required - anyone showing up without a ticket will be turned away.

 
 

Sunday, November 1, 10:30 am — 11:30 am (Room 1A14)

Recording & Mastering: RM7 - The Game Has Changed but You Don’t Know It: How to Make Recordings Sound Great on Streaming

Presenter:
Alan Silverman, Arf! Mastering - New York, NY, USA; NYU|Steinhardt Dept.of Music Technology

Abstract:
Records are engineered to sound their best in the real-world. To accomplish this on services like YouTube, Spotify, Apple Music, Tidal, and Pandora requires a different approach to mixing and mastering because of the way today’s streaming services treat audio. Few producers are aware of the game-changing technology under the hood. Recorded music can sound bigger and better than it has in the last decade, ironically, on audiophile systems as well, by applying an understanding of the new technology. Grammy-winning mastering engineer Alan Silverman demonstrates how to harness this potential to the fullest.

 
 

Sunday, November 1, 11:00 am — 12:30 pm (Room 1A12)

Networked Audio: N10 - AES67 Interoperability Testing

Chair:
Kevin Gross, AVA Networks - Boulder, CO, USA
Panelists:
Andreas Hildebrand, ALC NetworX GmbH - Munich, Germany
Peter Stevens, BBC Research & Development - London, UK
Nicolas Sturmel, Digigram S.A. - Montbonnot, France

Abstract:
The AES has now planned two plugfests for AES67 implementers and users. The first plugfest was held in October 2014 at IRT in Munich. A report describing this event was produced and published by the AESSC as AES-R12-2014. The second plugfest is planned for early November in Washington D.C. at NPR headquarters. This workshop will summarize the testing performed and will present results. A panel comprising plugfest participants will answer audience questions and the audience should get a good feel for where AES67 implementation stands.

 
 

Sunday, November 1, 11:00 am — 12:30 pm (Room 1A06)

Broadcast and Streaming Media: B13 - Technology and Storytelling: How Can We Best Use the Tools Available to Tell Our Stories

Panelists:
Sue Zizza, SueMedia Productions - Carle Place, NY, USA
David Shinn, National Audio Theatre Festivals - New York, NY

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.

 
 

Sunday, November 1, 2:00 pm — 5:00 pm (S-Foyer 1)

Broadcast and Streaming Media: B14 - SBE Certification Exams

Abstract:
The Society of Broadcast Engineers Certification Program, established in 1975, is a service contributing to the professional development of the broadcast engineer and advancement of the field of broadcast engineering. The SBE Program of Certification is administered by the SBE National Certification Committee, which continually develops exam questions based on changes in the industry and technology. Levels of SBE certification vary based on an individual's experience. Membership in the SBE is not required to hold SBE certification. Advance registration by Oct. 2 is required to take an SBE certification exam at the convention. More info and the application forms: http://sbe.org/certification

 
 

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