143rd AES CONVENTION Engineering Brief Details

AES New York 2017
Engineering Brief Details

EB01 - Posters—Part 1

Wednesday, October 18, 2:15 pm — 3:45 pm (Poster Area)

EB01-1 Frontal Sound Localization with Headphone Systems Using Characteristics of Each DeviceYuya Ono, University of Aizu - Aizu Wakamatsu, Japan; Akira Saji, University of Aizu - Aizuwakamatsu City, Japan; Jie Huang, University of Aizu - Aizuwakamatsu City, Japan
Generally, it is difficult to localize sounds on frontal positions with headphones because of the so called in-the-head localization. In this paper we proposed a method to create sound images on frontal positions by cancelling the frequency characteristics of sounds from headphones and adapting the frequency characteristics of sounds from frontal loudspeaker that are measured with HATS. We held auditory experiments that evaluating the similarity and naturalness between the sounds from frontal loudspeaker and headphones on the participants. As a result, we found the method can decrease the in-the-head localization
Engineering Brief 354 (Download now)

EB01-2 Design of Loudspeaker Crossover Networks Using Genetic AlgorithmChristian Claumarch, Aalborg University - Aalborg, Denmark; Kasper Kiis Jensen, Aalborg University - Aalborg, Denmark; Mikkel Krogh Simonsen, Aalborg University - Aalborg, Denmark
This brief proposes a method which allows for automation of the design of crossover networks for loudspeakers. The algorithm utilizes a combination of Genetic Algorithm and Least Squared Error Frequency Domain Design to find optimal type 1 FIR-filters fitting a target. The algorithm takes magnitude response, group delay, directivity index and displacement into account. Based on a two-way loudspeaker, the algorithm is capable of finding a crossover network which results in the final loudspeaker having a flat magnitude response from 105 Hz– 16 kHz ±  1.6 dB on-axis, where the directivity index is minimized and displacement limits are not exceeded
Engineering Brief 355 (Download now)

EB01-3 Microphone Array Impulse Response (MAIR) Library for Spatial Audio ResearchHyunkook Lee, University of Huddersfield - Huddersfield, UK; Connor Millns, University of Huddersfield - Huddersfield, UK
This engineering brief describes an open-access library of an extensive set of room impulse responses (RIRs) captured using numerous microphone arrays from 2-channel stereo to 9-channel surround with height. The RIRs were obtained for 13 loudspeakers placed in various positions on a stage in a reverberant concert hall. The library features five 2-channel stereo pairs, 10 main surround arrays, nine height microphone arrays for 3D main arrays and 15 4-channel configurations for surround and 3D ambience arrays, each with varied microphone polar patterns, directions, spacings, and heights. A dummy head and a first-order-Ambisonics microphone are also included. The library is provided with a rendering tool, with which the user can easily simulate different microphone combinations in both loudspeaker and binaural playback for 13 source positions. The library can be freely downloaded from the Resources section of the APL website: https://www.hud.ac.uk/apl
Engineering Brief 356 (Download now)

EB01-4 A Database of Head-Related Transfer Functions and Morphological MeasurementsRahulram Sridhar, Princeton University - Princeton, NJ, USA; Joseph G. Tylka, Princeton University - Princeton, NJ, USA; Edgar Choueiri, Princeton University - Princeton, NJ, USA
A database of head-related transfer function (HRTF) and morphological measurements of human subjects and mannequins is presented. Data-driven HRTF estimation techniques require large datasets of measured HRTFs and morphological data, but only a few such databases are freely available. This paper describes an on-going project to measure HRTFs and corresponding 3D morphological scans. For a given subject, 648 HRTFs are measured at a distance of 0.76 m in an anechoic chamber and 3D scans of the subject’s head and upper torso are acquired using structured-light scanners. The HRTF data are stored in the standardized “SOFA format” (spatially-oriented format for acoustics) while scans are stored in the Polygon File Format. The database is freely available online.
Engineering Brief 357 (Download now)

EB01-5 Phase Continuity in Amplitude-Phase Spatial Audio CodingFrançois Becker, Coronal Audio - Paris, France; Benjamin Bernard, Coronal Audio - Monaco, Monaco; Longcat Audio Technologies - Chalon-sur-Saone, France; Clément Carron, Coronal Audio - Lyon, France; Longcat Audio
In the context of amplitude-phase spatial audio coding, we give a proof of a phase discontinuity problem that affects all previous tridimensional stereo-compatible schemes. We solve it by using a dynamic mapping of spherical coordinates to the Scheiber sphere, which ensures phase continuity.
Engineering Brief 358 (Download now)

EB01-6 Evaluation of Binaural Renderers: A MethodologyGregory Reardon, New York University - New York, NY, USA; Agnieszka Roginska, New York University - New York, NY, USA; Patrick Flanagan, THX Ltd. - San Francisco, CA, USA; Juan Simon Calle, New York University - New York, NY, USA; THX; Andrea Genovese, New York University - New York, NY, USA; Gabriel Zalles, New York University - New York, NY, USA; Marta Olko, New York University - New York, NY, USA; Christal Jerez, New York University - New York, NY, USA; Platinum Sound Recording Studio
Recent developments in immersive audio technology have motivated a proliferation of binaural renderers used for creating spatial audio content. Binaural renderers leverage psychoacoustic features of human hearing to reproduce a 3D sound image over headphones. In this paper a methodology for the comparative evaluation of different binaural renderers is presented. The methodological approach is threefold. A subjective evaluation of 1) quantitative characteristics (such as front/back and up/down discrimination, localization) ; 2) qualitative characteristics (such as timbre, naturalness); and 3) overall preference. The main objective of the methodology is to help to elucidate the most meaningful factors for the performance of binaural renderers and to provide indications on possible improvements in the rendering process.
Engineering Brief 359 (Download now)

EB01-7 Simultaneous HRTF Measurement of Multiple Source Configurations Utilizing Semi-Permanent Structural MountsCal Armstrong, University of York - York, UK; Andrew Chadwick, University of York - York, UK; Lewis Thresh, University of York - York, UK; Damian Murphy, University of York - York, UK; Gavin Kearney, University of York - York, UK
A compact HRTF measurement rig has been designed and erected within the anechoic chamber at AudioLab, University of York. Utilizing 24 discrete elevations the efficient simultaneous HRTF measurement of 11 popular source configurations, ideally suited for the binaural rendering of Ambisonics, is undertaken. An overlapped exponential swept sine technique is used to make optimal use of a subject’s time. This report details the practical requirements, technical workflow and processing involved in the HRTF measurements, for inclusion to the SADIE database. The necessary modelling of low frequency cues is discussed.
Engineering Brief 360 (Download now)

EB01-8 Influence of Audience Noises on the Classical Music Perception on the Example of Anti-cough Candies Unwrapping NoiseAdam Pilch, AGH University of Science and Technology - Krakow, Poland; Bartlomiej Chojnacki, AGH University of Science and Technology - Kracow, Poland; Teresa Makuch, AGH University of Science and Technology - Cracow, Poland; Zuzanna Kusal, AGH University of Science and Technology - Kracow, Poland; Marcjanna Czapla, AGH University of Science and Technology - Kracow, Poland
A common problem in concert halls are people in the audience who distract other listeners by creating noises. Unwrapping anti-cough candies is an example of such undesirable behavior. The subject of the paper is to compare and analyze acoustic parameters of various candy wrappings in order to determine the discomfort they cause. The sounds generated while removing wrappings made of different materials were recorded in an anechoic chamber. The recordings were then analyzed in order to locate sounds in the audible frequency band in relation to musical sounds. Based on the results and a survey that was also carried out, an attempt was made to specify parameters of the noises perceived as most distracting.
Engineering Brief 361 (Download now)

EB01-9 A Simple Evaluating Method of a Reproduced Sound Field by a Measurement of Sound Intensities Using Virtual Source VisualizerMasataka Nakahara, ONFUTURE Ltd. - Tokyo, Japan; SONA Corp. - Tokyo, Japan; Akira Omoto, Kyushu University - Fukuoka, Japan; Onfuture Ltd. - Tokyo, Japan; Yasuhiko Nagatomo, Evixar Inc. - Tokyo, Japan
Recently, many kinds of technologies for restoring/reproducing 3D sound fields are proposed. However, it is little opportunity to compare these acoustic performances under a common condition. Though a subjective evaluation is one of the most effective methods for evaluating reproduced sound fields, it requires careful effort. Therefore, the authors propose an alternative method which requires only a physical measurement of sound intensities. Because the method is based on the intensity analysis, “sound images” are assumed to be “amplitude-based phantom sound sources” here. In order to verify effectiveness of the method, various types of reproduced fields were measured and analyzed. As a result, it is ascertained that the method can evaluate proper features of reproduced sound fields, regardless of their restoring techniques.
Engineering Brief 362 (Download now)

EB01-10 Physical Evaluations of Reproduced Sound Fields by a Measurements of Sound Intensities Using Virtual Source VisualizerTakashi Mikami, SONA Co. - Tokyo, Japan; Masataka Nakahara, ONFUTURE Ltd. - Tokyo, Japan; SONA Corp. - Tokyo, Japan; Akira Omoto, Kyushu University - Fukuoka, Japan; Onfuture Ltd. - Tokyo, Japan
In order to evaluate acoustic properties of reproduced sound fields, sound intensities were measured and analyzed in various types of multichannel studios by using a Virtual Source Visualizer (VSV hereafter). First, two different methods to reproduce sound fields are examined; 24ch amplitude-based phantom sound sources and Kirchhoff-Helmholtz-integral-based Boundary Surface Control principle. Secondly, sound fields created by four different types of 3D panners are examined; Dolby Atmos, DTS:X, Auro-3D and 22.2ch. Through these measurements, it was demonstrated that the VSV analyzes acoustic features of reproduced fields well, and interchangeabilities and differences of acoustic properties among different reproduced fields can be understood clearly. The session discusses accuracy and features of various types of reproduced sound fields which we measured and analyzed by the VSV.
Engineering Brief 363 (Download now)

EB01-11 WithdrawnN/A

Engineering Brief 382 (Download now)


EB02 - Recording & Production

Thursday, October 19, 11:15 am — 12:30 pm (Rm 1E12)

Palmyra Catravas, Union College - Schenectady, NY, USA

EB02-1 Engineered Remote-Sensing Audio Power Amplifier for High-Fidelity ApplicationsPeter Horowitz, Fourth Dimension Engineering - Columbia, MD, USA
The objective of this work is to minimize the deleterious effects of loudspeaker cable impedance when driving dynamic loudspeakers, accomplished primarily with a mathematical feedback analysis on the prominent role of the cables themselves within the audio baseband feedback loop. Presented are the measured system waveforms, along with computed root loci and transfer functions of a proof-of-principle remote-sensing (4-wire) 80 watt audio power amplifier. A single baseband feedback loop compares the incoming audio information (voltage) to the resultant voltage across the loudspeaker electrical terminals and minimizes the difference. Measured waveforms demonstrate notably superior replication of incoming information at the loudspeaker terminals over the audio band. The system is empirically robust for a wide range of dynamic loudspeaker and cable systems without any need for electronic adjustment. For example, with 35 meter 15/22 gauge cabling, a bandwidth of 72kHz, dynamic range of 110dB, phase linearity of <0.5°, and low impedance drive levels of <0.2O at the loudspeaker terminals are readily achieved simultaneously.
Engineering Brief 364 (Download now)

EB02-2 Building a Globally Distributed Recording StudioJohn Fiorello, RecordME - Torrington, CT, USA
The internet has played a significant role in changing consumer behavior in regards to the distribution and consumption of music. Record labels, recording studios, and musicians have felt the financial squeeze as physical media delivery has been depreciated. However, the internet also enables these studios, musicians, and record labels to re-orient their business model to take advantage of new content creation and distribution. By developing a hardware appliance that combines high-resolution audio recording and broadcasting with real-time, two-way video communication across the web, we can expand the geographic area that studios can serve, increase revenue for musicians, and change the value proposition traditional record labels have to offer.
Engineering Brief 365 (Download now)

EB02-3 Simultaneous Audio Capture at Multiple Sample Rates and Formats for Direct Comparison and EvaluationJordan Strum, ProStudioMasters - Montreal, QC, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Oles Protsidym, ProStudioMasters - Montreal, QC, Canada; Ieronim Catanescu, McGill University - Montreal, QC, Canada
In order to evaluate differences among recording formats and resolutions over a variety of classical, jazz, and popular musical material, a unique collection of audio assets was recorded. Live performances were captured using a single pair of microphones, positionally adjusted for each sound source. Preamplifier outputs were routed to 11 identical recording interfaces capturing various PCM and DSD formats simultaneously with 3 analogue tape recorders, the contents of which were then transferred to the above digital formats. These assets will be used to compare differences among recording formats and resolutions using identical performances, and to provide source material for listening tests as well as further research. The design and execution of this project will be discussed.
Engineering Brief 366 (Download now)

EB02-4 Undergraduate Curricular Development at the Electrical Engineering/Music Interface at Union CollegePalmyra Catravas, Union College - Schenectady, NY, USA
Curricular development at the interface of electrical engineering and music will be described, with a focus on the pedagogical use of audio and acoustics to reinforce basic fundamentals in electrical engineering. The effort, which has taken place over more than a decade, seeks to reinforce the foundation provided by the traditional, rigorous engineering curriculum at Union College, an undergraduate liberal arts college in upstate NY. A related specialized research laboratory – Phasor Lab – is located in the Peter Irving Wold Science and Engineering Center at Union.
Engineering Brief 367 (Download now)

EB02-5 Recording, Mixing and Mastering of Audio Using a Single Microphone Array and Audio Source Separation AlgorithmsJakub Zamojski, Zylia sp. z.o.o. - Poznan, Poland; Piotr Makaruk, Zylia - Poznan, Poland; Lukasz Januszkiewicz, Zylia Sp. z o.o. - Poznan, Poland; Tomasz Zernicki, Zylia sp. z o.o. - Poznan, Poland
The authors present a new way of recording of music bands using a single microphone array and audio source separation algorithms. In contrast to the traditional recording process, this novel approach allows for capturing all of musical instruments simultaneously using only one recording device, avoiding a multiple sets of spot microphones, cables, and D/A converters. Moreover, using a single microphone array and dedicated algorithms gives the sound engineer a unique set of “audio processing tools” that can be used in the post-production stage. This paper describes step-by-step a recording process of a music band playing ukulele using a 19-capsules spherical microphone array and dedicated software. The process includes the following stages: recording, sound source separation, mixing, and mastering.
Engineering Brief 368 (Download now)


EB03 - Applications in Audio

Thursday, October 19, 3:45 pm — 5:00 pm (Rm 1E12)

Bob Schulein, ImmersAV Technology - Schaumburg, IL, USA

EB03-1 Evolving the Audio EqualizerDavid Yonovitz, Key 49 - Del Mar, CA, USA
Current audio equalization techniques include Shelf, Parametric, and Graphic Equalizers. Each have inherent issues: dynamic spectrum input and the degradation of signal-to-noise ratio. The input spectrum is not static; yet, all the current equalizations are. To be effective, equalization must be dynamic, “tracking” the input signal spectrum. In the case of SNR, output noise is increased when no signal is present in the spectral band when adding gain. In an evolution of equalization, with input tracking capability, signal spectral components are identified and equalized; all other spectrum may be considered as noise and can be attenuated. The evolution of audio equalizers has progressed that negates the stated issues. Its implementation is realized in the Harmonic Tracking Equalizer (HTEq).
Engineering Brief 369 (Download now)

EB03-2 Immersive Audio: Optimizing Creative Impact without Increasing Production CostsConnor Sexton, Avid - Berkeley, CA, USA
Since its introduction in 2012, Dolby Atmos has gained widespread adoption in theatrical distribution for films, with over 2,000 Dolby Atmos enabled theaters worldwide. Now expanding into TV and gaming, this unique audio mixing format provides a new dimension of creative control over the immersive listening experience. Numerous audio workstations and consoles have been retrofitted for Dolby Atmos, but without native support, workflows have become cumbersome and complex. This paper will present best practices for native immersive audio production, from sound design to mixing to distribution. It will demonstrate how the latest audio production tools and techniques enabled content creators to capitalize on the creative power of immersive audio while streamlining the parallel authoring of traditional formats.
Engineering Brief 370 (Download now)

EB03-3 WithdrawnN/A

Engineering Brief 371 (Download now)

EB03-4 "Match Your Own Voice!": An Educational Tool for Vocal TrainingEvangelos Angelakis, National and Kapodistrian University of Athens - Athens, Greece; Panayiotis Velianitis, National and Kapodistrian University of Athens - Athens, Greece; Areti Andreopoulou, National and Kapodistrian University of Athens - Athens, Greece; Anastasia Georgaki, National and Kapodistrian University of Athens - Athens, Greece
In this paper, we discuss the development and preliminary evaluation of a new educational tool, intended for novice and advanced vocal students. The software, written in Max / MSP, aims to assist singing practice by providing users with a visual substitute to their subjective auditory feedback. Under the guidance of their professional vocal instructor, students can store in the software spectral representations of accurately produced sounds, creating personalized Reference Sound Banks (RSBs). When students practice on their own, the software can be put into practice, assisting them to match their current Voice Spectrum Harmonic Content to the stored RSBs one note at a time. Results of a preliminary evaluation showed that, when using this software, students achieve a larger number of accurately produced sounds in a smaller amount of time.
Engineering Brief 372 (Download now)

EB03-5 Measuring Micro-Dynamics—A First Step: Standardizing PSR, the Peak to Short-Term Loudness RatioIan Shepherd, Mastering Media Ltd - Cambridge, Cambridgeshire, UK; Eelco Grimm, HKU University of the Arts - Utrecht, Netherlands; Grimm Audio - Eindhoven, The Netherlands; Paul Tapper, Nugen Audio - UK; Michael Kahsnitz, RTW - Cologne, Germany; Ian Kerr, MeterPlugs Audio Inc. - Vancouver, BC, Canada
The “loudness war” still rages, but with major digital streaming services switching to loudness normalization by default, its end is near. Since absolute loudness is no longer effective at making music “stand out,” engineers are finding it much more effective to optimize microdynamics instead. The overall PLR (Peak to Loudness Ratio) of an audio track is widely recognized as a useful metric to assess the overall microdynamics of a section of audio and the likely results of normalization. However, short-term variations are also important, especially when judging the results of compression and limiting on audio quality, and these can be usefully assessed by a real-time property known as PSR (Peak to ShortTerm Loudness Ratio). PSR is found to be straightforward and intuitive to use, and several popular meters are already reporting it. This paper proposes a standardization of the term, to encourage consistency and adoption.
Engineering Brief 373 (Download now)


EB04 - Transducers

Friday, October 20, 4:30 pm — 5:45 pm (Rm 1E11)

Pascal Brunet, Samsung Research America - Valencia, CA USA; Audio Group - Digital Media Solutions

EB04-1 The Resonant Tuning Factor: A New Measure for Quantifying the Setup and Tuning of Cylindrical DrumsRob Toulson, University of Westminster - London, UK
A single circular drumhead produces complex and in-harmonic vibration characteristics. However, with cylindrical drums, which have two drumheads coupled by a mass of air, it is possible to manipulate the harmonic relationships through changing the tension of the resonant drumhead. The modal ratio between the fundamental and the batter head overtone therefore provides a unique and quantified characteristic of the drum tuning setup, which has been termed as the Resonant Tuning Factor (RTF). It may be valuable, for example, for percussionists to manipulate the RTF value to a perfect musical fifth, or to simply enable a repeatable tuning setup. This research therefore considers a number of user interfaces for analyzing the RTF and providing a tool for quantitative drum tuning.
Engineering Brief 374 (Download now)

EB04-2 Design and Implementation of a Practical Long-Throw High-Q CBT ArrayD. B. (Don) Keele, Jr., DBK Associates and Labs - Bloomington, IN, USA; Marshall Kay, Keysight Technologies - Apex, NC, USA
This paper describes the design and construction of a very-tall 5m experimental passive long-throw high-Q CBT array that provides coverage in a large church general-purpose activity room with a full-size basketball court. The room is 7.8 x 20 x 30 m (H x W x L). The 5 m tall 20° circular- arc array contains 80 ea 63.5 mm (2.5”) full-range drivers, and provides a tight 15° vertical beamwidth. The mechanically aimed no-DSP passive segmented design is composed of five straight-front boxes each containing 16 drivers. Series-parallel connections, resistive attenuators, and two power amplifiers provide the frequency-independent four-bank CBT shading. This paper also provides detailed simulation data of the array’s predicted beamwidth vs. frequency, directivity, vertical polar response, axial foot prints and predicted frequency response at three different downward tilt angles. The array provides very-even coverage along the entire length of the 30 m room.
Engineering Brief 375 (Download now)

EB04-3 Effects of Acoustic Center Position in SubwoofersMario Di Cola, Audio Labs Systems - Casoli, Italy; Paolo Martignon, Contralto Audio srl - Parma (PR), Italy; Merlijn van Veen, Merlijn van Veen - Soest, Utrecht, The Netherlands
As explained by J.Vanderkooy [1] the acoustic center of a direct radiating subwoofer unit is placed ahead respect to the driver membrane, at a distance depending on driver and cabinet dimensions. This has effects on acoustic simulations and it deserves some attention to avoid errors. Measurements are shown which confirm acoustic center position theoretical calculation and a discussion is made about its effect on the definition of models for accurate simulations.
Engineering Brief 376 (Download now)

EB04-4 Design and Implementation of a Constant-Directivity Two-Way 12” Woofer Wedge Loudspeaker SystemD. B. (Don) Keele, Jr., DBK Associates and Labs - Bloomington, IN, USA; Hugh Sarvis, Presonus Audio Electronics-Worx Audio Technologies - Baton Rouge, LA, USA
This paper describes the design and implementation of a two-way constant-directivity wedge loudspeaker system that houses a single 12” woofer and eight 2” drivers in a 20° circular arc mounted on a curved baffle that covers the LF driver. An individual system comprises a 20° wedge box with a four-channel plate amplifier with two bridged channels driving the woofer, and the two other channels individually driving each half of the eight-driver array. This basic wedge box is then used in multiples to form larger circular-arc arrays of one up to six boxes making arrays that provide various vertical beamwidths in the range of 15° to 90°. Appropriate amplifier gains are chosen to smooth the polar coverage for each array size.
Engineering Brief 377 (Download now)

EB04-5 A Tutorial on the Audibility of Loudspeaker Distortion at Bass FrequenciesJames Larson, Audioholics—Online A/V Magazine - South Elgin, IL, USA; Gene DellaSaia, Audioholics—Online A/V Magazine; D. B. (Don) Keele, Jr., DBK Associates and Labs - Bloomington, IN, USA
This tutorial paper goes into detail concerning the audibility and perception of loudspeaker distortion at low frequencies. It draws on many past references and publications to summarize many of the factors that contribute to low-frequency loudspeaker distortion. Items covered include: “What is distortion and how do we perceive it?,” causes of distortion, types of distortion and audibility: linear vs. nonlinear, THD vs. IM vs. intermodulation distortion etc., auditory masking and distortion thresholds, measurement methods including continuous sine wave, two-tone IM, tone-burst, and multi-tone log-spaced testing among others. In conclusion, this paper observes that distortion does occur, but by identifying the point at which distortion becomes audible, one can be prudent in choosing which distortions to ignore.
Engineering Brief 378 (Download now)


EB05 - Posters—Part 2

Saturday, October 21, 9:00 am — 10:30 am (Poster Area)

EB05-1 Impulse and Radiation Field Measurements for Single Exciter versus Exciter Array Flat-Panel LoudspeakersDavid Anderson, University of Rochester - Rochester, NY, USA; Michael Heilemann, University of Rochester - Rochester, NY, USA; Mark F. Bocko, University of Rochester - Rochester, NY, USA
Flat-panel loudspeakers with single exciters exhibit significant directivity shifts and many discrete resonances in frequency regions of low modal density. These phenomena are demonstrated through mechanical and acoustic measurements on an acrylic and a glass prototype panel, both with single exciters. The measurements are repeated for the acrylic panel using an array of exciters where the force magnitude of each exciter is specified to actuate only the lowest-index bending mode. The mechanical measurements demonstrate that no modes in the array-addressable frequency region above the first mode are actuated. Acoustically, measurements show omnidirectional radiation with a single low-frequency resonance, showing how the exciter array enables the flat panel to behave similarly to a conventional loudspeaker within the array-addressable frequency region.
Engineering Brief 379 (Download now)

EB05-2 Implementation of a Dipole Constant Directivity Circular-Arc ArrayKurtis Manke, Thompson Rivers University - Kamloops, BC, Canada; Richard Taylor, Thompson Rivers University - Kamloops, BC, Canada; Mark Paetkau, Thompson Rivers University - Kamloops, Canada; D. B. (Don) Keele, Jr., DBK Associates and Labs - Bloomington, IN, USA
We briefly present the theory for a broadband constant-beamwidth transducer (CBT) formed by a conformal circular-arc array of dipole elements previously developed in seminal works. This technical report considers a dipole CBT prototype with cosine amplitude shading of the source distribution. We show that this leads to a readily-equalizable response from about 100 Hz to 10 kHz with a far-field radiation pattern that remains constant above the cutoff frequency determined by the beam-width and arc radius of the array, and below the critical frequency determined by discrete element spacing at which spatial aliasing effects occur. Furthermore, we show that the shape of the radiation pattern is the same as the shading function, and remains constant over a broad band of frequencies.
Engineering Brief 381 (Download now)

EB05-3 Flexible Control of the Transducer and the Duct Resonance of a Speaker System Ducted to the Exterior of a Vehicle CabinTakashi Kinoshita, Bose Automotive G.K. - Tokyo, Tokyo, Japan; John Feng, Bose - Framingham, MA, USA
In order to reproduce lower frequency sound in a vehicle cabin efficiently, Zeljko Velican proposed a speaker system, where the backside of a transducer unit communicates with the exterior of a vehicle cabin via a tuned acoustic appliance. [1] Since this speaker system couples the interior and the exterior of a vehicle cabin, the efficiency and the frequency range of internal and external noise transmission are both important considerations. These two characteristics are strongly correlated with the two dominant resonances of the system. One is the mechanical resonance of the transducer which defines the lower limit of the sound reproduction frequency range. Another one is the Helmholtz resonance of the back-side acoustic appliance (enclosure and duct), which defines the frequency where, for example, noise transmission through the appliance is optimized. Choosing the appropriate acoustic parameters to balance those two dominant resonances is the key to optimal design this speaker system. But with the existing configuration [1], these two dominant acoustic resonances have strong mutual interaction via coupled design parameters, it can be difficult to find a good compromise between them. In this paper, a new speaker system configuration, consists of a transducer, an enclosure ducted to the exterior of the vehicle cabin, and a passive radiator to cover the duct, will be proposed and discussed. With this configuration, the two dominant resonances of the system can be controlled quasi-individually, therefore enhancing design flexibility for the practical use of such systems on a vehicle.
Engineering Brief 383 (Download now)

EB05-4 Multichannel Microphone Array Recording for Popular Music Production in Virtual RealityHashim Riaz, University of York - York, Yorkshire, UK; Mirek Stiles, Abbey Road Studios - London, UK; Cal Armstrong, University of York - York, UK; Andrew Chadwick, University of York - York, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK; Gavin Kearney, University of York - York, UK
There is a growing market for innovative ways to appreciate and listen to music through new Virtual Reality (VR) experiences made accessible through smartphones and VR headsets. However, production workflows for creating immersive musical experiences over VR are still in their infancy. This engineering report documents different microphone configurations and recording techniques applied in a higher-order Ambisonic processing framework to deliver an engaging and hyper-real interactive VR music experience. The report documents a live popular music recording undertaken at Abbey Road with traditional music recording techniques such as spot and stereo microphone setups and advanced techniques using dedicated VR multichannel microphone arrays.
Engineering Brief 384 (Download now)

EB05-5 WithdrawnN/A

Engineering Brief 385 (Download now)

EB05-6 Consonant Perception and Improved S/N Ratio Using Harmonic Tracking EqualizationAl Yonovitz, University of Montana - Missula, MT, USA; Silas Smith, University of Montana - Missula, MT, USA; David Yonovitz, Key 49 - Del Mar, CA, USA
Audio equalization techniques are often used to enhance signals and reduce noise. These include Shelf, Parametric, and Graphic Equalizers. These techniques modify spectral components within specified bands by applying gain or attenuation. Another promising technique utilizes the tracking of harmonics and sub-harmonics (HTEq). These harmonics may be individually changed in intensity. This study utilized 21 Consonant Vowel (CV) stimuli with a white noise masker (+6 dB S/N). Each stimulus was randomly presented to listeners. Confusion matrices determined consonant intelligibility and information transmission for distinctive features. Perceptually, after HTEq, the noise was minimally audible and required considerably less effort to identify consonants. The results indicated the distinctive feature transmission was not altered. Comparisons were made for consonants at various levels of noise reduction.
Engineering Brief 386 (Download now)

EB05-7 Bridging Fan Communities and Facilitating Access to Music Archives through Semantic Audio ApplicationsThomas Wilmering, Queen Mary University of London - London, UK; Centre for Digital Music (C4DM); Florian Thalmann, Queen Mary University of London - London, UK; György Fazekas, Queen Mary University of London - London, UK; Mark Sandler, Queen Mary University of London - London, UK
Semantic Audio is an emerging field in the intersection of signal processing, machine learning, knowledge representation, and ontologies unifying techniques involving audio analysis and the Semantic Web. These mechanisms enable the creation of new applications and user experiences for music communities. We present a case study focusing on what Semantic Audio can offer to a particular fan base, that of the Grateful  Dead, characterized by a profoundly strong affinity with technology and the internet. We discuss an application that combines information drawn from existing platforms and results from the automatic analysis of audio content to infer higher-level musical information, providing novel user experiences particularly in the context of live music events.
Engineering Brief 387 (Download now)

EB05-8 The ANU School of Music Recording Studios: Design, Technology, Research, and PedagogySamantha Bennett, Australian National University - Canberra, ACT, Australia; Matt Barnes, Australian National University - Canberra, Australia
This engineering brief addresses the refurbishment process of the School of Music, Australian National University recording studios to include focus on the historical, pedagogical and research requirements of a 21st  Century studio facility. The brief will first address issues of space, heritage and purpose before considering the acoustic (re)design process. Furthermore, the brief examines issues of technological integration and facilitation of analogue, digital and hybrid workflows. Finally, the brief considers the research and pedagogical remit of the refurbished facilities.
Engineering Brief 397 (Download now)


EB06 - Spatial Audio

Saturday, October 21, 1:30 pm — 3:15 pm (Rm 1E12)

Matthieu Parmentier, francetélévisions - Paris, France

EB06-1 How Streaming Object Based Audio Might WorkAdrian Wisbey, BBC Design and Engineering - London, UK
Object based media is being considered as the future platform model by a number of broadcasting and production organizations. This paper is a personal imagining of how object based broadcasting might be implemented with IP media as the primary distribution whilst still supporting traditional distributions such as FM, DAB and DVB. The examples assume a broadcaster supporting a number of linearly scheduled services providing both live (simulcast ) and on-demand (catch-up) content. An understanding of the basics of object based audio production and broadcasting by the reader is assumed. Whilst this paper specifically discusses audio or radio broadcasting many of the components and requirements are equally valid in a video environment.
Engineering Brief 398 (Download now)

EB06-2 DIY Measurement of Your Personal HRTF at Home: Low-Cost, Fast and ValidatedJonas Reijniers, University of Antwerp - Antwerpen, Belgium; Bart Partoens, University of Antwerp - Antwerp, Belgium; Herbert Peremans, University of Antwerp - Antwerpen, Belgium
The breakthrough of 3D audio has been hampered by the lack of personalized head-related transfer functions (HRTF) required to create realistic 3D audio environments using headphones. In this paper we present a new method for the user to personalize his/her HRTF, similar to the measurement in an anechoic room, yet it is low-cost and can be carried out at home. We compare the resulting HRTFs with those measured in an anechoic room. Subjecting the participants to a virtual localization experiment, we show that they perform significantly better when using their personalized HRTF, compared to a generic HRTF. We believe this method has the potential of opening the way for large scale commercial use of 3D audio through headphones.
Engineering Brief 399 (Download now)

EB06-3 Audio Localization Method for VR ApplicationJoo Won Park, Columbia University - New York, NY, USA
Audio localization is a crucial component in the Virtual Reality (VR) projects as it contributes to a more realistic VR experience to the users. In this paper a method to implement localized audio that is synced with user’s head movement is discussed. The goal is to process an audio signal real-time to represent three-dimensional soundscape. This paper introduces a mathematical concept, acoustic models, and audio processing that can be applied for general VR audio development. It also provides a detailed overview of an Oculus Rift- MAX/MSP demo.
Engineering Brief 400 (Download now)

EB06-4 Sound Fields Forever: Mapping Sound Fields via Position-Aware SmartphonesScott Hawley, Belmont University - Nashville, TN, USA; Sebastian Alegre, Belmont University - Nashville, TN, USA; Brynn Yonker, Belmont University - Nashville, TN, USA
Google Project Tango is a suite of built-in sensors and libraries intended for Augmented Reality applications allowing certain mobile devices to track their motion and orientation in three dimensions without the need for any additional hardware. Our new Android app, "Sound Fields Forever," combines locations with sound intensity data in multiple frequency bands taken from a co-moving external microphone plugged into the phone's analog jack. These data are sent wirelessly to a visualization server running in a web browser. This system is intended for roles in education, live sound reinforcement, and architectural acoustics. The relatively low cost of our approach compared to more sophisticated 3D acoustical mapping systems could make it an accessible option for such applications.
Engineering Brief 401 (Download now)

EB06-5 Real-time Detection of MEMS Microphone Array Failure Modes for Embedded MicroprocessorsAndrew Stanford-Jason, XMOS Ltd. - Bristol, UK
In this paper we describe an online system for real-time detection of common failure modes of arrays of MEMS microphones. We describe a system with a specific focus on reduced computational complexity for application in embedded microprocessors. The system detects deviations is long-term spectral content and microphone covariance to identify failures while being robust to the false negatives inherent in a passively driven online system. Data collected from real compromised microphones show that we can achieve high rates of failure detection.
Engineering Brief 402 (Download now)

EB06-6 A Toolkit for Customizing the ambiX Ambisonics-to-Binaural RendererJoseph G. Tylka, Princeton University - Princeton, NJ, USA; Edgar Choueiri, Princeton University - Princeton, NJ, USA
An open-source collection of MATLAB functions, referred to as the SOFA/ambiX binaural rendering (SABRE) toolkit, is presented for generating custom ambisonics-to-binaural decoders for the ambiX binaural plug-in. Databases of head-related transfer functions (HRTFs) are becoming widely available in the recently-standardized “SOFA format” (spatially-oriented format for acoustics), but there is currently no (easy) way to use custom HRTFs with the ambiX binaural plug-in. This toolkit enables the user to generate custom binaural rendering configurations for the plug-in from any SOFA-formatted HRTFs or to add HRTFs to an existing ambisonics decoder. Also implemented in the toolkit are several methods of HRTF interpolation and equalization. The mathematical conventions, ambisonics theory, and signal processing implemented in the toolkit are described.
Engineering Brief 403 (Download now)

EB06-7 withdrawnN/A

Engineering Brief 404 (Download now)


EB07 - Posters—Part 3

Saturday, October 21, 2:00 pm — 3:30 pm (Poster Area)

EB07-1 Cycle-Frequency Wavelet Analysis of Electro-Acoustic SystemsDaniele Ponteggia, Audiomatica Srl - Firenze (FI), Italy
A joint time-frequency analysis of the response of electro-acoustic systems has been long sought since the advent of PC based measurement systems. While there are several available tools to inspect the time-frequency response, when it comes to inspect resonant phenomena, there are always issues with time-frequency resolution. With a rather simple variable substitution in the Wavelet Analysis it is possible to switch one of the analysis axes from time to cycles. With this new cycle-frequency distribution it is then possible to analyze very easily resonances and decays. A PC based measurement tool capable of Cycle-Frequency analysis will be introduced.
Engineering Brief 388 (Download now)

EB07-2 Sharper Spectrograms with Fast Local SharpeningRobin Lobel, Divide Frame - Paris, France
Spectrograms have to make compromises between time and frequency resolution because of the limitations of the short-time Fourier transform (Gabor,1946). Wavelets have the same issue. As a result spectrograms often appear blurry, either in time, frequency, or both, A method called Reassignment was introduced in 1978 (Kodera et al.) to make spectrograms look sharper. Unfortunately it also adds visual noise, and its algorithm does not make it suitable for realtime scenarios. Fast Local Sharpening is a new method that attempts to overcome both these drawbacks.
Engineering Brief 389 (Download now)

EB07-3 An Interactive and Intelligent Tool for Microphone Array DesignHyunkook Lee, University of Huddersfield - Huddersfield, UK; Dale Johnson, The University of Huddersfield - Huddersfield, UK; Manchester, UK; Maksims Mironovs, University of Huddersfield - Huddersfield, West Yorkshire, UK
This engineering brief will present a new microphone array design app named MARRS (microphone array recording and reproduction simulator). Developed based on a novel psychoacoustic time-level trade-off algorithm, MARRS provides an interactive, object-based workflow and graphical user interface for localization prediction and microphone array configuration. It allows the user to predict the perceived positions of multiple sound sources for a given microphone configuration. The tool can also automatically configure suitable microphone arrays for the user’s desired spatial scene in reproduction. Furthermore, MARRS overcomes some of the limitations of existing microphone array simulation tools by taking into account  microphone height and vertical orientations as well as the target loudspeaker base angle. The iOS and Android app versions of MARRS can be freely downloaded from the Apple App Store and the Resources section of the APL website: https://www.hud.ac.uk/apl, respectively.
Engineering Brief 390 (Download now)

EB07-4 Real-Time Multichannel Interfacing for a Dynamic Flat-Panel Audio Display Using the MATLAB Audio Systems ToolboxArvind Ramanathan, University of Rochester - Rochester, NY, USA; Michael Heilemann, University of Rochester - Rochester, NY, USA; Mark F. Bocko, University of Rochester - Rochester, NY, USA
Flat-panel audio displays use an array of force actuators to render sound sources on a display screen. The signal sent to each force actuator depends on the actuator position, the resonant properties of the panel, and the source position on the screen. A source may be translated to different spatial locations using the shifting theorem of the Fourier transform. A real-time implementation of this source positioning is presented using the MATLAB Audio Systems Toolbox. The implementation includes a graphical interface that allows a user to dynamically position the sound source on the screen. This implementation may be combined with audio source separation techniques to align audio sources with video images in real-time as part of a multimodal display.
Engineering Brief 391 (Download now)

EB07-5 Perceived Differences in Timbre, Clarity, and Depth in Audio Files Treated with MQA Encoding vs. Their Unprocessed StateMariane Generale, McGill University - Montreal, QC, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
The purpose of this engineering brief is to detail a planned experiment in examining any perceived differences in timbre, clarity, and depth between WAV and Master Quality Authenticated (MQA) audio files. A study proposes examining the responses of engineers, musicians, and casual listeners on whether any changes to timbre, clarity, and depth are perceived between WAV and MQA. A blind listening test is considered in a controlled environment using both professional and consumer level loudspeakers and headphones. Additional interests include a comparison of responses between the target groups on different listening mediums.
Engineering Brief 392 (Download now)

EB07-6 The BACH Experience: Bring a Concert HomeSattwik Basu, University of Rochester - Rochester, NY, USA; Saarish Kareer, University of Rochester - Rochester, NY, USA
Inverse filtering of rooms to improve their frequency response or reverberation time is a well-researched topic in acoustical signal processing. With the aim of giving music lovers the experience of a concert hall in their own homes, we describe a system that employs signal processing techniques, including inverse filtering, to accurately reproduce concert hall acoustics in a home listening space. First, binaural impulse responses were measured at a few chosen seating positions in the concert hall. Next, the listening location along with its loudspeaker configuration is acoustically characterized and inverse filtered using MINT and Cross-talk Cancellation algorithms to produce a flat-frequency response. We observed that speech and music, after our inverse filtering method showed near-anechoic qualities which allowed us to subsequently impress the acoustical response of a wide range of concert halls upon the original audio. A demonstration will be provided using 4 loudspeakers for a quadraphonic sound reproduction at the listening area. In continuing work, to produce a sufficiently wide listening area, we are combining head tracking with adaptive inverse filtering to adjust to the listeners’ movements.
Engineering Brief 393 (Download now)

EB07-7 Early Reflection Remapping in Synthetic Room Impulse Responses: Theoretical FoundationGregory Reardon, New York University - New York, NY, USA
In audio-visual augmented and virtual reality applications, the audio delivered must be consistent with the physical or virtual environment, respectively, in which the viewer/listener is located. Artificial binaural reverberation processing can be used to match the listener’s/viewer's environment acoustics. Typical real-time artificial binaural reverberators render the binaural room impulse response in three distinct section for computational efficiency. Rendering the response using different techniques means that within the response the early reflections and late reverberation may not give the same room-acoustic impression. This paper lays the theoretical foundation for early reflection remapping. This is accomplished by acoustically characterizing the virtual room implied by the early reflections renderer and then later removing that room-character from the response through frequency-domain reshaping.
Engineering Brief 394 (Download now)

EB07-8 Acoustic Levitation—Standing Wave DemonstrationBartlomiej Chojnacki, AGH University of Science and Technology - Kracow, Poland; Adam Pilch, AGH University of Science and Technology - Krakow, Poland; Marcin Zastawnik, AGH University of Science and Technology - Krakow, Poland; ProperSound - The Spokesmen of Science; Aleksandra Majchrzak, AGH University of Science and Technology - Krakow, Poland
Acoustic levitation is a spectacular phenomenon, perfect for standing waves demonstration. There are a few propositions for such a construction in scientific literature, however they are often expensive and difficult to build. The aim of this project was to create a functional stand - easy to construct, with no need for much expensive software or hardware. Piezoelectric transducers, typical for ultrasonic washing machines, were used as a sound source; their directivity pattern and frequency characteristics have been measured. The final result of the project was a stand-alone acoustic levitator with very little need for calibration, and with no walls, so the effect can be observed easily. The paper presents whole design process and describes all functionalities of the final stand.
Engineering Brief 395 (Download now)

EB07-9 Developing a Reverb Plugin; Utilizing Faust Meets JUCE FrameworkSteve Philbert, University of Rochester - Rochester, NY, USA
Plug-ins come in many different shapes, sizes and sounds, but what makes one different from another? The coding of audio and the development of the graphical User Interface (GUI) play a major part in how the plugin sounds and how it functions. This paper details methods of developing a reverb plugin by comparing different programming methods based around the Faust meets JUCE framework launched in February of 2017. The methods include: Faust direct to a plugin, Faust meets JUCE compiled with different architectures, and C++ with JUCE Framework. Each method has its benefits; some are easier to use while others provide a better basis for customization.
Engineering Brief 396 (Download now)


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