AES San Francisco 2008
Live Sound Event Details
Wednesday, October 1, 9:00 am — 4:00 pm
Live Sound Symposium: Surround Live VI—Acquiring the Surround Field
The Lodge Ballroom, Regency Center
1290 Sutter St.
San Francisco, CA
Tel. +1 415 673 5716
Building from the five previous, highly successful Surround Live symposia, Surround Live Six, will once again explore in detail, the world of Live Surround Audio.
Frederick Ampel, President of consultancy Technology Visions, in cooperation with the Audio Engineering Society, brings this years event back to San Francisco for the third time. The event will feature a wide range of professionals from both the televised Sports arena, Public Radio, and the digital processing and encoding sciences.
Surround Live Six Platinum Sponsors are: Neural Audio and Sennheiser/K&H. Surround Live Six Gold Sponsor is: Ti-Max/Outboard Electronics
8:15am - 9:00 am – Coffee, Registration, Continental Breakfast
9:00 am – Keynote #1 – Kurt Graffy
9:40 am – Keynote #2 – J. Johnston
10:15 am - 10:25 am – Coffee Break
10:30 am - 12:30 pm – Presenters 1, 2, & 3 plus Live Demonstrations and Demo Video Clips with Surround Audio
12:30 pm - 1:00 pm – Lunch (provided for Ticketed Participants)
1:00 pm - 3:00 pm – Presenters 4, 5, & 6 with Live Demonstrations and Clips
3:00 pm - 3:15 pm – Break
3:15 pm - 4:30 pm – Panel Discussion and Interactive Q&A
4:30 pm - 5:00 pm – Organ Concert (Pending availability of Organist) featuring the 1909 Austin Pipe Organ
Scheduled to appear are:
•Fred Aldous - FOX Sports Audio Consultant / Sr. Mixer
•Tom Sahara - Sr. Director of Remote Operations & IT Turner Sports
•Mike Pappas – KUVO Radio – Denver
•Kurt Graffy – ARUP Acoustics – San Francisco –Co-Keynote
•James D. (JJ) Johnston - Chief Scientist, Neural Audio, Kirkland, WA.
•Jim Hilson – Dolby Laboratories – San Francisco, CA.
•Other possible presenters include Speed Network, NFL Films, and NPR.
The day’s events will include formal presentations, special demonstration materials in full surround, and interactive discussions with presenters. Seating is limited, and previous events have sold out quickly. Register quickly to insure you will be able to attend.
Further details will be added as they become available
Thursday, October 2, 9:00 am — 10:45 am
T1 - Electroacoustic Measurements
:Christopher J. Struck
, CJS Labs - San Francisco, CA, USA
This tutorial focuses on applications of electroacoustic measurement methods, instrumentation, and data interpretation as well as practical information on how to perform appropriate tests. Linear system analysis and alternative measurement methods are examined. The topic of simulated free field measurements is treated in detail. Nonlinearity and distortion measurements and causes are described. Last, a number of advanced tests are introduced.
This tutorial is intended to enable the participants to perform accurate audio and electroacoustic tests and provide them with the necessary tools to understand and correctly interpret the results.
Thursday, October 2, 9:00 am — 12:30 pm
P2 - Analysis and Synthesis of Sound
: Hiroko Terasawa
, Stanford University - Stanford, CA, USAP2-1 Spatialized Additive Synthesis of Environmental Sounds
, Orange Labs - Lannion, France, and Laboratoire de Mécanique et d’Acoustique, Marseille, France; Mitsuko Aramaki
, Institut de Neurosciences Cognitives de la Méditerranée - Marseille, France; Richard Kronland-Martinet
, Laboratoire de Mécanique et d’Acoustique - Marsielle, France; Grégory Pallone
, Orange Labs - Lannion, France
In virtual auditory environment, sound sources are typically created in two stages: the “dry” monophonic signal is synthesized, and then, the spatial attributes (like source directivity, width, and position) are applied by specific signal processing algorithms. In this paper we present an architecture that combines additive sound synthesis and 3-D positional audio at the same level of sound generation. Our algorithm is based on inverse fast Fourier transform synthesis and amplitude-based sound positioning. It allows synthesizing and spatializing efficiently sinusoids and colored noise, to simulate point-like and extended sound sources. The audio rendering can be adapted to any reproduction system (headphones, stereo, 5.1, etc.). Possibilities offered by the algorithm are illustrated with environmental sound.
Convention Paper 7509 (Purchase now)P2-2 Harmonic Sinusoidal + Noise Modeling of Audio Based on Multiple F0 Estimation
—Maciej Bartkowiak, Tomasz Zernicki
, Poznan University of Technology - Poznan, Poland
This paper deals with the detection and tracking of multiple harmonic series. We consider a bootstrap approach based on prior estimation of F0 candidates and subsequent iterative adjustment of a harmonic sieve with simultaneous refinement of the F0 and inharmonicity factor. Experiments show that this simple approach is an interesting alternative to popular strategies, where partials are detected without harmonic constraints, and harmonic series are resolved from mixed sets afterwards. The most important advantage is that common problems of tonal/noise energy confusion in case of unconstrained peak detection are avoided. Moreover, we employ a popular LP-based tracking method that is generalized to dealing with harmonically related groups of partials by using a vector inner product as the prediction error measure. Two alternative extensions of the harmonic model are also proposed in the paper that result in greater naturalness of the reconstructed audio: an individual frequency deviation component and a complex narrowband individual amplitude envelope.
Convention Paper 7510 (Purchase now)P2-3 Sound Extraction of Delackered Records
—Ottar Johnsen Frédéric Bapst
, Ecole d'ingenieurs et d'architectes de Fribourg - Fribourg, Switzerland; Lionel Seydoux
, Connectis AG - Berne, Switzerand
Most direct cut records are made of an aluminum or glass plate with a coated acetate lacquer. Such records are often crackled due to the shrinkage of the coating. It is impossible to read such records mechanically. We are presenting here a technique to reconstruct the sound from such record by scanning the image of the record and combining the sound from the different parts of the "puzzle." The system has been tested by extracting sounds from sound archives in Switzerland and in Austria. The concepts will be presented as well as the main challenges. Extracted sound samples will be played.
Convention Paper 7511 (Purchase now)P2-4 Parametric Interpolation of Gaps in Audio Signals
, Moscow State University - Moscow, Russia; Jeremy Todd
, iZotope, Inc. - Cambridge, MA, USA
The problem of interpolation of gaps in audio signals is important for the restoration of degraded recordings. Following the parametric approach over a sinusoidal model recently suggested in JAES by Lagrange et al., this paper proposes an extension to this interpolation algorithm by considering interpolation of a noisy component in a “sinusoidal + noise” signal model. Additionally, a new interpolator for sinusoidal components is presented and evaluated. The new interpolation algorithm becomes suitable for a wider range of audio recordings than just interpolation of a sinusoidal signal component.
Convention Paper 7512 (Purchase now)P2-5 Classification of Musical Genres Using Audio Waveform Descriptors in MPEG-7
, Microsoft Corporation - Seattle, WA, USA
Automated genre classification makes it possible to determine the musical genre of an incoming audio waveform. One application of this is to help listeners find music they like more quickly among millions of tracks in an online music store. By using numerical thresholds and the MPEG-7 descriptors, a computer can analyze the audio stream for occurrences of specific sound events such as kick drum, snare hit, and guitar strum. The knowledge about sound events provides a basis for the implementation of a digital music genre classifier. The classifier inputs a new audio file, extracts salient features, and makes a decision about the musical genre based on the decision rule. The final classification results show a recognition rate in the range 75% to 94% for five genres of music.
Convention Paper 7513 (Purchase now)P2-6 Loudness Descriptors to Characterize Programs and Music Tracks
, TC Group Research - Risskov, Denmark; Thomas Lund
, TC Electronic - Risskov, Denmark
We present a set of key numbers to summarize loudness properties of an audio segment, broadcast program, or music track: the loudness descriptors. The computation of these descriptors is based on a measurement of loudness level, such as specified by the ITU-R BS.1770. Two fundamental loudness descriptors are introduced: Center of Gravity and Consistency. These two descriptors were computed for a collection of audio segments from various sources, media, and formats. This evaluation demonstrates that the descriptors can robustly characterize essential properties of the segments. We propose three different applications of the descriptors: for diagnosing potential loudness problems in ingest material; as a means for performing a quality check, after processing/editing; or for use in a delivery specification.
Convention Paper 7514 (Purchase now)P2-7 Methods for Identification of Tuning System in Audio Musical Signals
—Peyman Heydarian, Lewis Jones, Allan Seago
, London Metropolitan University - London, UK
The tuning system is an important aspect of a piece. It specifies the scale intervals and is an indicator of the emotions of a musical file. There is a direct relationship between musical mode and the tuning of a piece for modal musical traditions. So, the tuning system carries valuable information, which is worth incorporating into metadata of a file. In this paper different algorithms for automatic identification of the tuning system are presented and compared. In the training process, spectral and chroma average, and pitch histograms, are used to construct reference patterns for each class. The same is done for the testing samples and a similarity measure like the Manhattan distance classifies a piece into different tuning classes.Convention Paper 7515 (Purchase now)P2-8 “Roughometer”: Realtime Roughness Calculation and Profiling
—Julian Villegas, Michael Cohen
, University of Aizu - Aizu-Wakamatsu, Fukushima-ken, Japan
A software tool capable of determining auditory roughness in real-time is presented. This application, based on Pure-Data (Pd), calculates the roughness of audio streams using a spectral method originally proposed by Vassilakis. The processing speed is adequate for many real-time applications and results indicate limited but significant agreement with an Internet application of the chosen model. Finally, the usage of this tool is illustrated by the computation of a roughness profile of a musical composition that can be compared to its perceived patterns of “tension” and “relaxation.”Convention Paper 7516 (Purchase now)
Thursday, October 2, 10:30 am — 1:00 pm
L1 - Sound Reinforcement of Acoustic Music
, Audio Systems GroupMark FrinkDan Mortensen
, Dansound Inc.Jim van Bergen
Amplifying acoustic music is a touchy subject, especially with musicians. It can be done, and it can be done well. Taste, subtlety, and restraint are the keywords. This live sound event brings four successful practitioners of the art with a discussion of what can make you successful, and what won't. There is one thing for sure: it's not rock-n-roll.
Thursday, October 2, 2:30 pm — 4:30 pm
L2 - The SOTA of Designing Loudspeakers for Live Sound
, Electroacoustic Design ServicesPanelists
, Danley Sound LabsAles Dravinec
, ADRaudioDave Gunness
, Fulcrum AcousticCharlie Hughes
, Excelsior Audio DesignPete Soper
, Meyer Sound
The loudspeakers we employ today for live sound (all levels, all types) are vastly improved over what we had on hand when R&R first exploded and pushed the limits of what was available back in the 1960s. Following a brief glimpse back in time (to provide a reality check on where we were when many of us started in this field) we will define where we are now. Along with advances made in enclosure design and fabrication, horn design, driver design, system engineering and fabrication, ergonomics and rigging, etc., we now are implementing various methods to improve the overall performance of the drivers and the loudspeaker systems we use, not to mention the advanced methods employed to optimize large systems, improve directivity, beam-steer, etc.
Much of this advancement, at least over the past 15 years or so, is directly related to our use of computers as a design tool for modeling, for complex measurements (both in the lab and in the field) as well as DSP for implementing various processing and monitoring functions. We will clarify what we can do with modern day loudspeakers/systems and where we still need to push further. We may even get our panelists to imagine where they believe we may be headed over the next 5–10 years.
Thursday, October 2, 2:30 pm — 4:30 pm
P4 - Acoustic Modeling and Simulation
: Scott Norcross
, Communications Research Centre - Ottawa, Ontario, CanadaP4-1 Application of Multichannel Impulse Response Measurement to Automotive Audio
, Fraunhofer Institute for Digital Media Technology - Ilmenau, Germany, and Technical University of Delft, Delft, The Netherlands; Diemer de Vries
, Technical University of Delft - Delft, The Netherlands
Audio reproduction in small enclosures holds a couple of differences in comparison to conventional room acoustics. Today’s car audio systems meet sophisticated expectations but still the automotive listening environment delivers critical acoustic properties. During the design of such an audio system it is helpful to gain insight into the temporal and spatial distribution of the acoustic field's properties. Because room acoustic modeling software reaches its limits the use of acoustic imaging methods can be seen as a promising approach. This paper describes the application of wave field analysis based on a multichannel impulse response measurement in an automotive use case. Besides a suitable preparation of the theoretical aspects, the analysis method is used to investigate the acoustic wave field inside a car cabin.
Convention Paper 7521 (Purchase now)P4-2 Multichannel Low Frequency Room Simulation with Properly Modeled Source Terms—Multiple Equalization Comparison
—Ryan J. Matheson
, University of Waterloo - Waterloo, Ontario, Canada
At low frequencies unwanted room resonances in regular-sized rectangular listening rooms cause problems. Various methods for reducing these resonances are available including some multichannel methods. Thus with introduction of setups like 5.1 surround into home theater systems there are now more options available to perform active resonance control using the existing loudspeaker array. We focus primarily on comparing, separately, each step of loudspeaker placement and its effects on the response in the room as well as the effect of adding additional symmetrically placed loudspeakers in the rear to cancel out any additional room resonances. The comparison is done by use of a Finite Difference Time Domain (FDTD) simulator with focus on properly modeling a source in the simulation. A discussion about the ability of a standard 5.1 setup to utilize a multichannel equalization technique (without adding additional loudspeakers to the setup) and a modal equalization technique is later discussed.
Convention Paper 7522 (Purchase now)P4-3 A Super-Wide-Range Microphone with Cardioid Directivity
—Kazuho Ono, Takehiro Sugimoto, Akio Ando
, NHK Science and Technical Research Laboratories - Tokyo, Japan; Tomohiro Nomura, Yutaka Chiba, Keishi Imanaga
, Sanken Microphone Co. Ltd. - Japan
This paper describes a super-wide-range microphone with cardioid directivity, which covers the frequency range up to 100 kHz. The authors have successfully developed the omni-directional microphone capable of picking up sounds of up to 100 kHz with low noise. The proposed microphone uses an omni-directional capsule adopted in the omni-directional super-wide-range microphone and a bi-directional capsule that is newly designed to fit the characteristics of the omni-directional one. The output signals of both capsules are synthesized as the output signals to achieve cardioid directivity. The measurement results show that the proposed microphone achieves wide frequency range up to 100 kHz, as well as low noise characteristics and excellent cardioid directivity.
Convention Paper 7523 (Purchase now)P4-4 Methods and Limitations of Line Source Simulation
, Ahnert Feistel Media Group - Berlin, Germany; Ambrose Thompson
, Martin Audio - High Wycombe, Bucks, UK; Wolfgang Ahnert
, Ahnert Feistel Media Group - Berlin, Germany
Although line array systems are in widespread use today, investigations of the requirements and methods for accurate modeling of line sources are scarce. In previous publications the concept of the Generic Loudspeaker Library (GLL) was introduced. We show that on the basis of directional elementary sources with complex directivity data finite line sources can be simulated in a simple, general, and precise manner. We derive measurement requirements and discuss the limitations of this model. Additionally, we present a second step of refinement, namely the use of different directivity data for cabinets of identical type based on their position in the array. All models are validated by measurements. We compare the approach presented with other proposed solutions.
Convention Paper 7524 (Purchase now)
Thursday, October 2, 5:00 pm — 6:45 pm
M1 - Basic Acoustics: Understanding the Loudspeaker
, University of Waterloo - Waterloo, Ontario, Canada
This presentation is for AES members at an intermediate level and introduces many concepts in acoustics. The basic propagation of sound waves in air for both plane and spherical waves is developed and applied to the operation of a simple, sealed-box loudspeaker. Topics such as the acoustic impedance, compact source operation, and diffraction are included. Some live demonstrations with a simple loudspeaker; microphone and measuring computer are used to illustrate the basic radiation principle of a typical electrodynamic driver mounted in a sealed box.
Thursday, October 2, 5:00 pm — 6:45 pm
L3 - AC Power and Grounding
:Bruce C. Olson
, Olson Sound Design - Minneapolis, MN, USAPanelists
:David StevensBill Whitlock
, Jensen Transformers - Chatsworth, CA, USA
How do you kill the hum without killing yourself? This panel will discuss how to provide AC power properly, avoid hum and not kill the performers, technicians, or yourself. A lot of the advice out there isn’t just wrong, it is potentially fatal. However, being safe is easy. The only question is, why doesn’t everyone know this! We will also discuss the use of generator sets, the myths and facts about grounding, and typical configurations.
Thursday, October 2, 5:00 pm — 6:45 pm
W5 - Engineering Mistakes We Have Made in Audio
, Oxford Digital Limited - UKPanelists
, Audio ImaginationJames D. (JJ) Johnston
, Neural Audio Corp.Mel Lambert
, Media & MarketingGeorge Massenburg
, Massenburg Design WorksJim McTigue
, Impulsive Audio
Six leading audio product developers will share the enlightening, thought-provoking, and (in retrospect) amusing lessons they have learned from actual mistakes they have made in the product development trenches.
Friday, October 3, 9:00 am — 12:00 pm
TT3 - Center for New Music and Audio Technology, UC Berkeley
The UC Berkeley Center for New Music and Audio Technologies (CNMAT) houses programs in research, pedagogy, and public performance that are focused on the creative interaction between music and technology. CNMAT's pedagogy program is highly integrated with the Department of Music's graduate program in composition, while the research program is linked with other disciplines and departments on campus such as architecture, mathematics, statistics, mechanical engineering, computer science, electrical engineering, psychology, cognitive science, physics, space sciences, the Center for New Media, and the Department of Theater, Dance, and Performance Studies. Presenters David Wessel (Co-Director, CNMAT) and Adrian Freed (Research Director, CNMAT) will give an overview of CNMAT research projects. For more information, visit http://cnmat.berkeley.edu/.
All visitors are required to sign a Non-Disclosure Agreement to enter the facility.
Maximum of 47 participants per tour.
$35 (members), $45 (nonmembers)
Friday, October 3, 9:00 am — 11:30 am
T4 - Perceptual Audio Evaluation
, Bang & Olufsen A/S - Struer, DenmarkNick Zacharov
, SenseLab - Delta, Denmark
The aim of this tutorial is to provide an overview of perceptual evaluation of audio through listening tests, based on good practices in the audio and affiliated industries. The tutorial is aimed at anyone interested in the evaluation of audio quality and will provide a highly condensed overview of all aspects of performing listening tests in a robust manner. Topics will include: (1) definition of a suitable research question and associated hypothesis, (2) definition of the question to be answered by the subject, (3) scaling of the subjective response, (4) control of experimental variables such as choice of signal, reproduction system, listening room, and selection of test subjects, (5) statistical planning of the experiments, and (6) statistical analysis of the subjective responses. The tutorial will include both theory and practical examples including discussion of the recommendations of relevant international standards (IEC, ITU, ISO). The presentation will be made available to attendees and an extended version will be available in the form of the text “Perceptual Audio Evaluation" authored by Søren Bech and Nick Zacharov.
Friday, October 3, 9:00 am — 10:45 am
L4 - White Space Issues
, Shure Incorporated - NiPanelists
, Production Radio Rentals - Yonkers, NY, USA
The DTV conversion will be complete on February 17, 2009. The impact of this and surrounding FCC decisions is of great concern to wireless microphone users. Will 700 MHz band mics retain type certification? Will proposed white space devices create new interference? Will there be an FCC crack-down on unlicensed microphone use? This panel will discuss the latest FCC rule decisions and decisions still pending.
Friday, October 3, 9:00 am — 11:30 am
P5 - Audio Equipment and Measurements
: John Vanderkooy
, University of Waterloo - Waterloo, Ontario, CanadaP5-1 Can One Perform Quasi-Anechoic Loudspeaker Measurements in Normal Rooms?
—John Vanderkooy, Stanley Lipshitz
, University of Waterloo - Waterloo, Ontario, Canada
This paper is an analysis of two methods that attempt to achieve high resolution frequency responses at low frequencies from measurements made in normal rooms. Such data is contaminated by reflections before the low-frequency impulse response of the system has fully decayed. By modifying the responses to decay more rapidly, then windowing a reflection-free portion, and finally recovering the full response by deconvolution, these quasi-anechoic methods purport to thwart the usual reciprocal uncertainty relationship between measurement duration and frequency resolution. One method works by equalizing the response down to dc, the other by increasing the effective highpass corner frequency of the system. Each method is studied with simulations, and both appear to work to varying degrees, but we question whether they are measurements or effectively simply model extensions. In practice noise significantly degrades both procedures.
Convention Paper 7525 (Purchase now)P5-2 Automatic Verification of Large Sound Reinforcement Systems Using Models of Loudspeaker Performance Data
—Klas Dalbjörn, Johan Berg
, Lab.gruppen AB - Kungsbacka, Sweden
A method is described to automatically verify individual loudspeaker integrity and confirm the proper configuration of amplifier-loudspeaker connections in sound reinforcement systems. Using impedance-sensing technology in conjunction with software-based loudspeaker performance modeling, the procedure verifies that the load presented at each amplifier output corresponds to impedance characteristics as described in the DSP system’s currently loaded model. Accurate verification requires use of load impedance models created by iterative testing of numerous loudspeakers.
Convention Paper 7526 (Purchase now)P5-3 Bend Radius
—Stephen Lampen, Carl Dole, Shulamite Wan
, Belden - San Francisco, CA, USA
Designers, installers, and system integrators, have many rules and guidelines to follow. Most of these are intended to maximize cable and equipment performance. Many of these are “rules-of-thumb,” simple guidelines, easy to remember, and often just as easily broken. One of these is the “rule-of-thumb” regarding the bending of cable, especially coaxial cable. Many may have heard the term “No tighter than ten times the diameter.” While this can be helpful, in a general way, there is a deeper and more complex question. What happens when you do bend cable? What if you have no choice? Often a specific choice of rack or configuration of equipment requires that cables be bent tighter than that recommendation. And what happens if you “unbend” a cable that has been damaged? Does it stay damaged or can it be restored? This paper outlines a series of laboratory tests to determine exactly what happens when cable is bent and what the reaction is. Further, we will analyze the effect of bending on cable performance, specifically looking at impedance variations and return loss (signal reflection). For high-definition video signals (HD-SDI) return loss is the key to maximum cable length, bit errors, and open eye patterns. So analyzing the effecting of bending will allow us to determine signal quality based on the bending of an individual cable. But does this apply to digital audio cables? Does the relatively low frequencies of AES digital signals make a difference? Can these cables be bent with less effect on performance? These tests were repeated on both coaxial cable of different sizes and twisted pairs. Flexible coax cables were tested, as well as the standard solid-core installation versions. Paired cables consisted of AES digital audio shielded cables, both install and flexible versions, were also tested.
Convention Paper 7527 (Purchase now)P5-4 Detecting Changes in Audio Signals by Digital Differencing
, Liberty Instruments Inc. - Liberty Township, OH, USA
A software application has been developed to provide an accessible method, based on signal subtraction, to determine whether or not an audio signal may have been perceptibly changed by passing through components, cables, or similar processes or treatments. The goals of the program, the capabilities required of it, its effectiveness, and the algorithms it uses are described. The program is made freely available to any interested users for use in such tests.
Convention Paper 7528 (Purchase now)P5-5 Research on a Measuring Method of Headphones and Earphones Using HATS
—Kiyofumi Inanaga, Takeshi Hara
, Sony Corporation - Tokyo, Japan; Gunnar Rasmussen
, G.R.A.S. Sound & Vibration A/S - Copenhagen, Denmark; Yasuhiro Riko
, Riko Associates - Tokyo, Japan
Currently various types of couplers are used for measurement of headphones and earphones. The coupler was selected according to the device under test by the measurer. Accordingly it was difficult to compare the characteristics of headphones and earphones. A measuring method was proposed using HATS and a simulated program signal. However, the method had some problems in the shape of ear hole, and the measured results were not reproducible. We tried to improve the reproducibility of the measurement using several pinna models. As a result, we achieved a measuring platform using HATS, which gives good reproducibility of measured results for various types of headphones and earphones and then makes it possible to compare the measured results fairly.
Convention Paper 7529 (Purchase now)
Friday, October 3, 9:00 am — 1:00 pm
P6 - Loudspeaker Design
: Alexander Voishvillo
, JBL Professional - Northridge, CA, USAP6-1 Loudspeaker Production Variance
, Equity Sound Investments - Bloomington, IN, USA; Laurie Fincham
, THX Ltd. - San Rafael, CA, USA
Numerous quality assurance philosophies have evolved over the last few decades designed to manage manufacturing quality. Managing quality control of production loudspeakers is particularly challenging. Variation of subcomponents and assembly processes across loudspeaker driver production batches may lead to excessive variation of sensitivity, bandwidth, frequency response, and distortion characteristics, etc. As loudspeaker drivers are integrated into production audio systems these variants result in broad performance permutation from system to system that affects all aspects of acoustic balance and spatial attributes. This paper will discuss traditional electro-dynamic loudspeaker production variation.
Convention Paper 7530 (Purchase now)P6-2 Distributed Mechanical Parameters Describing Vibration and Sound Radiation of Loudspeaker Drive Units
, University of Technology Dresden - Dresden, Germany; Joachim Schlechter
, KLIPPEL GmbH - Dresden, Germany
—Wolfgang Klippel, University of Dresden, Dresden, Germany; Joachim Schlechter, Klippel GmbH, Dresden, Germany
The mechanical vibration of loudspeaker drive units is described by a set of linear transfer functions and geometrical data that are measured at selected points on the surface of the radiator (cone, dome, diaphragm, piston, panel) by using a scanning technique. These distributed parameters supplement the lumped parameters (T/S, nonlinear, thermal), simplify the communication between cone, driver, and loudspeaker system design and open new ways for loudspeaker diagnostics. The distributed vibration can be summarized to a new quantity called accumulated acceleration level
(AAL), which is comparable with the sound pressure level (SPL) if no acoustical cancellation occurs. This and other derived parameters are the basis for modal analysis and novel decomposition techniques that make the relationship between mechanical vibration and sound pressure output more transparent. Practical problems and indications for practical improvements are discussed for various example drivers. Finally, the usage of the distributed parameters within finite and boundary element analyses is addressed and conclusions for the loudspeaker design process are made.
Convention Paper 7531 (Purchase now)P6-3 A New Methodology for the Acoustic Design of Compression Driver Phase-Plugs with Radial Channels
, Celestion International Ltd. - Ipswich, UK,and GP Acousics (UK) Ltd., Maidstone, UK; Jack Oclee-Brown
, GP Acousics (UK) Ltd. - Maidstone, UK, and University of Southampton, Southampton, UK
Recent work by the authors describes an improved methodology for the design of annular-channel, dome compression drivers. Although not so popular, radial channel phase plugs are used in some commercial designs. While there has been some limited investigation into the behavior of this kind of compression driver, the literature is much more extensive for annular types. In particular, the modern approach to compression driver design, based on a modal description of the compression cavity, as first pioneered by Smith, has no equivalent for radial designs. In this paper we first consider if a similar approach is relevant to radial-channel phase plug designs. The acoustical behavior of a radial-channel compression driver is analytically examined in order to derive a geometric condition that ensures minimal excitation of the compression cavity modes.
Convention Paper 7532 (Purchase now)P6-4 Mechanical Properties of Ferrofluids in Loudspeakers
—Guy Lemarquand, Romain Ravaud, Valerie Lemarquand, Claude Depollier
, Laboratoire d’Acoustique de l’Université du Maine - Le Mans, France
This paper describes the properties of ferrofluid seals in ironless electrodynamic loudspeakers. The motor consists of several outer stacked ring permanent magnets. The inner moving part is a piston. In addition, two ferrofluid seals are used that replace the classic suspension. Indeed, these seals fulfill several functions. First, they ensure the airtightness between the loudspeaker faces. Second, they act as bearings and center the moving part. Finally, the ferrofluid seals also exert a pull back force on the moving piston. Both radial and axial forces exerted on the piston are calculated thanks to analytical formulations. Furthermore, the shape of the seal is discussed as well as the optimal quantity of ferrofluid. The seal capacity is also calculated.
Convention Paper 7533 (Purchase now)P6-5 An Ironless Low Frequency Subwoofer Functioning under its Resonance Frequency
, Université du Maine - Le Mans, France, Orkidia Audio, Saint Jean de Luz, France; Guy Lemarquand
, Université du Maine - Le Mans, France; Bernard Nemoff
, Orkidia Audio - Saint Jean de Luz, France
A low frequency loudspeaker (10 Hz to 100 Hz) is described. Its structure is totally ironless in order to avoid nonlinear effects due to the presence of iron. The large diaphragm and the high force factor of the loudspeaker lead to its high efficiency. Efforts have been made for reducing the nonlinearities of the loudspeaker for a more accurate sound reproduction. In particular we have developed a motor totally made of permanent magnets, which create a uniform induction across the entire intended displacement of the coil. The motor linearity and the high force factor of this flat loudspeaker make it possible to function under its resonance frequency with great accuracy.
Convention Paper 7534 (Purchase now)P6-6 Line Arrays with Controllable Directional Characteristics—Theory and Practice
—Laurie Fincham, Peter Brown
, THX Ltd. - San Rafael, CA, USA
A so-called arc line array is capable of providing directivity control. Applying simple amplitude shading can, in theory, provide good off-axis lobe suppression and constant directivity over a frequency range determined at low-frequencies by line length and at high-frequencies by driver spacing. Array transducer design presents additional challenges–the dual requirements of close spacing, for accurate high-frequency control, and a large effective radiating area, for good bass output, are incompatible with the use of multiple full-range drivers. A novel drive unit layout is proposed and theoretical and practical design criteria are presented for a two-way line with controllable directivity and virtual elimination of spatial aliasing. The PC-based array controller permits real-time changes in beam parameters for multiple overlaid beams.
Convention Paper 7535 (Purchase now)P6-7 Loudspeaker Directivity Improvement Using Low Pass and All Pass Filters
, Excelsior Audio Design & Services, LLC - Gastonia, NC, USA
The response of loudspeaker systems employing multiple drivers within the same pass band is often less than ideal. This is due to the physical separation of the drivers and their lack of proper acoustical coupling within the higher frequency region of their use. The resultant comb filtering is sometimes addressed by applying a low pass filter to one or more of the drivers within the pass band. This can cause asymmetries in the directivity response of the loudspeaker system. A method is presented to greatly minimize these asymmetries through the use of low pass and all pass filters. This method is also applicable as a means to extend the directivity control of a loudspeaker system to lower frequencies.
Convention Paper 7536 (Purchase now)P6-8 On the Necessary Delay for the Design of Causal and Stable Recursive Inverse Filters for Loudspeaker Equalization
—Avelino Marques, Diamantino Freitas
, Polytechnic Institute of Porto - Porto, Portugal
The authors have developed and applied a novel approach to the equalization of non-minimum phase loudspeaker systems, based on the design of Infinite Impulse Response (recursive) inverse filters. In this paper the results and improvements attained on this novel IIR filter design method are presented. Special attention has been given to the delay of the equalized system. The boundaries to be posed on the search space of the delay for a causal and stable inverse filter, to be used in the nonlinear least squares minimization routine, are studied, identified, and related with the phase response of a test system and with the order of the inverse filter. Finally, these observations and relations are extended and applied to multi-way loudspeaker systems, demonstrating the connection of the lower and upper bounds of the delay with the loudspeaker’s crossover filters phase response and inverse filter order.
Convention Paper 7537 (Purchase now)
Friday, October 3, 9:00 am — 10:30 am
W6 - Audio Networking for the Pros
, ZP Engineering srlPanelists
, Peavey Digital ResearchGreg Shay
, Axia AudioJérémie Weber
, AuvitranAidan Williams
Several solutions are available on the market today for digital audio transfer over conventional data cabling, but only some of them allow usage of standard networking equipment. This workshop presents some commercially available solutions (Cobranet, Livewire, Ethersound, Dante), with specific focus on noncompressed, low-latency audio transmission for pro-audio and live applications using standard IEEE 802.3 network technology. The main challenges of digital audio transport will be outlined, including compatibility with common networking equipment, reliability, latency, and deployment. Typical scenarios will be proposed, with panelists explaining their own approaches and solutions.
Friday, October 3, 11:00 am — 1:00 pm
L5 - Practical Advice for Wireless Systems Users
:Freddy ChancellorHenry Cohen
, Production Radio Rentals - NYC, NY, USAMichael Pettersen
, Shure Incorporated - Niles, IL, USA
From houses of worship to wedding bands to community theaters, there are small- to medium-sized wireless microphone systems and IEMs in use by the millions. Unlike the Super Bowl or the Grammys, these smaller systems often do not have dedicated technicians, sophisticated frequency coordination, or in many cases even the proper basic attention to system setup. This live sound event will begin with a basic discussion of the elements of properly choosing components, designing systems, and setting them up in order to minimize the potential for interference while maximizing performance. Topics covered will include antenna placement, antenna cabling, spectrum scanning, frequency coordination, gain structure, system monitoring and simple testing/troubleshooting procedures. Briefly covered will also be planning for upcoming RF spectrum changes.
Friday, October 3, 2:30 pm — 4:00 pm
Compressors—A Dynamic Perspective
:Dave DerrWade GoekeDave HillHutch HutchisonGeorge MassenburgRupert Neve
A device that, some might say, is being abused by those involved in the “loudness wars,” the dynamic range compressor can also be a very creative tool. But how exactly does it work? Six of the audio industry’s top designers and manufacturers lift the lid on one of the key components in any recording, broadcast or live sound signal chain. They will discuss the history, philosophy and evolution of this often misunderstood processor. Is one compressor design better than another? What design features work best for what application? The panel will also reveal the workings behind the mysteries of feedback and feed-forward designs, side-chains, and hard and soft knees, and explore the uses of multiband, parallel and serial compression.
Friday, October 3, 2:30 pm — 4:15 pm
M3 - Sonic Methodology and Mythology
:Keith O. Johnson
- Pacifica, CA, USA
Do extravagant designs and superlative specifications satisfy sonic expectations? Can power cords, interconnects, marker dyes and other components in a controversial lineup improve staging, clarity, and other features? Intelligent measurements and neural feedback studies support these sonic issues as well as predict misdirected methodology from speculative thought. Sonic changes and perceptual feats to hear them are possible and we'll explore recorders, LPs, amplifiers, conversion, wire, circuits and loudspeakers to observe how they create artifacts and interact in systems. Hearing models help create and interpret tests intended to excite predictive behaviors of components. Time domain, tone cluster and fast sweep signals along with simple test devices reveal small complex artifacts. Background knowledge of halls, recording techniques, and cognitive perception becomes helpful to interpret results, which can reveal simple explanations to otherwise remarkable physics. Other topics include power amplifiers that can ruin a recording session, noise propagation from regulators, singing wire, coherent noise, eigensonics, and speakers prejudicial to key signatures. Waveform perception, tempo shifting, and learned object sounds will be demonstrated.
Friday, October 3, 2:30 pm — 4:30 pm
L6 - Source-Oriented Live Sound Reinforcement
, Technology VisionsPanelists
, ArupDave Haydon
, Out Board ElectronicsGeorge Johnsen
, Threshold Digital Research LabsVikram Kirby
, Thinkwell Design & ProductionRobin Whittaker
, Out Board Electronics
Directional amplification, also referred to as Source-Oriented Reinforcement (SOR), describes a practical technique to deliver amplified sound to a large listening area with even coverage while providing directional information to reinforce visual cues and create a realistic and non-contradictory auditory panorama. Audio demonstrations of the fundamental psychoacoustic techniques employed in a SOR design will be presented and limits discussed.
The panel of presenters will outline the history of SOR from the pioneering work of Ahnert, Steinke, and Fels with their Delta Stereophony System in the mid 1970s (later licensed to AKG), to Out Board’s current day TiMax Audio Imaging Delay Matrix, including the very latest ground breaking technology employed to enable control of precedence by radar tracking the actors on the stage.
Descriptions of venues and productions that have employed SOR will be included.
Friday, October 3, 2:30 pm — 5:00 pm
P10 - Nonlinearities in Loudspeakers
: Laurie Fincham
, THX Ltd. - San Rafael, CA, USAP10-1 Audibility of Phase Response Differences in a Stereo Playback System. Part 2: Narrow-Band Stimuli in Headphones and Loudspeakers
—Sylvain Choisel, Geoff Martin
, Bang & Olufsen A/S - Struer, Denmark
An series of experiments were conducted in order to measure the audibility thresholds of phase differences between channels using mismatched cross-over networks. In Part 1 of this study, it was shown that listeners are able to detect very small inter-channel phase differences when presented with wide-band stimuli over headphones, and that the threshold was frequency dependent. This second part of the investigation focuses on listeners’ abilities with narrow-band signals (from 63 to 8000 Hz) in headphones as well as loudspeakers. The results confirm the frequency dependency of the audibility threshold over headphones, whereas for loudspeaker playback the threshold was essentially independent of the frequency.
Convention Paper 7559 (Purchase now)P10-2 Time Variance of the Suspension Nonlinearity
, Technical University of Denmark - Lyngby, Denmark; Bo Rhode Petersen
, Aalborg University - Esbjerg, Denmark
It is well known that the resonance frequency of a loudspeaker depends on how it is driven before and during the measurement. Measurement done right after exposing it to high levels of electrical power and/or excursion giver lower values than what can be measured when the loudspeaker is cold. This paper investigates the changes in compliance the driving signal can cause, this includes low level short duration measurements of the resonance frequency as well as high power long duration measurements of the nonlinearity of the suspension. It is found that at low levels the suspension softens but recovers quickly. The high power and long term measurements affect the nonlinearity of the loudspeaker, by increasing the compliance value for all values of displacement. This level dependency is validated with distortion measurements and it is demonstrated how improved accuracy of the nonlinear model can be obtained by including the level dependency.
Convention Paper 7560 (Purchase now)P10-3 A Study of the Creep Effect in Loudspeakers Suspension
, Technical University of Denmark - Lyngby, Denmark; Knud Thorborg, Carsten Tinggaard
, Tymphany A/S - Taastrup, Denmark
This paper investigates the creep effect, the visco elastic behavior of loudspeaker suspension parts, which can be observed as an increase in displacement far below the resonance frequency. The creep effect means that the suspension cannot be modeled as a simple spring. The need for an accurate creep model is even larger as the validity of loudspeaker models are now sought extended far into the nonlinear domain of the loudspeaker. Different creep models are investigated and implemented both in simple lumped parameter models as well as time domain nonlinear models, the simulation results are compared with a series of measurements on three version of the same loudspeaker with different thickness and rubber type used in the surround.
Convention Paper 7561 (Purchase now)P10-4 The Influence of Acoustic Environment on the Threshold of Audibility of Loudspeaker Resonances
, Bang & Olufsen A/S - Struer, Denmark and University of Surrey, Guildford, Surrey, UK; Sylvain Choisel
, Bang & Olufsen A/S - Struer, Denmark
Resonances in loudspeakers can produce a detrimental effect on sound quality. The reduction or removal of unwanted resonances has therefore become a recognized practice in loudspeaker tuning. This paper presents the results of a listening test that has been used to determine the audibility threshold of a single resonance in different acoustic environments: headphones, loudspeakers in a standard listening room, and loudspeakers in a car. Real loudspeakers were measured and the resonances modeled as IIR filters. Results show that there is a significant interaction between acoustic environment and program material.
Convention Paper 7562 (Purchase now)P10-5 Confirmation of Chaos in a Loudspeaker System Using Time Series Analysis
, Queen Mary, University of London - London, UK; Ivan Djurek, Antonio Petosic
, University of Zagreb - Zagreb, Croatia; Danijel Djurek
, AVAC – Alessandro Volta Applied Ceramics, Laboratory for Nonlinear Dynamics - Zagreb, Croatia
The dynamics of an experimental electrodynamic loudspeaker is studied by using the tools of chaos theory and time series analysis. Delay time, embedding dimension, fractal dimension, and other empirical quantities are determined from experimental data. Particular attention is paid to issues of stationarity in the system in order to identify sources of uncertainty. Lyapunov exponents and fractal dimension are measured using several independent techniques. Results are compared in order to establish independent confirmation of low dimensional dynamics and a positive dominant Lyapunov exponent. We thus show that the loudspeaker may function as a chaotic system suitable for low dimensional modeling and the application of chaos control techniques.
Convention Paper 7563 (Purchase now)
Friday, October 3, 4:00 pm — 6:45 pm
B6 - History of Audio Processing
:Dick BurdenMarvin Caesar
, AphexGlen Clark
, Glen Clark & AssociatesMike Dorrough
, Dorrough ElectronicsFrank Foti
, OmniaGreg J. Ogonowski
, Orban/CRLBob Orban
, Orban/CRLEric Small
, Modulation Sciences
The participants of this session pioneered audio processing and developed the tools we still use today. A discussion of the developments, technology, and the “Loudness Wars” will take place. This session is a must if you want to understand how and why audio processing is used.
Friday, October 3, 5:00 pm — 6:30 pm
P12 - Amplifiers and Automotive Audio
P12-1 Imperfections and Possible Advances in Analog Summing Amplifier Design
, MMK Instruments - Belgrade, Serbia; Dragan Drincic
, Advanced School for Electrical & Computer Engineering - Belgrade, Serbia; Sasha Jankovic
, OXYGEN-Digital, Parkgate Studio - Sussex, UK
The major requirement in the design of the analog summing amplifier is the quality of the summing bus. The key problem in most common designs is the artifact of summing bus impedance, which cannot be considered as true physical impedance, because it has been generated by negative feedback. The loop gain of the amplifier used will limit the performance at higher audio frequencies where the loop gain is lower, increasing the channels cross talk. The inevitable effect of heavy feedback is the increased susceptibility of the amplifier to oscillate as well as sensitivity to RFI. The advanced solution, presented in this paper, could be seen in the usage of the transistor common-base pair (CB-CB) configuration as a summing bus. The CB pair offers inherent low-input impedance, low-noise, very good frequency response, and, very importantly, makes the application of total feedback not necessarily.
Convention Paper 7569 (Purchase now)P12-2 A Switchmode Power Supply Suitable for Audio Power Amplifiers
, Factor One Inc. - Keyport, NJ, USA
Power supplies for audio amplifiers have different requirements than typical commercial power supplies. A tabulation of power supply parameters that affect the audio application is presented and discussed. Different types of audio amplifiers are categorized and shown to have different requirements. Over time new technologies have emerged that affect the implementation of AC to DC converters used in audio amplifiers. A brief history of audio power supply technology is presented. The evolution of the newly proposed interleaved boost with LLC resonant half bridge topology from preceding technologies is shown. The operation of the new topology is explained and its advantages are shown by a simulation of the circuit.
Convention Paper 7570 (Purchase now)P12-3 On the Optimization of Enhanced Cascode
, Consultant - Miami, FL, USA
Twenty years ago enhanced cascode and other circuit topologies based on the same design principles were presented to audio amplifier designers. The circuit was supposed to be incorporated in transconductance gain stages and current sources. Enhanced cascode was used in some commercial products but have not received wide adoption. It was speculated that enhanced cascode has reduced phase margin and at times higher distortion being compared to conventional cascode. Enhanced cascode is analyzed on the basis of distortion and frequency response. It is shown how to make the most of enhanced cascode. Optimized novel circuit topology is presented.
Convention Paper 7571 (Purchase now)P12-4 An Active Load and Test Method for Evaluating the Efficiency of Audio Power Amplifiers
—Harry Dymond, Phil Mellor
, University of Bristol - Bristol, UK
This paper presents the design, implementation, and use of an “active load” for audio power amplifier efficiency testing. The active load can simulate linear complex loads representative of real-world amplifier operation with a load modulus between 4 and 50 ohms inclusive, load phase-angles between -60° and +60° inclusive, and operates from 20 to 20,000 Hz. The active load allows for the development of an automated test procedure for evaluating the efficiency of an audio power amplifier across a range of output voltage amplitudes, load configurations, and output signal frequencies. The results of testing a class-B and a class-D amplifier, each rated at 100 watts into 8 ohms, are presented.
Convention Paper 7572 (Purchase now)P12-5 An Objective Method of Measuring Subjective Click-and-Pop Performance for Audio Amplifiers
—Kymberly Christman (Schmidt)
, Maxim Integrated Products - Sunnyvale, CA, USA
Click-and-pop refers to any “clicks” and “pops” or other unwanted, audio-band transient signals that are reproduced by headphones or loudspeakers when the audio source is turned on or off. Until recently, the industry’s characterization of this undesirable effect has been almost purely subjective. Marketing phrases such as “low pop noise” and “clickless/popless operation” illustrate the subjectivity applied in quantifying click-and-pop performance. This paper presents a method that objectively quantifies this parameter, allowing meaningful, repeatable comparisons to be drawn between different components. Further, results of a subjective click-and-pop listening test are presented to provide a baseline for objectionable click-and-pop levels in headphone amplifiers.
Convention Paper 7573 (Purchase now)P12-6 Effective Car Audio System Enabling Individual Signal Processing Operations of Coincident Multiple Audio Sources through Single Digital Audio Interface Line
—Chul-Jae Yoo, In-Sik Ryu
, Hyundai Autonet - South Korea
There are three major audio sources in recent car environments: primary audio (usually music including radio), navigation voice prompt, and hands-free voice. Listening situations in cars include not only listening to a single audio source, but also listening to concurrent multiple audio sources—for example, navigation guided as listening music and
navigation guided or listening music as talking on a hands-free cell phone. In this paper a conventional external amplifier system connected with a head unit by three audio interface
lines was introduced. Then, an effective automotive audio system having single SPDIF interface line that is capable of concurrent processing of the above three kinds of audio sources was proposed. The new system leads to a reduced wire harness in car environments and also increases voice qualities by transmitting voice signals via an SPDIF digital line compared with that via analog lines.
Convention Paper 7574 (Purchase now)P12-7 Digital Equalization of Automotive Sound Systems Employing Spectral Smoothed FIR Filters
—Marco Binelli, Angelo Farina
, University of Parma - Parma, Italy
In this paper we investigate the usage of spectral smoothed FIR filters for equalizing a car audio system. The target is also to build short filters that can be processed on DSP processors with limited computing power. The inversion algorithm is based on the Nelson-Kirkeby method and on independent phase and magnitude smoothing, by means of a continuous phase method as Panzer and Ferekidis showd. The filter is aimed to create a "target" frequency response, not necessarily flat, employing a short number of taps and maintaining good performances everywhere inside the car's cockpit. As shown also by listening tests, smoothness, and the choice of the right frequency response increase the performances of the car audio systems.
Convention Paper 7575 (Purchase now)P12-8 Implementation of a Generic Algorithm on Various Automotive Platforms
—Thomas Esnault, Jean-Michel Raczinski
, Arkamys - Paris, France
This paper describes a methodology to adapt a generic automotive algorithm to various embedded platforms while keeping the same audio rendering. To get over the limitations of the target DSPs, we have developed tools to control the transition from one platform to another including algorithm adaptation and coefficients computing. Objective and subjective validation processes allow us to certify the quality of the adaptation. With this methodology, productivity has been increased in an industrial context.
Convention Paper 7576 (Purchase now)P12-9 Advanced Audio Algorithms for a Real Automotive Digital Audio System
—Stefania Cecchi, Lorenzo Palestini, Paolo Peretti, Emanuele Moretti, Francesco Piazza
, Università Politecnica delle Marche - Ancona, Italy; Ariano Lattanzi, Ferruccio Bettarelli
, Leaff Engineering - Porto Potenza Picena (MC), Italy
In this paper an innovative modular digital audio system for car entertainment is proposed. The system is based on a plug-in-based software (real-time) framework allowing reconfigurability and flexibility. Each plug-in is dedicated to a particular audio task such as equalization and crossover filtering, implementing innovative algorithms. The system has been tested on a real car environment, with a hardware platform comprising professional audio equipments, running on a PC. Informal listening tests have been performed to validate the overall audio quality, and satisfactory results were obtained.
Convention Paper 7577 (Purchase now)
Saturday, October 4, 9:00 am — 10:45 am
L7 - 10 Things to Get Right in PA and Sound Reinforceent
This Live Sound Event will discuss the 10 most important things to get right when designing/operating sound reinforcement and PA systems. However, as attendees at the event will learn, there are many more things to consider than just the 10 golden rules, and that the order of importance of these often changes depending upon the venue and type of system. We aim to provide a practical approach to sound systems design and operation and will be illustrated with many practical examples and case histories. Each panelist has many years of practical experience and between them can cover just about any aspect of sound reinforcement and PA systems design, operation, and technology. Come along to an event that aims to answer questions you never knew you had—but of course, to find out the 10 most important ones, you will have to attend the session!
Saturday, October 4, 9:00 am — 12:00 pm
P14 - Listening Tests & Psychoacoustics
: Poppy Crum
, Johns Hopkins University - Baltimore, MD, USAP14-1 Rapid Learning of Subjective Preference in Equalization
—Andrew Sabin, Bryan Pardo
, Northwestern University - Evanston, IL, USA
We describe and test an algorithm to rapidly learn a listener’s desired equalization curve. First, a sound is modified by a series of equalization curves. After each modification, the listener indicates how well the current sound exemplifies a target sound descriptor (e.g., “warm”). After rating, a weighting function is computed where the weight of each channel (frequency band) is proportional to the slope of the regression line between listener responses and within-channel gain. Listeners report that sounds generated using this function capture their intended meaning of the descriptor. Machine ratings generated by computing the similarity of a given curve to the weighting function are highly correlated to listener responses, and asymptotic performance is reached after only ~25 listener ratings.
Convention Paper 7581 (Purchase now)P14-2 An Initial Validation of Individualized Crosstalk Cancellation Filters for Binaural Perceptual Experiments
—Alastair Moore, Anthony Tew
, University of York - York, UK; Rozenn Nicol
, France Télécom R&D - Lannion, France
Crosstalk cancellation provides a means of delivering binaural stimuli to a listener for psychoacoustic research that avoids many of the problems of using headphone in experiments. The aim of this study was to determine whether individual crosstalk cancellation filters can be used to present binaural stimuli, which are perceptually indistinguishable from a real sound source. The fast deconvolution with frequency dependent regularization method was used to design crosstalk cancellation filters. The reproduction loudspeakers were positioned at ±90-degrees azimuth and the synthesized location was 0-degrees azimuth. Eight listeners were tested with three types of stimuli. In twenty-two out of the twenty-four listener/stimulus combinations there were no perceptible differences between the real and virtual sources. The results suggest that this method of producing individualized crosstalk cancellation filters is suitable for binaural perceptual experiments.
Convention Paper 7582 (Purchase now)P14-3 Reverberation Echo Density Psychoacoustics
—Patty Huang, Jonathan S. Abel, Hiroko Terasawa, Jonathan Berger
, Stanford University - Stanford, CA, USA
A series of psychoacoustic experiments were carried out to explore the relationship between an objective measure of reverberation echo density, called the normalized echo density (NED), and subjective perception of the time-domain texture of reverberation. In one experiment, 25 subjects evaluated the dissimilarity of signals having static echo densities. The reported dissimilarities matched absolute NED differences with an R2 of 93%. In a 19-subject experiment, reverberation impulse responses having evolving echo densities were used. With an R2 of 90% the absolute log ratio of the late field onset times matched reported dissimilarities between impulse responses. In a third experiment, subjects reported breakpoints in the character of static echo patterns at NED values of 0.3 and 0.7.
Convention Paper 7583 (Purchase now)P14-4 Optimal Modal Spacing and Density for Critical Listening
—Bruno Fazenda, Matthew Wankling
, University of Huddersfield - Huddersfield, West Yorkshire, UK
This paper presents a study on the subjective effects of modal spacing and density. These are measures often used as indicators to define particular aspect ratios and source positions to avoid low frequency reproduction problems in rooms. These indicators imply a given modal spacing leading to a supposedly less problematic response for the listener. An investigation into this topic shows that subjects can identify an optimal spacing between two resonances associated with a reduction of the overall decay. Further work to define a subjective counterpart to the Schroeder Frequency has revealed that an increase in density may not always lead to an improvement, as interaction between mode-shapes results in serious degradation of the stimulus, which is detectable by listeners.
Convention Paper 7584 (Purchase now)P14-5 The Illusion of Continuity Revisited on Filling Gaps in the Saxophone Sound
, AGH University of Science and Technology - Cracow, Poland
Some time-frequency gaps were cut from a recording of a motif played legato on the saxophone. Subsequently, the gaps were filled with various sonic material: noises and sounds of an accompanying band. The quality of the saxophone sound processed in this way was investigated by listening tests. In all of the tests, the saxophone seemed to continue through the gaps, an impairment in quality being observed as a change in the tone color or an attenuation of the sound level. There were two aims of this research. First, to investigate whether the continuity illusion contributed to this effect, and second, to discover what kind of sonic material filling the gaps would cause the least deterioration in sound quality.
Convention Paper 7585 (Purchase now)P14-6 The Incongruency Advantage for Sounds in Natural Scenes
, Veterans Affairs Northern California Health Care System - Martinez, CA, USA; Valeriy Shafiro
, Rush University Medical Center - Chicago, IL, USA
This paper tests identification of environmental sounds (dogs barking or cars honking) in familiar auditory background scenes (street ambience, restaurants). Initial results with subjects trained on both the background scenes and the sounds to be identified showed a significant advantage of about 5% better identification accuracy for sounds that were incongruous with the background scene (e.g., a rooster crowing in a hospital). Studies with naïve listeners showed this effect is level-dependent: there is no advantage for incongruent sounds up to a Sound/Scene ratio (So/Sc) of –7.5 dB, after which there is again about 5% better identification. Modeling using spectral-temporal measures showed that saliency based on acoustic features cannot account for this difference.
Convention Paper 7586 (Purchase now)
Saturday, October 4, 9:00 am — 10:30 am
P15 - Loudspeakers—Part 1
P15-1 Advanced Passive Loudspeaker Protection
, Kludge Audio - Williamsburg, VA, USA
In a follow-on to a previous conference paper (AES Convention Paper 5881), the author explores the use of polymeric positive temperature coefficient (PPTC) protection devices that have a discontinuous I/V curve that is the result of a physical state change. He gives a simple model for designing networks employing incandescent lamps and PPTC devices together to give linear operation at low levels while providing effective limiting at higher levels to prevent loudspeaker damage. Some discussion of applications in current service is provided.
Convention Paper 7588 (Purchase now)P15-2 Target Modes in Moving Assemblies of Compression Drivers and Other Loudspeakers
—Fernando Bolaños, Pablo Seoane
, Acústica Beyma S.A. - Valencia, Spain
This paper deals with how the important modes in a moving assembly of compression drivers and other loudspeakers can be found. Dynamic importance is an essential tool for those who work on modal analysis of systems with many degrees of freedom and complex structures. The important modes calculation or measurement in moving assemblies is an objective (absolute) method to find the relevant modes that act on the dynamics of these transducers. Our paper discusses axial modes and breath modes, which are basic for loudspeakers. The model generalized masses and the participation factors are useful tools to find the moving assemblies important modes (target modes). The strain energy of the moving assembly, which represents the amount of available potential energy, is essential as well.
Convention Paper 7589 (Purchase now)P15-3 Determining Manufacture Variation in Loudspeakers Through Measurement of Thiele/Small Parameters
—Scott Laurin, Karl Reichard
, Pennsylvania State University - State College, PA, USA
Thiele/Small parameters have become a standard for characterizing loudspeakers. Using fairly straightforward methods, the Thiele/Small parameters for twenty nominally identical loudspeakers were determined. The data were compiled to determine the manufacturing variations. Manufacturing tolerances can have a large impact on the variability and quality of loudspeakers produced. Generally, when more stringent tolerances are applied, there is less variation and drivers become more expensive. Now that the loudspeakers have been characterized, each one will be driven to failure. Some loudspeakers will be intentionally degraded to accelerate failures. The goal is to correlate variation in the Thiele/Small parameters with variation in speaker failure modes and operating life.
Convention Paper 7590 (Purchase now)P15-4 About Phase Optimization in Multitone Excitations
—Delphine Bard, Vincent Meyer
, University of Lund - Lund, Sweden
Multitone signals are often used as excitation for the characterization of audio systems. The frequency spectrum of the response consists of harmonics of the frequencies contained in the excitation and intermodulation products. Besides the choice of frequencies, in order to avoid frequency overlapping, there is also the need to chose adequate magnitudes and phases for the different components that constitute the multitone signal. In this paper we will investigate how the choice of the phases will impact the properties of the multitone signal, but also how it will affect the performances of a compensation method based on Volterra kernels and using multitone signals as an excitation.
Convention Paper 7591 (Purchase now)P15-5 Viscous Friction and Temperature Stability of the Mid-High Frequency Loudspeaker
—Ivan Djurek, Antonio Petosic
, University of Zagreb - Zagreb, Croatia; Danijel Djurek
, Alessandro Volta Applied Ceramics (AVAC) - Zagreb, Croatia
Mid-high frequency loudspeakers behave quite differently as compared to low-frequency units, regarding effects coming from the surrounding air medium. Previous work stressed high influence of the imaginary part of the viscous force, which significantly affects the resonance frequency of mid-high frequency loudspeakers. Viscous force is relatively highly dependent on temperature and humidity of the surrounding air, and in this paper we have evaluated how changes in temperature and humidity reflect to the loudspeaker's linearity, which may be significant for the quality of sound reproduction.
Convention Paper 7592 (Purchase now)P15-6 Calorimetric Evaluation of Intrinsic Friction in the Loudspeaker Membrane
—Antonio Petosic, Ivan Djurek
, University of Zagreb - Zagreb, Croatia; Danijel Djurek
, Alessandro Volta Applied Ceramics (AVAC) - Zagreb, Croatia
Friction losses in the vibrating system of an electrodynamic loudspeaker are represented by the intrinsic friction Ri
, which enters the equation of motion, and these losses are accompanied by irreversible release of the heat. A method is proposed for measurement of the friction losses in the loudspeaker's membrane by measurement of the thermocouple temperature probe glued to the membrane. Temperature on the membrane surface fluctuates stochastically as a result of thermo-elastic coupling in the membrane material. Evaluation of the amplitude in the temperature fluctuations enables an absolute and direct evaluation of intrinsic friction Ri
entering friction force F=Ri
·?(x), irrespective of the nonlinearity type and strength associated with the loudspeaker operation.
Convention Paper 7593 (Purchase now)P15-7 Phantom Powering the Modern Condenser Microphone: A Practical Look at Conditions for Optimized Performance
—Mark Zaim, Tadashi Kikutani, Jackie Green
, Audio-Technica U.S., Inc.
Phantom Powering a microphone is a decades old concept with powering conventions and methods that may have become obsolete, ineffective, or inefficient. Modern sound techniques, including those of live sound settings, now use many condenser microphones in settings that were previously dominated by dynamics. As a prerequisite for considering a modern phantom power specification or method, we study the efficiencies and requirements of microphones in typical multiple mic and high SPL settings in order to gain understanding of circuit and design requirements for the maximum dynamic range performance.
Convention Paper 7594 (Purchase now)
Saturday, October 4, 11:00 am — 1:00 pm
L8 - Good Mic Technique—It's Not Just for the Studio: Microphone Selection and Usage for Live Sound
, Shure Incorporated - Niles, IL, USAPanelists
:Richard BatagliaPhil Garfinkel
, Audix USAMark GilbertDan HealyDave Rat
, Rat Sound
While there are countless factors that contribute to a good sounding live event, selecting, placing, and using microphones well can make the difference between a pleasant event and a sonic nightmare. Every sound professional has their own approach to microphone technique. This live sound event will feature a panel of experts from microphone manufacturers and sound reinforcement providers who will discuss their tips, tricks, and experience for getting the job done right at the start of the signal path. We will address conventional and nonconventional techniques and share some interesting stories from the trenches hopefully giving everyone a few new ideas to try on their next event. Using good mic technique will ultimately give the live engineer more time and energy to concentrate on taming the rest of the signal chain and maybe even making it to catering!
Saturday, October 4, 11:30 am — 1:00 pm
P17 - Loudspeakers—Part 2
P17-1 Accuracy Issues in Finite Element Simulation of Loudspeakers
, PACSYS Limited - Nottingham, UK
Finite element-based software for simulating loudspeakers has been around for some time but is being used more widely now, due to improved solver functionality, faster hardware, and improvements in links to CAD software and other preprocessing improvements. The analyst has choices to make in what techniques to employ, what approximations might be made, and how much detail to model.
Convention Paper 7600 (Purchase now)P17-2 Nonlinear Loudspeaker Unit Modeling
—Bo Rohde Pedersen
, Aalborg University - Esbjerg, Denmark; Finn T. Agerkvist
, Technical University Denmark - Lyngby, Denmark
Simulations of a 6½-inch loudspeaker unit are performed and compared with a displacement measurement. The nonlinear loudspeaker model is based on the major nonlinear functions and expanded with time-varying suspension behavior and flux modulation. The results are presented with FFT plots of three frequencies and different displacement levels. The model errors are discussed and analyzed including a test with a loudspeaker unit where the diaphragm is removed.
Convention Paper 7601 (Purchase now)P17-3 An Optimized Pair-Wise Constant Power Panning Algorithm for Stable Lateral Sound Imagery in the 5.1 Reproduction System
, Yamaha Corporation - Shizuoka, Japan, and McGill University, Montreal, Quebec, Canada; Masahiro Ikeda, Akio Takahashi
, Yamaha Corporation - Shizuoka, Japan
Auditory image control in the 5.1 reproduction system has been a challenge due to the arrangement of loudspeakers, especially in the lateral region. To suppress typical artifacts in a pair-wise constant power algorithm, a new gain ratio between the Left and Left Surround channel has been experimentally determined. Listeners were asked to estimate the gain ratio between two loudspeakers for seven lateral positions so as to set the direction of the sound source. From these gain ratios, a polynomial function was derived in order to parametrically represent a gain ratio in an arbitrary direction. The result of validating the experiments showed that the new function produced stable auditory imagery in the lateral region.
Convention Paper 7602 (Purchase now)P17-4 The Use of Delay Control for Stereophonic Audio Rendering Based on VBAP
—Dongil Hyun, Tacksung Choi, Daehee Youn
, Yonsei University - Seoul, Korea; Seokpil Lee
, Broadcasting-Communication Convergence Research Center KETI - Seongnam, Korea; Youngcheol Park
, Yonsei University - Wonju, Korea
This paper proposes a new panning method that can enhance the performance of the stereophonic audio rendering system based on VBAP. The proposed system introduces a delay control to enhance the performance of the VBAP. Sample delaying is used to reduce the energy cancellation due to out-of-phase. Preliminary simulations and measurements are conducted to verify the controllability of ILD by delay control between stereophonic loudspeakers. By simulating ILD by the delay control, spatial direction at frequencies where energy cancellation occurred could be perceived more stable than the conventional VBAP. The performance of the proposed system is also verified by a subjective listening test.
Convention Paper 7603 (Purchase now)P17-5 Ambience Sound Recording Utilizing Dual MS (Mid-Side) Microphone Systems Based upon Frequency Dependent Spatial Cross Correlation (FSCC)—Part-2: Acquisition of On-Stage Sounds
—Teruo Muraoka Takahiro Miura, Tohru Ifukube
, University of Tokyo - Tokyo, Japan
In musical sound recording, a forest of microphones is commonly observed. It is for good sound localization and favorable ambience, however, the forest is desired to be sparse for less laborious setting up and mixing. For this purpose the authors studied sound-image representation of stereophonic microphone arrangements utilizing Frequency Dependent Spatial Cross Correlation (FSCC), which is a cross correlation of two microphone’s outputs. The authors first examined FSCCs of typical microphone arrangements for acquisition of ambient sounds and concluded that the MS (Mid-Side) microphone system with setting directional azimuth at 132-degrees is the best. The authors also studied conditions of on-stage sounds acquisition and found that the FSCC of a co-axial type microphone takes the constant value of +1, which is advantageous for stable sound localization. Thus the authors further compared additional sound acquisition characteristics of the MS system (setting directional azimuth at 120-degrees) and the XY system. We found the former to be superior. Finally, the author proposed dual MS microphone systems. One is for on-stage sound acquisition set directional azimuth at 120-degrees and the other is for ambient sound acquisition set directional azimuth at 132-degrees.
Convention Paper 7604 (Purchase now)P17-6 Ambisonic Loudspeaker Arrays
, Dolby Laboratories - San Francisco, CA, USA
The Ambisonic system is one of very few surround sound systems that offers the promise of reproducing full three-dimensional (periphonic) audio. It can be shown that arrays configured as regular polyhedra can allow the recreation of an accurate sound field at the center of the array. But the regular polyhedric shape can be impractical for real everyday usage because the requirement that the listener have his head located at the center of the array forces the location of the lower loudspeakers to be beneath the floor, or even the location of a loudspeaker directly beneath the listener. This is obviously impracticable, especially in domestic applications. Likewise, it is typically the case that the width of the array is larger than can be accommodated within the room boundaries. The infeasibility of such arrays is a primary reason why they have not been more widely deployed. The intent of this paper is to explore the efficacy of alternative array shapes for both horizontal and periphonic reproduction.
Convention Paper 7605 (Purchase now)P17-7 Optimum Placement for Small Desktop/PC Loudspeakers
, Audio Expert DOO - Skopje, Macedonia
A desktop/PC loudspeaker usually stands on a desk, so the direct sound from the loudspeaker interferes with the reflected sound from the desk. On the desk, a "perfect" loudspeaker with flat anechoic frequency response will not give a flat, but a comb-like resultant frequency response. Here is presented one simple and inexpensive solution to this problem—a small, conventional loudspeaker is placed on a holder. The holder is a horizontal pivoting telescopic arm that enables easy positioning of the loudspeaker. With one side, the arm is attached on the top corner of the PC monitor, and the other side is attached to the loudspeaker. The listener extends and rotates the arm in horizontal plane to such a position that no reflection from the desk or from the PC monitor reaches the listener, thus preserving the presumably flat anechoic frequency response of the loudspeaker.
Convention Paper 7606 (Purchase now)
Saturday, October 4, 2:00 pm — 4:00 pm
TT14 - WAM Studios, San Francisco
Women's Audio Mission is a San Francisco-based, non-profit organization dedicated to the advancement of women in music production and the recording arts. In a field where women are chronically under-represented (less than 5%), WAM seeks to "change the face of sound" by providing hands-on training, experience, career counseling, and job placement to women and girls in media technology for music, radio, film, television, and the internet. WAM believes that women's mastery of music technology and inclusion in the production process will expand the vision and voice of media and popular culture.
Maximum of 20 participants per tour.
$30 (members), $40 (nonmembers)
Saturday, October 4, 2:30 pm — 4:30 pm
B9 - Listener Fatigue and Longevity
, SIA AcousticsMarvin Caesar
, AphexJames Johnston
, Neural Audio Corp.Ted Ruscitti
, On-Air Research
This panel will discuss listener fatigue and its impact on listener retention. While listener fatigue is an issue of interest to broadcasters, it is also an issue of interest to telecommunications service providers, consumer electronics manufacturers, music producers, and others. Fatigued listeners to a broadcast program may tune out, while fatigued listeners to a cell phone conversation may switch to another carrier, and fatigued listeners to a portable media player may purchase another company’s product. The experts on this panel will discuss their research and experiences with listener fatigue and its impact on listener retention.
Saturday, October 4, 2:30 pm — 4:30 pm
L9 - Digital and Networked Audio in Sound Reinforcement
, Crown InternationalPanelists
:Steve GrayRick KreifeldtSteve MacateeDemetrius PalavosDavid Revel
This event promises a discussion of the challenges and planning involved with deploying digital audio in the sound reinforcement environment. The panel will cover not just the use of digital audio but some of the factors that need to be covered during the design and application of audio systems. While no one solution fits every application, after this panel discussion you will be better able to understand what needs to be considered.
Saturday, October 4, 2:30 pm — 4:30 pm
T12 - Damping of the Room Low-Frequency Acoustics (Passive and Active)
, University of Dayton - Dayton, OH, USAJim Wischmeyer
, Modular Sound Systems, Inc. - Lake Barrington, IL, USA
As the result of its size and geometry, a room excessively amplifies sound at certain frequencies. This is the result of standing waves (acoustic resonances/modes) of the room. These are waves whose original oscillation is continuously reinforced by their own reflections. Rooms have many resonances, but only the low-frequency ones are discrete, distinct, unaffected by the sound absorbing material in the room, and accommodate most of the acoustic energy build-up in the room.
Saturday, October 4, 2:30 pm — 4:00 pm
P20 - Loudspeakers—Part 3
P20-1 Preliminary Results of Calculation of a Sound Field Distribution for the Design of a Sound Field Effector Using a 2-Way Loudspeaker Array with Pseudorandom Configuration
, Musashi Institute of Technology - Tokyo, Japan; Kaoru Ashihara
, Advanced Industrial Science and Technology - Tsukuba, Japan; Shogo Kiryu
, Musashi Institute of Technology - Tokyo, Japan
We have been developing a loudspeaker array system that can control a sound field in real time for live concerts. In order to reduce the sidelobes and to improve the frequency range, a 2-way loudspeaker array with pseudorandom configuration is proposed. Software is being developed to determine the configuration. For now, the configuration is optimized for a focused sound. The software calculates the ratio between the sound pressure of the focus point and the average of the sound pressure around the focus. It was shown that the sidelobes can be reduced with a pseudorandom configuration.
Convention Paper 7616 (Purchase now)P20-2 Design and Implementation of a Sound Field Effector Using a Loudspeaker Array
—Seigo Hayashi, Tomoaki Tanno
, Musashi Institute of Technology - Tokyo, Japan; Toru Kamekawa
, Tokyo National University Arts and Music - Tokyo, Japan; Kaoru Ashihara
, Advanced Industrial Science and Technology - Tokyo Japan; Shogo Kiryu
, Musashi Institute of Technology - Tokyo, Japan
We have been developing an effector that uses a 128-channel two-way loudspeaker array system for live concerts. The system was designed to realize the change of the sound field within 10 ms. The variable delay circuits and the communication circuit between the hardware and the control computer are implemented in one FPGA. All of the delay data that have been calculated in advance are stored in the SDRAM that is mounted on the FPGA board, and only the simple command is sent from the control computer. The system can control up to four sound focuses independently.
Convention Paper 7617 (Purchase now)P20-3 Wave Field Synthesis: Practical Implementation and Application to Sound Beam Digital Pointing
—Paolo Peretti, Laura Romoli, Lorenzo Palestini, Stefania Cecchi, Francesco Piazza
, Universita Politecnica delle Marche - Ancona, Italy
Wave Field Synthesis (WFS) is a digital signal processing technique introduced to achieve an optimal acoustic sensation in a larger area than in traditional systems (Stereophony, Dolby Digital). It is based on a large number of loudspeakers and its real-time implementation needs the study of efficient solutions in order to limit the computational cost. To this end, in this paper we propose an approach based on a preprocessing of the driving function component, which does not depend on the audio streaming. Linear and circular geometries tests will be described and the application of this technique to digital pointing of the sound beam will be presented.
Convention Paper 7618 (Purchase now)P20-4 Highly Focused Sound Beamforming Algorithm Using Loudspeaker Array System
—Yoomi Hur, Seong Woo Kim
, Yonsei University - Seoul, Korea; Young-cheol Park
, Yonsei University - Wonju, Korea; Dae Hee Youn
, Yonsei University - Seoul, Korea
This paper presents a sound beamforming technique that can generate a highly focused sound beam using a loudspeaker array. For this purpose, we find the optimal weight that maximizes the contrast of sound power ratio between the target region and the other regions. However, there is a limitation to make the level of non-target region low with the directly derived weights, so the iterative pattern synthesis technique, which was introduced for antenna array, is investigated. Since it is assumed that there are imaginary signal powers in the non-target regions, the system makes efforts to further improve the contrast ratio iteratively. The performance of the proposed method was evaluated, and the results showed that it could generate highly focused sound beam than conventional method.
Convention Paper 7619 (Purchase now)P20-5 Super-Directive Loudspeaker Array for the Generation of a Personal Sound Zone
—Jung-Woo Choi, Youngtae Kim, Sangchul Ko, Jung-Ho Kim
, Samsung Electronics Co. Ltd. - Gyeonggi-do, Korea
A sound manipulation technique is proposed for selectively enhancing a desired acoustic property in a zone of interest called personal sound zone. In order to create the personal sound zone in which a listener can experience high sound level, acoustic energy is focused on only a selected area. Recently, two performance measures indicating acoustic properties of the personal sound zone—acoustic brightness and contrast—were employed to optimize driving functions of a loudspeaker array. In this paper first some limitations of individual control method are presented, and then a novel control strategy is suggested such that advantages of both are combined in a single objective function. Precise control of a sound field with desired shape of energy distribution is made possible by introducing a continuous spatial weighting technique. The results are compared to those based on the least-square optimization technique.
Convention Paper 7620 (Purchase now)
Saturday, October 4, 5:00 pm — 6:45 pm
L10 - Automixing for Live Sound
:Michael 'Bink' Knowles
, Freelance Live Sound MixerPanelists
, Dan Dugan Sound DesignMac Kerr
, Freelance Live Sound MixerGordon Moore
This seminar offers the live sound mixer a chance to audition a selection of automixers within the context of a panel discussion. Panelists will argue the strengths and weaknesses of automixer topologies and algorithms with respect to their sound quality and ease of use in the field. Analog and digital systems will be compared. Real world applications will be presented and discussed. Questions from the audience will be encouraged.
Saturday, October 4, 5:00 pm — 6:45 pm
T14 - Electric Guitar-The Science Behind the Ritual
:Alex U. Case
, University of Massachusetts - Lowell, MA, USA
It is an unwritten law that recording engineers’ approach the electric guitar amplifier with a Shure SM57, in close against the grille cloth, a bit off-center of the driver, and angled a little. These recording decisions serve us well, but do they really matter? What changes when you back the microphone away from the amp, move it off center of the driver, and change the angle? Alex Case, Sound Recording Technology professor to graduates and undergraduates at UMass Lowell breaks it down, with measurements and discussion of the variables that lead to punch, crunch, and other desirables in electric guitar tone.
Sunday, October 5, 9:00 am — 11:00 am
M4 - Acoustics and Multiphysics Modeling
, Comsol - Palo Alto, CA, USA
This Master Class covers acoustics and multiphysics modeling using Comsol. The Acoustics Module is specifically designed for those who work in classical acoustics with devices that produce, measure, and utilize acoustic waves. Application areas include the design of loudspeakers, microphones, hearing aides, noise control, sound barriers, mufflers, buildings, and performance spaces.
Sunday, October 5, 9:00 am — 11:00 am
P23 - Audio DSP
: Jon Boley
, LSB AudioP23-1 Determination and Correction of Individual Channel Time Offsets for Signals Involved in an Audio Mixture
—Enrique Perez Gonzalez, Joshua Reiss
, Queen Mary University of London - London, UK
A method for reducing comb-filtering effects due to delay time differences between audio signals in sound mixer has been implemented. The method uses a multichannel cross-adaptive effect topology to automatically determine the minimal delay and polarity contributions required to optimize the sound mixture. The system uses real time, time domain transfer function measurements to determine and correct the individual channel offset for every signal involved in the audio mixture. The method has applications in live and recorded audio mixing where recording a single sound source with more than one signal path is required, for example when recording a drum set with multiple microphones. Results are reported that determine the effectiveness of the proposed method.
Convention Paper 7631 (Purchase now)P23-2 STFT-Domain Estimation of Subband Correlations
—Michael M. Goodwin
, Creative Advanced Technology Center - Scotts Valley, CA, USA
Various frequency-domain and subband audio processing algorithms for upmix, format conversion, spatial coding, and other applications have been described in the recent literature. Many of these algorithms rely on measures of the subband autocorrelations and cross-correlations of the input audio channels. In this paper we consider several approaches for estimating subband correlations based on a short-time Fourier transform representation of the input signals.
Convention Paper 7632 (Purchase now)P23-3 Separation of Singing Voice from Music Accompaniment with Unvoiced Sounds Reconstruction for Monaural Recordings
—Chao-Ling Hsu, Jyh-Shing Roger Jang
, National Tsing Hua University - Hsinchu, Taiwan; Te-Lu Tsai
, Institute for Information Industry - Taipei, Taiwan
Separating singing voice from music accompaniment is an appealing but challenging problem, especially in the monaural case. One existing approach is based on computational audio scene analysis, which uses pitch as the cue to resynthesize the singing voice. However, the unvoiced parts of the singing voice are totally ignored since they have no pitch at all. This paper proposes a method to detect unvoiced parts of an input signal and to resynthesize them without using pitch information. The experimental result shows that the unvoiced parts can be reconstructed successfully with 3.28 dB signal-to-noise ratio higher than that achieved by the currently state-of-the-art method in the literature.
Convention Paper 7633 (Purchase now)P23-4 Low Latency Convolution In One Dimension Via Two Dimensional Convolutions: An Intuitive Approach
, Garritan Corp. - Orcas, WA, USA
This paper presents a class of algorithms that can be used to efficiently perform the running convolution of a digital signal with a finite impulse response. The impulse is uniformly partitioned and transformed into the frequency domain, changing the one dimensional convolution into a two dimensional convolution that can be efficiently solved with nested short length acyclic convolution algorithms applied in the frequency domain. The latency of the running convolution is the time needed to acquire a block of data equal in size to the uniform partition length.
Convention Paper 7634 (Purchase now)
Sunday, October 5, 11:00 am — 1:00 pm
L11 - Loudspeaker System Optimization
:Bruce C. Olson
, Olson Sound Design - Minneapolis, MN, USAPanelists
, Renkus HeinzTBA
The panel will discuss recommended ways to optimize loudspeaker systems for use in the typical venues frequented by local bands and regional sound companies. These industry experts will give you practical advice on getting your system to sound good in the usual setup time that is typically available. OK, maybe not typical, we assume you can get more than 5 minutes for the task. Once the system is optimized properly, all you have to do is make the band sound good. This advice is targeted at all the band engineers, as well as system tech’s for small sound companies, maybe even some of you big guys as well.
Sunday, October 5, 11:30 am — 1:00 pm
Platinum Road Warriors
:Eddie MappPaul “Pappy” MiddletonHoward Page
An all-star panel of leading front-of-house engineers will explore subject matter ranging from gear to gossip, in what promises to be an insightful, amusing, and enlightening 90 minute session. Engineers for superstar artists will discuss war stories, technical innovations, and heroic efforts to maintain the eternal “show must go on” code of the road. Ample time will be provided for an audience Q&A session.
Sunday, October 5, 2:30 pm — 5:00 pm
L12 - Innovations in Live Sound—A Historical Perspective
, Pro Media | UltraSoundPanelists
, SoundcraftKen Lopez
, University of Southern CaliforniaJohn Meyer
, Meyer Sound
New techniques and products are often driven by changes in need and available technology. Today’s sound professional has a myriad of products to choose from. That wasn’t always the case. What drove the creation of today’s products? What will drive the products of tomorrow? Sometimes a look back is the best way to get a peek ahead. A panel of industry pioneers and trailblazers will take a look back at past live sound innovations with an emphasis on the needs and constraints that drove their development and adoption.
Sunday, October 5, 5:00 pm — 6:45 pm
T20 - Radio Frequency Interference and Audio Systems
, Audio Systems Group, Inc.
This tutorial begins by identifying and discussing the fundamental mechanisms that couple RF into audio systems and allow it to be detected. Attention is then given to design techniques for both equipment and systems that avoid these problems and methods of fixing problems with existing equipment and systems that have been poorly designed or built.