AES Rome 2013
Engineering Brief Details
EB1 - E-Brief Posters—Part 1
Saturday, May 4, 11:00 — 12:30 (Foyer)
EB1-1 Timecode-Aware Loudness Monitoring: Accelerate Engineers’ Everyday Workflow—Arnaud Laborie, Trinnov Audio - Bry sur Marne, France; Samuel Tracol, Trinnov Audio - Bry sur Marne, France; Arnaud Destinay, Trinnov Audio - Bry sur Marne, France; Jacques Di Giovanni, Trinnov Audio - Bry sur Marne, France
While EBU R-128 loudness normalization is in the process of being adopted by a majority of European countries, most real-time loudness meters aren’t still completely adapted to mixing engineers’ workflows, as continuous project measurements are always required to keep consistent loudness values. By slaving the loudness measurement to an incoming time code, every loudness and true peak values are constantly recorded and time stamped, allowing their calculation at any time. Engineers no longer need to manually pause, resume, or even start a measurement over to keep a relevant loudness monitoring.
Engineering Brief 78 (Download now)
EB1-2 Control of the Audio Signal Using Thermal Parameter for Protection of the Voice Coil—Oanjin Kim, Samsung Electronics Co., Ltd. - Suwon, Korea; Keeyeong Cho, Samsung Electronics Co., Ltd. - Suwon, Korea; Jongwoo Kim, Samsung Electronics Co., Ltd. - Suwon, Korea
This engineering brief presents a procedure for a signal processing method to protect a voice coil from overheating. The basic concept is that of estimating temperature of the voice coil with heat transfer model and controlling the output level before the voice coil is burnt. This paper mainly focuses on the calculation method and various considerations in level control. A basic heat transfer model and a commonly used method in calculation of the thermal parameters are introduced.
Engineering Brief 79 (Download now)
EB1-3 Automatic Segmentation of Concert Recordings via a Heuristic Approach—Andrew Ayers, University of Miami - Coral Gables, FL, USA
In the age of digital recordings, many institutions maintain large databases of concert recordings. While segmentation of these concert recordings for mastering and production is a time-consuming task for humans, this paper presents a novel heuristic algorithm to automate that process. Building on other work in audio segmentation, technique from the music informatics community is used to detect events, classify them, and segment entire concert recordings unsupervised. A brief review of previous work and the methodology used in this approach are provided, as well as the results obtained on a corpus of sixteen concerts.
Engineering Brief 80 (Download now)
EB1-4 Simulation of a Near Field Loudspeaker System on Headphones—Erich Meier, amoenus audio by Erich Meier - Bern, Switzerland
The vast majority of recorded music is produced for reproduction via loudspeakers positioned at the standard 60° stereo triangle. To achieve the same sound impression on headphones with its near-field 180° transducer positions, the sound of the standard stereo 60° triangle has to be simulated. We describe a circuit using different serial and parallel delay-and-filter paths for direct and cross-feed channels, with the goal to achieve accurate near-field loudspeaker-sound and also a good externalized localization.
Engineering Brief 81 (Download now)
EB1-5 Influence of First Reflections in Listening Room on Subjective Listener Impression of Reproduced Sound—Hidetaka Imamura, Tokyo University of the Arts - Tokyo, Japan; Atsushi Marui, Tokyo University of the Arts - Adachi-ku, Tokyo, Japan; Toru Kamekawa, Tokyo University of the Arts - Tokyo, Japan; Masataka Nakahara, SONA Corp. - Tokyo, Japan
This study investigated the perceptual factors regarding room acoustics such as spatial impression and timbre preferences, with focus on the arrival direction and pattern of the early reflections. Impulse responses were recorded with varying wall reflection and evaluated in a subjective test. Although no significant difference in timbre preferences and some evaluation terms were found in subjective listening test, the variation of early reflections did significantly influence listeners spatial impression. The method and results of the analysis is reported in the presentation.
Engineering Brief 82 (Download now)
EB1-6 Workload Estimation for Low-Delay Segmented Convolution—Malte Spiegelberg, HAW Hamburg - Hamburg, Germany
Zero-delay convolution usually follows a hybrid approach with convolution processing steps in both the time and the frequency domain [Gardner, J. Audio Eng. Soc., vol. 43, 127-136 (1995 Mar.)]. Implementations are likely to ask for dynamic coding, and related workload estimations are focused on efficiency and are limited to the hybrid approach. This paper considers simpler implementations of segmented convolution that work in the frequency domain only and that achieve acceptable low delay for real-time applications when processing several seconds of impulse-response in FIR mode. Workload and memory demand are estimated for this approach in the context of likely application parameters.
Engineering Brief 88 (Download now)
EB2 - E-Brief Papers—Part 1
Monday, May 6, 09:00 — 10:45 (Sala Foscolo)
Chair:
Etienne Corteel, Sonic Emotion Labs - Paris, France
EB2-1 Measurement of Sound Quality Differences in Individual CD Media Using Residual Waveform Comparison—Akira Nishimura, Tokyo Univeristy Information Sciences - Chiba-shi, Japan
To measure miniscule differences of sound quality that might exist between different CD media we compared residuals of two DA- and AD-converted waveforms from different discs on which the same data were recorded under clock synchronization conditions between DA and AD converters. This method clarifies the existence of differences in sound quality, except for sampling clock fluctuation. The results showed no media-dependent difference in sound quality. The main source of the residual waveform was change in the audio circuit transfer function associated with time since turning on the CD player.
Engineering Brief 83 (Download now)
EB2-2 Binaural Room Simulation for Acoustic Testing—Scott Levine, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Brett Leonard, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
Often, in testing with acoustic conditions as the independent variable, challenges arise with the ease and speed of altering acoustic conditions. This study compares two possibilities for testing different acoustic conditions. In this test, physically varied acoustic treatment is compared to binaural room simulation. Explorations of these two methods are conducted employing an in situ, task-based paradigm presented to highly trained listeners. Results indicate significant differences in acoustic conditions within binaural simulations; however do not provide corresponding data to actual acoustic alteration.
Engineering Brief 84 (Download now)
EB2-3 The Advantages of Using Active Crossovers in High-End Wireless Speakers—David Jones, CSR Limited - Manchester, UK
With the availability of standardized wireless interfaces and high performance codecs, wireless loudspeakers can be designed that suit the consumer demands of compactness and ease of use. This paper will examine the performance benefits of using active crossovers and digital equalization in an amplification subsystem based on a high performance digital input switching amplifier. Measurements of distortion and damping factors will be compared in an example signal chain and the influence these parameters have on the perceived audio quality of the speaker system will be discussed.
Engineering Brief 85 (Download now)
EB2-4 Comparative Analysis of Different Loudness Meters Based on Voice Detection and Gating—Alessandro Travaglini, Fox International Channels Italy - Guidonia Montecelio (RM), Italy
After decades of extensive investigation, the international broadcasting community, represented by technical associations and bodies, has set precise standards aimed to objectively assess loudness levels of programs. Although all standards rely on the same algorithm as described in ITU-R BS1770, there are still two possible ways to implement such metering, including voice detection and gating. These two different implementations might, in some cases, provide measurements that significantly differ from each other. Furthermore, while the gating feature is uniquely defined in the updated version of BS1770-3, voice detection is not currently specified in any standard and its implementation is the independent choice of manufacturers. This paper analyses this scenario by comparing the results and robustness provided by three different loudness meters based on voice detection. In addition, those values are compared with measurements obtained by using BS1770-3 compliant loudness meters, including tables, comments, and conclusions.
Engineering Brief 86 (Download now)
EB2-5 Assessing the Standardization of an Existing iOS Control Application to AES64-2012 Network Protocol—Joan Amate, Master Audio - Barcelona, Spain
The recent publication of AES64-2012 standard has motivated the comparison of Master Audio’s own IP network control protocol against the new standard, in order to assess its interoperability or adaptability. This brief analyzes what AES64 means for manufacturers with existing control protocols who are willing to seek standardization. The protocol used for this assessment was developed for controlling self-amplified PA systems (built-in amplifier and processing), and is fully functional under Windows and iOS (iPad). Finally, a brief guide on how to face standardization is given from the manufacturer point of view.
Engineering Brief 87 (Download now)
EB2-6 Innovation in Audio: Update on Patent Activity in the Audio Field—Elliot Cook, Finnegan, Henderson, Farabow, Garrett & Dunner, LLP - Reston, VA, USA; Joseph E. Palys, Finnegan, Henderson, Farabow, Garrett & Dunner, LLP - Reston, VA, USA
This paper provides a sampling of recent patents relating to innovations in the audio field. The innovations come from a range of AES member companies and cover a diverse spectrum of technologies, such as music composition software, loudspeaker design, headphones, digital signal processing, microphones, and musical comprehension. In addition, unique statistical information regarding patent litigation in the audio field is provided. This information is based on original research regarding litigation involving audio patents. AES members may find this information helpful to better understand how audio innovations play a role in their industry.
Engineering Brief 89 (Download now)
EB2-7 An Examination of Early Analog and Digital Sampling—The Robb Wave Organ circa 1927—Michael Murphy, Ryerson University - Toronto, ON, Canada; Eric Kupp, Ryerson University - Toronto, ON, Canada
This paper examines Frank Morse Robb's work in the late 1920s and early 1930s on his Wave Organ, the first successful electronic organ. The Robb Wave Organ originally functioned by creating a visual representation of an analog pipe organ waveform through means of an oscilloscope and engraving that representation onto metal tone wheels. Later versions of the organ featured a digital, almost PCM-style, waveform representation on the tone wheels. This predates the theoretical description of PCM by Alec Reeves, as well as the PCM patent filed by Oliver and Shannon in 1946. These sample-based methods of tone generation were unique to the Robb Wave Organ, and this paper serves to place the organ within its contemporaries of that time period, most notably its primary competitor, the Hammond organ, launched in 1935.
Engineering Brief 90 (Download now)
EB3 - E-Brief Papers—Part 2
Tuesday, May 7, 11:15 — 13:00 (Sala Foscolo)
Chair:
Brett Leonard, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
EB3-1 Spatial Sound Reinforcement Using Wave Field Synthesis—Etienne Corteel, Sonic Emotion Labs - Paris, France; Hubert Westkemper, Independent Tonmeister - Naples, Italy; Cornelius Ihssen, Sonic Emotion Labs - Oberglatt, Switzerland; Khoa-Van Nguyen, Sonic Emotion Labs - Paris, France
Spatial audio in sound reinforcement remains an open topic, requiring good level coverage and at the same time good localization accuracy over a very large listening area, typically the entire audience. Wave Field Synthesis offers high localization accuracy over an extended listening area but the number of required loudspeakers, their placement on stage, and the level coverage that results from it can be problematic. The paper addresses these issues, presenting a case study of a sound reinforcement system based on Wave Field Synthesis for sound reinforcement for the play "The Panic" written by Rafael Spregelburd and directed by Luca Ronconi. The paper describes an improved Wave Field Synthesis rendering for sound reinforcement involving two arrays of loudspeakers at different heights. The paper addresses the practical implementation of the system in a theater and the overall installation: miking, real time tracking of actors, and loudspeakers used.
Engineering Brief 91 (Download now)
EB3-2 Sync-AV–Workflow Tool for File-Based Video Shootings—Andreas Fitza, University of Applied Science Mainz - Mainz, Germany
The Sync-AV workflow tool eases the sorting and synchronization of video and audio footage without the need of expensive special hardware. It supports the preproduction, shooting, and postproduction. It consists of three elements: a script-information and metadata-gathering iOS app that is synchronized with a server-back-end and that can be used to exchange information on-set; a server database with a web-front-end that can sort files by their metadata and show dailies and that can be used to distribute and manage information during the pre-production; and a local import client that manages the footage ingest and sorts the files together. The client also takes care of the synchronization of the video that contains audio and separately recorded audio files and it renames the files and implements the metadata.
Engineering Brief 92 (Download now)
EB3-3 On the Optimum Microphone Array Configuration for Height Channels—Hyunkook Lee, University of Huddersfield - Huddersfield, UK; Christopher Gribben, University of Huddersfield - Huddersfield, West Yorkshire, UK
To date no experimental data have been presented on the optimum microphone array configuration for new surround formats employing height channels. A series of subjective listening tests were conducted to investigate how the spacing between base and height microphones affects perceived spatial impression and overall preference. Four different spacings of 0, 0.5, 1, and 1.5 m were compared for various sound sources using a 9-channel loudspeaker setup. For sources with more continuous temporal characteristics, the spacing between the layers did not have any significant effect on spatial impression, whereas for more transient sources the 0 m layer appeared to produce a greater spatial impression than more spaced layers. Furthermore, the 0 m layer was more or similarly preferred to the spaced layers depending on source type.
Engineering Brief 93 (Download now)
EB3-4 Free Improv—The Hard Way—Justin Paterson, London College of Music, University of West London - London, UK
As a fringe genre, “Free Improvisation” does not normally attract large production budgets. Often time-constrained, the subsequent technological approach to the production tends to emphasize the naturalistic and neglects many of the tools and techniques that are commonplace in contemporary popular music. The author produced the album, The Making of Quiet Things by The Number (featuring Keith Tippett). This album consciously employed a range of contemporary approaches such as creative and corrective automation, reverberation-matching, audio editing, and extreme compression, while maintaining an overall impression of minimal mediation. This paper considers and contextualizes such an approach, reflecting on the practice and its implications for the genre.
Engineering Brief 94 (Download now)
EB3-5 International Experiences from a New Sound System Approach—Thomas Lagö, QirraSound Technologies LLC - Las Vegas, NV, USA; Alan Boyer, QirraSound Technologies LLC - Las Vegas, NV, USA
QirraSound's new sound system approach benefits from a high level of intelligibility and substantially lower feedback. These properties help in placing loudspeakers behind the performers and thus minimizing the need for monitor speakers. Substantial empiric testing has been done in applications in multiple countries and applications and results from these tests will be presented. Listeners and performers report increased feel and intelligibility and even people with hearing loss and/or sensitivity to high sound levels can enjoy the sound. It has also been noticed that the ability to talk while music is playing is much better than with classical systems. An overall outline of these test results and feedback will be reported.
Engineering Brief 95 (Download now)
EB3-6 Nonlinear Guitar Loudspeaker Simulation—Thomas Schmitz, University of Liege - Liege, Belgium; Jean J. Embrechts, University of Liege - Liege, Belgium
In this study we simulated in real time the sound of a guitar amplifier loudspeaker, including its non-linear behavior. The simulation method is based on a non-linear convolution of the signal emitted by the instrument with the Volterra kernels, which were measured in anechoic conditions with a sine-sweep technique. The model has been implemented in a "VST" (Virtual Studio Technology) audio plugin. The loudspeaker simulation can be performed in real time with the Volterra kernels up to the third order and offers a good accuracy. Informal tests revealed that the simulated and the real sound were very close, although approximately 50 percent of the tested musicians were still able to hear a small difference.
Engineering Brief 96 (Download now)
EB3-7 Using Low-Latency Net-Based Solutions to Extend the Audio and Video Capabilities of a Studio Complex—Paul Ferguson, Edinburgh Napier University - Edinburgh, UK
Two low-latency IP-based systems, RedNet and LOLA, were selected by Edinburgh Napier University to link a new music building with existing broadcast and drama facilities and to allow low-latency audio and video collaboration with other institutions/organizations around the world. First to be examined will be their use of Focusrite RedNet and Audinate Dante to expand existing AES10 (MADI) point to point links. Second, an overview will be provided of the University's ongoing research with JANET and GARR (the UK and Italy National Research and Education Networks) into the use of the Italian LOLA system (LOw LAtency audio visual streaming system) to provide long-distance audio and video links for rehearsal and performance involving musicians in different countries.
Engineering Brief 97 (Download now)
EB4 - E-Brief Posters—Part 2
Tuesday, May 7, 14:30 — 16:00 (Foyer)
EB4-1 Spatial Acoustic Synthesis—Celambarasan Ramasamy, Mind Theatre - Pasadena, CA, USA
This method leverages binaural sound synthesis to present a novel way for listeners to experience musical performances on a headphone. By separating out the notes from a musical instrument and synthesizing them individually in the space around the listener, the musical instrument is turned from a point source into a flowing musical volume that engulfs the listener. This can lead to interesting ways of thinking about a musical performance in terms of the distribution of the individual musical notes around the listener without being tied down to the notion of an individual musical instrument.
Engineering Brief 98 (Download now)
EB4-2 Enhancing the Learning of Stereo Microphone Techniques through the Use of a Simulated Learning Environment—Colin Dodds, Perth College, University of the Highlands and Islands - Perth, Scotland, UK
A key set of skills for aspiring recording engineers to acquire is that of good stereo microphone techniques. Within the world of education it is relatively straightforward to present the underpinning theory, but helping students gain the tacit knowledge necessary to achieve quality results can be difficult when access to suitable groups of musicians, spaces, and equipment is limited. A suitable learning environment was simulated within a computer program and tasks set to support the learning of stereo microphone techniques. A trial carried out on first year undergraduate sound production students revealed that using the simulated learning environment enhanced both the students’ knowledge of and ability to apply stereo microphone techniques.
Engineering Brief 99 (Download now)
EB4-3 The Good Vibrations Problem—Derry FitzGerald, Dublin Institute of Technology - Dublin, Ireland
Many of the Beach Boys’ records were mono only as this was Brian Wilson's preferred format. However, starting in the mid-1990s, stereo mixes of many of these classics were created by synchronizing the tracks from the instrumental multitrack with those of the vocal multitrack. Unfortunately, for a number of tracks, including “Good Vibrations,” elements of the multitracks were missing, making a true stereo mix impossible. This paper deals with how stereo extraction mixes were created for a number of Beach Boys’ songs using sound source separation techniques to separate sources from the original mono recordings, which were then panned to create stereo mixes. These mixes were used in reissues of Beach Boys albums in 2012.
Engineering Brief 100 (Download now)
EB4-4 Architectural Acoustics and Electroacoustics in the Asisium Theatre: An Integrated Construction Work—Luca Quaranta, CP Progetti S.r.l. - Rome, Italy
Based on the multi-functional needs, from the conference room to the cinema with Dolby Surround sound effects, we performed an analysis of the room’s acoustic response based on the model of a 3-D simulation, referenced on acoustic parameters according to ISO-3382, thus directing the choices of architectural design to obtain optimum acoustical conditions both on stage and in the audience. Design of the electroacoustic provides various control technologies for the equalization and a 6.1 Dolby Digital distribution of the amplified audio signal. In this contribution we present choices and design criteria, materials and audio devices used to obtain the final result, to underline the key role of architectural acoustics and electroacoustic integrated design.
Engineering Brief 101 (Download now)
EB4-5 Expressive Physical Modeling of Keyboard Instruments: From Theory to Implementation—Stefano Zambon, Viscount International S.p.A. - Mondaino (RN), Italy; Leonardo Gabrielli, Universitá Politecnica delle Marche - Ancona, Italy; Balazs Bank, Budapest University of Technology and Economics - Budapest, Hungary
Physics-based algorithms for sound synthesis have been extensively studied in the past decades. Nevertheless, their use in commercial synthesizers is still limited due to the difficulty in achieving realistic and easily controllable sounds with current technology. In this Engineering Brief we present an overview of the models used in Physis Piano, a digital piano recently introduced in the market with dedicated physics-based algorithms for acoustic pianos, electric pianos (e.g., Rhodes, Wurlitzer, and Clavinet), and chromatic percussions (e.g. vibraphone, marimba, xylophone). The synthesis algorithms, which are based on standard techniques such as Modal Synthesis and Digital Waveguides, have been highly customized in order to faithfully reproduce the sound features of the original instruments and are easily controllable by a set of meaningful, user-friendly parameters.
Engineering Brief 102 (Download now)
