Thursday, September 29, 10:45 am — 12:15 pm (Rm 403B)
EB1-1 A Ground Plane Measurement Comparison Between Two Floor-Standing Loudspeaker Systems: A Conventional Three-Way Studio Monitor vs. A Ground-Plane Constant Beamwidth Transducer (CBT) Line Array—D.B. (Don) Keele, Jr., DBK Associates and Labs - Bloomington, IN, USA
This paper compares two different types of floor-standing loudspeaker systems. Both were measured over an acoustically reflective hard surface in a large space. The first is a high-performance conventional three-way 12”-woofer studio monitor and the second is a ground-plane circular-arc CBT line array loudspeaker. Measurements included direct-field frequency responses in front of the systems at 20 grid locations ranging over different distances/heights and response vs. distance at seated and standing heights. Horizontal off-axis and near-field responses were also gathered along with ceiling illumination responses at several launch angles. The measurements reveal that the CBT system has vastly more even coverage at all these locations compared to the three-way monitor and in addition eliminates the detrimental effects of floor bounce.
Engineering Brief 279 (Download now)
EB1-2 A 3D Sound Localization System Using Two Side Loudspeaker Matrices—Ryo Kaneta, University of Aizu Graduate School - Aizuwakamatsu-city, Fukushima-prefecture, Japan; Akira Saji, University of Aizu - Aizuwakamatsu City, Japan; Jie Huang, University of Aizu - Aizuwakamatsu City, Japan
We have been researching about 3D spatial sound localization using loudspeakers. However, loudspeaker systems using VBAP methods, usually need a large listening space and a lot of loudspeakers. In this paper we propose a loudspeaker matrix system to improve sound localization on 3D sound system for personal use. Four loudspeakers were set on all vertices of 25 centimeters regular square, 2 matrices on both sides of the listener. Then, we have held audio experiments to confirm the effect of the matrix system. As a result, the loudspeaker matrices system can improve sound localization especially with higher elevation.
Engineering Brief 280 (Download now)
EB1-3 Source-Distance Based Panning Algorithm (SDAP) for 3D Surround Sound—Matthew Wong, Rochester Institute of Technology - Rochester, NY, USA; Sungyoung Kim, Rochester Institute of Technology - Rochester, NY, USA
The Source-Distance Based Amplitude Panning (SDAP) algorithm offers a new approach in determining gain amounts for distributed loudspeakers in a three-dimensional (3D) space. Similar to the 3D implementation of Vector Based Amplitude Panning (VBAP), this method is based on the use of non-overlapping triangular regions formed by the known locations of sets of three loudspeakers. Unlike VBAP, however, this method compares the location of the panning vector to the surface formed by the triangular region and uses Barycentric coordinates to determine the speakers' respective amplitudes. In addition, SDAP removes the possibility of negative amplitudes as may appear in VBAP. Informal listening test results showed that the perceived position of the sound source was perceptually well matched to the target position.
Engineering Brief 281 (Download now)
EB1-4 STEAK: Backward-Compatible Spatial Telephone Conferencing for Asterisk—Dennis Guse, Technische Universität Berlin - Berlin, Germany; Frank Haase, Technische Universität Berlin - Berlin, Germany
In this paper we present our implementation of a telephone conferencing system that renders a spatial representation via binaural synthesis. The implementation extends the open-source software Asterisk and complies with established Voice-over-IP standards. The implementation only requires clients to be capable of receiving and reproducing the rendered binaural signals (two channels). Furthermore, the implementation is backward-compatible as clients not fulfilling these requirements are provided with mono-rendered signals without additional spatial information. The implemented system is released as open-source software and will enable researchers to investigate the (dis-)advantages of spatial conferencing under real-world conditions. The project name is Spatial TelephonE conferencing for AsterisK (STEAK)
Engineering Brief 282 (Download now)
EB1-5 Process of HRTF Individualization by 3D Statistical Ear Model—Slim Ghorbal, 3D Sound Labs - Rennes, France; CentraleSupélec - Rennes, France; Renaud Séguier, 3D Sound Labs - Port Marly, France; Xavier Bonjour, 3D Sound Labs - Cesson-Sevigne, France
The use of HRTFs is of well-known importance when it comes to binaural listening. However, easily capturing accurate data is a key point that has not been solved yet. In this paper we present a process for individualizing the HRTFs of an individual using ear photographs. These are fitted to a statistical 3D ear model coupled to a statistical HRTF model. This coupling allows to instantly generate from a given parameterization of the ear model a corresponding set of HRTFs. The accuracy of the results is a direct consequence of the quality and the size of the underlying databases.
Engineering Brief 283 (Download now)
EB1-6 Evaluation of Portable Loudspeakers Using Virtual Listening Test—Ziyun Liu, Nanjing University - Nanjing, China; Yong Shen, Nanjing University - Nanjing, Jiangsu Province, China; Pei Yu, Nanjing University - Nanjing, China; Hao Yin, Nanjing University - Nanjing, China
The audio quality of portable wireless speakers for consumer use has improved in recent years. In this study perceptual evaluations of several portable loudspeaker systems were performed using virtual listening test methods. A preference test was designed to assess the comparative performance of these loudspeakers. The recordings of each loudspeaker were done in a controlled situation and then the listening panel took the listening test by headphone playbacks. Considering the wide use of bass-boost effect in these products, perceptual qualities of both high playback level and low level were investigated. Several objective measures were also applied to these loudspeakers. The results of the listening test and objective measures were compared and discussed.
Engineering Brief 284 (Download now)
Friday, September 30, 1:30 pm — 3:15 pm (Rm 409B)
Amandine Pras, Paris Conservatoire (CNSMDP) - Paris, France; Stetson University - DeLand, FL, USA
EB2-1 A Broadcast Film Leader with Audio Channel, Frequency, and Synchronism Test Properties—Luiz Fernando Kruszielski, Globo TV Network - Rio de Janeiro, Brazil; Rodrigo Meirelles, Globo TV Network - Rio de Janeiro, Brazil
Universal film leaders, commonly known as “countdowns,” have been an important tool to synch audio and video. In a broadcast production, the material goes through several stages where audio and video are edited and processed, and time is a very precious resource. Also, it is important to minimize possible errors in the production chain. We propose a film leader format that, in a single 10 second clip, would be possible to do a preliminary check on aspects such as surround and stereo channel identification, relative channel level and frequency response, as well as synchronism. The proposed film leader has been tested and integrated in a Brazilian Television Network with very good results.
Engineering Brief 286 (Download now)
EB2-2 Live vs. Edited Studio Recordings: What Do We Prefer?—Amandine Pras, Paris Conservatoire (CNSMDP) - Paris, France; Stetson University - DeLand, FL, USA
This pilot study examines a common belief in written classical music that a live recording conveys a more expressive musical performance than a technically flawless studio production. Two tonmeister students of the Paris Conservatoire recorded a six-dance baroque suite and a four-movement romantic sonata in concert and in studio sessions with the same microphone techniques and in the same venue for both conditions. Twenty listeners completed an online survey to rate three versions of the dances and movements, i.e., the concert performance, the firstt studio take, and the edited version. Results show that listeners preferred the edited versions (44%) more often than the firstt studio takes (29%) and the concert performances (27%).
Engineering Brief 287 (Download now)
EB2-3 Rondo360: Dysonics’ Spatial Audio Post-Production Toolkit for 360 Media—Robert Dalton, Dysonics - San Francisco, CA, USA; Jimmy Tobin, Dysonics - San Francisco, CA, USA; CCRMA - Stanford, CA, USA; David Grunzweig, Dysonics - San Francisco, CA, USA
Rondo360 is Dysonics’ toolkit for spatial audio post-production, supporting multiple workflows including multichannel, Ambisonics, and Dysonics’ own native 360 Motion-Tracked Binaural (MTB) format. Rondo360 works with all input formats—live or prerecorded—from traditional or sound field microphones, and exports to a wide array of formats depending on desired content distribution. Rondo360 integrates seamlessly with all DAWs by adding a final layer onto the creator’s existing workflow, and it comes bundled with a suite of custom mastering tools (Mixer, Compressor, Limiter, and Reverb) that work on multichannel sound field content. With support for RondoMotion, Dysonics' wireless head-tracking device, creators can monitor their 360 mixes in real-time. Rondo360 also provides an intuitive audio/video sync and export functionality along with live broadcasting support.
Engineering Brief 288 (Download now)
EB2-5 Mixing Hip-Hop with Distortion—Paul "Willie Green" Womack, Willie Green Music - Brooklyn, NY, USA
The grit and grime of Hip-Hop doesn't have to be metaphorical. With the vast array of saturation tools available, distortion is no longer just something to remove from recordings; and the huge and aggressive sounds in Hip-Hop music can benefit specifically. From subtly warming drums and keyboards to mangling vocals and samples, this brief will demonstrate techniques for creatively distorting urban music. Exploring tape emulation, parallel vocal distortion, drum crushing, and more, I will investigate how a bit of dirt can drastically affect a mix.
Engineering Brief 290 (Download now)
EB2-6 Smart Audio Is the Way Forward for Live Broadcast Production—Peter Poers, Junger Audio GmbH - Berlin, Germany
Today’s broadcast facilities are facing ever-increasing demands on their resources as they strive to keep up with consumers who expect more content on more devices both where and when they want it. To attract and retain viewers, consistent, stable, and coherent audio is a vital requirement. One aspect that is particularly important to pay attention to is speech intelligibility. This is most critical and difficult in a live broadcast situation. The Smart Audio concept is to utilizing real time processing algorithms that are both intelligent and adaptive. Devices need to be fully interoperable with others in the broadcast environment and need to seamlessly integrate with both playout automation systems and logging and monitoring processes. The Engineering Brief will present some dedicated and proofed algorithms and practical use cases for Smart Audio
Engineering Brief 291 (Download now)
EB2-7 Towards Improving Overview and Metering through Visualization and Dynamic Query Filters for User Interfaces Implementing the Stage Metaphor for Music Mixing—Steven Gelineck, Aalborg University Copenhagen - Copenhagen, Denmark; Anders Kirk Uhrenholt, Copenhagen University - Copenhagen, Denmark
This paper deals with challenges involved with implementing the stage metaphor control scheme for mixing music. Recent studies suggest that the stage metaphor outperforms the traditional channel-strip metaphor in several different ways. However, the implementation of the stage metaphor poses issues including clutter, lack of overview and monitoring of levels, and EQ. Drawing upon suggestions in recent studies, the paper describes the implementation of a stage metaphor prototype incorporating several features for dealing with these issues, including level and EQ monitoring using brightness, shape, and size. Moreover we explore the potential of using Dynamic Query filtering for localizing channels with certain properties of interest. Finally, an explorative user evaluation compares different variations of the prototype, leading to a discussion of the importance of each feature.
Engineering Brief 292 (Download now)
Saturday, October 1, 10:45 am — 11:30 am (Rm 403A)
Dylan Menzies, University of Southampton - Southampton, UK
EB3-1 A Perceptual Approach to Object-Based Room Correction—Dylan Menzies, University of Southampton - Southampton, UK; Filippo Maria Fazi, University of Southampton - Southampton, Hampshire, UK
Object-based audio offers some advantages over conventional channel-based reproduction. Objects can be adapted based on conditions at each reproduction site, in order to improve the overall quality or according to listener preferences. In particular, if the direct and reverberant parts of objects are separately available, more freedom is available to compensate for the effects of the reproduction room. An overview is provided here of a practical approach to such room correction that can modify the object stream in real-time based on captured acoustic properties of the room.
Engineering Brief 295 (Download now)
EB3-2 The Physical Limit of Microspeakers—Kang Hou, GoerTek Electronics - Santa Clara, CA, USA; Ming Hui Shao, GoerTek Audio Technologies - China
Audio playback in portable devices might be the most challenging and least satisfactory in acoustic fields. The rising of new audio hardware and software bring some silver lights and push the components to it limits. The physical limit and some practical design guidelines of micro-speakers are discussed in this paper.
Engineering Brief 296 (Download now)
EB3-3 Line Arrays: a General Study of Space Dependent Frequency Response—Mario Di Cola, Audio Labs Systems - Casoli, Italy; Paolo Martignon, Dott., Audio Labs Systems - Parma (PR), Italy
Vertical line arrays are nowadays the standard solution for large scale sound reinforcement for several well known advantages. But they carry also critical issues, like distance dependent frequency response, governed by parameters like single box height, HF vertical dispersion, as well as array length and curvature. On the field this leads to solutions that goes from array shape optimization to multichannel DSP processing (relying on a prediction software). The authors felt the necessity to investigate distance dependent frequency response with a simple and quite general (Matlab) model, not based on single element measurements but on parametric curved sub-arrays, in order to explain the very nature of involved phenomena with simplicity and generality and graph results in a meaningful way.
Engineering Brief 297 (Download now)
Saturday, October 1, 1:30 pm — 3:00 pm (Rm 403B)
EB4-1 SAE Parametric Equalizer Training: Development of a Technical Ear Training Program Using Max—Mark Bassett, SAE Institute Byron Bay - Byron Bay, NSW, Australia; University of Sydney - Sydney, NSW, Australia; William L. Martens, University of Sydney - Sydney, NSW, Australia
Spectral-based technical ear training (TET) programs generally require the user to identity, by means of matching or absolute identification, one or more parameters of an equalizer applied to a stimulus signal. Numerous TET programs have been developed to date, targeted at either consumers, employees (for in-house training), or students (delivered within educational institutions). Corey’s 2010 suite of programs featured the first commercially available TET programs developed using Max software, deployed as stand-alone applications on CD-ROM. This paper details the development of a new TET program developed in Max, successfully deployed in the Apple App Store. “SAE Parametric Equalizer Training” is a TET application designed to teach students to identify the center frequency of a parametric equalizer applied to any imported audio files.
Engineering Brief 298 (Download now)
EB4-2 Implementation and Demonstration of Applause and Hand-Clapping Feedback System for Live Viewing—Kazuhiko Kawahara, Kyushu University - Fukuoka, Japan; Akiho Fujimori, Kyushu University - Fukuoka-ken, Japan; Yutaka Kamamoto, NTT Communication Science Laboratories - Kanagawa, Japan; Akira Omoto, Kyushu University - Fukuoka, Japan; Onfuture Ltd. - Tokyo, Japan; Takehiro Moriya, NTT Communication Science Labs - Atsugi-shi, Kanagawa-ken, Japan
Recent progress of network capacity enables real-time distribution of high-quality content of multimedia contents. This paper reports on our attempt to transmit the applause and hand-clapping in music concerts. We built a system that has an efficient implementation scheme for low-delay coding of applause and hand-clapping sounds. The system relayed applause and hand-clapping by viewers back to the performance site to provide these sounds in a synthesized and simulated manner. With this system, we conducted an experimental concert using a network distributed site. We observed some interactions between the performers and the receiver site audience. Responses to our questionnaire distributed to the audience and performers also confirmed that applause and hand-clapping feedback were effective for improving the sense of unity established in live viewings.
Engineering Brief 299 (Download now)
EB4-3 Preliminary Experimental Study on Deep Neural Network-Based Dereverberation—Ji Hyun Park, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Kwang Myung Jeon, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Chanjun Chun, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Ji Sang Yoo, Kwangwoon University - Seoul, Korea; Hong Kook Kim, Gwangju Institute of Science and Tech (GIST) - Gwangju, Korea
This paper deals with the issues associated with the dereverberation of speech or audio signals using deep neural networks (DNNs). They include feature extraction for DNNs from both clean and reverberant signals and DNN construction for generating dereverberant signals. To evaluate the performance of the proposed dereverberation method, artificially processed reverberant speech signals are obtained and a feed-forward DNN is constructed. It is shown that log spectral distortion (LSD) after applying DNN-based dereverberation is reduced by around 1.9 dB, compared with that of reverberant speech signals.
Engineering Brief 300 (Download now)
EB4-4 JSAP: A Plugin Standard for the Web Audio API with Intelligent Functionality—Nicholas Jillings, Birmingham City University - Birmingham, UK; Yonghao Wang, Birmingham City University - Birmingham, UK; Joshua D. Reiss, Queen Mary University of London - London, UK; Ryan Stables, Birmingham City University - Birmingham, UK
In digital audio, software plugins are commonly used to implement audio effects and synthesizers, and integrate them with existing software packages. While these plugins have a number of clearly defined formats, a common standard has not been developed for the web, utilizing the Web Audio API. In this paper we present a standard framework that defines the plugin structure and host integration of a plugin. The project facilitates a novel method of cross-adaptive processing where features are transmitted between plugin instances instead of audio routing, saving on multiple calculations of features. The format also enables communication and processing of semantic data with a host server for the collection and utilization of the data to facilitate intelligent music production decisions.
Engineering Brief 301 (Download now)
Saturday, October 1, 3:15 pm — 4:15 pm (Rm 403A)
Matthew Boerum, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT)
EB5-1 Digital Waveguide Network Reverberation in Non-Convex Rectilinear Spaces—Aidan Meacham, Stanford University - Stanford, CA, USA; Lauri Savioja, Aalto University - Aalto, Finland; Sara R. Martin, Norwegian University of Science and Technology - Trondheim, Norway; Julius O. Smith, III, Stanford University - Stanford, CA, USA
We present a method to simulate the late reverberation of a non-convex rectilinear space using digital waveguide networks (DWNs). In many delay-line-based reverberators, diffraction effects and even occlusion are often neglected due to the need for hand-tuned, non-physical mechanisms that complicate the extreme computational economy typical of such systems. We contend that a target space can be decomposed into rectangular solids following a succinct set of geometric rules, each of which correspond to a simple DWN reverberator. By defining the interactions between these systems, an approximation of diffraction and occlusion can be achieved while maintaining structural simplicity. This approach provides a promising engine for real-time synthesis of late reverberation with an arbitrary number of sources and receivers and dynamic geometry.
Engineering Brief 303 (Download now)
EB5-2 Max as an Interactive, Multi-Modal Learning and Teaching Tool for Audio Engineering—Mark Bassett, SAE Institute Byron Bay - Byron Bay, NSW, Australia; University of Sydney - Sydney, NSW, Australia
A solid understanding of fundamental audio engineering concepts, specifically signal flow, is paramount to the successful operation of audio technologies. Failure to fully understand the signal path of a system can lead to students learning audio technology “functions by rote, making them inherently non-transferrable” and may also lead to the development of inaccurate conceptual models. This paper discusses the use of Max software to teach fundamental audio engineering concepts to first-year Bachelor of Audio students. Although not designed as a learning and teaching tool, Max is perfectly suited for this purpose as it is interactive, adaptive and facilitates multiple modes of learning and interaction.
Engineering Brief 304 (Download now)
EB5-3 A Real-Time Simulation Environment for Use in Psychoacoustic Studies of Aircraft Community Noise—Kenneth Faller, II, California State University, Fullerton - Fullerton, CA, USA; Stephen Rizzi, NASA Langley Research Center - Hampton, VA, USA; Aric Aumann, Science Applications International Corporation - Hampton, VA, USA
The Exterior Effects Room (EER) is a psychoacoustic test facility located at the NASA Langley Research Center, with a real-time simulation environment that includes a three-dimensional sound-reproduction system. The main purpose of the EER is to support research investigating human response to aircraft community noise. To compensate for the spectral coloration of the installation and room effects, the system required real-time application of equalization filters. The efforts taken to design, implement, and analyze the equalization filters for use in the real-time sound-reproduction system is described. Acoustic performance of the system was assessed for its crossover performance and stationary and dynamic conditions.
Engineering Brief 305 (Download now)
EB5-4 Blind VST: A Perceptual Testing Tool for Professional Mixing Evaluation—Matthew Boerum, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT)
The Blind VST tool is presented to aid perceptual testing of real time applied digital audio signal processing in professional mixing situations. The software was designed in MaxMSP and runs as a standalone application or MaxMSP external. It hosts any Virtual Studio Technology (VST) plug-in for blind, singular control of any and all accessible VST parameters. Visual biasing from the identification and response of the VST’s graphical user interface (GUI) is removed. All user data is recorded in real time. The author proposes this tool as a freely distributed academic resource. It is best integrated as an add-on module for training and research applications when determining audible signal quality, comparison and the analysis of audio descriptors.
Engineering Brief 306 (Download now)