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AES Munich 2009
Live Sound Event Details

Thursday, May 7, 09:00 — 10:00

Acoustics and Sound Reinforcement

Thursday, May 7, 09:00 — 11:30

P1 - Audio for Telecommunications

Chair: Damian Murphy

P1-1 20 Things You Should Know Before Migrating Your Audio Network to IPSimon Daniels, APT - Belfast, Northern Ireland, UK
For many years, synchronous networks have been considered the industry standard for audio transport worldwide. Balanced analog copper circuits, microwave, and synchronous based systems such as V.35/X.21 or T1/E1 have been the traditional choice for studio transmitter and inter-studio links in professional audio broadcast networks. Readily available from all major service providers, the popularity of synchronous links has been largely due to the fact that they offer dedicated, reliable, point-to-point and bi-directional communication at guaranteed data and error rates. However, the reign of synchronous links as the preferred choice for STLs is currently coming under threat from a new challenger, in the form of IP-based network technology.
Convention Paper 7651 (Purchase now)

P1-2 Deploying Large Scale Audio IP NetworksKevin Campbell, APT - Belfast, Northern Ireland, UK
This paper will examine the key considerations for those interested in deploying large-scale ip audio networks. It will include an overview of the main challenges and draw on the experience of national public broadcasters who have already migrated to IP. We will provide an overview of the key concerns such as jitter, delay, and link reliability that are valid for an IP network of any size. However, this paper will focus mainly on the issues arising from the greater complexity and scale of large national and country-wide deployments. The paper will use illustrations and network applications from real-world deployments to illustrate the points. Paper presented by Hartmut Foerster
Convention Paper 7652 (Purchase now)

P1-3 A Spatial Filtering Approach for Directional Audio CodingMarkus Kallinger, Henning Ochsenfeld, Giovanni Del Galdo, Fabian Kuech, Dirk Mahne, Richard Schultz-Amling, Oliver Thiergart, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
In hands-free telephony, spatial filtering techniques are employed to enhance intelligibility of speech. More precisely, these techniques aim at reducing the reverberation of the desired speech signal and attenuating interferences. Additionally, it is well-known that the spatially separate reproduction of desired and interfering sources enhance intelligibility of speech. For the latter task, Directional Audio Coding (DirAC) has proven to be an efficient method to capture and reproduce spatial sound. In this paper we propose a spatial filtering processing block, which works in the parameter domain of DirAC. Simulation results show that compared to a standard beamformer the novel technique offers significantly higher interference attenuation, while introducing comparably low distortion of the desired signal. Additional subjective tests of speech intelligibility confirm the instrumentally obtained results.
Convention Paper 7653 (Purchase now)

P1-4 A New Bandwidth Extension for Audio Signals without Using Side-InformationKha Le Dinh, Chon Tam Le Dinh, Roch Lefebvre, Université de Sherbrooke - Sherbrooke, Quebec, Canada
The use of narrow bandwidth (300 – 3400 Hz) in the current telephone network limits the perceptual quality of telephone conversations. Changing to wideband network is a solution that can help to improve quality, but it will need a long time to upgrade. Thus, bandwidth extension can be seen as an alternative solution during the transition time. A new bandwidth extension method is presented in this paper. Without using any side-information, the proposed method can be applied as a post-processing step at the terminal devices, maintaining the compatibility to the current telephone network, and thus, no modification is needed in the network nodes. Experimental results show that the proposed solution can help to improve significantly the perceptual quality of narrowband telephone signal.
Convention Paper 7654 (Purchase now)

P1-5 Feature Selection vs. Feature Space Transformation in Music Genre Classification FrameworkHanna Lukashevich, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany
Automatic classification of music genres is an important task in music information retrieval research. Nearly all state-of-the-art music genre recognition systems start from the feature extraction block. The extracted acoustical features often could tend to be correlated or/and redundant, which can cause various difficulties in the classification stage. In this paper we present a comparative analysis on applying supervised Feature Selection (FS) and Feature Space Transformation (FST) algorithms to reduce the feature dimensionality. We discuss pros and cons of the methods and weigh the benefits of each one against the others.
Convention Paper 7655 (Purchase now)

Thursday, May 7, 09:00 — 11:00

T2 - An Introduction to Digital Audio Effects

Christoph M. Musialik, Algorithmix GmbH
Joshua D. Reiss, Queen Mary, University of London - London, UK
Udo Zölzer, Helmut-Schmidt-Universität - Hamburg, Germany

In this tutorial we discuss the ways by which signal processing techniques are used to produce effects acting on digital audio signals. The audio effects are systematically classified and discussed, with emphasis on how and why they are used. Practical examples of common effects are provided, along with block diagrams, pseudo-code, and sound examples. During the tutorial, a few effects will be created from scratch and the audience will be provided with the basic background knowledge to design their own effects.

Thursday, May 7, 10:00 — 11:30

P3 - Recording, Reproduction, and Delivery

P3-1 Audio Content Annotation, Description, and Management Using Joint Audio Detection, Segmentation, and Classification TechniquesChristos Vegiris, Charalambos Dimoulas, George Papanikolaou, Aristotle University of Thessaloniki - Thessaloniki, Greece
The current paper focuses on audio content management by means of joint audio segmentation and classification. We concentrate on the separation of typical audio classes, such as silence/background noise, speech, crowded speech, music, and their combinations. A compact feature-vector subset is selected by a Correlation feature selection subset evaluation algorithm after the use of EM clustering algorithm on an initial audio data set. Time and spectral parameters are extracted using filter-banks and wavelets in combination with sliding windows and exponential moving averaging techniques. Features are extracted on a point-to-point basis, using the finest possible time resolution, so that each sample can be individually classified to one of the available groups. Clustering algorithms like EM or Simple K-means are tested to evaluate the final point-to-point classification result, therefore the joint audio detection-classification indexes. The extracted audio detection, segmentation, and classification results can be incorporated into appropriate description schemes that would annotate audio events/segments for content description and management purposes.
Convention Paper 7661 (Purchase now)

P3-2 Ambience Sound Recording Utilizing Dual MS (Mid-Side) Microphone Systems Based upon Frequency Dependent Spatial Cross Correlation (FSCC) [Part 3: Consideration of Microphones’ Locations]Teruo Muraoka, Takahiro Miura, Tohru Ifukube, University of Tokyo - Tokyo, Japan
In order to achieve ambient and exactly sound-localized musical recording with fewer number of microphones, we studied sound acquisition performances of microphone arrangements utilizing their Frequency Dependent Spatial Cross Correlation (FSCC). The result is that an MS microphone is best for this purpose. The setting of the microphone's directional azimuth at 132 degrees is the best for ambient sound acquisition and setting of that at 120 degrees is best for on-stage sound acquisition. We conducted actual concert recordings with a combination of those MS microphones (Dual MS microphone systems) and obtained satisfactory results. Successively, we studied the proper setting positions of those microphones. For ambient sound acquisition, suspending the microphone at the center of a concert hall is favorable, and for on-stage sound acquisition, locating it at almost above the conductor’s position will also be satisfactory. Process of the studies will be reported.
Convention Paper 7662 (Purchase now)

P3-3 A Comparative Approach to Sound Localization within a 3-D Sound FieldMartin J. Morrell, Joshua D. Reiss, Queen Mary, University of London - London, UK
In this paper we compare different methods for sound localization around and within a 3-D sound field. The first objective is to determine which form of panning is consistently preferred for panning sources around the loudspeaker array. The second objective and main focus of the paper is localizing sources within the loudspeaker array. We seek to determine if the sound sources can be located without movement or a secondary reference source. The authors compare various techniques based on ambisonics, vector base amplitude panning and time delay based panning. We report on subjective listening tests that show which method of panning is preferred by listeners and rate the success of panning within a 3-D loudspeaker array.
Convention Paper 7663 (Purchase now)

P3-4 The Effect of Listening Room on Audio Quality in Ambisonics ReproductionOlli Santala, Helsinki University of Technology - Espoo, Finland; Heikki Vertanen, Helsinki University of Technology - Espoo, Finland, University of Helsinki, Helsinki, Finland; Jussi Pekonen, Jan Oksanen, Ville Pulkki, Helsinki University of Technology - Espoo, Finland
In multichannel reproduction of spatial audio with first-order Ambisonics the loudspeaker signals are relatively coherent, which produces prominent coloration. The coloration artifacts have been suggested to depend on the acoustics of the listening room. This dependency was researched with subjective listening tests in an anechoic chamber with an octagonal loudspeaker setup. Different virtual listening rooms were created by adding diffuse reverberation with 0.25 seconds RT60 using a 3-D 16-channel loudspeaker setup. In the test, the subjects compared the audio quality in the virtual rooms. The results suggest that optimal audio quality was obtained when the virtual room effect and the direct sound were on equal level at the listening position.
Convention Paper 7664 (Purchase now)

P3-5 Ontology-Based Information Management in Music ProductionGyorgy Fazekas, Mark Sandler, Queen Mary, University of London - London, UK
In information management, ontologies are used for defining concepts and relationships of a domain in question. The use of a schema permits structuring, interoperability, and automatic interpretation of data, thus allows accessing information by means of complex queries. In this paper we use ontologies to associate metadata, captured during music production, with explicit semantics. The collected data is used for finding audio clips processed in a particular way, for instance, using engineering procedures or acoustic signal features. As opposed to existing metadata standards, our system builds on the Resource Description Framework, the data model of the Semantic Web, which provides flexible and open-ended knowledge representation. Using this model, we demonstrate a framework for managing information, relevant in music production.
Convention Paper 7665 (Purchase now)

Thursday, May 7, 12:00 — 13:30

Opening Ceremonies
Keynote Speech

Awards Presentation
Please join us as the AES presents special awards to those who have made outstanding contributions to the Society in such areas of research, scholarship, and publications, as well as other accomplishments that have contributed to the
enhancement of our industry. The awardees are:
Bronze Medal Award:
• Ivan Stamac
Fellowship Award:
• Martin Wöhr
Board of Governors Award:
• Jan Berg
• Klaus Blasquiz
• Kimio Hamasaki
• Shinji Koyano
• Tapio Lokki
• Jiri Ocenasek
• John Oh
• Jan Abildgaard Pedersen
• Joshua Reiss

Keynote Speaker

This year’s Keynote Speaker is Gerhard Thoma. Thoma has been leading the department of acoustics projects at BMW for more than 20 years. His speech will highlight many aspects of perception and acoustics from an unusual point of view: What does a driver in a car need to hear, what does he should not hear, and how can the acoustics and sounds of a car help to significantly enhance driving pleasure and safety?

Thursday, May 7, 13:30 — 15:30

W2 - New Technologies for Audio Over IP

Jeremy Cooperstock, McGill University - Montreal, Quebec, Canada
Steve Church, Telos
Christian Diehl, Mayah
Manfred Lutzky, Fraunhofer IIS
Greg Massey, APT

This workshop is intended to provide an "under the hood" discussion of various low-latency codecs as well as a comparison of their pros and cons for different applications. Codecs including AAC-ELD and ULD will be discussed, along with techniques such as adaptive jitter buffer management.

Thursday, May 7, 14:00 — 15:00

Fiber Optics for Audio (Formative Meeting)

Formative Meeting

Thursday, May 7, 14:00 — 16:00

W3 - Intelligent Digital Audio Effects

Christian Uhle, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Alexander Lerch, Z-Plane - Berlin, Germany
Josh Reiss, Queen Mary, University of London - London, UK
Udo Zölzer, Helmut-Schmidt-Universität - Hamburg, Germany

Intelligent Digital Audio Effects (I-DAFx) process audio signals in a signal-adaptive way by using some kind of high-level analysis of the input. Beat tracking, for example, enables the automated adaption of time delays or of the LFO rate in tremolos, auto-wahs, and vibrato effects. A harmonizer can adapt the additional intervals to the melody that is played. Automatic mixing is approached by analyzing the signal content in all channels to control the panning. These are examples of techniques that are in the scope of this workshop. It presents an overview of I-DAFx and of methods of semantic audio analysis used in these devices. Practical examples are described and sound examples are demonstrated.

Thursday, May 7, 14:00 — 18:30

P4 - Recording, Reproduction, and Delivery

Chair: Joerg Wuttke

Siegfried Linkwitz, Linkwitz Lab

P4-1 An Expert in Absentia: A Case-Study for Using Technology to Support Recording Studio PracticeAndrew King, University of Hull - Scarborough, North Yorkshire, UK
This paper examines the use of a Learning Technology Interface (LTI) to support the completion of a recording workbook with audio examples over a ten-week period. The VLE provided contingent support to studio users for technical problems encountered in the completion of four recording tasks. Previous research has investigated how students collaborate and problem-solve during a short session in the recording studio using technology as a contingent support tool. In addition, online message boards have been used to record problems encountered when completing a prescribed task (critical-incident recording). A mixed-methods case study approach was used in this study. The students interactions within the LTI were logged (i.e., frequency, time, duration, type of support) and their feedback was elicited via a user questionnaire at the end of the project. Data for this study demonstrates that learning technology can be a successful support tool and also highlights the frequency and themes concerning the types of recording practice information accessed by the learners.
Convention Paper 7669 (Purchase now)

P4-2 Recording and Reproduction over Two Loudspeakers as Heard Live—Part 1: Hearing, Loudspeakers, and RoomsSiegfried Linkwitz, Linkwitz Lab - Corte Madera, CA, USA; Don Barringer, Linkwitz Lab - Arlington, CA, USA
Innate hearing processes define the realism that can be obtained from reproduced sound. An unspecified system with two loudspeakers in a room places considerable limitations upon the degree of auditory realism that can be obtained. It has been observed that loudspeakers and room must be hidden from the auditory scene that is evoked in the listener’s brain. Requirements upon the polar response and the output volume capability of the loudspeaker will be discussed. Problems and solutions in designing a three-way, open baffle loudspeaker with piston drivers will be presented. Loudspeakers and listener must be symmetrically placed in the room to minimize the effects of reflections upon the auditory illusion.
Convention Paper 7670 (Purchase now)

P4-3 Recording and Reproduction over Two Loudspeakers as Heard Live—Part 2: Recording Concepts and PracticesDon Barringer, Linkwitz Lab - Arlington, VA, USA; Siegfried Linkwitz, Linkwitz Lab - Corte Madera, CA, USA
For a half century, the crucial interaction between recording engineer and monitor loudspeakers during two-channel stereophonic recording has not been resolved, leaving the engineer to cope with uncertainties. However, recent advances in defining and improving this loudspeaker-room-listener interface have finally allowed objectivity to inform and shape the engineer’s choices. The full potential of the two-channel format is now accessible to the recording engineer, and in a room that is just as normal as most consumers’ rooms. The improved reproduction has also allowed a deeper understanding of the merits and limits of spaced and coincident/near-coincident microphone arrays. As a result of these and earlier observations, a four-microphone array was conceived that exploits natural hearing processes to achieve greater auditory realism from two loudspeakers. A number of insights have emerged from the experiments.
Convention Paper 7671 (Purchase now)

P4-4 Vision and Technique behind the New Studios and Listening Rooms of the Fraunhofer IIS Audio LaboratoryAndreas Silzle, Stefan Geyersberger, Gerd Brohasga, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Dieter Weninger, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany, Innovationszentrum für Telekommunikationstechnik GmbH IZT, Erlangen, Germany; Michael Leistner, Fraunhofer Institute for Building Physics IBP - Stuttgart, Germany
The new audio laboratory rooms of the Fraunhofer IIS and their technical design are presented here. The vision behind them is driven by the very high demands of a leading edge audio research organization with more than 100 scientists and engineers. The 300 m2 sound studio complex was designed with the intention of providing capabilities that are in combination far more extensive than those available in common audio research or production facilities. The reproduction room for listening tests follows the strict recommendations of ITU-R BS 1116. The results of the qualification measurements regarding direct sound, reflected sound, and steady state sound field will be shown and the construction efforts needed to achieve these values are explained. The connection from all the computers in the server room to more than 70 loudspeakers in the reproduction rooms, other audio interfaces, and the projection screens is done by an audio and video routing system. The architecture of the advanced control software of this routing system is presented. It allows easy and flexible access for each class of user to all the possibilities made available by this completely new system.
Convention Paper 7672 (Purchase now)

P4-5 Advances in National Broadcaster Networks: Exploring Transparent High Definition IPTVMatthew O’Donnell, British Sky Broadcasting - Upminster, UK
British commercial broadcasters are increasing their ability to determine the quality of distribution of audio-over-IP by acquiring and installing next generation national Gigabit networks. This paper explores how broadcasters can use the advances in broadband technology to transparently integrate supplemental on-demand IPTV services with traditional broadcasting transport, which has led to broadcasters being confident in achieving scalable carrier-class quality of service for delivery of high definition media direct to the customer’s set top box.
Convention Paper 7673 (Purchase now)

P4-6 Multi-Perspective Surround Sound Audio RecordingMark J. Sarisky, The University of Texas at Austin - Austin, TX, USA
With the advent of Blu-Ray Disc Audio (BD-Audio), high resolution uncompressed audio recordings can be presented as a consumer product in a variety of surround sound formats. This paper proposes a new take on the recording of live and studio music in surround sound that allows the consumer to benefit from the large capacity of the BD-Audio disc and enjoy the recording from multiple listening perspectives.
Convention Paper 7674 (Purchase now)

P4-7 Sound Intensity-Based Three-Dimensional PanningAkio Ando, Kimio Hamasaki, NHK Science and Technical Research Laboratories - Setagaya, Tokyo, Japan
Three-dimensional (3-D) panning equipment is essential for the production of 3-D audio content. We have already proposed an algorithm to enable such panning. It generates the input signal to be fed into multichannel loudspeakers so as to realize the same physical properties of sound at the receiving point as those created by a single loudspeaker model of the virtual source. A sound pressure vector is used as the physical property. This paper proposes a new method that uses sound intensity instead of the sound pressure vector and shows that both conventional “vector base amplitude panning” and our previous method come very close to achieving coincidence of sound intensity. A new panning method using four loudspeakers is also proposed.
Convention Paper 7675 (Purchase now)

P4-8 A Practical Comparison of Three Tetrahedral Ambisonic MicrophonesDan Hemingson, Mark Sarisky, The University of Texas at Austin - Austin, TX, USA
This paper compares two low-cost tetrahedral ambisonic microphones, an experimental microphone, and a Core Sound TetraMic with a Soundfield MKV or SPS422B serving as a standard for comparison. Recordings were made in natural environments of live performances, in a recording studio, and in an anechoic chamber. The results of analytical and direct listening tests of these recordings are discussed in this paper. A description of the experimental microphone and the recording setup is included.
Convention Paper 7676 (Purchase now)

P4-9 A New Reference Listening Room for Consumer, Professional, and Automotive Audio ResearchSean Olive, Harman International - Northridge, CA, USA
This paper describes the features, scientific rationale, and acoustical performance of a new reference listening room designed for the purpose of conducting controlled listening tests and psychoacoustic research for consumer, professional, and automotive audio products. The main features of the room include quiet and adjustable room acoustics, a high-quality calibrated playback system, an in-wall loudspeaker mover, and complete automated control of listening tests performed in the room.
Convention Paper 7677 (Purchase now)

Thursday, May 7, 14:00 — 18:00

P5 - Loudspeakers

Chair: John Vanderkooy, University of Waterloo - Waterloo, Ontario, Canada

P5-1 Estimating the Velocity Profile and Acoustical Quantities of a Harmonically Vibrating Loudspeaker Membrane from On-Axis Pressure DataRonald M. Aarts, Philips Research Europe - Eindhoven, The Netherlands, Technical University of Eindhoven, Eindhoven, The Netherlands; Augustus J. Janssen, Philips Research Europe - Eindhoven, The Netherlands
Formulas are presented for acoustical quantities of a harmonically excited resilient, flat, circular loudspeaker in an infinite baffle. These quantities are the sound pressure on-axis, far-field, directivity and the total radiated power. These quantities are obtained by expanding the velocity distribution in terms of orthogonal polynomials. For rigid and non-rigid radiators, this yields explicit, series expressions for both the on-axis and far-field pressure. In the reverse direction, a method of estimating velocity distributions from (measured) on-axis pressures by matching in terms of expansion coefficients is described. Together with the forward far-field computation scheme, this yields a method for assessment of loudspeakers in the far-field and of the total radiated power from (relatively near-field) on-axis data (generalized Keele scheme).
Convention Paper 7678 (Purchase now)

P5-2 Testing and Simulation of a Thermoacoustic Transducer PrototypeFotios Kontomichos, Alexandros Koutsioubas, John Mourjopoulos, Nikolaos Spiliopoulos, Alexandros Vradis, Stamatis Vassilantonopoulos, University of Patras - Patras, Greece
Thermoacoustic transduction is the transformation of thermal energy fluctuations into sound. Devices fabricated by appropriate materials utilize such a mechanism in order to achieve acoustic wave generation by direct application of an electrical audio signal and without the use of any moving components. A thermoacoustic transducer causes local vibration of air molecules resulting in a proportional pressure change. The present paper studies an implementation of this alternative audio transduction technique for a prototype developed on silicon wafer. Measurements of the performance of this hybrid solid state device are presented and compared to the theoretical principles of its operation, which are evaluated via simulations.
Convention Paper 7679 (Purchase now)

P5-3 Analysis of Viscoelasticity and Residual Strains in an Electrodynamic LoudspeakerIvan Djurek, Antonio Petosic, University of Zagreb - Zagreb, Croatia; Danijel Djurek, Alessandro Volta Applied Ceramics (AVAC) - Zagreb, Croatia
An electrodynamic loudspeaker was analyzed in three steps: (a) as a device supplied by the market, (b) removed upper suspension, and (c) dismantled assembly consisting only of vibrating spider and voice coil. In three steps, resonant frequency and stiffness were measured dynamically for driving currents up to 100 mA, whereas stiffness was also measured quasi-statically by the use of calibrated masses. It was found that widely quoted effect of decreasing resonant frequency, as plotted against driving current, comes from the residual strain in the vibrating material, and significant contribution is associated with the spider. When driving current increases residual strain is gradually compensated, giving rise to the minimum of stiffness, and further increase of resonant frequency is attributed to a common nonlinearity in the forced vibrating system.
Convention Paper 7680 (Purchase now)

P5-4 Forces in Cylindrical Metalized Film Audio CapacitorsPhilip J. Duncan, University of Salford, Greater Manchester, UK; Nigel Williams, Paul S. Dodds, ICW Ltd. - Wrexham, Wales, UK
This paper is concerned with the analysis of forces acting in metalized polypropylene film capacitors in use in loudspeaker crossover circuits. Capacitors have been subjected to rapid discharge measurements to investigate mechanical resonance of the capacitor body and the electrical forces that drive the resonance. The force due to adjacent flat current sheets has been calculated in order that the magnitude of the electro-dynamic force due to the discharge current can be calculated and compared with the electrostatic force due to the potential difference between the capacitor plates. The electrostatic force is found to be dominant by several orders of magnitude, contrary to assumptions in previous work where the electro-dynamic force is assumed to be dominant. The capacitor is then modeled as a series of concentric cylindrical conductors and the distribution of forces within the body of the capacitor is considered. The primary outcome of this is that the electrostatic forces act predominantly within the inner and outer turn of the capacitor body, while all of the forces acting within the body of the capacitor are balanced almost to zero. Experimental results where resonant acoustic emissions have been measured and analyzed are presented and discussed in the context of the model proposed.
Convention Paper 7682 (Purchase now)

P5-5 On the Use of Motion Feedback as Used in 4th Order SystemsStefan Willems, Denon & Marantz Holding, Premium Sound Solutions - Leuven Belgium; Guido D’Hoogh, Retired
Class D amplification allows the design of compact very high power amplifiers with a high efficiency. Those amplifiers are an excellent candidate for being used in compact high-powered subwoofers. The drawback of compact subwoofers is the nonlinear compression of the air inside the (acoustically) small box. Fourth order systems are beneficial over 2nd order systems due to their increased efficiency. To combine the best of both worlds, 4th order design and acoustically small enclosures, a feedback mechanism has been developed to reduce the nonlinear distortion found in compact high-powered subwoofers. Acceleration feedback on woofer systems is traditionally used in 2nd order systems. This paper discusses the use of an acceleration and velocity feedback system applied to a 4th order system.
Convention Paper 7683 (Purchase now)

P5-6 Mapping of the Loudspeaker Emission by the Use of Anemometric MethodDanijel Djurek, Alessandro Volta Applied Ceramics (AVAC) - Zagreb, Croatia; Ivan Djurek, Antonio Petosic, University of Zagreb - Zagreb, Croatia
Lateral wire anemometry (LWA) has been developed for recording of air vibration. Standard anemometry is founded upon the hot wire method, and wire temperature changes in the oscillating air velocity in the range 800-1000 °C, which is less suitable because of the proper heat emission from the wire. LWA deals only with the initial slope of the changing wire resistance, and subsequent Fourier analysis enables measurements of periodic air velocity. The probe has been developed for precise mapping of the air velocity field in the front of the membrane, and local power emission of the membrane may be evaluated in the region fitted to 0.15 cm2.
Convention Paper 7684 (Purchase now)

P5-7 Flat Panel Loudspeaker Consisting of an Array of Miniature TransducersDaniel Beer, Stephan Mauer, Sandra Brix, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Jürgen Peissig, Sennheiser Electronic GmbH & Co. KG - Wedemark, Germany
Multichannel audio reproduction systems like the Wave Field Synthesis (WFS) use a large number of small and closely spaced loudspeakers. The successful use of WFS requires, among other things, the ability of an "invisible” integration of loudspeakers in a room. Flat panel loudspeakers compared with conventional cone loudspeakers provide advantages in the space saved room integration because of their low manufactured depth. In this way flat panel loudspeakers can be found in furniture, media devices, or like pictures hung on the wall. Besides the integration, flat loudspeakers should provide at least the same good acoustical performance as conventional loudspeakers. This is indeed a problem, because the low depth negatively influences the acoustical quality of reproduction in the lower and middle frequency range. This paper demonstrates a new flat panel loudspeaker consisting of an array of miniature transducers.
Convention Paper 7685 (Purchase now)

P5-8 Subwoofer Loudspeaker System with Dynamic Push-Pull DriveDrazenko Sukalo, DSLab–Device Solution Laboratory - Munich, Germany
This paper examines the influence of mutual coupling between two driver-diaphragms driven by two electrical signals, each with a 90° phase shift on the voice-coil impedance curve. A new model of the system is described, and the effects are observed using the electrical circuit simulator PSpice. Finally, predicted and measured values are presented.
Convention Paper 7686 (Purchase now)

Thursday, May 7, 16:00 — 17:00

Hearing and Hearing Loss Prevention

Thursday, May 7, 16:00 — 18:30

W4 - Microphones—What to Listen For—What Specs to Look For

Eddy B. Brixen
Jean-Marie Greijsen, Polyhymnia International
David Josephson, Josephson Engineering
Douglas McKinnie, Middle Tennessee State University
Mikkel Nymand, DPA Microphones
Ossian Ryner, DR, Danish Broadcasting

When selecting microphones for a specific music recording, it is worth knowing what to expect and what to listen for. Accordingly it is good to know what specifications that would be optimum for that microphone. This workshop explains the process of selecting a microphone both from the aesthetical as well as the technical point of view. Also explained and demonstrated: what to expect when placing the microphone. This is not a “ I feel like . . .” presentation. All presenters on the panel are serious and experienced engineers and tonmeisters. The purpose of this workshop is to encourage and teach young engineers and students to take advantage by taking a closer look at the specifications the next time they are going to pick a microphone for a job.

Thursday, May 7, 16:30 — 18:30

W5 - Professional Audio Networking in Sound Reinforcement and Broadcast Applications

Umberto Zanghieri, ZP Engineering srl
Bradford Benn, Crown International
David Revel, Technical Multimedia Design
Greg Shay, Axia Audio
Jérémie Weber, Auvitran

Several solutions are available on the market today for digital audio transfer over conventional data cabling. This workshop presents some commercially available solutions, 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 reliability, latency, and deployment. Typical usage scenarios will be proposed, with specific emphasis on live sound reinforcement and broadcast applications.

This event promises a discussion of the challenges and planning involved with deploying digital audio in such scenarios.

The workshop will include a brief overview of potential evolutions related to pro audio networking.

Thursday, May 7, 16:30 — 18:00

P7 - Spatial Audio Processing

P7-1 Low Complexity Binaural Rendering for Multichannel SoundKangeun Lee, Changyong Son, Dohyung Kim, Samsung Advanced Institute of Technology - Suwon, Korea
The current paper is concerned with an effective method to emulate the multichannel sound in a portable environment where low power is required. The goal of this paper is to show the complexity of binaural rendering of the multichannel to stereo sound systems in cases of portable devices. To achieve this, we proposed the modified discrete cosine transform (MDCT) based binaural rendering, combined with the Dolby Digital decoder (AC-3) that is a multichannel audio decoder. A reverberation algorithm is added to the proposed algorithm for closing to real sound. This combined structure is implemented on a DSP processer. The complexity and quality are compared with a conventional head-related transfer function (HRTF) filtering method and Dolby headphone that are the most current in commercial binaural rending technology, demonstrating significant complexity reduction and comparable sound quality to the Dolby headphone.
Convention Paper 7687 (Purchase now)

P7-2 Optimal Filtering for Focused Sound Field Reproductions Using a Loudspeaker ArrayYoungtae Kim, Sangchul Ko, Jung-Woo Choi, Jungho Kim, SAIT, Samsung Electronics Co., Ltd. - Gyeonggi-do, Korea
This paper describes audio signal processing techniques in designing multichannel filters for reproducing an arbitrary spatial directivity pattern with a typical loudspeaker array. In designing the multichannel filters, some design criteria based on, for example, least-squares methods and the maximum energy array are introduced as non-iterative optimization techniques with a lower computational complexity. The abilities of the criteria are first evaluated with a given loudspeaker configuration for reproducing a desired acoustic property in a spatial area of interest. Also, additional constraints are considered to impose for minimizing the error between the amplitudes of actual and the desired spatial directivity pattern. Their limitations in practical applications are revealed by experimental demonstrations, and finally some guidelines are proposed in designing optimal filters.
Convention Paper 7688 (Purchase now)

P7-3 Single-Channel Sound Source Distance Estimation Based on Statistical and Source-Specific FeaturesEleftheria Georganti, Philips Research Europe - Eindhoven, The Netherlands, University of Patras, Patras, Greece; Tobias May, Technische Universiteit Eindhoven - Eindhoven, The Netherlands; Steven van de Par, Aki Härmä, Philips Research Europe - Eindhoven, The Netherlands; John Mourjopoulos, University of Patras - Patras, Greece
In this paper we study the problem of estimating the distance of a sound source from a single microphone recording in a room environment. The room effect cannot be separated from the problem without making assumptions about the properties of the source signal. Therefore, it is necessary to develop methods of distance estimation separately for different types of source signals. In this paper we focus on speech signals. The proposed solution is to compute a number of statistical and source-specific features from the speech signal and to use pattern recognition techniques to develop a robust distance estimator for speech signals. Experiments with a database of real speech recordings showed that the proposed model is capable of estimating source distance with acceptable performance for applications such as ambient telephony.
Convention Paper 7689 (Purchase now)

P7-4 Implementation of DSP-Based Adaptive Inverse Filtering System for ECTF EqualizationMasataka Yoshida; Haruhide Hokari; Shoji Shimada, Nagaoka University of Technology - Nagaoka, Niigata, Japan
The Head Related Transfer Function (HRTF) and the inverse Ear Canal Transfer Function (ECTF) must be accurately determined if stereo earphones are realized out-of-head sound localization (OHL) with high presence. However, the characteristics of ECTF depend on the type of earphone used and the number of earphone mounting and demounting operations. Therefore, we present a DSP-based adaptive inverse filtering system for ECTF equalization in this paper. The buffer composition and size of DSP were studied so as to implement operation processing. As a result, we succeeded in constructing a system that was able to work in the audio-band of 15 kHz with the sampling frequency of 44.1 kHz. Listening tests clarified that the effective estimation error of the adaptive inverse-ECTF for OHL was less than –11 dB with convergence time of about 0.3 seconds.
Convention Paper 7690 (Purchase now)

P7-5 Improved Localization of Sound Sources Using Multi-Band Processing of Ambisonic ComponentsCharalampos Dimoulas, George Kalliris, Konstantinos Avdelidis, George Papanikolaou, Aristotle University of Thessaloniki - Thessaloniki, Greece
The current paper focuses on the use of multi-band ambisonic-processing for improved sound source localization. Energy-based localization can be easily delivered using soundfield microphone pairs, as long as free field conditions and the single omni-directional-point-source model apply. Multi-band SNR-based selective processing improves the noise tolerance and the localization accuracy, eliminating the influence of reverberation and background noise. Band-related sound-localization statistics are further exploited to verify the single or multiple sound-sources scenario, while continuous spectral fingerprinting indicates the potential arrival of a new source. Different sound-excitation scenarios are examined (single /multiple sources, narrowband / wideband signals, time-overlapping, noise, reverberation). Various time-frequency analysis schemes are considered, including filter-banks, windowed-FFT and wavelets with different time resolutions. Evaluation results are presented.
Convention Paper 7691 (Purchase now)

P7-6 Spatial Audio Content Management within the MPEG-7 Standard of Ambisonic Localization and Visualization DescriptionsCharalampos Dimoulas, George Kalliris, Kostantinos Avdelidis, George Papanikolaou, Aristotle University of Thessaloniki - Thessaloniki, Greece
The current paper focuses on spatial audio video/imaging and sound field visualization using ambisonic-processing, combined with MPEG-7 description schemes for multi-modal content description and management. Sound localization can be easily delivered using multi-band ambisonic processing under free-field and single point-source excitation conditions, offering an estimate on the achieved accuracy. Sound source forward propagation models can be applied in case that confident localization accuracy has achieved, to visualize the corresponding sound field. Otherwise, 3-D audio/surround sound reproduction simulation can be used instead. In any case, sound level distribution colormap-videos and highlighting images can be extracted. MPEG-7 adapted description schemes are proposed for spatial-audio audiovisual content description and management, facilitating a variety of user-interactive postprocessing applications.
Convention Paper 7692 (Purchase now)

Thursday, May 7, 18:30 — 19:30

Heyser Lecture
followed by
Technical Council

The Richard C. Heyser distinguished lecturer for the 126th AES Convention is Gunnar Rasmussen, a pioneer in the construction of acoustic instrumentation, particularly of microphones, transducers, vibration and related devices. He was employed at Brüel & Kjær Denmark as an electronics engineer immediately after his graduation in 1950. After holding various positions in development, testing, and quality control, he spent one year in the United States working for Brüel & Kjær in sales and service.

After his return to Denmark in the mid-1950s he began the development of a new measurement microphone. This resulted in a superior mechanical stability, increased temperature, and long term stability. The resulting one-inch pressure microphone soon became the de facto standard microphone for acoustical measurements to replace the famous W.E. 640AA standardized microphone.

The optimized mechanical design of the new generation of measurement microphones opened up the possibility for reducing the size of the microphones, first to a ½” microphone and then to ¼” and 1/8” microphones with essentially the same superior mechanical, temperature and long term stability. Notably the ½” microphone is still the most widely used measurement tool today. Since the beginning of the 1960’s, this microphone design has been preferred for all types of acoustic measurements and has formed the basis for the IEC 1094 series of international standards for measurement microphones.

Gunnar Rasmussen received the Danish Design Award in 1969 for his novel design of the microphones that were exhibited at the New York Museum of Modern Art. He also developed the first acoustically optimized sound level meter, where the shape of the body was designed to minimize the effect of reflections from the casing to the microphone. This type 2203 Sound Level meter was for many years seen as the archetype of sound level meters and its characteristic shape became the symbol of a sound level meter.

Other major inventions and designs include the Delta Shear accelerometer, the dual piston pistonphone calibrator for precision calibration, the face-to-face sound intensity probe and hydrophones, occluded ears, artificial mouth, etc. Rasmussen is also the author of numerous papers on acoustics and vibration and has served as chairman and vice-chairman of various international organizations and standard committees. In 1990 he received the CETIM medal for his contribution to the field of intensity techniques. He is also a Fellow of the Acoustical Society of America.

In 1994 Rasmussen started his own company, G.R.A.S. Sound and Vibration. Originally a company specializing in precision Outdoor Microphones for permanent noise monitoring around airports, it is now one of the world’s leading companies in acoustic front-ends and transducers forming a wide range of general purpose and specialized microphones, electro-acoustic measurement devices such as ear couplers, precision calibration tools and multi-dimensional sound intensity probes. The title of his lecture is, “The Reproduction of Sound Starts at the Microphone.”

The microphones may be developed for many specific purposes: for communication, recording or precision measurements. Quality may have different meaning for different applications. Price may be a dominating factor. Carbon microphones were dominating up to the 1950s. Electret microphones have taken the place of carbon microphones with great improvement in quality and performance at low prices. The MEMS microphones are on the way.

The challenge in the high quality microphone development is to match or exceed the human ear in perception of sound for measurement purposes. Without measurements we cannot qualify our progress. We are still trying to match the frequency band, the dynamic range, the phase linearity of the human ear and to obtain very good reproducibility in all situations where humans are involved. We need microphones for development, for standardized measurements and for legal related measurements. Where are we today?

Friday, May 8, 09:00 — 10:00

Spatial Audio

Friday, May 8, 10:00 — 11:00

Signal Processing

Friday, May 8, 10:00 — 13:30

TT3 - Herkulessaal der Residenz

A comparison of microphone settings for a live broadcast of a symphonic concert in 5.1 and stereo at the Bavarian Radio will be presented. The participants will have the opportunity to compare the settings themselves at the console and to make their own experiences with a 5.1 mix via multitrack recording under different ambience-mic-arrays. Presenters are Wolfram Graul and Klemens Kamp. Tour is limited to 10 people and transportation is not provided.

Price: Free

Friday, May 8, 10:30 — 12:00

P10 - Audio for Telecommunications

P10-1 Harmonic Representation and Auditory Model-Based Parametric Matching and its Application in Speech/Audio AnalysisAlexey Petrovsky, Elias Azarov, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Alexander Petrovsky, Bialystok Technical University - Bialystok, Poland
The paper presents new methods for the selection of sinusoids and transients components in hybrid sinusoidal modeling of speech/audio. The instantaneous harmonic parameters (magnitude, frequency, and phase) are calculated as the result of the narrow band filtering of speech/audio. The frequency-modulated filters synthesis with the closed form impulse response has been proposed. The filter frequency bounds can be determined during the components frequency tracking and can be adjusted according to the fundamental frequency modulations. It can be implemented speech/audio harmonic/noise decomposition. The transient components modeling are presented by matching pursuit with frame-based psychoacoustic optimized wavelet packet dictionary. The choice of most relevant coefficients is based on maximizing the matching between the auditory excitation scalograms of original and modeled signals.
Convention Paper 7705 (Purchase now)

P10-2 Perceptual Compression Methods for Metadata in Directional Audio Coding Applied to Audiovisual TeleconferenceToni Hirvonen, Institute of Computer Science (ICS) of the Foundation for Research and Technology - Hellas, Greece; Jukka Ahonen, Ville Pulkki, TKK - Finland
In teleconferencing application of Directional Audio Coding, the transmitted data consists of monophonic audio signal and directional metadata measured in frequency bands depending on time. In reproduction, each frequency channel of the signal is reproduced to corresponding direction with corresponding diffuseness. This paper examines methods for reducing the data rate of the metadata. The compression methods are based on psychoacoustic studies about the accuracy of directional hearing, and further developed and validated. Informal tests with one-way reproduction, as well as usability testing where an actual teleconference was arranged, were utilized for this purpose. The results indicate that the data rate can be as low as approximately 3 kbit/s without a significant loss in the reproduced spatial quality.
Convention Paper 7706 (Purchase now)

P10-3 Speaker Detection and Separation with Small Microphone ArraysMaximo Cobos, Jose J. Lopez, David Martinez, Universidad Politécnica de Valencia - Valencia, Spain
Small microphone arrays are desirable for many practical speech processing applications. In this paper we describe a system for detecting several sound sources in a room and enhancing a predominant target source using a pair of close microphones. The system consists of three main steps: time-frequency processing of the input signals, source localization via model fitting, and time-frequency masking for interference reduction. Experiments and results using recorded signals in real scenarios are discussed.
Convention Paper 7707 (Purchase now)

P10-4 Directional Audio Coding with Stereo Microphone InputJukka Ahonen, Ville Pulkki, TKK - Finland; Fabian Kuech, Giovanni Del Galdo, Markus Kallinger, Richard Schultz-Amling, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
The use of stereo microphone configuration as input to teleconference application of Directional Audio Coding (DirAC) is presented. DirAC is a method for spatial sound processing, in which the direction of the arrival of sound and diffuseness are analyzed and used for different purposes in reproduction. So far, omnidirectional microphones arranged in an array have been used to generate input signals for one- and two-dimensional sound field analysis in DirAC processing. In this study the possibility to use domestic stereo microphones with DirAC analysis is investigated. Different methods to derive omnidirectional and dipole signals from stereo microphones for directional analysis are presented and their applicability is discussed.
Convention Paper 7708 (Purchase now)

P10-5 Robust Noise Reduction Based on Stochastic Spatial FeaturesMitsunori Mizumachi, Kyushu Institute of Technology - Fukuoka, Japan
This paper proposes a robust noise reduction method relying on stochastic spatial features. Almost all of noise reduction methods have both strong and weak sides in the real world. In this paper time evolution of direction of arrival (DOA) and its stochastic reliability are the clues for selecting a suitable approach of noise reduction under time-variant noisy environments, where a DOA is an important spatial feature in beamforming for noise reduction. On the other hand, single channel approaches for noise reduction may be reasonable when DOA estimates are not reliable. Then, either spectral subtraction or beamforming is selected out for achieving robust noise reduction depending on a DOA estimate and its reliability. The proposed method had an advantage in noise reduction compared with a conventional approach.
Convention Paper 7709 (Purchase now)

Friday, May 8, 11:00 — 12:00

Microphones and Applications

Friday, May 8, 11:00 — 13:30

T6 - Loudness—Light at the End of the Tunnel

Florian Camerer, ORF, EBU Group P/LOUD
Eelco Grimm, Dutch Loudness Committee
Mike Kahsnitz, rtw
Ralph Kessler, Pinguin Engineering
Thomas Lund, tc electronic
Andrew Mason, BBC R&D

Audio levels in broadcasting have become increasingly diverse and different over the last decades. Despite clear guidelines and recommended practices the general use of peak measurement in audio metering and the development of more and more sophisticated level processors have led to over-compression of audio signals with the questionable aim of being louder than the competition. This attitude has especially impacted the audio quality of advertisements and promos with very little dynamic range. Already considered a hopeless situation, the introduction of loudness level metadata and especially the introduction of an international standard of loudness measurement (ITU-R BS.1770) is a light at the end of the tunnel. A few broadcasters and even whole countries have addressed the loudness issue thoroughly, and their experience shows that it is possible to solve that problem to the advantage of the consumer. It is long overdue to establish a new paradigm in audio levelling: the switch from peak normalization to loudness normalization. With widespread adoption of this approach consistent loudness not only within a channel, but also between different channels will be within reach—thus finding the “Holy Grail” of audio broadcasting.

In this session the current situation from the perspective of the EBU Group “P/LOUD” will be examined. Vendors will present their approaches to loudness metering.

Friday, May 8, 12:00 — 13:00

Loudspeakers and Headphones

Friday, May 8, 13:30 — 17:30

TT5 - Stadtmuseum Musikinstrumente

Museum of the City of Munich Instruments

The extraordinary collection of the Sammlung Musik-Münchner Stadtmuseum presents exhibits highlighting the construction of musical instruments from different cultures as well as a wide survey of the musical activities of mankind. On show are about 1500 musical instruments from Africa, Asia, the precolonial Americas, and Europe out of a total 6,000 objects. During the guided tour of the collections visitors have the opportunity to play the complete gamelans from the Indonesian Islands of Java and Bali.

Price: EUR 20

Friday, May 8, 13:30 — 17:30

TT6 - Herkulessaal der Residenz

A comparison of microphone settings for a live broadcast of a symphonic concert in 5.1 and stereo at the Bavarian Radio will be presented. The participants will have the opportunity to compare the settings themselves at the console and to make their own experiences with a 5.1 mix via multitrack recording under different ambience-mic-arrays. Presenters are Wolfram Graul and Klemens Kamp. Tour is limited to 10 people and transportation is not provided.

Price: Free

Friday, May 8, 13:30 — 15:00

P12 - Loudspeakers

P12-1 Reduction of Distortion in Conical Horn Loudspeakers at High LevelsSverre Holm, University of Oslo - Oslo, Norway; Rune Skramstad, Paragon Arrays - Drammen, Norway
Many horns have audible distortion at high levels. We measured a horn consisting of 6 conical sections with a 10-inch element at 99 dB SPL. A closed back gave maximum 2.4 percent second harmonic and 3.4 percent third harmonic distortion in the 100–1000 Hz range, while an open construction had 1.25 percent and 0.6 percent. A new semi-permeable back chamber reduced this to 0.7 percent and 0.35 percent. We hypothesize that the distortion is partly due to the non-linear compliance of air in the back chamber, and partly is due to the element’s interaction with the front and back loading of the horn, and that the new construction loads the element in a more optimal way.
Convention Paper 7717 (Purchase now)

P12-2 Comparison of Different Methods for the Subjective Sound Quality Evaluation of Compression DriversJosé Martínez, Acustica Beyma S.L. - Valencia, Spain; Joan Croañes, Escola Politecnica Superior de Gandia - Valencia, Spain; Jorge Francés Monllor Jaime Ramis, Universidad de Alicante - Alicante, Spain
In this paper an approach to the problem of sound quality evaluation of radiating systems is considered, applying a perceptual model. One of the objectives is to use the parameter proposed by Moore [. . .] to test if it provides satisfactory results when it is applied to the quality evaluation of indirect radiation loudspeakers. Three compression drives have been used for these proposals. Recordings with different test signals at different input voltages have been done. Using this experimental base, an approach to the problem from different points of view is done: [. . .] Taking in consideration classic sound quality parameters such as roughness, sharpness, and tonality. [. . .] Applying the parameter suggested by Moore obtained from the application of a perceptual model. Moreover, a psychoacoustic experiment has been made on a population of 25 people. The results, although preliminary and strongly dependant on the reference signal used to obtain Rnonlin, show a good correlation with the Rnonlin values.
Convention Paper 7718 (Purchase now)

P12-3 Membrane Modes in Transducers with the Direct D/A ConversionLibor Husník, Czech Technical University in Prague - Prague, Czech Republic
Operating principle of systems with the direct acoustic D/A conversion, which are sometimes called digital loudspeakers, brings new features to the field of transducer design. There are many design possibilities to these systems, using different transduction principles and spatial arrangement of constituting parts. This paper deals with the single-acting condenser transducer, suitable for micromachining applications, in which the membrane is driven by a partitioned back electrode. While in conventional transducers the electric force between the back electrode and the membrane is evenly distributed, in digital transducers it is no longer the case. Consequences to membrane vibrations for some cases of excitation by various distributions of forces representing given binary combinations from the dynamic level are presented.
Convention Paper 7719 (Purchase now)

P12-4 Increasing Active Radiating Factor of High-Frequency Horns by Using Staggered Arrangement in Loudspeaker Line ArrayKang An, Yong Shen, Aiping Zhang, Nanjing University - Nanjing, China
Active Radiating Factor (ARF) is an important parameter to analyze the loudspeaker line array when considering the gap between each two transducers, especially for high-frequency horns. As ARF is desired to be as high as possible, the staggered arrangement of horns is introduced in this paper. The responses in vertical direction and horizontal direction are analyzed. Compared with the conventional arrangement, the negative effects of gaps are reduced and responses are improved in simulation.
Convention Paper 7720 (Purchase now)

Friday, May 8, 14:00 — 15:00

Network Audio Systems

Friday, May 8, 14:00 — 16:00

T8 - Microphone History

Jörg Wuttke, Schoeps, Technical Director Emeritus
Ulrich Apel, Microtech Gefell GmbH
Sean Davies, S.W. Davies
Stephan Peus, Neumann GmbH

This tutorial will be presented in 3 parts.

Stephan Peus' presentation, "35 Years of Microphone Development at Neumann—What Touched Us, What Moved Us," gives an insight to specific development topics and to some very special test procedures including: microphone’s transient response: insights beyond frequency response or polar pattern; RF susceptibility: already a topic before the era of mobile phones; capsule distortion measurement: difficult procedure giving a lot of interesting results; dynamic range and self noise level of studio microphones: a remarkable development within the 35 years in question.

Ulrich Apel will report on "The Importance of Vacuum for Condenser Microphones." He will speak on such topics as: the electron-tube was and is still an important step in the development of condenser microphones; the construction of special-made tubes for use in mics such as RE084k, Hiller MSC2, Telefunken AC701k, EF804, Valvo EF86, 6072, etc.; and special measuring capabilities to select tubes regarding noise, stability. and sound.

Sean Davies' presentation is "Microphone History: The Why, The How, and The Who." The developments in microphone technology are reviewed from the earliest telephone based type through the decades as far as the 1970s. The “Why” section looks at the reasons behind the different designs, e.g., directional characteristics, output signal levels, diffraction effects, frequency range. The “How” examines the solutions proposed for the “Why” section, and the “Who” identifies the landmark designs and the designers behind them.

Friday, May 8, 15:00 — 16:00

Human Factors in Audio Systems

Friday, May 8, 15:00 — 16:30

W9 - Mixing Sports in 5.1–Part 1

Gerhard Stoll, IRT Munich
Dennis Baxter, Sound for the Olympics - USA
Beat Joss, tpc AG - Switzerland
Ales Koman, Slovenia TV - Slovenia

Sports has proven to be a major driving force for the introduction of HDTV into the market. Events like the Football World Championships 2006, the Football European Championships in 2008, and the 2008 Bejing Olympics provide an excellent opportunity to showcase the strengths of high resolution pictures. Teaming up with HD picture is High Definition Surround Sound and a generally more elaborate and creative sound design. The challenges are manifold:

• from capturing a roaring stadium crowd of 80,000 to hearing the details of every ball kick, the so-called “close ball”;
• from producing a surround mix to still serving your stereo-viewers with an appropriate downmix and an intelligible commentator;
• from getting your signal to and through your broadcast center unharmed to arriving at the consumer in sync with the picture;
• from making meaningful use of LFE to using the center channel not only for commentary but for the "sound of sports" as well.

A panel of experienced protagonists will discuss these issues and other challenges. Examples will give the audience the chance to judge the effectiveness themselves.

In Part 2 of this workshop a multitrack recording of one of the finals of the Swiss Ice-Hockey Championship 2009 will be mixed live for the audience to witness the approach to creating a compelling surround sound field, which includes the atmosphere of thousands of bawling, booing, and applauding fans and the “sounds of the match,” which you see on your screen.

Friday, May 8, 16:30 — 18:30

W10 - Audio Network Control Protocols

Richard Foss, Rhodes University - Grahamstown, South Africa
John Grant, Nine Tiles
Robby Gurdan, UMAN
Rick Kreifeldt, Harman Professional
Philip Nye, Engineering Arts

With the advent of digital audio networking, there has been a need to manage the connection of devices on networks and also to control and access various parameters of the devices. A number of standard and proprietary protocols have been developed. In this workshop a panel of experts who have helped develop some of these protocols will discuss their approaches and attempt to define a way forward, whereby devices with differing protocol implementations can communicate.

Friday, May 8, 17:00 — 18:30

W11 - Mixing Sports in 5.1–Part 2

Gerhard Stoll, IRT Munich
Dennis Baxter, Sound for the Olympics - USA
Beat Joss, tpc AG - Switzerland
Ales Koman, Slovenia TV - Slovenia

Sports has proven to be a major driving force for the introduction of HDTV into the market. Events like the Football World Championships 2006, the Football European Championships in 2008, and the 2008 Bejing Olympics provide an excellent opportunity to showcase the strengths of high resolution pictures. Teaming up with HD picture is High Definition Surround Sound and a generally more elaborate and creative sound design. The challenges are manifold:

• from capturing a roaring stadium crowd of 80,000 to hearing the details of every ball kick, the so-called “close ball”;
• from producing a surround mix to still serving your stereo-viewers with an appropriate downmix and an intelligible commentator;
• from getting your signal to and through your broadcast center unharmed to arriving at the consumer in sync with the picture;
• from making meaningful use of LFE to using the center channel not only for commentary but for the "sound of sports" as well.

A panel of experienced protagonists will discuss these issues and other challenges. Examples will give the audience the chance to judge the effectiveness themselves.

In Part 2 of this workshop a multitrack recording of one of the finals of the Swiss Ice-Hockey Championship 2009 will be mixed live for the audience to witness the approach to creating a compelling surround sound field, which includes the atmosphere of thousands of bawling, booing, and applauding fans and the “sounds of the match,” which you see on your screen.

Friday, May 8, 19:30 — 21:30


Isar Brau, Munchen Pullach

This year the Banquet will take place in a small old railway station, above the valley of the River Isar. The railway opened in 1891 and steam trains took people from the city to many beautiful places in the south of Munich. Today the steam trains have been replaced and the line is now part of the S-Bahn, so the old station is not needed anymore and has been turned into a traditional Bavarian style restaurant with its own micro-brewery. What could be more natural than making this location a pleasant place for a “get together” in a lovely atmosphere?

The welcome beer from the micro brewery and other drinks will be followed by a fine buffet with Bavarian delicacies. At the end of a long day at the Convention, these “Schmankerl” will be a good way to relax and enjoy the evening with old and new friends and colleagues. Come and savour Munich’s lifestyle. The ticket price includes all food and drinks and the bus to the restaurant and back.

55 Euros for AES members; 65 Euros for nonmembers
Tickets will be available at the Special Events desk.

Saturday, May 9, 09:00 — 11:30

P16 - Spatial Rendering–Part 1

Chair: Andreas Silzle

P16-1 An Alternative Ambisonics Formulation: Modal Source Strength Matching and the Effect of Spatial AliasingFranz Zotter, Hannes Pomberger, University of Music and Dramatic Arts - Graz, Austria; Matthias Frank, Graz University of Technology - Graz, Austria
Ambisonics synthesizes sound fields as a sum over angular (spherical/cylindrical harmonic) modes, resulting in the definition of an isotropically smooth angular resolution. This means, virtual sources are synthesized with outstanding smoothness across all angles of incidence, using discrete loudspeakers that uniformly cover a spherical or circular surface around the listening area. The classical Ambisonics approach models the fields of these discrete loudspeakers in terms of a sampled continuum of plane-waves. More accurately, the contemporary concept of Ambisonics uses a continuous angular distribution of point-sources at finite distance instead, which is considerably easier to imagine. This also improves the accuracy of holophonic sound field synthesis and the analytic description of the sweet spot. The sweet spot is a limited area of faultless synthesis emerging from angular harmonics truncation. Additionally, playback with loudspeakers causes spatial aliasing. In this sense, it allows for a successive consideration of the major shortcomings of Ambisonics: the limited sweet spot size and spatial aliasing. To elaborate on this concept this paper starts with the solution of the nonhomogeneous wave equation for a spherical point-source distribution, and ends with a novel study on spatial aliasing in Ambisonics.
Convention Paper 7740 (Purchase now)

P16-2 Sound Field Reproduction Employing Non-Omnidirectional LoudspeakersJens Ahrens, Sascha Spors, Deutsche Telekom Laboratories, Techniche Universität Berlin - Berlin, Germany
In this paper we treat sound field reproduction via circular distributions of loudspeakers. The general formulation of the approach has been recently published by the authors. In this paper we concentrate on the employment of secondary sources (i.e., loudspeakers) whose spatio-temporal transfer function is not omnidirectional. The presented approach allows us to treat each spatial mode of the secondary source’s spatio-temporal transfer function individually. We finally outline the general process of incorporating spatio-temporal transfer functions obtained from microphone array measurements.
Convention Paper 7741 (Purchase now)

P16-3 Alterations of the Temporal Spectrum in High-Resolution Sound Field Reproduction of Different Spatial BandwidthsJens Ahrens, Sascha Spors, Deutsche Telekom Laboratories, Techniche Universität Berlin - Berlin, Germany
We present simulations of the wave field reproduced by a discrete circular distribution of loudspeakers. The loudspeaker distribution is driven either with signals of infinite spatial bandwidth (as it happens in wave field synthesis), or the loudspeaker distribution is driven with signals of finite spatial bandwidth (as it is the case in near-field compensated higher order Ambisonics). The different spatial bandwidths lead to different accuracies of the desired component of the reproduced wave field and to spatial aliasing artifacts with essentially different properties. Our investigation focuses on the potential consequences of the artifacts on human perception.
Convention Paper 7742 (Purchase now)

P16-4 Cooperative Spatial Audio Authoring: Systems Approach and Analysis of Use CasesJens-Oliver Fischer, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Francis Gropengiesser, TU Ilmenau - Ilmenau, Germany; Sandra Brix, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany
Today’s audio production process is highly parallel and segregated. This is especially the case in the field of audio postproduction for motion pictures. The introduction of spatial audio systems like 5.1, 22.2 or Wave Field Synthesis results in even more production steps, namely the spatial authoring, to accomplish a rich experience for the audience. This paper proposes a system that enables the audio engineers to work together on the same project. The proposed system is planned to be implemented for an existing spatial authoring software but can be utilized by any other application that organizes its data in a tree structured way. Three major use cases, i.e., Single User, Work Space, and Work Group, are introduced and analyzed.
Convention Paper 7743 (Purchase now)

P16-5 Spatial Sampling Artifacts of Wave Field Synthesis for the Reproduction of Virtual Point SourcesSascha Spors, Jens Ahrens, Deutsche Telekom Laboratories, Techniche Universität Berlin - Berlin, Germany
Spatial sound reproduction systems with a large number of loudspeakers are increasingly being used. Wave field synthesis is a reproduction technique using a large number of densely placed loudspeakers (loudspeaker array). The underlying theory, however, assumes a continuous distribution of loudspeakers. Individual loudspeakers placed at discrete positions constitute a spatial sampling process that may lead to sampling artifacts. These may degrade the perceived reproduction quality and will limit the application of active control techniques like active room compensation. The sampling artifacts for the reproduction of plane waves have already been discussed in previous papers. This paper derives the spatial sampling artifacts and anti-aliasing conditions for the reproduction of virtual point sources on linear loudspeaker arrays using wave field synthesis techniques.
Convention Paper 7744 (Purchase now)

Saturday, May 9, 10:30 — 13:00

LS1 - Neumann & Müller and d & b

Stefan Goertz
Michael Kennedy

Sound System Design and Commissioning in Critical Acoustic Environments

The interaction of sound systems with special attention to the excitation of diffuse sound will be examined in theory and practical demonstrations using speech intelligibility measurements as an indicator. Software-aided line and subwoofer array designs will be discussed followed by a live demonstration of the tuning and
alignment process.

Saturday, May 9, 11:30 — 13:00

W13 - AES 42 Digital Microphones

Gregor Zielinsky, Sennheiser
Malgorzata Albinska-Frank
Stephan Flock, Direct Out
Stephan Peus, Neumann
Helmut Wittek, Schoeps

The panel will discuss daily work with digital microphones and their peripheral devices.

While the first digital microphones were launched a few years ago, their use is still within an exclusive community of users.

During the last two years, prices of digital microphones have dropped, while the choice of mics—as well as choices of interfaces—have risen very strongly. More and more companies are incorporating the AES 42 standard.

In this panel manufacturers and users discuss possibilities and workflows of digital microphones. Experienced users will give their view on the AES 42 standard.

Saturday, May 9, 13:30 — 17:30

TT10 - Staatsoper München

Munich’s “first” opera house, the Nationaltheater, shows its audio equipment. One studio for live-sound, one for production and broadcast, and one for recording are integrated in a digital network. Additionally, the back-stage installations will be shown.

Price: EUR 20

Saturday, May 9, 13:30 — 15:30

W14 - 5.1 High Profile Mixing

Akira Fukada, Senior Engineer, NHK - Tokyo, Japan
Ulrike Schwarz, Engineer, Bavarian Radio - Munich, Germany
Jean-Marie Geijsen, Director & Balance Engineer, Polyhymnia - Al Baarn, The Netherlands
Sascha Paeth, Owner/Engineer, Gate Studios - Wolfsburg, Germany
Ronald Prent, Residential Surround Engineer, Galaxy Studios - Mol, Belgium

It is very interesting how the perception of music can be altered by a mixing engineer. A conductor or a musician changes the figure of written music, the composer's work, by his or her interpretation and expression. For the recording of music it is rather important what the music conveys to the engineer. In the process of recording and mixing the engineer will approach and embrace the music like an artist or musician.

In this workshop engineers who have various cultural and musical backgrounds present their different mixing results. What did each engineer consider and what did they aim at? We believe that considering the result is a very important element for understanding music and art.

Saturday, May 9, 13:30 — 15:00

P19 - Event, Stage, and Sound Reinforcement

Chair: Francis Rumsey, University of Surrey - Guildford, Surrey, UK

P19-1 Comparative Evaluation of Howling Detection Criteria in Notch-Filter-Based Howling SuppressionToon van Waterschoot, Marc Moonen, Katholieke Universiteit Leuven - Leuven Belgium
Notch-filter-based howling suppression (NHS) is one of the most popular methods for acoustic feedback control in public address and hands-free communication systems. The NHS method consists of two stages: howling detection and notch filter design. While the design of notch filters is based on well-established filter design techniques, there is little agreement in the NHS literature on how the howling detection subproblem should be tackled. Moreover, since the NHS literature mainly consists of patents, only few experimental results have been reported. The aim of this paper is to describe a unifying framework for howling detection and to provide a comparative evaluation of existing and novel howling detection criteria.
Convention Paper 7752 (Purchase now)

P19-2 Professional Wireless Microphone Systems: Current Situation and Upcoming Changes in Regulatory Issues in Europe and USAFrank Ernst, Beyerdynamic GmbH & Co. KG - Heilbronn, Germany
Professional wireless microphones have been in use for almost 50 years now. The operation is based on a frequency sharing with TV broadcast transmitters. With the transition to digital TV, this situation changes. Digital TV is more spectrum efficient. After the transition is completed, areas of the spectrum, will be cleared from TV broadcast and will be available for new services. These cleared spectrum areas are referred to as “digital dividend” or white spaces. By the allocation of these bands to other services, valuable resources for the operation of professional wireless microphones will be lost. This paper will give an overview on the current situation for professional wireless microphones and the upcoming changes with the transition to digital TV.
Convention Paper 7753 (Purchase now)

P19-3 Sound Field Reconstruction: An Improved Approach for Wave Field SynthesisMihailo Kolundzija, Christof Faller, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland; Martin Vetterli, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland, University of California at Berkeley, Berkeley, CA, USA
Wave field synthesis (WFS) is a prevalent approach to multiple-loudspeaker sound reproduction for an extended listening area. Although powerful as a theoretical concept, its deployment is hampered by practical limitations due to diffraction, aliasing, and the effects of the listening room. Reconstructing the desired sound field in the listening area, accounting for the medium propagation characteristic, is another approach termed as sound field reproduction (SFR). It is based on the essential band-limitedness of the sound field, which enables a continuous matching of the reconstructed and the desired sound field by their matching on a discrete set of points spaced below the Nyquist distance. We compare the two approaches in a common single-source free-field setup, and show that SFR provides improved sound field reproduction compared to WFS in a wide listening area around a defined reference line.
Convention Paper 7754 (Purchase now)

Saturday, May 9, 14:00 — 15:00

Perception and Subjective Evaluation of Audio Signals

Saturday, May 9, 15:00 — 16:00

Electro Magnetic Compatibility

Saturday, May 9, 15:00 — 18:30

LS2 - Sennheiser & Yamaha – Live Sound Workshop

Svenja Dunkel
Horst Hartmann
Oliver Voges
Jürgen Wilhelm
Gregor Zielinsky

The workshop will cover two important aspects of PA:

(1) Digital mixing consoles for PA and monitors.

(2) A complete soundcheck of a live rockband on stage. The visitors will experience how high-end professionals do their soundcheck, including FOH, monitors, and wireless techniques.

The special task will be to get the tricky sound of the location under control. The hall Atrium 2 is very reverberant and is not an optimized condition. This is what pros have to work with every day. The band "Rauschenberger" is a new upcoming group from Hannover around singer and leader Rauschenberger, who has a splendid and very characteristic voice. After the soundcheck, there will be a half-hour concert.

Saturday, May 9, 15:00 — 18:30

P21 - Spatial Rendering–Part 2

Chair: Sascha Spors, Technical University of Berlin - Berlin, Germany

P21-1 Score File Generators for Boids-Based Granular Synthesis in CsoundEnda Bates, Dermot Furlong, Trinity College - Dublin, Ireland
In this paper we present a set of score file generators and granular synthesis instruments for the Csound language. The applications use spatial data generated by the Boids flocking algorithm along with various user-defined values to generate score files for grainlet additive synthesis, granulation, and glisson synthesis instruments. Spatialization is accomplished using Higher Order Ambisonics and distance effects are modeled using the Doppler Effect, early reflections, and global reverberation. The sonic quality of each synthesis method is assessed and an original composition by the author is presented.
Convention Paper 7761 (Purchase now)

P21-2 Acoustical Rendering of an Interior Space Using the Holographically Designed Sound ArrayWan-Ho Cho, Jeong-Guon Ih, KAIST - Daejeon, Korea
It was reported that the filter for the acoustic array can be inversely designed in a holographic way, which was demonstrated in a free-field. In this study the same method using the boundary element method (BEM) was employed to render the interior sound field in an acoustically desired fashion. Because the inverse BEM technique can deal with arbitrary shaped source or bounding surfaces, one can simultaneously consider the effect of irregular radiation surface and reflection boundaries having impedances such as walls, floor, and ceiling. To examine the applicability, a field rendering example was tested to control the relative spatial distribution of sound pressure in the enclosed field.
Convention Paper 7762 (Purchase now)

P21-3 Validation of a Loudspeaker-Based Room Auralization System Using Speech Intelligibility MeasuresSylvain Favrot, Jörg M. Buchholz, Technical University of Denmark - Lyngby, Denmark
A novel loudspeaker-based room auralization (LoRA) system has been proposed to generate versatile and realistic virtual auditory environments (VAEs) for investigating human auditory perception. This system efficiently combines modern room acoustic models with loudspeaker auralization using either single loudspeaker or high-order Ambisonics (HOA) auralization. The LoRA signal processing of the direct sound and the early reflections was investigated by measuring the speech intelligibility enhancement by early reflections in diffuse background noise. Danish sentences were simulated in a classroom and the direct sound and each early reflection were either auralized with a single loudspeaker, HOA or first-order Ambisonics. Results indicated that (i) absolute intelligibility scores are significantly dependent on the reproduced technique and that (ii) early reflections reproduced with HOA provide a similar benefit on intelligibility as when reproduced with a single loudspeaker. It is concluded that speech intelligibility experiments can be carried out with the LoRA system either with the single loudspeaker or HOA technique.
Convention Paper 7763 (Purchase now)

P21-4 Low Complexity Directional Sound Sources for Finite Difference Time Domain Room Acoustic ModelsAlexander Southern, Damian Murphy, University of York - York, UK
The demand for more natural and realistic auralization has resulted in a number of approaches to the time domain implementation of directional sound sources in wave-based acoustic modeling schemes such as the Finite Difference Time Domain (FDTD) method and the Digital Waveguide Mesh (DWM). This paper discusses an approach for implementing simple regular directive sound sources using multiple monopole excitations with distributed spatial positioning. These arrangements are tested along with a discussion of the characteristic limitations for each setup scenario.
Convention Paper 7764 (Purchase now)

P21-5 Binaural Reverberation Using a Modified Jot Reverberator with Frequency-Dependent and Interaural Coherence MatchingFritz Menzer, Christof Faller, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland
An extension of the Jot reverberator is presented, producing binaural late reverberation where the interaural coherence can be controlled as a function of frequency such that it matches the frequency-dependent interaural coherence of a reference binaural room impulse response (BRIR). The control of the interaural coherence is implemented using linear filters outside the reverberator’s recursive loop. In the absence of a reference BRIR, these filters can be calculated from an HRTF set.
Convention Paper 7765 (Purchase now)

P21-6 Design and Limitations of Non-Coincidence Correction Filters for Soundfield MicrophonesChristof Faller, Illusonic LLC - Lausanne, Switzerland; Mihailo Kolundzija, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland
The tetrahedral microphone capsule arrangement in a soundfield microphone captures a so-called A-format signal, which is then converted to a corresponding B-format signal. The phase differences between the A-format signal channels due to non-coincidence of the microphone capsules cause coloration and errors in the corresponding B-format signals and linear combinations thereof. Various strategies for designing B-format non-coincidence correction filters are compared and limitations are discussed.
Convention Paper 7766 (Purchase now)

P21-7 Generalized Multiple Sweep MeasurementStefan Weinzieri, Andre Giese, Alexander Lindau, TU Berlin - Berlin, Germany
A system identification by impulse response measurements with multiple sound source configurations can benefit greatly from time-efficient measurement procedures. An optimized method by interleaving and overlapping of multiple exponential sweeps (MESM) used as excitation signals was presented by Majdak et al. (2007). For single system identifications, however, much higher signal-to-noise ratios (SNR) can be reached with sweeps whose magnitude spectra are adapted to the background noise spectrum of the acoustical environment, as proposed by Müller & Massarani (2001). We investigated on which conditions and to what extent the efficiency of multiple sweep measurements can be increased by using arbitrary, spectrally adapted sweeps. An extension of the MESM approach toward generalized sweep spectra is presented, along with a recommended measurement procedure and a prediction of the efficiency of multiple sweep measurements depending on typical measurement conditions.
Convention Paper 7767 (Purchase now)

Saturday, May 9, 15:00 — 18:30

P22 - Microphones and Headphones

Chair: William Evans, University of Surrey - Guildford, Surrey, UK

P22-1 Frequency Response Adaptation in Binaural HearingDavid Griesinger, Consultant - Cambridge, MA, USA
The pinna and ear canals act as listening trumpets to concentrate sound pressure on the eardrum. This concentration is strongly frequency dependent, typically showing a rise in pressure of 20 dB at 3000 Hz. In addition, diffraction and reflections from the pinna substantially alter the frequency response of the eardrum pressure as a function of the direction of a sound source. In spite of these large departures from flat response, listeners usually report that a uniform pink power spectrum sounds frequency balanced, and loudspeakers are manufactured to this standard. But on close listening frontal pink noise does not sound uniform. The ear clearly uses adaptive correction of timbre to achieve these results. This paper discusses and demonstrates the properties and limits of this adaptation. The results are important for our experience of live music in halls and in reproduction of music through loudspeakers and headphones.
Convention Paper 7768 (Purchase now)

P22-2 Concha Headphones and Their Coupling to the EarLola Blanchard, Bang & Olufsen ICEpower s/a - Lyngby, Denmark; Finn T. Agerkvist, Technical University of Denmark - Lyngby, Denmark
The purpose of the study is to obtain a better understanding of concha headphones. Concha headphones are the small types of earpiece that are placed in the concha. They are not sealed to the ear and therefore, there is a leak between the earpiece and the ear. This leak is the reason why there is a significant lack of bass when using such headphones. This paper investigates the coupling between the headphone and the ear, by means of measurement in artificial ears and models. The influence of the back volume is taken into account.
Convention Paper 7769 (Purchase now)

P22-3 Subjective Evaluation of Headphone Target Frequency ResponsesGaëtan Lorho, Nokia Corporation - Finland
The effect of headphone frequency response equalization on listeners’ preference was studied for music and speech reproduction. The high-quality circum-aural headphones selected for this listening experiment were first equalized to produce a flat frequency response. Then, a set of filters was created based on two parameters defining the amplitude and center frequency of the main peak found around 3 kHz in the free-field and diffuse-field equalization curves. Two different listening tests were carried out to evaluate these equalization candidates using a different methodology and a total of 80 listeners. The results of this study indicate that a target frequency response with a 3 kHz peak of lower amplitude than in the diffuse-field response is preferred by listeners for both music and speech.
Convention Paper 7770 (Purchase now)

P22-4 Study and Consideration on Symmetrical KEMAR HATS Conforming to IEC60959Kiyofumi Inanaga, Homare Kon, Sony Corporation - Tokyo, Japan; Gunnar Rasmussen, Per Rasmussen, G.R.A.S. Sound & Vibration A/S - Holte, Denmark; Yasuhiro Riko, Riko Associates - Yokohama, Japan
KEMAR is widely recognized as a leading model of head and torso simulators (HATS) for different types of acoustic measurements meeting requirements of a global industrial standard, ANSI S3.36/ASA58-1985 and IEC 60959:1990. One of the KEMAR HATS pinna models has a reputation for good reproducibility of measured results in examining headphones and earphones. However, it requires free filed compensation in order to conduct the measurements; thus, the head-related transfer function (HRTF) of HATS fitted with the pinna model must be corrected. Because headphones and earphones are usually designed symmetrically, we developed a prototype of Symmetrical KEMAR HATS based on the original KEMAR mounted with the pinna model with good reproducibility. We measured and evaluated a set of HRTFs from the sound source to both ears. Our study concluded that the HATS we developed carries symmetrical characteristics and is also suitable to be utilized as a tool to measure the qualities of variety of acoustic devices along with the conventional KEMAR and it can serve as a new common platform for different types of electroacoustic measurements.
Convention Paper 7771 (Purchase now)

P22-5 Spatio-Temporal Gradient Analysis of Differential Microphone ArraysMihailo Kolundzija, Christof Faller, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland; Martin Vetterli, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland, University of California at Berkeley, Berkeley, CA, USA
The literature on gradient and differential microphone arrays makes a distinction between the two, and nevertheless shows how both types can be used to obtain the same response. A more theoretically sound rationale for using delays in differential microphone arrays has not yet been given. This paper presents a gradient analysis of the sound field viewed as a spatio-temporal phenomenon, and gives a theoretical interpretation of the working principles of gradient and differential microphone arrays. It shows that both types of microphone arrays can be viewed as devices for approximately measuring the spatio-temporal derivatives of the sound field. Furthermore, it also motivates the design of high-order differential microphone arrays using the aforementioned spatio-temporal gradient analysis.
Convention Paper 7772 (Purchase now)

P22-6 The Analog Microphone Interface and its HistoryJoerg Wuttke, Joerg Wuttke Consultancy - Pfinztal, Germany, Schoeps GmbH, Karlsruhe, Germany
The interface between microphones and microphone inputs has special characteristics and requires special attention. The low output levels of microphones and the possible need for long cables have made it necessary to think about noise and interference of all kinds. A microphone input is also the electrical load for a microphone and can have an adverse influence on the its performance. Condenser microphones contain active circuitry that required some form of powering. With the introduction of transistorized circuitry in the 1960s, it became practical for this powering to be incorporated into microphone inputs. Various methods appeared in the beginning; 48-Volt phantom powering is now the dominant standard, but this standard method is still not always implemented correctly.
Convention Paper 7773 (Purchase now)

P22-7 Handling Noise Analysis in Large Cavity Microphone Windshields—Improved SolutionPhilippe Chenevez, CINELA - Paris, France
Pressure gradient microphones are well known to be highly sensitive to vibrations. Respectable suspensions are made to create the best isolation possible, but when the microphone is placed inside a large cavity windshield, the external skin behaves as a drum excited by the vibrations of the support (boom or stand). As a consequence structure-borne noise is also transmitted acoustically to the microphone, due to its hard proximity effect. Some theoretical aspects and practical measurements are presented, in conjunction with a proposed improved solution.
Convention Paper 7774 (Purchase now)

Saturday, May 9, 15:30 — 17:00

Career/Job Fair

The Career Fair will feature several companies from the exhibit floor. All attendees of the convention, students and professionals alike, are welcome to come talk with representatives from the companies and find out more about job and internship
opportunities in the audio industry. Bring your resume!

Saturday, May 9, 18:30 — 19:00

Live Concert

The band featured in the Live Sound Workshop LS2, Rauschenberger, will continue to play after the workshop finishes, in a concert open to all attendees.

The band "Rauschenberger" is a new upcoming group from Hannover around singer and leader Rauschenberger, who has a splendid and very characteristic voice.

Sunday, May 10, 09:00 — 11:30

W19 - Audio over IP

Heinz-Peter Reykers, WDR Köln
Joost Bloemen, Technica Del Arte BV
Jeremy Cooperstock, McGill University - Montreal, Quebec, Canada
Gerald List, TransTel Communications GmbH
Peter Stevens, BBC R&D

In radio broadcasting most transmissions and communication between remote broadcast venues and the radio house are based on ISDN lines. Within TV the communication lines are also based on ISDN circuits. But the technology of ISDN networks is gradually being exchanged by IP networks and packet-based transmission. What are the consequences of this migration for the broadcasters? In this workshop the focus will be on practical demonstrations. The topics range from SIP-Servers and infrastructure issues to audio examples regarding latency and other related parameters. Quality of Service and current problems as well as possible solutions will be examined.

Sunday, May 10, 09:00 — 12:30

P25 - Sound Design and Processing

Chair: Michael Hlatky

P25-1 Hierarchical Perceptual MixingAlexandros Tsilfidis, Charalambos Papadakos, John Mourjopoulos, University of Patras - Patras, Greece
A novel technique of perceptually-motivated signal-dependent audio mixing is presented. The proposed Hierarchical Perceptual Mixing (HPM) method is implemented in the spectro-temporal domain; its principle is to combine only the perceptually relevant components of the audio signals, derived after the calculation of the minimum masking threshold, which is introduced in the mixing stage. Objective measures are presented indicating that the resulting signals have enhanced dynamic range and lower crest factor with no unwanted artifacts, compared to the traditionally mixed signals. The overall headroom is improved, while clarity and tonal balance are preserved.
Convention Paper 7789 (Purchase now)

P25-2 Source-Filter Modeling in Sinusoid DomainWen Xue, Mark Sandler, Queen Mary, University of London - London, UK
This paper presents the theory and implementation of source-filter modeling in sinusoid domain and its applications on timbre processing. The technique decomposes the instantaneous amplitude in a sinusoid model into a source part and a filter part, each capturing a different aspect of the timbral property. We show that the sinusoid domain source-filter modeling is approximately equivalent to its time or frequency domain counterparts. Two methods are proposed for the evaluation of the source and filter, including a least-square method based on the assumption of slow variation of source and filter in time, and a filter bank method that models the global spectral envelope in the filter. Tests show the effectiveness of the algorithms for isolation frequency-driven amplitude variations. Example applications are given to demonstrate the use of the technique for timbre processing.
Convention Paper 7790 (Purchase now)

P25-3 Analysis of a Modified Boss DS-1 Distortion PedalMatthew Schneiderman, Mark Sarisky, University of Texas at Austin - Austin, TX, USA
Guitar players are increasingly modifying (or paying someone else to modify) inexpensive mass-produced guitar pedals into boutique units. The Keeley modification of the Boss DS-1 is a prime example. In this paper we compare the measured and perceived performance of a Boss DS-1 before and after applying the Keeley All-Seeing-Eye and Ultra mods. This paper sheds light on psychoacoustics, signal processing, and guitar recording techniques in relation to low fidelity guitar distortion pedals.
Convention Paper 7791 (Purchase now)

P25-4 Phase and Amplitude Distortion Methods for Digital Synthesis of Classic Analog WaveformsJoseph Timoney, Victor Lazzarini, Brian Carty, NUI Maynooth - Maynooth, Ireland; Jussi Pekonen, Helsinki University of Technology - Espoo, Finland
An essential component of digital emulations of subtractive synthesizer systems are the algorithms used to generate the classic oscillator waveforms of sawtooth, square, and triangle waves. Not only should these be perceived to be authentic sonically, but they should also exhibit minimal aliasing distortions and be computationally efficient to implement. This paper examines a set of novel techniques for the production of the classic oscillator waveforms of analog subtractive synthesis that are derived from using amplitude or phase distortion of a mono-component input waveform. Expressions for the outputs of these distortion methods are given that allow parameter control to ensure proper bandlimited behavior. Additionally, their implementation is demonstrably efficient. Last, the results presented illustrate their equivalence to their original analog counterparts.
Convention Paper 7792 (Purchase now)

P25-5 Soundscape Attribute IdentificationMartin Ljungdahl Eriksson, Jan Berg, Luleå University of Technology - Luleå, Sweden
In soundscape research, the field’s methods can be employed in combination with approaches involving sound quality attributes in order to create a deeper understanding of sound images and soundscapes and how these may be described and designed. The integration of four methods are outlined, two from the soundscape domain and two from the sound engineering domain.
Convention Paper 7793 (Purchase now)

P25-6 SonoSketch: Querying Sound Effect Databases through PaintingMichael Battermann, Sebastian Heise, Hochschule Bremen (University of Applied Sciences) - Bremen, Germany; Jörn Loviscach, Fachhochschule Bielefeld (University of Applied Sciences) - Bielefeld, Germany
Numerous techniques support finding sounds that are acoustically similar to a given one. It is hard, however, to find a sound to start the similarity search with. Inspired by systems for image search that allow drawing the shape to be found, we address quick input for audio retrieval. In our system, the user literally sketches a sound effect, placing curved strokes on a canvas. Each of these represents one sound from a collection of basic sounds. The audio feedback is interactive, as is the continuous update of the list of retrieval results. The retrieval is based on symbol sequences formed from MFCC data compared with the help of a neural net using an editing distance to allow small temporal changes.
Convention Paper 7794 (Purchase now)

P25-7 Generic Sound Effects to Aid in Audio RetrievalDavid Black, Sebastian Heise, Hochschule Bremen (University of Applied Sciences) - Bremen, Germany; Jörn Loviscach, Fachhochschule Bielefeld (University of Applied Sciences) - Bielefeld, Germany
Sound design applications are often hampered because the sound engineer must either produce new sounds using physical objects, or search through a database of sounds to find a suitable sample. We created a set of basic sounds to mimic these physical sound-producing objects, leveraging the mind's onomatopoetic clustering capabilities. These sounds, grouped into onomatopoetic categories, aid the sound designer in music information retrieval (MIR) and sound categorization applications. Initial testing regarding the grouping of individual sounds into groups based on similarity has shown that participants tended to group certain sounds together, often reflecting the groupings our team constructed.
Convention Paper 7795 (Purchase now)

Sunday, May 10, 10:30 — 12:30

LS3 - Yamaha and d & b

Stefan Goertz
Arthur Koll


Line Source to Point Source Transformation—Technical Concept of the d&b T-Series

Increasing demands on flexibility, scalability, and efficiency in sound reinforcement applications encouraged one loudspeaker development configurable for point and line source applications by an easy mechanical
modification. Several implemented design technologies will be discussed before the listening demonstration of the performance of the system under critical acoustic conditions comparing Q-Series line arrays.


Modern IT-Compatible Audio Networks

IT-compatible audio networks and standards: Cobranet and Ethersound. Both formats will be discussed regarding their advantages and limitations. Advanced network strategies like VLAN programming offers high channel counts and a wide range of additional services via Gigabit audio networks. Now video, intercom, remote control, and DMX services may be included in a modern network infrastructure. Of course, such a network has to be as safe and stable as possible, so the redundancy concepts developed by the IT industry like link aggregation / trunking / spanning tree will be discussed.

Sunday, May 10, 12:00 — 14:00

W20 - Standards-Based Audio Networks Using IEEE 802.1 AVB

Edward Clarke, XMOS Semiconductor - Bristol, UK
Rick Kreifeldt, Harman International - UT, USA

Recent work by IEEE 802 working groups will allow vendors to build a standards-based network with the appropriate quality of service for high quality audio performance and production. This new set of standards, developed by the IEEE 802.1 Audio Video Bridging Task Group, provides three major enhancements for 802-based networks: (1) Precise timing to support low-jitter media clocks and accurate synchronization of multiple streams; (2) A simple reservation protocol that allows an endpoint device to notify the various network elements in a path so that they can reserve the resources necessary to support a particular stream; and (3) Queuing and forwarding rules that ensure that such a stream will pass through the network within the delay specified by the reservation. These enhancements require no changes to the Ethernet lower layers and are compatible with all the other functions of a standard Ethernet switch (a device that follows the IEEE 802.1Q bridge specification). As a result, all of the rest of the Ethernet ecosystem is available to developers, in particular, the various high speed physical layers (up to 10 gigabit/sec in current standards, even higher speeds are in development), security features (encryption and authorization), and advanced management (remote testing and configuration) features can be used. This workshop will outline the basic protocols and capabilities of AVB networks, describe how such a network can be used, and provide some simple demonstrations of network operation (including a live comparison with a legacy Ethernet network).

Sunday, May 10, 13:00 — 15:00

LS4 - Sennheiser

Matthias Fehr, DKE (German Comission for Electronics and IT-Techn.)
Hubert Eckart, DTHG (German Society of Theatre, Event, etc.)
Bruno Marx, APWTP (Association of Professional Wireless Production Technologies)
Walter Möller, BLM, TKLM (Technical Conference of State Media Authority)
Florian von Hofen, VPLT (Society of Private Light and Soundtechique)

Loss of Spectrum for Wireless Production Tools

The WRC 07, World Radio Conference 2007, has identified the Spectrum 790 – 862MHz for IMT, International Mobile Telecom. In Germany more than 630,000 users operate in that spectrum: they need a new home for their activities.

The discussion will focus on the alternative spectrum that has to be opened and the transition time that can be expected.

For how long can the quality of wireless production tools be guaranteed?
Experts of the Federal Network Agency, the IRT, and the APWPT user organization and from the Manufacturers will give Spectrum users an outlook on expected changes.

Sunday, May 10, 14:30 — 17:00

T10 - Audio System Grounding & Interfacing—An Overview

Bill Whitlock

Although the subject has a black art reputation, this tutorial replaces myth and hype with insight and knowledge, revealing the true causes of system noise and ground loops. Although safety must be the top priority, some widely used cures are both illegal and deadly. Both balanced and unbalanced interfaces are vulnerable to noise coupling, but the unbalanced interface is exquisitely so due to an intrinsic problem. Because balanced interfaces are widely misunderstood, their theoretically perfect noise rejection is severely degraded in most real-world systems. Some equipment, because of an innocent design error, has a built-in noise problem. A simple, no-test-equipment, troubleshooting method can pinpoint the location and cause of system noise. Ground isolators in the signal path solve the fundamental noise coupling problems. Also discussed are unbalanced to balanced connections, RF interference, and power line treatments such as technical power, balanced power, isolation transformers, and surge suppressors.