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Friday, May 4 13:30 Room 142/143 previously released in mono or stereo due to the Standards Committee Meeting on SC-02-02 Digital high effort necessary or simply because the origi- Input/Output Interfacing nal multitrack recordings are unavailable. Conse- quently the need for tools that allow an automated Session P1 Saturday, May 5 09:00 – 12:00 or semi-automated stereo to surround up-mix is Room I growing. In this paper a novel approach is de- scribed that is based on technology that is part of SPATIAL AUDIO PERCEPTION AND PROCESSING, the MPEG Surround standard. The basic algo- PART 1 rithm and some proposed extensions are outlined and potential use-cases described. Finally, the Chair: Thomas Sporer, Frauenhofer IDMT, Ilmenau, subjective quality of the presented approach is Germany compared to existing solutions. Convention Paper 6991 09:00 10:00 P1-1 Using Transient Suppression in Blind Multichannel Up-Mix Algorithms—Andreas P1-3 Coding of “2+2+2”Surround Sound Content Walther, Christian Uhle, Sacha Disch, Using the MPEG Surround Standard— Fraunhofer Institute IIS, Erlangen, Germany Andreas Ehret,1 Alexander Gröschel,1 Heiko Purnhagen,2 Jonas Roedén2 In the field of blind up-mixing, many algorithms 1Coding Technologies, Nuremberg, Germany exist for generating multichannel sound from 2Coding Technologies, Stockholm, Sweden mono or stereo sources. One of the important blind up-mix scenarios is the ambience-based An increasing number of recordings is available up-mix. This approach aims at extracting the am- in the so-called 2+2+2 surround sound format, bient parts of a given signal and their reproduc- where in addition to two front and two rear loud- tion by taking the best possible advantage of a speakers, a third pair of loudspeakers is placed multichannel loudspeaker setup. Depending on at an elevated position above the front speakers. the number of channels and the signal charac- In 2006, the MPEG Surround standard was final- teristics of the input signal, the quality of the ex- ized as an efficient stereo backward-compatible tracted ambience can vary. In this paper tran- surround sound coding format. The present pa- sient suppression is suggested as a method for per studies the applicability of MPEG Surround improving an extracted ambience signal. Two for efficient coding of “2+2+2” content. Several methods for suppressing transient components alternative approaches are outlined and evaluat- are proposed and contrasted to existing tech- ed by means of subjective listening tests. niques. The ability of those methods to improve Convention Paper 6992 the perceived quality of the ambience signal and overall up-mix is evaluated in a subjective listen- 10:30 ing test. Convention Paper 6990 P1-4 Quality Taxonomies for Auditory Virtual Environments—Andreas Silzle, Ruhr-Universität 09:30 Bochum, Bochum, Germany P1-2 A Novel Approach to Up-Mix Stereo to The aim of the here developed new taxonomies is Surround Based on MPEG Surround to describe the components involved in the quality Technology—Heiko Purnhagen,1 Andreas process of Auditory Virtual Environments (AVE) Ehret,2 Jonas Rödén,1 Alexander Gröschel2 and to quantify the relations between them for dif- 1Coding Technologies, Stockholm, Sweden ferent applications. The taxonomy should allow an 2Coding Technologies, Nuremberg, Germany overview and identify the relations that are most important in the software development process With the increasing number of installed surround and the design of listening experiments. For the sound systems in consumer homes and cars, the first time the multivariate relations between the demand for surround sound content is rising. How- quality elements in the physical domain and the ever, the vast majority of content is still only avail- quality features in the perceptual domain of such a able in stereo. Furthermore, in many cases it is dif- quality taxonomy are evaluated for three different ficult to create a surround mix for content AVE applications. This evaluation and quantifica- ¯ Audio Engineering Society 122nd Convention Program, 2007 Spring 1

Technical Program tion is done by means of an expert survey (DEL- Panelists: Sylvain Choisel, Bank & Olufsen a/s, Struer, PHI method) to objectify the results. Principal com- Denmark ponent analysis reveals that five dimensions are Garmt Dijksterhuis, Unilever Food & Health necessary to describe about 95 percent of the Research Institute, Vlaardingen, The variance in the data. This indicates that the select- Netherlands ed seven quality features are clearly distinguish- Natanya Ford, Bang and Olufsen a/s, Struer, able for the experts, but not orthogonal to each Denmark other. Most of the quality features are introduced William Martens, McGill University, Montreal, in meaningful terms in the audio engineering field Quebec, Canada and therefore usable without training for the partic- ipating experts. The results of the expert survey As knowledge about the listener experience of audio ap- are compared to listening test results, which use plications is fundamental in research and product design the same quality features. The bottom line is that within the audio field, methods that can be used for eval- the expert survey is not only a much faster method uation of perceived sound quality are essential to to get a good overview about a specific application explore. In order to capture and quantify listener experi- as compared to the listening test, but it also ence, different methodological approaches are utilized. reveals more information about it. One of the approaches involves development of individ- Convention Paper 6993 ual vocabulary of test subjects. This considers a collec- tion of techniques that can be used for evaluating the 11:00 detailed perceptual characteristics of products or sys- tems through listening tests. This workshop aims to pro- P1-5 Individual Localization Behavior for Perception 1 vide guidance to the researcher and experimenter of Virtual Sound Sources—Cornelius Bradter, regarding the nature of descriptive analysis by means of Klaus Hobohm2 1 individual vocabulary development techniques and their University of Applied Science, Berlin, Germany application in audio. 2Film and Television Academy, Potsdam, Germany Test results normally indicate very large variability Workshop 2 Saturday, May 5 in perception of lateral virtual sound sources with a 09:00 – 12:00 Room H 5-channel loudspeaker setup. Three recent studies indicate that up to a third of stimuli from a side THE PRACTICE OF AUDIO FORENSICS loudspeaker pair were perceived surprisingly accu- rately as virtual sound sources. In other cases Cochairs: Syliva Moosmüller, Acousics Research sound sources were perceived as coming from Institute, Vienna, Austria only one loudspeaker or from its vicinity. Therefore, Richard Sanders, UCDHSC, Denver we specified prototypical localization behaviors. We Colorado, CO, USA examined effects on localization by reproduction rooms, exact position of test persons in relation to loudspeakers, test persons’ head movements, and Panelists: Durand Begault, NASA Ames Research trading between time delay and level differences. Center, Mountainview, CA, USA Convention Paper 6994 Eddy B. Brixen, EBB-consult, Smørum, Denmark 11:30 Catalin Grigoras, Forensic Examiner, Bucharest, Romania P1-6 Comparison of Different Sound Capture Gordon Reid, CEDAR Audio, Cambridge, UK and Reproduction Techniques in a Virtual Acoustic Window—Timo Haapsaari, Werner This workshop will examine the common challenges of De Bruijn, Aki Härmä, Philips Research an audio forensic professional. Topics will include voice Laboratories, Eindhoven, The Netherlands identification, analog and digital media authenticity, audio enhancement, and adaptive filtering and running a foren- In this paper we describe a two-way audio com- sic business. This three hour workshop will include Q & munication system using arrays of microphones A during and after each 30-minute presentation. Audio and loudspeakers to create a virtual acoustic and visual examples will be shown where appropriate. window. We compare three different methods for Software will be demonstrated in real-time and input capturing the sound: wave field sampling using a along with questions from the attendees will be encour- line array of microphones, an adaptive beam- aged. All of the presenters are considered to be experts former, and close-talk microphones. For sound in audio forensics and some related fields. reproduction, we employ wave field synthesis. In the paper we review the acoustic and perceptual requirements for a real-time virtual acoustic win- Saturday, May 5 09:00 Room 142/143 dow system and report results of a set of listen- Standards Committee Meeting on SC-05-02 Audio ing experiments performed with the system. Connectors Convention Paper 6995 Session P2 Saturday, May 5 09:30 – 10:30 Workshop 1 Saturday, May 5 Room K 09:00 – 11:30 Room P AUDIO RECORDING AND REPRODUCTION AUDIO QUALITY EVALUATION—DESCRIPTIVE Chair: Nadja Wallaszkovits, Austrian Academy of ANALYSIS 2: INDIVIDUAL VOCABULARY Sciences, Vienna, Austria DEVELOPMENT Chair: Jan Berg, Luleå University of Technology, 09:30 Luleå, Sweden P2-1 Recording of Acoustical Concerts Using a

2 Audio Engineering Society 122nd Convention Program, 2007 Spring Soundfield Microphone—Markus Schellstede,1 give an overview of the receiver penetration in various Christof Faller2 countries. 1Sunhoe Pro Audio, Bern, Swtizerland The panel will also discuss future directions 2Illusonic LLC, Chavannes, Switzerland Based on extensive practical experience of one TECHNICAL TOUR 1 of the authors, recording of acoustical concerts, ORF Radio Broadcasting Studios using a soundfield microphone, without spot or Saturday, May 5, 09:30 – 12:30 support microphones is discussed. The focus is stereo recording of classical music. Strategies ORF Radio is the National Public Broadcaster for Austria, for positioning of the microphone, B-Format with three nationwide, nine regional, and two MW/SW decoding, and mastering are presented. The so- stations. The tour to the Radio Broadcasting House gives obtained final mix is largely based on the natural access to the music recording studios, continuity studios mix of sound reaching the microphone. This is in for three different radio stations, and insight into the practi- contrast to more conventional recording tech- cal 5.1-Surround sound operation of the satellite radio niques that usually use a large number of spot channel OE1DD. and support microphones. Last but not least, lim- itations and cost considerations are discussed. Convention Paper 6996 Saturday, May 5 09:30 Room 633 Technical Committee Meeting on Electro-Magnetic Compatibility 10:00 P2-2 Ambience Sound Recording Utilizing Dual MS (Mid-Side) Microphone Systems Based Session P3 Saturday, May 5 10:00 – 11:30 upon Frequency Dependent Spatial Cross Foyer IK Correlation (FSCC)—Teruo Muraoka, Takahiro Miura, Tohru Ifukube, University of Tokyo, POSTERS: LOW BIT-RATE AUDIO CODING Tokyo, Japan In musical sound recording for CD production or broadcasting, a forest of microphones is com- 10:00 monly observed. They are for good sound local- P3-1 Enhanced MPEG-4 Low Delay AAC—Low Bit- ization and favorable ambience, however it is Rate High-Quality Communication—Markus desirable to make the forest sparse for less labo- Schnell,1 Ralf Geiger,1 Markus Schmidt,1 rious setting up and mixing. Previously, the Manuel Jander,1 Markus Multrus,1 Gerald authors examined ambience representation of Schuller,2 Jürgen Herre1 stereophonic microphone arrangements utilizing 1Fraunhofer IIS, Erlangen, Germany frequency dependent spatial cross correlation 2Fraunhofer IDMT, Ilmenau, Germany (FSCC). FSCC is defined as a cross correlation of outputs by two microphones that of MS micro- The MPEG-4 Low Delay Advanced Audio Cod- phone system is most favorable. Based upon the ing (AAC-LD) scheme has recently evolved into result, we devised a combination of two MS a popular algorithm for audio communication. It microphone systems, one for picking up stage produces excellent audio quality at bit rates be- sounds and the other for ambience representa- tween 64 kbit/s and 48 kbit/s per channel. This tion. In addition to minor stage microphones, the paper introduces an enhancement to AAC-LD authors achieved satisfactory musical recording. that reduces the bit rate demand by 25-to-33 Convention Paper 6997 percent. This is achieved by adding both a delay-optimized version of the Spectral Band Replication (SBR) tool and by utilizing a dedicat- ed low delay filterbank. The introduced tech- Workshop 3 Saturday, May 5 niques maintain the high audio quality and offer 09:30 – 11:30 Room G an algorithmic delay low enough for use in two way communication systems. This paper SURROUND FOR BROADCASTING: AN OVERVIEW describes the coder enhancements including a detailed discussion of algorithmic delay issues, Cochairs: Kimio Hamasaki, NHK Science and a performance assessment, and possible Technical Research Laboratories, Tokyo, Japan applications. Lars Jonsson, SR/Swedish Radio Convention Paper 6998

Panelists: Steve Church, Telos 10:00 Eric Lundbeck, SVT - Swedish Television Tony Spath, Dolby Laboratories, UK P3-2 On the Design of Low Power MPEG-4 HE- Gerhard Stoll, IRT, Munich, Germany AAC Encoder—Wen-Chieh Lee, Chung-Han Yang, Cheng-Lun Hu, National Chiao Tung This workshop will review proprietary systems and University, Hsinchu, Taiwan MPEG Surround/spatial standards. Which systems are in use? Which systems will come? Spectral Band Replication (SBR) has been com- TV and radio broadcasters are using distribution sys- bined with MPEG AAC as a bandwidth extension tems for transmitting 5.1 surround sound with different tool. The resulting scheme is referred to as the audio coding methods. The workshop will review the cur- MPEG-4 High Efficient (HE) AAC or aacPlus. rent state of the art of used systems and report from With the SBR module taking care of the high fre- broadcasters’ experience of starting up new services and quency contents, the conventional AAC ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 3 Technical Program encoder can compress the low frequency part 10:00 using most of the available bits. The SBR para- meters are all calculated by SBR encoder in P3-5 A Hybrid Warped Linear Prediction (WLP) complex domain in the architecture of conven- AAC Audio Coding Algorithm—Jaeseong Lee, tional QMF. If the components in the SBR Young-Cheol Park, Dae-Hee Youn, Hong-Goo encoder can be implemented in the real domain, Kang, Yonsei University, Seoul, Korea the computational complexity of HE-AAC will be We propose a hybrid warped linear prediction reduced by half. The paper proposes a low pow- (WLP) AAC audio coding algorithm. The pro- er MPEG-4 HE-AAC encoder to reduce the com- posed algorithm employs a warped linear predic- putational complexity. Both subjective and objec- tion (WLP) processor to construct a perceptual tive experiments are conducted to demonstrate pre- and post-filter for the MPEG-4 AAC. The the quality of the low power HE-AAC encoder on WLP residue is applied to the MDCT filter-bank, critical music tracks. and the signal-to-mask ratio (SMR) of the corre- Convention Paper 6999 sponding block is modified to set a masking threshold for the WLP residues. In the decoder, 10:00 the reconstructed residual signal is passed to a modified WLP synthesis filter to restore the P3-3 High Quality, Low Power QMF Bank Design audio signal. Subjective tests show that the pro- for SBR, Parametric Coding, and MPEG posed audio codec operating at 50 kbps has Surround Decoders—Hsin-Yao Tseng, comparable perceptual quality to the convention- Han-Wen Hsu, Chi-Min Liu, National Chiao Tung al MPEG-4 AAC operating at the 58 kbps. University, Hsin-Chu, Taiwan Convention Paper 7002 Due to the alias-free properties, the complex quadrature mirror filter (QMF) bank has been 10:00 used in MPEG-4 audio standard on SBR, para- metric, and surround coding. The high com- P3-6 Comparison of Stereo Redundancy Reduction plexity overhead from the complex QMF bank Schemes for an Ultra Low Delay Audio Coder—Tobias Albert,1 Gerald Schuller,2 Stefan and the complex data processing in the 2 2 2 decoder leads to the development of a low Wabnik, Ulrich Krämer, Jens Hirschfeld 1Fraunhofer IIS, Erlangen, Germany power decoder, which adopts the real QMF 2 bank as the basic building module to reduce Fraunhofer IDMT, Ilmenau, Germany the complexity. However, the artifacts from the In the Fraunhofer Ultra Low Delay Audio Coder aliasing in the real QMF bank are the major (ULD) a pre-filter that is controlled by a psychoa- concern. This paper studies the artifacts from coustic model is followed by a quantizer and a the real QMF bank and proposes a novel QMF predictive coder to code signals in the time- bank design to achieve both low complexity domain. The output of the predictor is entropy and high quality. Also, this paper applies the coded and transmitted. Predictor and entropy novel QMF bank to develop the high-quality coder form the lossless redundancy reduction and low-power SBR, parametric, and MPEG part of the coder. Our goal is to improve the surround decoders and shows the merits in lossless redundancy-reduction part for stereo complexity and quality. signals. We present and evaluate six different Convention Paper 7000 alternatives for the stereo redundancy reduction, and we combine those alternatives to obtain a 10:00 higher compression ratio. Convention Paper 7003 P3-4 Low Power Stereo Perceptual Audio Coding Based on Adaptive Masking Threshold Reuse—Evelyn Kurniawati, Sapna George, ST 10:00 Microelectronics Asia Pacific Pte. Ltd., Singapore P3-7 Speech Codec Enhancements Utilizing Time The term perceptual audio coder refers to audio Compression and Perceptual Coding—Maciej compression schemes that exploit the proper- Kulesza, Andrzej Czyzewski, Gdansk University ties of human auditory perception. The idea is of Technology, Gdansk, Poland to allocate the quantization noise elegantly be- A method for encoding wideband speech signal low the masking threshold to make it impercep- employing standardized narrowband speech tible to the ear. The process requires consider- codecs is presented as well as experimental able computational effort, especially due to the results concerning detection of tonal spectral psychoacoustics analysis and bit allocation- components. The speech signal sampled with a quantization process. This paper proposes a higher sampling rate than it is suitable for nar- new method to simplify the psychoacoustics rowband coding algorithm is compressed in modeling process by adaptively reusing the order to decrease the amount of samples. Next, computed masking threshold depending on the the time-compressed representation of a signal signal characteristics. The method also devices is encoded using a narrowband speech codec. a scheme to patch the potential spectral hole The time expansion procedure is applied to the problems that might occur when the quantiza- speech signal after transmission and decoding in tion parameters are reused. This proposal can order to restore original time relations. Finally, be applied to generic stereo perceptual audio the wideband speech signal is presented to the encoders where low computational complexity user. The method for spectral envelope estima- is required. tion involving perceptual criteria is described. Convention Paper 7001 The algorithms for tonal components detection

4 Audio Engineering Society 122nd Convention Program, 2007 Spring were evaluated and compared during experi- Sunday, May 6, 10:30 – 13:30 and 14:30 – 16:45 ments carried-out. University for Music and Performing Arts, Vienna Convention Paper 7004 CELEBRATE GOOD VIBES! 10:00 Hosted by the University for Music and Performing Arts, Vienna and supported by the partners Fraunhofer Insti- P3-8 Design and Implementation of a Web-Based tute for Digital Media Technology IDMT, Sennheiser, Software Framework for Real Time Intelligent Studer, and Merging Technologies, this Special Event Audio Coding Based on Speech/Music will present live concerts and their reproduction in Wave Discrimination—Jose Enrique Muñoz Exposito, Field Synthesis alongside. Nicolas Ruiz Reyes, Sebastian Garcia-Galan, Attendees will be able to directly compare the sound Pedro Vera Candeas, University of Jaén, Jaén, inside the hall with its recorded and live mixed counter- Spain part. Therefore, renowned musicians—including Alvaro In this paper a software framework based on Pierri (classical guitar) and the tango orchestra Band-O- client-server architecture is implemented for real Neon—will play pieces from their repertoire in the time intelligent audio coding. A speech/music Joseph Haydn Concert Hall. Next door, in the Batiken discrimination scheme analyzes the input audio Hall, the recorded and mixed session will be played back signal and takes a decision about the nature of using a WFS system with more than 90 loudspeakers. the audio signal (speech or music) on a frame by Different microphone techniques can be compared frame basis. According to the decision of the and evaluated. Neumann/Sennheiser microphones, a speech/music discriminator, a suitable coder is Vista 5 console by Studer, the Merging Technologies selected at each frame. The designed software Pyramix Digital Audio Workstation, and the Spatial Audio framework makes use of the speech and audio Workstation by Fraunhofer IDMT will provide a profes- coders incorporated into the MPEG4 audio stan- sional recording and mixing environment. dard (HVXC or CELP for speech frames and This Special Event will be held in four sessions on May TwinVQ or AAC for music frames) to evaluate 5 and May 6 at the University of Music Vienna. A bus the performance of an intelligent multimode au- shuttle service will be offered between the Austria Center dio coder. The framework supports several types and the University. of audio features (timbral texture features and Tickets will be available from the Technical Tour rhythmic content features) and classifiers (classi- counter. Meeting point is the main entrance of the Aus- cal Statistical Pattern Recognition (SPR) classi- tria Center. Please note that the times listed above fiers, Multilayer Perceptron Neural Networks include travel time. (MLPNN), Support Vector Machines (SVM), Additional Demo Sessions in Wave Field Synthesis will Fuzzy Expert Systems (FES), Hidden Markov also be available at the following times: Models (HMM)). Comparison between a • Saturday, May 5, 17:15 – 18:15 speech/music discrimination based-intelligent • Sunday, May 6, 17:15 – 18:15 audio coder and MPEG4-AAC has been per- formed using audio signals representative of the Saturday, May 5 10:30 Room 633 two corresponding classes (speech and music). Technical Committee Meeting on Archiving, Subjective and objective tests have been Restoration, and Digital Libraries accomplished aiming at assessing the behavior of the intelligent audio coding scheme. Saturday, May 5 11:00 Room 142/143 Convention Paper 7005 Standards Committee Meeting on SC-05-05 Grounding and EMC Practices 10:00 P3-9 Quantization of Laguerre-Based Stereo Linear SPECIAL EVENT Predictors—Albertus C. den Brinker,1 Arijit Awards Presentation and Keynote Address Biswas2 Saturday, May 5, 12:00 – 13:30, Room K 1Philips Research Laboratories, Eindhoven, Opening Remarks: The Netherlands • Executive Director Roger Furness 2Technische Universiteit Eindhoven, Eindhoven, • President Wieslaw Woszczyk The Netherlands • Convention Chair Werner Deutsch Recently a quantization scheme for stereo linear Program: prediction systems was proposed and was test- • AES Awards Presentation ed using random data as input. This research is • Introduction of Keynote Speaker by Convention extended in the current paper by incorporating Vice Chair Florian Camerer Laguerre filters in the stereo linear prediction • Keynote Address by Helmut Voitl, Documentary scheme. First, it is shown that the associated and Feature Film Director, “Sound and Emotion: normalized reflection matrices (NRM) can be The Power of Audio in Storytelling obtained efficiently. Second, the system was tested using stereo audio data in order to gain Awards Presentation an insight into the required bit rates for practical Please join us as the AES presents special awards to applications. those who have made outstanding contributions to the Convention Paper 7006 Society in such areas of research, scholarship, and pub- lications, as well as other accomplishments that have contributed to the enhancement of our industry. SPECIAL EVENT Recording and Mixing in Wave Field Synthesis Keynote Speaker Saturday, May 5, 10:30 – 13:30 and 14:30 – 16:45 This year’s Keynote Speaker is Helmut Voitl. Voitl was ¯ Audio Engineering Society 122nd Convention Program, 2007 Spring 5 Technical Program born in Dresden and grew up in Vienna, studying photog- the cradle of sound recording, nobody would raphy and film at the Graphische Lehr- und Versuch- have imagined a recording and entertaining sanstalt. He scripted and directed his first short film in industry which today serves one of the greatest 1966 and directed his first documentary series in 1968. markets of the world. Sound recording was the Voitl teamed up with Elisabeth Guggenberger in 1973, result of scientific interest predominantly aimed and they began making films for numerous TV produc- at studying the nature of spoken language. And tions commissioned by ORF, SRG, ZDF, BR, and NDR. it was scholars—linguists, anthropologists, and With the productions Russkiy Chleb and Schto delat, musicologists—who systematically employed Towarisch? in 1989 and a documentary of the Ukraine in sound recording technology from its very begin- 1990 began an involvement with the former Soviet ning. Consequently, the academic world played Union, finding its climax in the re-enactment series Arctic a key role in founding sound archives from North-East (ORF, 1992–1996), a dramatized and staged around 1900 onward, and the longevity of sound reconstruction of the Austrian polar expedition of recordings was given special emphasis, specifi- 1872–1874 that led to the discovery of the high arctic cally in the archives of Vienna and Berlin. archipelago Franz Joseph Land. Shooting took place at The emerging phonographic industry, howev- original locations in the Russian High Arctic. Another pro- er, shaped the development of sound recording duction in Russia followed in 1998, Russia’s Holy War, technology since that time; yet the permanence and a year later A-Watch, which dealt with the environ- of the record that once had attracted the scholar- mental state of affairs in Austria. Currently Helmut Voitl is ly world was not among the driving forces, par- working on a feature film project called White Clouds ticularly not in the development of magnetic tape Island, also situated in the High Arctic. recording. Only in the late 1950s, when libraries Voitl’s keynote address is entitled: “Sound and Emotion: had already been collecting sound recordings as The Power of Audio in Storytelling.” With the significant cultural sources on a greater scale, moment of our birth, our sense of hearing recedes into the did preservation start to become an issue. second row, into the unconsciousness. Nevertheless it Today, the world-wide holdings of audio record- cannot be switched off and is a vital means for our ings are estimated to amount to some 100 mil- understanding of the world around us, as well as a power- lion hours, many of them still on analog or digital ful transport medium for emotional content. Filmmakers single carriers, which sooner or later are prone have long used these powers for translating their story- to decay. Current thinking suggests, however, telling to the audience in sometimes sublime, sometimes that obsolescence of replay equipment is an drastic ways. In documentary filmmaking, the situation is even greater threat to the long-term survival of no different. In this talk, the various ways how sound can the audio heritage. be used to augment, embelish or even contrast the visual This constitutes substantial challenges to AES, impression will be discussed, always with the prerequisite of which the greatest may be: while little can be of being an equal partner and a welcome tool for the final done to counteract the present terrifying speed of result—the experience of watching a film. withdrawal from the manufacture of replay equip- ment and spare parts, how can we maintain the Session P4 Saturday, May 5 13:30 – 18:30 knowledge and the skills needed for the mainte- Room I nance of equipment and for the optimal retrieval of signals from our audio documents? AUDIO ARCHIVING, STORAGE, RESTORATION, This is a presentation only; no Convention AND CONTENT MANAGEMENT Paper will be available for purchase.

Chair: Dietrich Schüller, Austrian Academy of 14:00 Sciences, Vienna, Austria P4-2 150 Years of Time-Base in Acoustic 13:30 Measurement and 100 Years of Audio's Best Publicity Stunt— 2007 as a Commemorative P4-1 Sound Archiving—A Challenge for the Audio Year—George Brock-Nannestad, Patent Engineering Society [Invited Paper]—Dietrich Tactics, Gentofte, Denmark Schuller, Austrian Academy of Sciences, Vienna, Austria Léon Scott's invention of the phonoautograph in 1857 made a long time-base available for In the course of the past 20 years, issues related recording of vibrations, and it was also the first to sound archiving and restoration have increas- time an air-borne sound was recorded. Although ingly conquered a stable position at AES conven- his invention formed the basis, both for sound tions. These issues are also permanently dealt recording and reproduction and for acoustical with and further developed by the respective science as we know it, it has been largely forgot- groups within the AES Technical and Standards ten. Neither Scott nor the instrument maker Committees. In 2001, the 20th AES Conference Koenig are mentioned in the series “Benchmark held in Budapest was devoted to Archiving, Papers of Acoustics” Today we take sound Restoration, and New Methods of Recording. archives for granted, but the whole sound The world’s first sound archive, the Phono- archive movement would not have received any grammarchiv, was founded in 1899 by the then attention in the general public, if one particular Imperial Academy of Sciences, and that, inter event had not occurred: the sealed deposit in alia, may be one of the reasons why the Vienna 1907 of important shellac records and a gramo- AES Convention has made “Archiving” one of its phone in the vaults below the Paris Opera main topics. house. They were intended to remain untouched This paper will introduce to a full afternoon of for 100 years, and they have survived to this specialized presentations, reminding us that, at day. The paper will provide the documentation

6 Audio Engineering Society 122nd Convention Program, 2007 Spring for these historical events that form the basis of Preprocessing of Bias Signal for Wow so many of our professional activities. and Flutter Correction—Nadja Wallaszkovits,1 Convention Paper 7007 Franz Pavuza,1 Heinrich Pichler2 1Phonogrammarchiv Austrian Academy of 14:30 Sciences, Vienna, Austria 2Audio Consultant, Vienna, Austria P4-3 Knowledge: The Missing Element in Archiving and Restoration?—Sean W. Davies, SW This paper discusses the practical implementa- Davies, Ltd., Aylesbury, UK tion of high frequency bias signal retrieval from analog magnetic tapes at original replay This admittedly provocative title nevertheless speeds by using slightly modified standard calls attention to a situation that already exists playback facilities. Based on an implementation and may become critical in the future. No sound within an archival workflow and prior studies of recording can be considered in isolation from the bias signal retrieval from analog magnetic tape, technical system that produced it. A proper work- the authors focus on a comparison of reproduc- ing knowledge of such a system is an essential tion and preprocessing methods of the bias sig- requirement for any person working on the trans- nal. Previous approaches are compared to the fer of such a recording. This paper examines the authors’ proposed method. The signal prepro- range of such required knowledge and the cessing in the analog as well as digital domain means by which it may be taught to personnel is outlined and, based on analysis of bias sig- likely to be involved with archival material. nals from professional and semi-professional Convention Paper 7008 recordings, the various practical problems are discussed: level instability and unknown fre- 15:00 quency of the recorded bias signal, frequency variations mainly with semiprofessional devices P4-4 Noncontact Phonographic Disk Digitization of older types of recording equipment due to Using Structured Color Illumination—Louis the instability of the bias oscillator, as well as Laborelli, Jean-Hugues Chenot, Alain Perrier, effects of signal distortions, interferences, sig- INA (Institut National de l’Audiovisuel), Bry sur nal aliasing problems, and ultrasonic artifacts. Marne, France The practical applicability within a standard We propose an innovative contact-less optical archival transfer is discussed. playing device for 78 rpm and 33 rpm lateral Convention Paper 7011 modulation phonographic disks using structured color illumination. An area of the disk is illuminat- 16:30 ed by a beam of rays, with color depending on the direction of incidence, that are reflected by P4-7 Improved Magneto-Optical 1/4-Inch Audio the groove wall toward a camera image. In con- Tape Player for Preservation—Marcel trast with standard methods that measure the Guwang, Hi-Stor Technologies, Colomiers, velocity of the groove at a single location, direct France access to the audio signal value is obtained here This improved 1/4-inch audio tape player fea- directly from pictures through color decoding, and tures a multitrack magneto-optical reader to re- the whole height of the groove wall is exploited. duce preservation cost through speed, adjust- This color coding allows for the detection of oc- ment automation, and compatibility. The benefits cluding dust and automated interpolation of the of this 32-channel head, connected to a digital missing audio signal. Results on distortion, S/N signal processor, are high speed capability, com- ratio, and bandwidth are presented. patibility, and automatic detection of any number Convention Paper 7009 of audio tracks, real time automatic adjustment of the best digital playback azimuth, and filtering of 15:30 crosstalk and partial track erasures. Convention Paper 7012 P4-5 Improvement of Cylindrical Record This paper will be presented by Jean-Hughes Reproduction Utilizing Inharmonic Frequency Chenot. Analysis GHA—Teruo Muraoka, Shota Nakagomi, Tohru Ifukube, University of Tokyo, Tokyo, Japan 17:00 Cylindrical records were important sound media P4-8 Analysis and Restoration of Faulty Audio CDs— Hélène Galiègue,1 Jean-Marc Fontaine,2 around the beginning of 20th century, and a lot of 1 historical recordings were made by using them. Laurent Daudet 1Université Pierre et Marie Curie, Paris, France We have engaged in the research of cylindrical 2 record reproduction and noise-reduction of dam- Laboratoire d’Acoustique Musicale, Paris, France aged SP records utilizing inharmonic frequency Many audio CDs (mostly CD-Rs but also CD- analysis of GHA (Generalized Harmonic Analysis). ROMs) have defects that appear due to bad Surface noise of cylindrical records are more seri- manufacturing, careless use, or simple aging of ous than SP records, so we challenged its noise its physical constituents. Here, we study such reduction by modifying GHA noise-reduction. audio CDs that are still readable with a standard Convention Paper 7010 player (computer CD/DVD drive or standalone audio player with digital output), but whose de- 16:00 fects are not fully handled by error correction codes, resulting in a highly distorted signal. This P4-6 Method Comparison of Pick-Up and paper is two-fold: first, we characterize these ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 7 Technical Program errors on a few example discs; and second, we saine has found that well over 95 percent of passenger study different means to restore the audio con- seat listeners receive a soundstage with images that are tent, by fusion of multiple reads and interpolation asymmetrically clustered toward the near side of a car schemes. with individual images sometimes being placed at or on Convention Paper 7013 the near door loudspeakers. Surround system systems with center channel transducers seldom produce an opti- 17:30 mal solution, and the majority of driver side listeners receive a smaller degree of the same problem. In the P4-9 Techniques for the Authentication of Digital opinion of Nousaine near side spatial bias is a nearly uni- Audio Recordings—Eddy B. Brixen, EBB- versal problem in autosound design and implementation. consult, Smørum, Denmark In considering this issue, individual panelists will present In forensic audio one important task is the their individual interpretation of the situation along with authentication of audio recordings. Standards methods and techniques for solving the condition. and procedures already exists regarding analog recordings. In the field of digital recording and Exhibitor Seminar Saturday, May 5 digital media the conditions are different. A rock 13:30 Room 357 solid methodology is needed here, but does not D.A.V.I.D. exist yet. This paper reviews existing techniques and presents some results regarding an addi- tional number of tools, the ENF criterion, which Presenter: Axel Holzinger should be considered to become a standard within the AES as well as in the forensic commu- “Visual Radio”— Radio Delivery of Multimedia nity as a whole. Content to Multiple Platforms Convention Paper 7014 The upcoming delivery platforms like DMB, DVB-H, DVB- S, and wireless Internet open new dimensions for radio 18:00 broadcasters to bring content to their listeners. New P4-10 Using Multiple Feature Extraction with D.A.V.I.D. broadcast server functions make it easy to Statistical Models to Categorize Music by extend the classical radio workflow and provide “visible” Genre—Benjamin Fields, Goldsmiths College, content to the new mobile devices—at any time. University of London, London, UK TECHNICAL TOUR 2 In recent years, large capacity portable personal ORF TV Broadcasting Studios music players have become widespread in their Saturday, May 5, 13:30 – 16:30 use and popularity. Coupled with the exponen- tially increasing processing power of personal ORF TV is the National Public Broadcaster for Austria with computers and embedded devices, the way peo- two main TV channels (ORF 1+ 2), two dedicated theme ple consume and listen to music is ever chang- channels (TW1 – tourism and weather, ORF Sport Plus), ing. To facilitate the categorization of these per- regional programs, and diverse off-air activities, as well as sonal music libraries, a system is employed being by far the most popular Web-presence in the country. using MPEG-7 feature vectors as well as Mel- The focus of the tour to ORF TV will be the three newly re- Frequency Cepstral Coefficients classified furbished postproduction studios with outstanding acoustic through multiple trained Hidden Markov Models design and full surround sound capability, unique for Euro- and other statistical methods. The output of pean broadcast studios. Example clips will be demonstrat- these models is then compared and a genre ed, and the tour will also cover the general infrastructure choice is made based on which model gives the shortly after a major relaunch of a substantial part of ORF’s best fit. Results from these tests are analyzed programs. and ways to improve the performance of a genre sorting system are discussed. Saturday, May 5 13:30 Room 633 Convention Paper 7015 Technical Committee Meeting on Coding of Audio Signals

Workshop 4 Saturday, May 5 Workshop 5 Saturday, May 5 13:30 – 16:30 Room G 14:00 – 16:00 Room P

NEAR SIDE BIAS INTERACTIVE AUDIO AND HUMAN PERCEPTION: CAN WE CONNECT KNOWLEDGE WITH PRACTICE? Chair: Tom Nousaine, TN Communications, MI, USA Chair: Renato S. Pellegrini, sonic emotion ag, Panelists: David Carlstrom, DCX Oberglatt, Switzerland Roger Dressler, Dolby Laboratories Dan Field, Harman Specialty Group Panelists: Durand Begault, NASA Ames Research Jon Lane, Nissan Center, Mountainview, CA, USA Geoff Martin, Bang and Olufsen a/s Douglas S. Brungart, WPAFB Richard Stroud, Stroud Audio Bill L. Chapin, Ausim-3D, Palo Alto, CA, USA Mark Ziemba, Panasonic Jyri Huopaniemi, Nokia, Helsinki, Finland John-Marc Jot, Creative Advanced In testing nearly 700 OEM autosound systems, over 200 Technology Center, Ottawa, Ontario, Canada competitive aftermarket autosound systems and several William Martens, McGill University, Montreal, hundred aftermarket loudspeaker systems Tom Nou- Quebec, Canada

8 Audio Engineering Society 122nd Convention Program, 2007 Spring Olivier Warusfel, IRCAM, Paris, France Mousse T), and Svenja Dunkel (radio engineer and pro- duction manager for Echo Award, Comet, and produc- The term “interactive audio” is used here in its most gen- tions from Sennheiser). The presenters will show their eral sense to mean real-time modification of an audio way of working with both the band and the soundcheck. signal in which attributes of a reproduced sound source The rhythm section of “Hot Pants Road Club” plus their (such as the source’s apparent position, timbral charac- lead singer is a top band from Austria, well known for ter, etc.) can be controlled by a user’s actions (using their funky, groovy sound and the always slightly “Vie- interaction interfaces such as trackers, game-controllers, neese” style of their frontman Harry Ahamer. joysticks, etc.). Interactive audio is a growing field in Presentation schedule: today’s audio environments. The complexity of interac- tive environments in computer games and simulations 14.00 – 15.00 — O. Voges, lecture about FOH sound continues to grow. Understanding the perceptual effects of this increase in complexity is becoming a greater chal- 15.00 – 16.00 — W. Gittens, lecture about monitor lenge. This workshop investigates these perceptual sound and In-Ear effects by exploring the design of interactive spaces and 16.15 – 18.30 — Live and practical work by highlighting what is already known from established techniques for generating virtual environments. The Live Sound Seminar, led by Gregor Zielinsky, is pro- duced in cooperation with Neumann & Mueller, who sup- Tutorial 1 Saturday, May 5 port all sound, light, stage, and rigging equipment. This 14:00 – 16:00 Room 560/561 presentation has been held in many places, e.g., Bejing, Moskau, Mumbay, Istanbul, Barcelona. REALIZATION OF 7.1 MIXES TECHNICAL TOUR 3 Technical Museum Presenter: Jeff Levison, DTS, Inc., Agoura Hills, CA, USA Saturday, May 5, 14:00 – 17:00 The increasing popularity of multichannel playback sys- Beside mechanical musical instruments developed in the tems has required the development of a variety of early twenties of the last century and operated by sophisti- recording and mixing techniques. Artists and engineers cated light- or air-controlled devices, this visit will comprise are more aware of the impact of speaker placement on the original first Austrian TV-studio (1950) and a demon- the balance between envelopment and imaging— espe- stration of a laser vibrometer with a discussion of the re- cially for the surround channels d for the reproduction (or sults obtained with it (research on fortepiano soundboards illusion) of reverberation and other acoustic environmen- and violins; in cooperation with the Vienna University of tal aspects. Increasing the number of channels repro- Music some students used the vibrometer for duced can ease the dilemma of this envelopment/imag- investigations of cettle drums, violins and cellos), an intro- ing compromise, and now new high definition playback duction to a restoration project (a Neo-Bechstein grand) methods, such as Blu-ray and HD DVD, offer eight chan- where electroacoustical research was necessary to find the nels of discrete reproduction with simultaneous high appropriate material for the strings, and a guided tour quality video. Besides using these extra channels for through the gallery of musical instruments. extra surround channels, the possibility exists for the in- clusion of height channels for greater spaciousness and Session P5 Saturday, May 5 14:30 – 17:30 vertical pan positioning. This tutorial will present a group Room K of realized examples of 7.1 with four surround channels and alternate mixes with 5.1 plus height. Comparisons will be made between stereo, 5.1, and SPATIAL AUDIO PERCEPTION AND PROCESSING, these new 7.1 mixes. Proposals of other higher order PART 2 systems and the possible improvements to three-dimen- sional audio presentation will be discussed. Chair: Gerhard Stoll, IRT, Munich, Germany 14:30 Live Sound Seminar 1 Saturday, May 5 14:00 – 18:30 Room E P5-1 On the Application of Sound Source Separation to Wave-Field Synthesis—Máximo SENNHEISER: The Soundcheck Is the Show! Cobos, Jose López, Technical University of Valencia, Valencia, Spain Chair: Gregor Zielinsky Wave-Field Synthesis (WFS) is a spatial sound Presenters: Svenja Dunkel system that can synthesize an acoustic field in Wayne "Heights" Glittens an extended area by means of loudspeaker ar- Oliver Voges rays. Spatial positioning of virtual sources is pos- sible but requires separated signals for each The Sennheiser Live Mixing Event presents a complete source to be feasible. Despite the fact that most soundcheck with a live rock band on stage, with full of the music is recorded in separated tracks for sound and light equipment. Participants will learn about each instrument, in the stereo mix-down process how professional soundchecks are done in real life. This this information is lost. Unfortunately, most of the will include the FOH/PA soundcheck as well as the moni- existing recorded material is in stereo format. In tor soundcheck. All participants will receive an In-Ear this paper we propose to use sound source sep- wireless receiver, to be able to follow the in-ear check. aration techniques to overcome this problem. The presentation is held by Wayne “Heights” Gittens Existing algorithms are yet far from perfect re- (monitor engineer for Herbert Groenemyer, Xavier sulting in audible artifacts that clearly reduce the Naidoo, Söhne Mannheims, etc.), Oliver Voges (FOH en- quality of the resynthesized sources in practice. gineer for Echo Awards, The Dome, Naturally 7, Scooter, Despite these artifacts, when separated ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 9 Technical Program sources are mixed again by a WFS system they interleaving and overlapping sweeps in an opti- are masked by other sounds. The utility of differ- mized way and retrieves the impulse responses ent separation algorithms and the subjective of the measured systems. In this paper the results are discussed. MESM and its parameter optimization is Convention Paper 7016 described. As an example of an application of MESM, the measurement duration of an HRTF 15:00 set with 1550 positions is compared to the unop- timized method. Using MESM, the measurement P5-2 Reproduction of Arbitrarily Shaped Sound duration could be reduced by a factor of four Sources with Wave Field Synthesis— without a reduction of the signal-to-noise ratio. Physical and Perceptual Effects—Marije Convention Paper 7019 Baalman, Technische Universität Berlin, Berlin, Germany 16:30 Current Wave Field Synthesis (WFS) implemen- P5-5 A Fast Multipole Boundary Element Method tations only allow for point sources and plane for Calculating HRTFs—Wolfgang Kreuzer, waves. In order to reproduce arbitrarily shaped Zhensheng Chen, Austrian Academy of sound sources with WFS several aspects need Sciences, Vienna, Austria to be considered, such as the WFS-operator for source points outside of the horizontal plane, Methods for measuring head related transfer discretization of the object surface and diffrac- functions (HRTFs) for an individual person are tion of the sound around the sounding object rather long and complicated. To avoid this prob- itself, which can be modeled by introducing sec- lem a numerical model using the Boundary Ele- ondary sources at the edges of the object. This ment Method (BEM) is introduced. In general, paper discusses those issues, describes the such methods have the drawback that the com- implementation in software and shows results of putations for high frequencies are very time- and both objective and subjective evaluation. resource-consuming. To reduce these costs the Convention Paper 7017 BEM-model is combined with a fast multipole method (FMM) and a reciprocal approach. 15:30 Convention Paper 7020

P5-3 The Effect of Head Diffraction on Stereo 17:00 Localization in the Mid-Frequency Range— Eric Benjamin, Phil Brown, Dolby Laboratories, P5-6 A Hybrid Artificial Reverberation Algorithm— San Francisco, CA, USA Rebecca Stewart,1 Damian Murphy2 1Queen Mary, University of London, London, UK In a previous paper, the present author de- 2University of York, York, UK scribed anomalous localization in intensity stereo at frequencies above the frequency at Convolution based reverberation allows for an which the head is approximately one wavelength accurate reproduction of a space, but yields no in diameter. Conventional analysis of stereo flexibility in defining that space, while filterbank- localization has usually depended on an asymp- based reverberation allows computational effi- totic shadowless model of the head’s diffraction. ciency and flexibility but lacks accuracy. A hybrid Measurements of the ear signals heard by the artificial reverberation algorithm that uses ele- subjects in localization experiments showed that ments of both convolution and filterbank rever- there were large differences between what was beration is investigated. An impulse response is predicted by the simple model, and what was truncated to contain only the early reflections found in actual circumstances. We present a and is convolved with input audio; the output au- simple model for the head’s diffraction in the dio then is combined with audio processed range of 1200 Hz to 5000 Hz and show that it through a filterbank to simulate the late reflec- produces results which correspond more closely tions. The parameters defining the filterbank are to real-world localization. derived from the impulse response being ana- Convention Paper 7018 lyzed. It is shown that this hybrid reverberator can produce a high-quality reverberation compa- 16:00 rable to convolution reverberators. Convention Paper 7021 P5-4 Multiple Exponential Sweep Method for Fast Measurement of Head Related Transfer Functions—Piotr Majdak, Peter Balazs, Bernhard Laback, Austrian Academy of Exhibitor Seminar Saturday, May 5 Sciences, Vienna, Austria 14:30 Room 357

Presenting sounds in virtual environments CUBE-TEC INTERNATIONAL requires filtering of free field signals with head related transfer functions (HRTF). The HRTFs Presenter: Stefan Zachau describe the filtering effects of pinna, head, and torso measured in the ear canal of a subject. The measurement of HRTFs for many positions “World Class Noise Reduction for Pro Tools” in space is a time-consuming procedure. To Cube-Tec audio restoration and mastering tools (VPI’s) speed up the HRTF measurement the multiple are used by some of the world’s most prestigious facili- exponential sweep method (MESM) was devel- ties. This presentation covers a selection of restoration oped. MESM speeds up the measurement by VPI’s that have recently been converted for use on the

10 Audio Engineering Society 122nd Convention Program, 2007 Spring Digidesign Pro Tools platform. The presentation will tem appropriate for encoding the audio signals include tools for hum & buzz removal, noise reduction, with minimum errors by the fixed-point process- de-crackling, disturbance tone removal, and much more. ing. Tests showed that the fixed-point system optimized using the proposed model had sound Saturday, May 5 14:30 Room 633 quality comparable to the floating-point encoding Technical Committee Meeting on Audio for system. Telecommunications Convention Paper 7023

Saturday, May 5 14:30 Room 142/143 15:00 Standards Committee Meeting on SC-04-01 P6-3 Enhanced Bass Reinforcement Algorithm Acoustics and Sound Source Modeling for Small-Sized Transducer—Han-gil Moon, Manish Arora, Chiho Chung, Seong-cheol Jang, Session P6 Saturday, May 5 15:00 – 16:30 Samsung Electronics Co. Ltd., Suwon, Foyer IK Gyeonggi-Do, Korea Nowadays, mobile devices such as cell phones POSTERS: SIGNAL PROCESSING, or MP3 players using small-sized loudspeaker SOUND QUALITY DESIGN systems to supply sound events to users is very popular. The main reasons why small-sized transducers are being used are due to the de- 15:00 sign and the size of the devices. Unfortunately, P6-1 On Development of New Audio Codecs—Imre their design and size restrain the transducers Varga, Siemens Networks, Munich, Germany from high quality of low frequency performance. To breakthrough this physical barrier of poor low This paper presents the works recently complet- frequency generation, the well-known psychoa- ed or on-going in 3GPP and ITU-T on the devel- coustical background “missing fundamental illu- opment of new audio codecs. The main applica- sion” is exploited. In this paper the method of tions are wideband speech telephony, audio enhancing bass perception using virtual pitch is conferencing, and mobile multimedia applica- presented. In our demonstration, listeners can tions including Packet-Switched Streaming feel the deep bass with fewer artifacts. (PSS), Multimedia Messaging (MMS), and Multi- Convention Paper 7024 media Broadcast/Multicast Service (MBMS). In the standardization process, terms-of-reference 15:00 describing design constraints and performance requirements, test plans, selection rules are P6-4 Subordinate Audio Channels—Tim Jackson,1 finalized first. Next, extensive subjective listening Keith Yates,1 Francis Li2 testing is conducted. The codec selection is 1Manchester Metropolitan University, based on the selection test results and the Manchester, UK selection rules. Characterization phase of testing 2University of Salford, Salford, Greater allows obtaining the full amount of information Manchester, UK on the codec behavior. In this paper we propose a model for a back- Convention Paper 7022 ward-compatible subordinate audio channel within a host digital audio signal. Embedding and 15:00 extraction methods are presented and objective- P6-2 Fixed-Point Processing Optimization of the perceptual assessment results reported. The MPEG Audio Encoder Using a Statistical method is designed so as to minimize perceptual Model—Keun-Sup Lee,1 Young-Cheol Park,2 degradation to the host signal and maintain com- Dae Hee Youn3 patibility with existing systems. The implementa- 1Samsung Electronics Co. Ltd., Suwon, Korea tions utilize the discrete cosine transform and 2Yonsei University, Wonju, Korea the masking properties of the human auditory 3Yonsei University, Seoul, Korea system. Performance evaluation is assessed using the objective perceptual measure, objec- Audio applications for portable devices have two tive difference grade. Test results support that critical restrictions: small size and low power both the host and subordinate audio channels consumption. Therefore, fixed-point implementa- can maintain good audio fidelity without signifi- tions are essential for those applications. Even cant perceptual degradation. with a fixed-point processor, however, the data Convention Paper 7025 width can still be an issue because it can affect both the hardware cost and power consumption. 15:00 In this paper we propose a statistical model for the MPEG AAC audio encoder that can provide P6-5 Room Equalization Based on Acoustic and an optimal precision for the implementation. The Human Perceptual Features—Lae-Hoon Kim,1 hardware with the optimal precision, being com- Mark Hasegawa-Johnson,2 Jun-Seok Lim,3 pared with the floating-point system, is supposed Koeng-Mo Sung1 to have perceptually insignificant errors at its 1Seoul National University, Seoul, Korea output. To have an optimal precision for the AAC 2University of Illinois at Urbana-Champaign, encoder, we estimate the maximum allowable Urbana, IL, USA amount of fixed-point arithmetic errors in the bit- 3Sejong University, Seoul, Korea allocation process using the statistical model. Room equalization has the potential to create Finally, we present an architecture for the sys- ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 11 Technical Program improved audio display for homes, cars, and 15:00 professional applications. In this paper, the sig- nal is inverse filtered using an inverse filter com- P6-8 Artificial Speech Synthesis Using LPC— puted by using newly introduced regularized Manjunath D. Kadaba, Uvinix Computing Solu- optimal multipoint frequency-warped linear tions Bangalore/Karnataka, Bangalore, India prediction coefficients. We present experimental Speech analysis and synthesis with Linear Pre- results that show that the proposed room equal- dictive Coding (LPC) exploit the predictable ization algorithm improves equalization on the nature of speech signals. Cross-correlation, au- equalizable parts, thus enlarging the region of tocorrelation, and autocovariance provide the perceptually effective equalization. mathematical tools to determine this pre- Convention Paper 7026 dictability. If we know the autocorrelation of the speech sequence, we can use the Levinson- 15:00 Durbin algorithm to find an efficient solution to the least mean-square modeling problem and P6-6 Parametric Loudspeaker Equalization— use the solution to compress or resynthesize Results and Comparison with Other Methods the speech. —German Ramos, Jose J. Lopez, Technical Convention Paper 7029 University of Valencia, Valencia, Spain The results obtained by a loudspeaker equaliza- tion method are presented and compared with oth- Tutorial 2 Saturday, May 5 er equalization methods. The main characteristic 15:00 – 18:00 Room H of the proposed method resides on the fact that the equalizer structure is planned from the begin- STEREOPHONIC TECHNIQUES ning as a chain of SOS (Second Order Sections), where each SOS is a low-pass, high-pass, or parametric filter defined by its parameters (fre- Presenter: Ron Streicher, Pacific Audo-Visual quency, gain, and Q), and designed by a direct Enterprises, Pasadena, CA, USA search method. This filter structure, combined with What is stereo? Why and how do we hear with spatial the subjectively motivated error function acuity? How can we realistically capture and reproduce employed, allows obtaining better results from a the stereo sound field with just two microphones and two subjective point of view and requiring lower com- loudspeakers? These are but a few of the questions dis- putational cost. The results have been compared cussed in this in-depth tutorial. with different FIR (finite impulse response) and IIR The session begins with a discussion and demonstra- (infinite impulse response) filter design methods, tion of how the human ear-brain hearing system works. with and without warped structures. In all cases, This is followed by a historical overview of the develop- for the same computational cost, the presented ment stereophonic recording. The main body of the method obtains a lower error function value. session presents a comprehensive analysis of the vari- Convention Paper 7027 ous common stereophonic microphone configurations and concludes with numerous recorded examples for 15:00 evaluation and comparison of the techniques discussed. P6-7 A Zero-Pole Vocal Track Model Estimation Method Accurately Reproducing Spectral Zeros—Damián Marelli,1 Peter Balazs2 Exhibitor Seminar Saturday, May 5 1University of Vienna, Vienna, Austria 15:30 Room 357 2Austrian Academy of Sciences, Vienna, Austria Model-based speech coding consists in modeling LAWO – NETWORKING AUDIO SYSTEMS the vocal tract as a linear time-variant system. The all-pole model produced by the computation- Presenter: Felix Krueckels ally efficiency linear predictive coding method provides a good representation for the majority of “Integration of Workstation Audio Tools in Modern speech sounds. However, nasal and fricative Broadcast Live Situations” sounds, as well as stop consonants, contain spectral zeros, which requires the use of a zero- Lawo will be presenting a new approach in integrating pole model. Roughly speaking, a zero-pole mod- audio editing tools in live broadcast and production el estimation method typically does a nonpara- desks previously provided only by workstations. You will metric estimation of the vocal tract impulse be enthusiastic about a full integration of audio editing response and tunes the zero-pole model to fit this tools in the broadcast mixing desks mc266 and mc290. estimation in a square sense. In this paper we All common applications of the mc2-series are also avail- propose an alternative strategy. We tune the able for this integration. See and hear how the merge of zero-pole model to directly fit the power spectrum two completely independent worlds is realized in the of the speech signal in a logarithmic scale, to be Lawo products. The second part of the lecture is devoted consistent with the way the human ear perceives to the networking of several Lawo systems with total stu- sounds. In this way, we avoid the error intro- dio control. duced by the vocal tract impulse response esti- mation and obtain a model that is more accurate at reproducing spectral zeros in a logarithmic scale. A drawback of the proposed method, how- Saturday, May 5 15:30 Room 633 ever, is its computational complexity. Technical Committee Meeting on High Resolution Convention Paper 7028 Audio

12 Audio Engineering Society 122nd Convention Program, 2007 Spring Workshop 6 Saturday, May 5 Saturday, May 5 16:30 Room 142/143 16:30 – 18:30 Room G Standards Committee Meeting on SC-04-01 Acoustics and Sound Source Modeling LOUDNESS METERING SYSTEMS AND INTERNATIONAL ACTIVITIES SPECIAL EVENT Wave Field Synthesis Demonstration Chair: Lars Jonsson, SR/Swedish Radio Saturday, May 5, 16:45 – 18:45 Sunday, May 6, 16:45 – 18:45 Panelists: Kimio Hamasaki, NHK, Japan University for Music and Performing Arts, Vienna Erik Lundbeck, SVT Swedish TV Thomas Lundh, TC Electronics, Denmark CELEBRATE GOOD VIBES! Ralph Kessler, Pinguin, Germany Andrew Mason, BBC, UK Tony Spath, Dolby, UK Hosted by the University of Music and Performing Arts Gerhard Spikofski, IRT, Germany Vienna and supported by their partners Fraunhofer Insti- Matti Zemack, SR, Swedish Radio tute for Digital Media Technology IDMT, Sennheiser, Studer, and Merging Technologies, this Demo Session This workshop will cover loudness metering and auto- will present the reproduction of preliminarily recorded matic level adjustments and present an overview of sys- concerts in Wave Field Synthesis. tems and international activities. The session will cover Attendees will be able to listen to the recordings that the latest progress in standards made by the ITU and the were done in the afternoon. EBU to recommend solutions for broadcasting. The gen- Different microphone techniques and mixing concepts eral problem to solve is common to all new digital distrib- can be compared and evaluated. Neumann/Sennheiser ution systems. Old analog audio metering methods have microphones, a Vista 5 console by Studer, the Merging become obsolete and can no longer create a smooth Technologies Pyramix Digital Audio Workstation, and the perceived level for listeners. Manufacturers making pro- Spatial Audio Workstation by Fraunhofer IDMT will pro- posals along with the new standards are presenting their vide a professional mixing environment. Additionally products. A general discussion among the panel mem- Wave Field Synthesis demos including film, radio drama, bers and the audience will point out the direction for the and music will be presented. future. This Special Event will be held in two sessions on May 5 and May 6 at the University of Music Vienna. A bus Tutorial 3 Saturday, May 5 shuttle service will be offered between the Austria Center 16:30 – 18:30 Room 560/561 and the University. Tickets will be available from the Technical Tour REMAPPING AND DOWNMIXING counter. Meeting point is the main entrance of the Aus- tria Center. Please note that the times listed above in- clude travel time. Presenter: Jeff Levison, DTS, Inc., Agoura Hills, CA, USA Additional Demo Sessions in Wave Field Synthesis will For the new Blu-ray and HD-DVD systems remapping of also be available at the following times: loudspeakers is possible. What this means is that a giv- • Saturday, May 5, 17:15 – 18:15 en mix loudspeaker position can be described and then • Sunday, May 6, 17:15 – 18:15 on playback the room loudspeaker positions are used to balance the power distribution if there is a difference between the source and the playback speaker positions STUDENT EVENT (or the number of speakers). Several examples will be Opening and Student Delegate Assembly Meeting – played to demonstrate various remapping and downmix- Part 1 ing scenarios in a real-world manner. Saturday, May 5, 17:00 – 18:30 Room P Chair: Ainslie Harris Exhibitor Seminar Saturday, May 5 16:30 Room 357 Vice Chair: Suzana Jakovic The first Student Delegate Assembly (SDA) meeting is SCHOEPS the official opening of the convention’s student program and a great opportunity to meet with fellow students from Presenter: Helmut Wittek all corners of the world. This opening meeting of the Stu- dent Delegate Assembly will introduce new events and “Double M/S Recording in Stereo and Surround” election proceedings, announce candidates for the com- ing year’s election for the Europe/International Regions, announce the finalists in the recording competition cate- “Double M/S” is a technique for two-channel or multi- gories, hand out the judges’ sheets to the nonfinalists, channel stereo recording. It uses a compact microphone and announce any upcoming events of the convention. arrangement that requires only three recorder channels, Students and student sections will be given the opportu- while allowing full postproduction processing. During the nity to introduce themselves and their activities, in order seminar, recorded examples from a range of scenarios to stimulate international contacts. will be presented. The new Schoeps VST Plug-In will All students and educators are invited to participate in also be introduced and its capabilities demonstrated. this meeting. Election results and Recording Competition and Poster Awards will be given at the Student Delegate Saturday, May 5 16:30 Room 633 Assembly Meeting–2 on Tuesday, May 8, at 09:00. Technical Committee Meeting on SC-04-03 Loudspeaker Modeling Measurement ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 13 Technical Program Exhibitor Seminar Saturday, May 5 This paper extends the study of head move- 17:30 Room 357 ments during listening by including various lis- tening tasks where the listeners evaluate spatial MICROTECH GEFELL impression and timbre, in addition to the more common task of judging source location. Subjec- Presenter: Ulrich Apel tive tests were conducted in which the listeners were allowed to move their heads freely while listening to various types of sound and asked to “Surround Microphones Not Only for Concert Halls” evaluate source location, apparent source width, Microtech-Gefell presents microphones with special rain- envelopment, and timbre. The head movements proof windscreens that are applicable not only in closed were recorded with a head tracker attached to premises but for recordings of atmos on locations; even the listener’s head. From the recorded data, the for film and television. Herewith Microtech-Gefell translat- maximum range of movement, mean position ed their outdoor-and environmental experiences into and speed, and maximum speed were calculat- their studio-microphones. An application of this wind- ed along each axis of translational and rotational and rainshield and an operation of new switchable micro- movement. The effects of various independent phone UM930 will be shown. variables, such as the attribute being evaluated, the stimulus type, the number of repetition, and SPECIAL EVENT the simulated source location were examined Mixer Party through statistical analysis. The results showed Saturday, May 5, 18:30 – 19:30 that while there were differences between the head movements of individual subjects, across A mixer party will be held on Saturday evening to enable all listeners the range of movement was greatest convention attendees to meet in a social atmosphere when evaluating source width and envelopment, after the opening day’s activities to catch up with friends less when localizing sources, and least when and colleagues from the industry. There will be a cash judging timbre. In addition, the range and speed bar and snacks. of head movement was reduced for transient sig- nals compared to longer musical or speech Session P7 Sunday, May 6 09:00 – 12:00 phrases. Finally, in most cases for the judgment Room I of spatial attributes, head movement was in the direction of source direction. PSYCHOACOUSTICS, PERCEPTION, Convention Paper 7031 AND LISTENING TESTS, PART 1 10:00 Chair: Stefan Weinzierl, Technical University of Berlin, P7-3 Binaural Resynthesis for Comparative Berlin, Germany Studies of Acoustical Environments— Alexander Lindau, Torben Hohn, Stefan 09:00 Weinzierl, Technical University of Berlin, Berlin, Germany P7-1 Some Effects of the Torso on Head-Related Transfer Functions—Ole Kirkeby,1 Eira A framework for comparative studies of binaurally Seppälä,1 Asta Kärkkäinen,1 Leo Kärkkäinen,1 resynthesized acoustical environments is present- Tomi Huttunen2 ed. It consists of a software-controlled, automated 1Nokia Research Center, Helsinki, Finland head and torso simulator with multiple degrees of 2University of Kuopio, Kuopio, Finland freedom, an integrated measurement device for the acquisition of binaural impulse responses in A numerical method based on the ultra-weak high spatial resolution, a head-tracked real-time variational formulation (UWVF) is used to cal- convolution software capable to render multiple culate three sets of Head-Related Transfer acoustic scenes at a time, and a user interface to Functions (HRTFs). The three sets are made conduct listening tests according to different test by combining a hard head with a hard torso, a designs. Methods to optimize the measurement moderately absorbing torso, and no torso. Each process are discussed, as well as different ap- set is sampled for every 50 Hz from DC to 24 proaches to data reduction. Results of a percep- kHz at 21,872 points almost evenly distributed tive evaluation of the system are shown, where in the far-field, thus providing a spatial resolu- acoustical reality and binaural resynthesis of an tion of approximately one degree everywhere. acoustic scene were confronted in direct A/B Since the results of the numerical simulations comparison. The framework permits, for the first are not contaminated by the response of an time, to study the perception of a listener instanta- electroacoustic chain it is possible to compare neously relocated to different binaurally rendered the HRTFs of a head and torso model to the acoustical scenes. HRTFs of the head only without the risk of Convention Paper 7032 interpreting a measurement artifact as a physi- cal phenomenon. Convention Paper 7030 10:30 P7-4 Acoustic Factors of Auditory Distance 09:30 Perception by the Blind While Walking— Takahiro Miura, Shuichi Ino; Teruo Muraoka, P7-2 An Investigation into Head Movements Made Tohru Ifukube, University of Tokyo, Tokyo, Japan When Evaluating Various Attributes of Sound —Chungeun Kim, Russell Mason, Tim Brookes, The ability by which the blind that can recognize University of Surrey, Guildford, Surrey, UK surrounding objects solely by hearing is called

14 Audio Engineering Society 122nd Convention Program, 2007 Spring “obstacle sense.” By analyzing and modeling its Panelists: Durand Begault, NASA Ames Research mechanism, this model will be utilized for realiz- Center, Mountainview, CA, USA ing an acoustic VR environment as well as train- Florian Camerer, ORF - Austrian TV, Vienna, ing systems for the vision-impaired through Austria acoustic analysis. In this paper the authors par- George Massenburg, GML, Inc., TN, USA ticularly focused on various sorts of acoustic fac- tors that may contribute to perceive the distance The design of controllers and/or work surfaces for music from the subject to the obstacle especially while recording and mixing will be examined from the experi- walking. We also investigated the factors based enced user’s perspective. This examination is motivated on the psychophysical experiments and acousti- by the belief that there may be better ways to organize cal analysis methods. In addition, the authors controllers, ways that prioritize access to controls in a discussed the contribution of these factors to the manner that is based upon what experienced users know blind persons’ auditory distance perception. about how they do their work. The list of workshop pan- Convention Paper 7033 elists, which includes representatives from live sound, postproduction, and mastering will have a brainstorming session about new control layouts using tools designed 11:00 to stimulate “thinking outside of the box.” Also planned P7-5 Listener Loudspeaker Preference Ratings are breakout sessions, within which small groups will be Obtained in situ Match those Obtained via a lead to focus on particular applications and/or problems, Binaural Room Scanning Measurement and with results that hopefully contribute some new and use- Playback System—Sean Olive,1 Todd Welti,1 ful ideas. William Martens2 1Harman International Industries, Inc., Northridge, CA, USA Tutorial 4 Sunday, May 6 2McGill University, Montreal, Quebec, Canada 09:00 – 10:30 Room P

Binaural room scanning (BRS) is a method of cap- SOUND ARCHIVING turing, storing, and reproducing via a head-track- ing headphone display system the binaural room Presenter: Dietrich Schüller, Austrian Academy of impulse response of one or more loudspeakers in Sciences, Vienna, Austria a listening room. This paper reports the results of the first test in a series of validation tests of a cus- The world’s audio heritage is estimated to amount to tom BRS system that was developed for research 100 million hours of recorded documents. A consider- and evaluation of different loudspeakers and differ- able part is at severe risk of not surviving in the long- ent listening spaces. The test examined whether term, as it is still kept on analog or digital single carriers, listeners’ loudspeaker preference ratings made in which sooner or later are prone to deterioration. An even a listening room with reflective walls (in situ) were greater threat is the fast withdrawal from the manufac- comparable to ratings made in response to BRS ture of specific replay equipment and spare parts. This reproductions of those loudspeakers located in the will lead to a situation where still well-preserved record- same room. Virtually the same results were ob- ings cannot be retrieved anymore because of the lack of tained in these two cases. replay equipment. Convention Paper 7034 This tutorial concentrates on two basic documents for long-term audio preservation, released by the Technical 11:30 Committee of IASA, the International Association of Sound and Audiovisual Archives: P7-6 Perceptually-Motivated Audio Morphing: Brightness—Duncan Williams, Tim Brookes, • IASA-TC 03, The Safeguarding of the Audio Her- University of Surrey, Guildford, Surrey, UK itage: Ethics, Principles and Preservation Strategy • IASA-TC 04, Guidelines on the Production and A system for morphing the brightness of two Preservation of Digital Audio Objects sounds independently from their other perceptu- The tutorial will also survey the respective AES Stan- al or acoustic attributes was coded, based on dards that concentrate on the storage and handling of the spectral modeling synthesis additive/residual various types of audio carriers. model. A multidimensional scaling analysis of lis- tener responses showed that the brightness con- trol was perceptually independent from the other STUDENT EVENT controls used to adjust the morphed sound. A Recording Competition—Stereo timbre morpher, providing perceptually meaning- Sunday, May 6, 09:00 – 12:30 Room G ful controls for additional timbral attributes, can now be considered for further work. The Student Recording Competition is a highlight at Convention Paper 7035 each convention. A distinguished panel of judges par- ticipates in critiquing finalists in each of the categories in an interactive presentation during the convention. Student members can submit stereo and surround Workshop 7 Sunday, May 6 recordings in the categories classical, jazz, folk/world 09:00 – 11:00 Room H music, and pop/rock. Meritorious awards will be pre- sented at the closing Student Delegate Assembly Meet- USER-CENTERED DESIGN OF CONTROLLERS ing on Tuesday. FOR PRO AUDIO 09:00 – Jazz Stereo 09:45 – Pop/ Rock Stereo Chair: William Martens, McGill University, 10:30 – Folk/ World Music Montreal, Quebec, Canada 11:15 – Classical Stereo ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 15 Technical Program Judges: 22.2 multichannel sound system than with other Classical: Kimio Hamasaki, Ulrich Vette, TBA sound systems. Pop: Andres Mayo, Darcy Proper, Thomas Rabitsch Convention Paper 7037 Jazz: Christian Heck, Jan Erik Kongshaug, TBA World/Folk: Eva Bauer-Oppelland, Akira Fukada, TBA 10:30 We would like to acknowledge the generosity of our P8-3 Up-Mixing and Localization—Localization sponsors for the student recording competition: AKG, Performance of Up-Mixed Consumer Audio-Technica, DTS, Magix, PMC, RME, Schoeps, Multichannel Formats—Ben Shirley,1 Sennheiser. Richard Chaffey2 1University of Salford, Salford, Greater Sunday, May 6 09:00 Room 633 Manchester, UK Technical Committee Meeting on Automotive Audio 2AVE Systems, Hersham, Surrey, UK A number of listening tests were carried out to Sunday, May 6 09:00 Room 142/143 assess localization of sound in derived surround Standards Committee Meeting on SC-02-01 Digital sound fields. Two up-mixed consumer multi- Audio Measurement Techniques channel formats that use matrix decoding of 3/2 multichannel surround channels to increase the Session P8 Sunday, May 6 09:30 – 11:00 surround channel array (Dolby Pro Logic IIx and Room K DTS Neo:6) were compared to original 3/2 multi- channel material to determine the degree of spa- tial performance improvement. Noise bursts MULTICHANNEL SOUND, PART 1 were panned to 11 different locations, 13 sub- jects participated in the tests, and results were Chair: Karl Petermichl, ORF, Vienna, Austria analyzed to assess any improvement in localiza- tion in each of the assessed surround systems. Convention Paper 7038 09:30 P8-1 5.1 Radio—Too Much Too Soon?—Jon McClintock, APT, Belfast, Ireland, UK Exhibitor Seminar Sunday, May 6 5.1 multichannel audio is arguably the next nat- 09:30 Room 357 ural progression for radio. Although digital radio offered an incremental improvement over stereo LEAR CORPORATION/DIRAC RESEARCH for listeners, there has not been a fundamental change in radio since the migration from AM to Presenters: Nilo Casimiro Ericsson, Mathias FM. For radio to survive in a highly competitive Johansson, Armin Prommersberger environment, with an audience that is increas- ingly judgmental on delivery mediums and con- “Dirac Live™—A New Approach to Room Correction tent, then as an industry radio needs to embrace Algorithms” 5.1. This paper explores the principles that are vital to the success of multichannel audio for A method for improving car audio systems through digital radio, the enabling technology, and outlines vari- correction of the impulse/frequency responses of individ- ous projects that have been undertaken. ual loudspeakers is presented. The ability to simultane- Convention Paper 7036 ously optimize different time and frequency domain criteria and flexible spatial averaging are the main bene- 10:00 fits of the new approach. The solution is featured in the new BMW M5 Individual High End Sound System, P8-2 Wide Listening Area with Exceptional Spatial recently reviewed by Auto, Motor und Sport (number Sound Quality of a 22.2 Multichannel Sound 8/2007 pp. 146–148) stating: “the most impressive car System—Kimio Hamasaki, Toshiyuki stereo sound in the world.” An A/B comparison to con- Nishiguchi, Reiko Okumura, Yasushige ventional filtering is available in LEAR’s demo car in our Nakayama, NHK Science and Technical common exhibition booth. Research Laboratories, Tokyo, Japan While issues regarding the sweet spot in 5.1 sur- TECHNICAL TOUR 4 round sound have been discussed, a 22.2 multi- ORF TV Broadcasting Studios channel sound system has been developed for Sunday, May 6, 09:30 – 12:30 ultrahigh-definition TV. One of its features is ex- ORF TV is the National Public Broadcaster for Austria pansion of the listening area with exceptional with two main TV channels (ORF 1+ 2), two dedicated sound quality. Although the wideness of listening theme channels (TW1 – tourism and weather, ORF area was reported in previous papers, its evalua- Sport Plus), regional programs, and diverse off-air ac- tion was performed using only sound clips of a tivities, as well as being by far the most popular Web- symphony orchestra without a picture. There- presence in the country. The focus of the tour to ORF fore, subjective evaluations were performed for TV will be the three newly refurbished postproduction comparing the impression of various spatial at- studios with outstanding acoustic design and full sur- tributes at different listening positions using con- round sound capability, unique for European broad- tents with pictures in both large and small cast studios. Example clips will be demonstrated, and rooms. These evaluations demonstrate that the tour will also cover the general infrastructure viewers have better impressions of various spa- shortly after a major relaunch of a substantial part of tial attributes in a wider listening area with the ORF”s programs.

16 Audio Engineering Society 122nd Convention Program, 2007 Spring Sunday, May 6 10:00 Room 633 Several recording techniques and equipment are Technical Committee Meeting on Multichannel used in outdoor and indoor recordings for motion and Binaural Audio Technologies pictures. The choices are usually characterized from subjectivity and technical limitations irrele- Session P9 Sunday, May 6 10:30 – 12:00 vant to the desired final sound quality. Our goal Foyer IK is to present results of comparative recordings in order to give answers to every-day practice problems that arise. Overhead and underneath POSTERS: AUDIO RECORDING AND REPRODUCTION booming and the use of wireless microphones are compared through third octave frequency analysis. 10:30 Convention Paper 7041

P9-1 Intelligent Editing of Studio Recordings with 10:30 the Help of Automatic Music Structure Extraction— György Fazekas, Mark Sandler, P9-4 Semi-Automatic Mono to Stereo Up-Mixing Queen Mary, University of London, London, UK Using Sound Source Formation—Mathieu Lagrange,1 Luis Gustavo Martins,2 George In a complex sound editing project, automatic Tzanetakis1 exploration and labeling of the semantic music 1University of Victoria, Victoria, British Columbia, structure can be highly beneficial as a creative Canada assistance. This paper describes the develop- 2INESC Porto, Porto, Portugal ment of new tools that allow the engineer to nav- igate around the recorded project using a hierar- In this paper we propose an original method to chical music segmentation algorithm. include spatial panning information when con- Segmentation of musical audio into intelligible verting monophonic recordings to stereophonic sections like chorus and verses will be dis- ones. Sound sources are first identified using cussed briefly followed by a short overview of perceptively motivated clustering of spectral the novel segmentation approach by timbre- components. Correlations between these individ- based music representation. Popular sound-edit- ual sources are then identified to build a middle ing platforms were investigated to find an opti- level representation of the analyzed sound. This mal way of implementing the necessary allows the user to define panning information for features. The integration of music segmentation major sound sources thus enhancing the stereo- and the development of a new navigation toolbar phonic immersion quality of the resulting sound. in Audacity, an open-source multitrack editor, Convention Paper 7042 will be described in more detail. Convention Paper 7039 Live Sound Seminar 2 Sunday, May 6 10:30 – 12:30 Room E 10:30 P9-2 Constant Complexity Reverberation for Any NEUMAN & MUELLER: 3 Orchestras & Stars — Reverberation Time—Tobias May,1,2 Daniel World Cup Opening in Munich Schobben1 1Philips Research Laboratories, Eindhoven, The Presenters: Michael Kennedy Netherlands Rudolf Pirc 2Carl-von-Ossietzky University Oldenburg, Oldenburg, Germany As part of the opening celebrations for the Soccer World Cup in Germany, a concert was given in Munich on June A new artificial reverberation system is pro- 6, 2006 in the Munich Olympic Stadium. Together on posed, which is based on perceptually relevant stage were the three principle Munich Orchestras: the components in reverberated audio and, as such, Munich Philharmonic, the Bavarian State Opera Orches- allows for a very efficient implementation. The tra, and the Bavarian Broadcasting Symphony Orches- system first separates the signal into transient tra. The concert program included Placido Domingo, and steady-state components. The transient sig- soprano Diana Damrau, the pianist Lang Lang, and the nal is reverberated by using an efficient time- German band “Die Söhne Mannheims” with Xavier varying recursive filter while the steady-state sig- Naidoo. nal is processed separately with an all-pass The presentation describes the realization of the filter. In contrast to common reverberation sys- sound reinforcement system, which featured J and Q tems, the complexity of the recursive filter is series D&B loudspeakers, three Yamaha PM1 D sys- determined solely by the duration of the tran- tems at Front of House, a PM5D at the monitor position, sients and is therefore independent of the rever- and two separate DME systems. All interconnections beration time. between the mixing consoles and the DME systems Convention Paper 7040 were realized in the digital domain.

10:30 Exhibitor Seminar Sunday, May 6 P9-3 Outdoor and Indoor Recording for Motion 10:30 Room 357 Picture. A Comparative Approach on Microphone Techniques—Christos Goussios, D.A.V.I.D. Christos Sevastiadis, George Kalliris, Aristotle University of Thessaloniki, Thessaloniki, Greece Presenter: Ingo Hahn ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 17 Technical Program “Bi-Medial Production” “Double M/S” is a technique for two-channel or multi- channel stereo recording. It uses a compact microphone DigaSystem offers a wide variety of bi-medial production arrangement that requires only three recorder channels, tools embedded in a scalable content management sys- while allowing full postproduction processing. During the tem. This includes support of standard broadcast audio seminar, recorded examples from a range of scenarios and video formats, as well as formats for new delivery will be presented. The new Schoeps VST Plug-In will channels like mobile devices and new media. This lec- also be introduced and its capabilities demonstrated. ture gives an overview about bi-medial production work- flows within DigaSystem. STUDENT EVENT Education Fair Tutorial 5 Sunday, May 6 11:00 – 12:30 Sunday, May 6, 12:00 – 14:00 Foyer D Room H Institutions offering studies in audio (from short cours- MXF—THE MATERIAL EXCHANGE FORMAT es to graduate degrees) will be represented in a “table top” session. Information on each school’s respective Presenter: Tim Harris, Snell & Willcox programs will be made available through displays and academic guidance. There is no charge for schools to This tutorial takes an audio-oriented look at the MXF file participate. Admission is free and open to all conven- format. It will start off with a general introduction—to the tion attendees. basics of MXF—what MXF is, why it was invented, by whom it was developed, and how it was standardized. Session P10 Sunday, May 6 12:30 – 17:00 The tutorial will then focus on the MXF synchronization Room I model, and the capabilities of the format to combine au- dio, video, data and metadata in a versatile way. PSYCHOACOUSTICS, PERCEPTION, Carriage of essence within MXF will then be explained AND LISTENING TESTS, PART 2 with a particular focus on audio. Attention will be drawn to the involvement of the AES in providing the underlying international reference for the Broadcast Wave format. Chair: Sean Olive, Harman International Industries, Metadata annotation of essence, particularly recent work Inc., Northridge, CA, USA on the MXF Master Format Guidelines for multi-lingual annotation, will be explained. The talk will be interspersed with demonstrations of 12:30 how MXF software could be used in real-world work- flows. P10-1 Verbal Elicitation and Scale Construction for Evaluating Perceptual Differences between Four Multichannel Microphone Techniques— Exhibitor Seminar Sunday, May 6 William L. Martens, Sungyoung Kim, McGill 11:00 Room 441 University, Montreal, Quebec, Canada A verbal elicitation task using triadic compar- SENNHEISER isons was completed by eight listeners to ex- plore the adjectives that describe the audible dif- Presenter: Gregor Zielinsky ferences between solo piano performances captured using four different multichannel micro- “High Quality Surround Recording with the New MKH phone techniques. Although the elicited terms 8000 Microphone” differed somewhat between listeners, a set of five bipolar adjective pairs were found to repre- In this seminar Sennheiser presents the brand-new MKH sent the most salient differences between the 8000 Studio Microphone Series for the first time ever. On auditory imagery associated with multichannel- the examples of classical symphony and big band multi- loudspeaker reproduction of the piano perfor- track recordings, this “hands-on” seminar will show sev- mances. These adjectives were used as the an- eral surround microphone techniques using the MKH chors for five attribute rating scales on which the 8000. The different recordings will be explained and can same eight listeners rated each of the 32 stimuli be compared and judged by the listeners. that had been presented for triadic comparison. Stepwise multiple regression showed that rat- Sunday, May 6 11:00 Room 633 ings on three of the five attributes successfully Technical Committee Meeting on Acoustics predicted those listeners’ preference ratings for and Sound Reinforcement the same stimuli. Convention Paper 7043 Sunday, May 6 11:00 Room 142/143 Standards Committee Meeting on SC-04-04 13:00 Microphone Measurement and Characterization P10-2 Broadcast Loudness: Mixing, Monitoring, Exhibitor Seminar Sunday, May 6 and Control—Alessandro Travaglini, FOX 11:30 Room 357 International Channels (Italy), Rome, Italy In the satellite broadcast era, to manage trans- SCHOEPS mission from a digital platform that broadcasts dozens of channels and to guarantee loudness Presenter: Helmut Wittek consistency to viewers has became a primary, yet difficult task. In this paper I describe my work “Double M/S Recording in Stereo and Surround” in balancing the broadcasted loudness of the

18 Audio Engineering Society 122nd Convention Program, 2007 Spring Italian satellite TV platform SKY Italia. In fact, for 14:30 a few years, its transmissions suffered very audible loudness inconsistency, due to several P10-5 On the Audibility of Comb Filter Distortions factors, such as numbers of channels, various —Stefan Brunner, Hans-Joachim Maempel, content offerings, different mastering levels in Stefan Weinzierl, Technical University of Berlin, between programs, and interstitials, outsourcing Berlin, Germany productions from many external facilities, etc. Superpositions of delayed and undelayed ver- The project lasted for over one year, and the re- sions of the same signal can occur at different sults are a much improved audio quality and a stages of the audio transmission chain. Some- more balanced loudness consistency throughout times it is a deliberate measure to provide audio all the channels involved. material with certain spatial or timbral qualities. Convention Paper 7044 Often it is a result of multiple microphone signals, sound reflections on walls or latencies in digital 13:30 signal processing leading to comb-filter-shaped, linear distortions. The measurement of a hearing P10-3 Sound Levels of TV Advertisements Relative threshold for this type of distortion with its depen- to the Adjacent Programs and Cross-National dence on reflection delay, relative level, and the Comparison of the Way of Their Insertion type of audio content can be the basis for bound- into Programs—Eiichi Miyasaka, Akiko Kimura, aries in everyday recording practice below which Musashi Institute of Technology, Yokohama, undesired timbral distortions can be neglected. Kanagawa, Japan Therefore, a listening test was conducted to de- A couple of perceptual experiments were con- termine the just noticeable difference for three ducted in order to investigate the relationship be- stimulus categories (speech, a snare drum roll, tween the physical sound levels of advertise- and a piano phrase) and different time delays be- ments (CMs) and the corresponding auditory tween direct and delayed signal from 0.1 ms to 15 perception relative to the reference speech. Ex- ms, equivalent to 0.03 to 5.15 m of sound path periment 1 for three types of CMs aired in difference. The results show that comb-filter dis- Japanese broadcasting with different sound lev- tortions can still be audible if the level of the first els show that all CMs sound louder than a refer- reflection is more than 20 dB lower than the level ence speech irrespective of large sound level of the direct sound. differences. Experiment 2 for three types of CMs Convention Paper 7047 with similar sound levels and the standard devia- tions show that some of them sound louder than 15:00 the reference. Next, ways of insertion of CMs into programs were investigated for news pro- P10-6 VirtualPhone—A Rapid Virtual Audio grams broadcast in Japan, the UK, and the US. Prototyping Environment—Nick Zacharov, The results show that silent periods with different Nokia Corporation, Tampere, Finland durations were commonly inserted between As the complexity of mobile phones increases main programs and CMs in UK and in US, while with the evolution of digital convergence, there is no silent period was introduced in Japanese increased demand to ensure high audio quality commercial broadcasting. for all applications. VirtualPhone is a graphical Convention Paper 7045 user interface based software environment al- lowing for the rapid prototyping of mobile phone 14:00 audio and its subsequent calibrated auralization. This paper describes the framework of the Virtu- P10-4 Influence of Interaction on Perceived Quality alPhone application, illustrates its usage and in Audio Visual Applications: Subjective performance compared to other conventional Assessment with n-Back Working Memory prototyping schemes. Task, II—Ulrich Reiter, Mandy Weitzel, Convention Paper 7048 Technische Universität Ilmenau, Ilmenau, Germany The mechanisms of human audio visual percep- 15:30 tion are not fully understood yet. For interactive audio visual applications running on devices with P10-7 Evaluation of HE-AAC, AC-3, and E-AC-3 limited computational power it is desirable to Codecs— Leslie Gaston, Richard Sanders, know which of the stimuli to be rendered in an University of Colorado at Denver and Health audio visual room simulation have the greatest Sciences Center, Denver, CO, USA impact upon the perceived quality of the system. The Recording Arts Program at the University of We have conducted experiments to determine Colorado at Denver and Health Sciences Center the effect of interaction upon the precision with (UCDHSC) performed an independent evalua- which test subjects are able to discriminate be- tion of three audio codecs: Dolby Digital (AC-3 at tween different parameter values of auditory at- 384 kbps), Advanced Audio Coding Plus (HE- tributes. This paper details one of these experi- AAC at 160 kbps), and Dolby Digital Plus (E-AC- ments and compares different approaches for 3 at 224 and 200 kbps). UCDHSC performed the analysis of the obtained data. The results double-blind listening tests during the summer of show a noticeable bias toward faulty ratings dur- 2006, which adhered to the standards of ITU-R ing the involvement in a task, although the BS.1116 (that provides guidelines for multichan- analysis using significance tests did not com- nel critical listening tests). The results of this test pletely confirm this effect. illustrate a clear delineation between the AC-3 Convention Paper 7046 codec and the others tested. We will present ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 19 Technical Program the test procedures and findings in this paper. audio bit rate reduction codecs for broadcast Convention Paper 7049 applications. Codecs under test include offerings from Dolby, DTS, implementations of MPEG 16:00 AAC, and of the new MPEG Surround codec. The bit rates ranged from 64 kbit/s to 1.5 Mbit/s. P10-8 Perceptual Evaluation of Mobile Multimedia The subjective tests, including choice of method, Loudspeakers—Gaetan Lorho, Nokia selection of test material, and statistical analysis Corporation, Helsinki, Finland of results are described. The conclusions are An experiment was conducted to compare the that the hope for consistently high quality at a perceptual characteristics of stereo loudspeaker relatively low bit rate of, say, 256 kbit/s has not systems found in mobile multimedia devices. An yet been fulfilled, and that some audio material individual vocabulary development approach still demands at least 448 kbit/s. It has also been was employed for this descriptive analysis. Ten observed that later developments of some systems and five musical programs were select- codecs perform less well than earlier versions. ed for the experiment. Sixteen listeners devel- Convention Paper 7052 oped their own set of attributes in three hours and performed a comparative evaluation of the 13:00 ten systems for several program items using the P11-2 The Design and Analysis of First Order attribute scales they developed. A total of 111 Ambisonic Decoders for the ITU Layout— attributes was generated in this experiment, David Moore, Jonathan Wakefield, University of which could be divided in several perceptual Huddersfield, Huddersfield, West Yorkshire, UK groups relating to spatial, timbral, loudness, sound disturbance and sound articulation Ambisonic decoders for irregular layouts can be aspects. The principle of this sensory profiling designed using heuristic search algorithms. method is described and some results of the These methods provide an alternative to solving subjective experiment are presented. complex mathematical equations. New fitness Convention Paper 7050 function objectives for search algorithms are pre- sented that ensure derived decoders meet the 16:30 requirements of the Ambisonic system more closely than previous work. The resulting new P10-9 A Rapid Listening Test Environment— decoder coefficients are compared to other pub- Helping Managers Make Better Decisions— lished coefficients, and a detailed performance Nick Zacharov, Nokia Corporation, Tampere, analysis of first order decoders for the ITU layout Finland is given. This analysis highlights common poor As the complexity of mobile phones increases performance characteristics that these decoders with the evolution of digital convergence, there is hold. Proposed future work will attempt to increased demand to ensure high audio quality address these issues by looking at techniques for all applications. This paper presents a set of for producing decoders with a more even error software applications that allow for the rapid def- distribution around the listener and investigating inition, administration, analysis, and reporting of methods for removing the bias toward meeting listening tests without the need for extensive certain objectives. technical knowledge of the field. A through Convention Paper 7053 description of the concepts behind the client/server architecture of the software is pre- 13:30 sented followed by some example applications. P11-3 A New Digital Module for Variable Acoustics Last, a performance comparison of listening and Wave Field Synthesis: Design and tests performed using more traditional methods Applications—Diemer de Vries,1 Jasper van versus the presented method is made. Dorp Schuitman,1,2 At van den Heuvel2 Convention Paper 7051 1Delft University of Technology, Delft, The Netherlands 2 Session P11 Sunday, May 6 12:30 – 15:30 Acoustic Control Systems, Garderen, The Room K Netherlands A new digital module has been developed that MULTICHANNEL SOUND, PART 2 creates variable acoustics for multipurpose halls according to the Acoustic Control Systems Chair: Ulrike Schwarz, BR, Germany (ACS) concept. Additionally, it is capable of gen- erating wave fields according to the Wave Field 12:30 Synthesis (WFS) concept. The design concepts P11-1 EBU Tests of Multichannel Audio Codecs— and criteria, the technical specifications, and Andrew Mason,1 David Marston,1 Franc some first applications of the module will be Kozamernik,2 Gerhard Stoll3 explained and discussed. 1British Broadcasting Corporation, Tasworth, Convention Paper 7054 Surrey, UK 2EBU, Geneva, Switzerland 14:00 3 IRT, Munich, Germany P11-4 Artificial Reverberator with Location Control The latest project of one of the European Broad- in Multichannel Recording—Hwan Shim, casting Union technical groups has been the Jeong-Hun Seo, Koeng-Mo Sung, Seoul National assessment of the sound quality of multichannel University, Seoul, Korea

20 Audio Engineering Society 122nd Convention Program, 2007 Spring In this paper a novel artificial reverberator is pro- Workshop 8 Sunday, May 6 posed. Compared with conventional algorithms 12:30 – 14:30 Room P focused to append appropriate timber and rever- berance, the proposed algorithm is designed to ADVANCED TECHNIQUES FOR BUILDING produce realistic reverberation by controlling INTERACTIVE ENVIRONMENTS each location of sound sources. The new algo- rithm proposed in this paper controls perceived Chair: Damian Murphy, University of York, York, UK direction by panning the direct sound and con- trols perceived distance by adjusting the energy Panelists: Stefan Bilbao, University of Edinburgh, decay curve of reverberation, which is obtained Edinburgh, Scotland, UK by a location-clustering method and gain of the Renato Pellegrini, sonic emotion, Oberglatt direct sound. In addition, the algorithm enhances (Zurich), Switzerland Listener Envelopment (LEV) to make late rever- Ville Pulkki, Helsinki University of Technology, beration incoherent among channels. Helsinki, Finland Convention Paper 7055 Lauri Savioja, Helsinki University of Technology, Helsinki, Finland

14:30 Interactive environments are defined by the unpre- dictable actions of their users and by the real-time pro- P11-5 Spatial Audio Rendering Using Sparse and cessing required to generate them. Until recently, many Distributed Arrays—Aki Härmä, Steven van de audio processing techniques that could be used in inter- Par, Werner de Bruijn, Philips Research Europe, active environments require computational or bandwidth Eindhoven, The Netherlands resources that prohibit their implementation in real time. A widely distributed but multichannel audio Such techniques include methods to create audio con- reproduction system can be used to create tent, to model the environment, to communicate audio dynamic spatial effects for various entertainment data between remote participants, and to render the spa- and communication applications. In this paper tial audio output. The workshop will introduce state-of- we focus on the follow-me audio effect where the-art techniques for each of these four areas and dis- the sound source appears moving with the ob- cuss how suitable each is to particular applications in server who is walking through a hallway or going terms of practicality as well as computational and finan- from one room to another in the home environ- cial cost. The example techniques are physical modeling, ment. We give an overview of the array theory interactive virtual reality environment modeling, wave for the sparse distributed loudspeaker systems, field synthesis, and directional audio coding for each of study the binaural properties of the sound field the four areas respectively. rendered with a sparse line array, and compare two different dynamic rendering techniques in a Live Sound Seminar 3 Sunday, May 6 new type of a listening test. 12:30 – 14:00 Room E Convention Paper 7056 d & b

15:00 Presenters: Holger Blum P11-6 Magic Arrays—Multichannel Microphone Ralf Zuleeg Array Design Applied to Multiformat This session is comprised of two parts. Part one covers Compatibility— Michael Williams, Sounds of the dimensioning of line arrays and answers the question Scotland, Paris, France “Is more really more?“ This paper describes the principles and design Part two covers the advantage of digital signal wiring and procedure of multiformat-compatible microphone how to achieve optimal sonic quality and high reliability. arrays for a range of different segment coverage angles and for omnidirectional, hypocardioid, Exhibitor Seminar Sunday, May 6 cardioid, and supercardioid microphones. At pre- 12:30 Room 357 sent the only practical solution available for the main microphone array for a multiple format MICROTECH GEFELL recording is to use different microphone arrays for each of the required formats. This paper Presenter: Ulrich Apel shows how this jungle of main microphone ar- rays can be replaced by a single 5-channel mi- crophone array that will supply signals that are “Surround Microphones Not Only for Concert Halls” directly compatible with five standard formats: Microtech-Gefell presents microphones with special rain- mono, 2- and 3-channel “stereo,” 4-channel proof windscreens that are applicable not only in closed “quadraphony,” and “multichannel” with the full premises but for recordings of atmos on locations; even five channels. The specific reproduction format for film and television. Herewith Microtech-Gefell translat- can be chosen either during the production ed their outdoor-and environmental experiences into process as a function of the desired support me- their studio-microphones. An application of this wind- dia, or by the consumer from a multichannel me- and rainshield and an operation of new switchable micro- dia product according to their own particular lis- phone UM930 will be shown. tening configuration. Convention Paper 7057

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Audio Engineering Society 122nd Convention Program, 2007 Spring 21 Technical Program Session P12 Sunday, May 6 13:00 –14:30 tem identification using the voice coil current as Foyer IK feedback from the loudspeaker (plant). This is investigated in a simulation study for finding use- ful system identification algorithms. Two different POSTERS: MICROPHONES AND LOUDSPEAKERS identification techniques (ARMA and FIR) are AND AUDIO IN COMPUTERS (GAMES, INTERNET, compared. The stability of the nonlinearities is DESKTOP COMPUTER AUDIO) tested in a measurement series. Convention Paper 7060 13:00 13:00 P12-1 The Relation between Active Radiating Factor and Pressure Responses of Loudspeaker P12-4 Finite Element Analysis of Near Field Beam Line Arrays—Yong Shen, Dayi Ou, Kang An, Forming in Safety Relevant Work Spaces— Nanjing University, Nanjing, China Roman Beigelbeck,1 Heinrich Pichler2 1Austrian Academy of Sciences, Wiener Active Radiating Factor (ARF) is an important Neustadt, Austria parameter to analyze the loudspeaker line array 2Consultant, Vienna, Austria when considering the gaps between each of two radiating transducers. The relation between ARF Due to their unique features, loudspeaker arrays of the loudspeaker line array and the differential are an interesting alternative to standard loud- chart of its pressure responses in two distances speaker setups or headphone-based solutions in (PRD) is analyzed. Some valuable conclusions safety relevant workspaces such as air traffic about ARF and PRD are found. A method to control rooms. Consequentially, near field beam estimate ARF by measuring pressure responses forming in small spaces plays an important role comes out. for this field of application. In this paper the Convention Paper 7058 sound design based on a set of loudspeaker [Paper was not presented, but Convention arrays featuring their interaction with a typical air Paper 7058 is available for purchase.] traffic control room infrastructure is investigated by means of finite element modeling. Guided by 13:00 these results, optimized array parameters can be determined. Representative three-dimension- P12-2 Alternative Encoding Techniques for Digital al near field directional diagrams in front of the Loudspeaker Arrays—Fotios Kontomichos, arrays are shown to visualize the sound field in Nicolas–Alexander Tatlas, John Mourjopoulos, different cases. Finally, these theoretical values University of Patras, Patras, Greece are compared with practical results. Convention Paper 7061 Recent developments in digital loudspeakers have resulted in the introduction of digital trans- ducer arrays (DTA). In most implementations, 13:00 DTA loudspeakers are driven by PCM encoded P12-5 Creating Directed Microphones from audio signals, usually resampled and requan- Undirected Microphones—Emil Milanov, Elena tized to an appropriate number of bits, in Milanova, Acoustical Engineers, Sofia, Bulgaria accordance to the number of the transducers constituting the DTA topology. However, given In this paper we examine the possibility of creat- that DTAs generally increase harmonic distortion, ing directed microphones from undirected micro- especially for off-axis listening positions, opti- phones. The result is achieved only by using mization in signal encoding and bit-to-transducer acoustical elements and is valid for all types of assignment, is necessary. Here, a number of microphones, regardless of their way of work novel, alternative strategies are examined, con- (electro-dynamical, condenser, optical, electro- cerning the input signal encoding via PCM-to- mechanical, etc.). By using alternations of the PWM conversion, as well as techniques for bit- membrane usage, a force is obtained, which is assignment on the transducers of a DTA. These equivalent to the effect of simultaneously operat- tests are supported by simulation results and ing undirected and bidirected (eight) micro- comparisons, for different operating parameters. phones. The result is a microphone with a space Convention Paper 7059 characteristic equivalent to the Pascal curve (i.e., directed microphone with the traditional 13:00 shapes of the space characteristic—cardioid, super cardioid and hyper cardioid). The shape of P12-3 Online Identification of Linear Loudspeaker the space characteristic curve is near to theoreti- Parameters—Bo Rohde Pedersen,1 Per Rubak2 cal and does not depend on the acoustic ele- 1Aalborg University, Esbjerg, Denmark ments of the microphone. The microphone is 2Aalborg University, Aalborg, Denmark directed, but does not have a proximity effect. Convention Paper 7062 Feed forward nonlinear error correction of loudspeakers can improve sound quality. For creating an efficient feed forward strategy identi- 13:00 fication of the loudspeaker parameters is need- P12-6 Transducer with the Direct D/A Conversion ed. The strategy of the compensator is that the Using the Optoacoustic Principle—Libor nonlinear behavior of the loudspeakers has rela- Husník, Czech Technical University in Prague, tively small drift and only the linear loudspeaker Prague, Czech Republic parameters must be identified. In music systems this can be done with online transducer-less sys- Transducers with the direct D/A conversion,

22 Audio Engineering Society 122nd Convention Program, 2007 Spring sometimes called digital transducers, either 13:00 loudspeakers or earphones, are searching their ways into being. There have been several at- P12-10 Steganographic Approach to Copyright tempts to design such devices, but none of them Protection of Audio—Suthikshn Kumar, PES left research laboratories and made its way to Institute of Technology, Bangalore, India commercial use as yet. Most of them use “classi- Steganography is the technique of hiding data in cal” electroacoustic transduction principles, i.e., images and music. It is one of the powerful electrodynamic or electrostatic. In this paper the mechanisms by which useful copyright informa- possibility to use optoacoustic transduction prin- tion is hidden in the audio. In this paper we pro- ciples is explored. First, the principles of physical pose the use of steganography and public key phenomena used in this transducer are revised. cryptography to store the copyright information Then, some construction details in light of their and authenticate the original audio. A tool called usage in the digital earphone are described. Steger is being developed that automatically Convention Paper 7064 determines the original copyright holders of the audio content. This tool is useful in Digital Rights Management (DRM) enabling end-user systems 13:00 such as PDAs, mobile phones, PCs, handheld P12-7 Demystifying the Measurement of Impulse devices, consumer electronics, etc. Response in Condenser Microphones—Part Convention Paper 7067 I—Christian Langen, Schoeps Mikrofone, Karlsruhe, Germany Good impulse response is an important reason Workshop 9 Sunday, May 6 for preferring condenser microphones in audio 13:00 – 15:30 Room H applications that require high quality. However, it is difficult to characterize the impulse THE HOWS AND WHYS OF SIGMA DELTA response of a microphone precisely. We cannot CONVERTERS create an acoustic impulse that approximates the Dirac delta function closely enough that a Chair: Joshua Reiss, Queen Mary, University of microphone will emit only its own impulse London, London, UK response. Electrical spark discharges, pistol shots, and pressure-step methods all approxi- Panelists: Jamie Angus, University of Salford, Salford, mate the Dirac distribution, but due to their limi- Greater Manchester, Salford, UK tations one must still deconvolve the impulse Lars Risbo, Texas Instruments, Denmark, responses of the excitation signal and that of Lyngby, Denmark the microphone itself. Since every known Brian Trotter, Cirrus Logic, Inc., Austin, TX, method for performing such deconvolution has USA further pitfalls of its own, a novel time-domain Sigma delta modulation is the most popular form of ana- method of deconvolution is introduced. log-to-digital conversion used in audio applications. They Convention Paper 7065 are also commonly used in D/A converters, sample rate converters, and digital power amplifiers. This workshop will discuss methods of operation, design, and use of sig- 13:00 ma delta modulators. The theory behind their operation P12-8 Toward Multimodal Interfaces for Intrusion will be introduced and explained. We will discuss the Detection—Miguel Garcia-Ruiz,1 Miguel Vargas issues with their use and how they can be resolved. We’ll Martin,2 Bill Kapralos2 explain how their performance is assessed and how best 1University of Colima, Colima, Mexico to navigate through the specifications on a sigma delta 2University of Ontario Institute of Technology, modulator’s data sheet. Finally, practical examples will Oshawa, Ontario, Canada be given to illustrate the concepts presented.

Network intrusion detection has generally been Tutorial 6 Sunday, May 6 dealt with using sophisticated software and sta- 13:00 – 15:30 Room G tistical analysis tools. However, occasionally network intrusion detection must be performed manually by administrators, either by detecting BROADCAST CASE STUDIES the intruders in real-time or by revising network logs, making this a tedious and time consuming Presenters: Dennis Baxter, Audio for the Olympics labor. To support this, intrusion detection analy- Akira Fukada, NHK tokyo sis has been carried out using visual, auditory Gaute Nistov, NRK Oslo or tactile sensory information in computer inter- Three experts will relate stories of their individual experi- faces. However, little is known about how to ences in broadcasting. best integrate the sensory channels for analyz- ing intrusion detection. We propose a multi- Dennis Baxter (at 14:00) will tell of his experience modal human-computer interface to analyze broadcasting the Olympics. The Olympics uses produc- malicious attacks during forensic examination tion teams from all over the world including the host of network logs. We describe a sonification pro- country, which is currently China. Most of these produc- totype that generates different sounds accord- tion teams are considered to be the best at their particu- ing to a number of well-known network attacks. lar sports coverage. For example YLE (Finland) and Convention Paper 7066 NRK (Norway) produce Cross Country Skiing and Winter Biathlon, the BBC has produced Tennis and New Zealand covers sailing. The host country, ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 23 Technical Program 2004–Greece, 2006–Italy, 2008– China, is favored by TECHNICAL TOUR 5 the Host Broadcaster to participate as much as possible ORF Radio Broadcasting Studios but often these broadcaster lack the experience. One of Sunday, May 6, 13:00 – 16:00 the greatest challenges in the broadcast production of the Olympics is a consistency in production. With sound ORF Radio is the National Public Broadcaster for Austria, mixers from over 30 different countries they bring with with three nationwide, nine regional, and two MW/SW sta- them various levels of technical skills and personal expe- tions. The tour to the Radio Broadcasting House gives rience that influence the way the sound is produced. access to the music recording studios, continuity studios Additionally sound mixing is very personal and subjective for three different radio stations, and insight into the practi- and not easily definable as to what is right or wrong. The cal 5.1-Surround sound operation of the satellite radio following factors influence sound mixing: (1) Cultural channel OE1DD. interpretation of television production; (2) Psychological – Television has been dominated by video and engineers Sunday, May 6 13:00 Room 633 and technicians who do not understand audio. Often Technical Committee Meeting on Audio Forensics there has been a lack of resources and support, and sound engineers sometimes just give up! (3) Personal STUDENT EVENT prejudices – Most North American sound mixers disap- Design Competition prove of live sound sweetening; (4) Ego; (5) Experience. Sunday, May 6, 13:30 – 16:00 Room 631/632 The presentation will explore these areas and the sub- jectivity of sound production. The design competition is a competition for audio pro- jects made by students at any university or recording Gaute Nistov’s topic (at 13:00) is location recording school, which will challenge students with an opportu- from the bottom of the North Sea to the Pyramids of nity to showcase their technical skills. This is not for Egypt. On the 2nd of October 2006 singer Katie Melua recording projects or theoretical papers. Designs will performed a concert over 300 meters below sea level be judged by a panel of industry experts in design and inside one of the concrete shafts that anchors the Troll manufacturing. Multiple prizes will be awarded. gas rig to the sea bed. In addition to being Europe’s highest selling European female artist last year this gig in Judges: Christoph Musialik, Juha Backman, Dan Lavry the North Sea also secured Katie a world record for the deepest underwater concert. The special acoustic prop- Exhibitor Seminar Sunday, May 6 erties of the shaft and the very strict security measures 13:30 Room 357 on the platform were among the challenges for this extra- ordinary production. D.A.V.I.D. Only weeks later in Cairo a performance of Norwegian playwright Henrik Ibsen’s “Peer Gynt” was staged in front Presenter: Andreas Hildebrand of the Great Pyramids of Giza. The combined effort of more than 30 actors and singers, the Cairo Symphony Orchestra, and a 60-plus strong choir at the outdoor are- “Regionalization in Distributed Radio Broadcast” na, posed a very different set of requirements for a sound Today’s daily challenges in distributed PlayOut are the production that had to accommodate both a live transmis- driving considerations in workflow management for many sion locally on the night as well as recording for postpro- broadcasters. For this, D.A.V.I.D. provides a distributed duction. Nistov was in charge of the TV-sound production broadcast server concept, including background on both occasions and will discuss the technical solutions processes aligning media and scheduling content auto- used with an emphasis on production planning. matically and just in time, ready to be broadcasted when Akira Fukada (at 14:45) talks of his challenges broad- needed at multiple operational sites. casting two concerts in Japan. Two special concerts will be presented having taken place in demanding places in Sunday, May 6 13:30 Room 142/143 Japan: One of them is the concert at which the “field of Standards Committee Meeting on SC-02-08 Audio- summer” was performed at the city center of Hiroshima. File Transfer and Exchange This is a concert which looks back upon the 60th anniversary of the atomic bombing. The piece was per- Workshop 10 Sunday, May 6 formed at the exact place where the atomic bomb of 14:00 – 15:30 Room 560/561 Hiroshima was dropped. For Japan, that is a holy place. Therefore, there are many regulations and the concert was held in the severe environment of hot summer. It ACOUSTIC CONCEPTS AND TIME MANAGEMENT was not only broadcast live in 5.1 surround sound, but FOR SURROUND RECORDINGS—JAZZ offered simultaneously all over the world on the Internet. The second concert took place inside of a mountain. This Chair: Ulrich Vette, University of Music and mountain has a huge base rock and the sound per- Performing Arts Vienna, Austria formed there produces characteristic reverberation. In recordings of jazz combos and big bands it is a Composer Isao Tomita and Fukada planned the con- common concept to record instruments acoustically sep- cert using the sound properties of this space. First, arated by screens or even in multiple rooms. In a live music was performed by allotting a player to various recording situation this is not possible in the same way, places of a mountain. And those sounds are projected to so often close microphone positioning is the solution. By the base rock using PA. The reflective sound is record- using path length delay compensation a different record- ed with the surround microphone installed in the space. ing approach is possible: The use of main and ambience The performance had a distinctive sound due to this microphones leads to a more realistic spatial image and mountain’s effect. However, during this concert it, unfor- allows more natural sound colors of instruments than a tunately, rained; nevertheless the sound of the rain mix with only spot microphones. made for an exceptional sound effect caught by the sur- This acoustical concept will be demonstrated with a round microphone.

24 Audio Engineering Society 122nd Convention Program, 2007 Spring big band concert recorded at the Porgy and Bess Jazz “Digital Microphones—Purpose, Meaning, Club in Vienna and a studio recording of a Jazz Quartet. and Aspects” This seminar presents: the advantages of a fully digital sig- Sunday, May 6 14:00 Room 633 nal chain; digital microphone technology in today’s digital Technical Committee Meeting on Transmission studio environment; the AES 42 Interface: The “chicken or and Broadcasting egg” question; status quo concerning the implementation of the AES 42 standard; synchronization—the basic for any Live Sound Seminar 4 Sunday, May 6 digital recordings; gain setting with digital microphones; 14:30 – 16:00 Room E how is the workflow in the recording changed by using digi- tal microphones?; several examples of application. YAMAHA Sunday, May 6 15:00 Room 633 Presenter: Ruben van der Goor Technical Committee Meeting on Loudspeakers and Headphones This presentation will give an overview, explanation, and the pros and cons of existing networks. Integration into Yamaha-based applications will also be demonstrated. Exhibitor Seminar Sunday, May 6 15:30 Room 357 Exhibitor Seminar Sunday, May 6 14:30 Room 357 DPA MICROPHONES

CUBE-TEC INTERNATIONAL Presenters: Eddy Bøgh Brixen, Mikkel Nymand

Presenter: Stefan Zachau “DPA Decca Tree & Surround Mount” The DPA Decca Tree (D3)/Surround Mount (S5) is a “Quadriga: The Archive Solution” highly versatile and stylish microphone mount for 2 to 5+ microphones. The unique building block design with an Cube-Tec presents the latest version of QUADRIGA, the integrated cable option provides extreme flexibility allow- world standard in automated quality controlled audio ing the possibility of numerous configurations. At this archival systems. QUADRIGA workstations are capable seminar the system is demonstrated by experienced of simultaneously capturing multiple mono, stereo, or audio engineers. multichannel sources. The presentation shows the ingest process, Cube-Workflow/DOBBIN integration, and explains the new RF64/MBWF capabilities. Sunday, May 6 15:30 Room 142/143 Standards Committee Meeting on SC-03-02 Transfer Technologies Workshop 11 Sunday, May 6 15:00 – 18:00 Room P Workshop 12 Sunday, May 6 16:00 – 17:30 Room 560/561 MUSIC 2.0: MUSIC AND THE SEMANTIC WEB— CONSUMING AND PRODUCING MUSIC IN THE 21ST CENTURY ACOUSTIC CONCEPTS AND TIME MANAGEMENT FOR SURROUND RECORDINGS—JAZZ Chair: Mark Sandler, Queen Mary, University of London, London, UK Chair: Ulrich Vette, University of Music and Performing Arts Vienna, Panelists: Oscar Celma, Universitat Pompeu Fabra In recordings of jazz combos and big bands it is a com- Lucas Gonze, Yahoo! Music mon concept to record instruments acoustically separat- Yves Raimond, Queen Mary, University of ed by screens or even in multiple rooms. In a live record- London ing situation this is not possible in the same way, so often close microphone positioning is the solution. By Music and audio have evolved to encompass the Internet using path length delay compensation a different record- and all it offers for purchase and consumption of music; ing approach is possible: The use of main and ambience this is often called digital music. But the Internet is evolv- microphones leads to a more realistic spatial image and ing, too. With Web 2.0 products already on offer, how will allows more natural sound colors of instruments than a this affect digital music? Come to think of it, what exactly mix with only spot microphones. is Web 2.0? This acoustical concept will be demonstrated with a This workshop will provide some answers to these big band concert recorded at the Porgy and Bess Jazz questions. Topics to be covered by leading researchers Club in Vienna and a studio recording of a jazz quartet. include: Semantic Web and Web 2.0; Semantic Audio basics; current Music 2.0 projects; future Music 2.0 projects. Tutorial 7 Sunday, May 6 16:00 – 18:00 Room H Exhibitor Seminar Sunday, May 6 15:00 Room 411 CHAMPAGNE LIFESTYLE—BEER MONEY

GEORG NEUMANN GMBH Presenters: Simon Bishop, Freelance Sound Recordist, UK Richard Merrick, Freelance Sound Presenter: Stephan Peus Recordist, UK ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 25 Technical Program Simon Bishop and Richard Merrick contrast location comings, such as spatial aliasing, these plane audio acquisition from both ends of the budget spectrum. waves enlarged the sweet spot of the listening Discussing and comparing techniques and tricks collec- area and decreased the localization error of the tively acquired from 60 man-years of experience, from sound source. Also, we suggested a grouped being awash with money and equipment to begging, bor- reflections algorithm (GRA) for reproducing early rowing, and hunting on eBay. Light-hearted, but informa- reflections. This sequence of early reflections tive, both will prove that it’s not the size of your nail, but increased the spaciousness of the listening room the skill of the guy with the hammer! environment. The result, obtained by applying this method, was implemented by linear arrays STUDENT EVENT of 32 loudspeakers constructed in an anechoic Career/Job Fair room. For backward compatibility with standard Sunday, May 6, 16:00 – 18:00 Foyer D five-channel surround titles, a new hybrid sound field processing algorithm using WFS and GRA The Career Fair will feature several companies from the method was implemented. exhibit floor. All attendees of the convention, students Convention Paper 7069 and professionals alike, are welcome to come talk with representatives from the companies and find out more 16:30 about job and internship opportunities in the audio indus- try. Bring your resume! P13-3 Reproduction of Virtual Reality with Multichannel Microphone Techniques—Timo Sunday, May 6 16:00 Room 633 Hiekkanen, Tero Lempiäinen, Martti Mattila, Ville Technical Committee Meeting on Audio for Games Veijanen, Ville Pulkki, Helsinki University of Technology, Espoo, Finland Session P13 Sunday, May 6 16:30 – 18:00 The perceptual differences between virtual reali- Foyer IK ty and its reproduction with different simulated multichannel microphone techniques were mea- POSTERS: MULTICHANNEL SOUND sured using listening tests. The virtual reality was generated using the image-source method 16:30 and 16 loudspeakers in a 3-D arrangement in an anechoic chamber. Two spaced and two P13-1 Headphones Technology for Surround Sound coincident microphone techniques were tested, Monitoring—A Virtual 5.1 Listening Room— namely Fukada tree, Decca tree, 1st order Renato Pellegrini,1 Clemens Kuhn,1 Mario Ambisonics, and 2nd order Ambisonics. The Gebhardt2 spaced techniques utilized the 5.0 setup, and 1sonic emotion ag, Obergltt (Zurich), Switzerland Ambisonics techniques utilized the quadraphonic 2Beyerdynamic GmbH & Co. KG, Heilbronn, setup. The perceptual difference was measured Germany with ITU impairment scale. This paper presents a headphone technology for Convention Paper 7070 professional surround monitoring with virtual 5.1 reproduction. Using perceptually motivated bin- aural signal processing and ultra sonic head Exhibitor Seminar Sunday, May 6 tracking, this system enables the simulation of a 16:30 Room 357 loudspeaker set-up with correct localization and room impression. As a professional recording APT and mixing tool it provides the advantages of a portable headphone solution but avoids the Presenter: Jon McClintock known drawbacks such as inside-head localiza- tion, limited room perception, and turning of the sonic image with the listener’s head. The combi- “Surround Sound Broadcast Projects in Europe” nation of three technologies—binaural reproduc- tion, room simulation, and head tracking— Sunday, May 6 17:00 Room 633 enables the reproduction of a virtual reference Technical Committee Meeting on Signal Processing listening room for applications in studios, record- ing trucks, and mobile recording set-ups. Sunday, May 6 17:00 Room 142/143 Convention Paper 7068 Standards Committee Meeting on SC-03-04 Storage and Handling of Media 16:30 SPECIAL EVENT P13-2 Hybrid Sound Field Processing for Wave Open House of the Technical Council Field Synthesis System—Hyunjoo Chung,1 1 2 3 and the Richard C. Heyser Memorial Lecture Hwan Shim, JunSeok Lim, Jae Hyoun Yoo, Sunday, May 6, 18:15 – 19:15 Room K Koeng-Mo Sung1 1Seoul National University, Seoul, Korea Lecturer: Gerhard Steinke 2Sejong University, Seoul, Korea 3Electronics and Telecommunications Research The Heyser Series is an endowment for lectures by emi- Institute (ETRI), Yusung-gu Daejeon, Korea nent individuals with outstanding reputations in audio engi- neering and its related fields. The series is featured twice Using the wave field synthesis (WFS) method, annually at both the United States and European AES con- the sound of a primary source was reproduced ventions. Established in May 1999, The Richard C. Heyser by plane waves. Although having some short- Memorial Lecture honors the memory of Richard Heyser, a

26 Audio Engineering Society 122nd Convention Program, 2007 Spring scientist at the Jet Propulsion Laboratory, who was award- wine will be complemented by discreet music. As always, ed nine patents in audio and communication techniques one of the main pleasures of attending the banquet is to and was widely known for his ability to clearly meet old friends and colleagues and make new ones. present new and complex technical ideas. Heyser was also an AES governor and AES Silver Medal recipient. STUDENT EVENT The Richard C. Heyser distinguished lecturer for the Student Party 122nd AES Convention is Gerhard Steinke. He began Sunday, May 6, 20:00 –22:000 his career at Radio Dresden as a sound engineer in Club Ost 1947. In 1953 he moved to Berlin’s Radio and Television Schwindgasse 1, 1040 Wien Research Centre (RFZ), where he established a labora- tory for acoustical-musical boundary problems in This year the student’s traditional get-together takes broadcasting. In 1956, Steinke set up the first subjective place in one of Vienna’s renowned music locations— listening test group to assess sound recordings, studios, Club Ost. The Band “Vienna Funk Agency” performs and impairments in the broadcasting chain. This concept funk and disco classics and for those who still need more and the associated findings are included in various inter- there will be DJ’s heating up the party. But don’t miss the national standards (OIRT, ITU-R, and EBU) and docu- beginning of the Recording Competition Surround the ments (SSF, AES) on listening tests and test rooms. He next morning :). was also responsible for the introduction of stereophonic Of course everybody is welcome to this party—you broadcasting in East Germany and established an don’t have to be a student to party like a student. experimental electronic music studio with the new Sub- harchord synthesizer in 1962. In 1971 he became the TECHNICAL TOUR 6 director of the Research and Development Department Wiener Musikverein—The Golden Hall and the 4 New of Sound and Video System Technology of RFZ. Togeth- Halls er with co-inventors he developed the “Delta Stereopho- Monday, May 7, 08:30 – 11:00 ny” sound reinforcement system and a home processor for multichannel sound. He moved to Deutsche Telekom Creating the four new halls in Musikverein Vienna has in 1990 where he set up the research and development been the largest building project since the famous “Gold- group for new sound transmission systems. en Hal” was established in 1870. The original plan was Gerhard Steinke lectured sound technology and the creation of four new rehearsal halls, since Musikvere- electronic music at Berlin’s University of Music in the in had already met capacity problems. The excavation Tonmeister discipline for 27 years. Since his retirement he for a new Vienna Metro line next to the Musikverein published further numerous papers and lectures, and con- Building triggered the construction of the project, reach- tributed documents to the Surround Sound Forum of the ing down to as much as 16 meters below ground. In Tonmeister Society (VDT) and to the AES. 2000, the architects Holzbauer and Irresberger upgraded For his work in the field of standards he received the the project by “materializing” each hall: The Glass Hall Honorary Golden Medal of the OIRT and was awarded the (Magna Auditorium), the Metal Hall, The Stone Hall, and Bèkesy Medal for his contributions to audio by the Hungari- The Wooden Hall. This materialization provided a high an Acoustical Society. representative character as well as a big challenge for Steinke is a life member and fellow of the AES and the acoustical consultant. The opening of The Four New served as vice president, Europe Region of the AES Halls took place in March 2004. The technical tour is from 1991 – 1993, where he initiated the inauguration of guided by K. Bernd Quiring who designed the acoustics new AES sections in the Eastern European countries. He of these halls. Acoustical measures are presented and a is also member of the VDT. life chamber music example will be presented in all halls The title of his lecture is, “What Is Needed to Have the in order to provide an opportunity for personal judgments Audio-Eldorado at Home?” of the participants. Considering the various legitimate demands for high audio quality by the recipient at home, especially of the Session P14 Monday, May 7 09:00 – 13:00 music enthusiasts, his presentation will review to what Room I degree the requirements in standards for the whole transmission chain can be observed—from microphone/ studio up the individual listening conditions at home— MICROPHONES AND LOUDSPEAKERS, PART 1 and which important problems are disturbing the compli- ance. On one side, economical constraints cannot allow Chair: Thomas Gmeiner, AKG Acoustics GmbH, that highest production quality can be brought to the lis- Vienna, Austria tener via the networks in each case. On the other side, it can be shown that a lot of necessary improvements could be realized within the transmission process for the 09:00 reconstruction of the original perception that a listener P14-1 Refinements of Transmission Line has had of the original sound source and sound aesthet- Loudspeaker Models—Juha Backman, Nokia ic in the sense of the best possible approximation. Corporation, Espoo, Finland Steinke’s presentation will be followed by a reception hosted by the AES Technical Council. Simple waveguide models of loudspeaker enclo- sures describe well enclosures with simple interi- SPECIAL EVENT or geometry, but their accuracy is limited if used Banquet with more complex internal structures. A refine- Sunday, May 6, 20:00 – 23:30 ment of a transmission line loudspeaker model is Wappensaal at the Wiener Rathaus (Vienna Town Hall) discussed, presenting one-dimensional wave- guide approximations for bends and corners. This year’s banquet will be held in the Wappensaal at the Bends and corners are represented as area Wiener Rathaus (Vienna Town Hall), which is a beautiful changes in the line, approximated as one-dimen- room in a very famous landmark building. Fine food and sional line segments with parameters adjusted ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 27 Technical Program to match the exact solutions for sharp (rectangu- Control of the directivity of loudspeaker systems lar) corners in a waveguide. Besides modeling, is important in applications of sound reproduc- the paper discusses the sound transmission tion with public address systems. The use of characteristics of commonly used bend types loudspeaker arrays shows great advantages to and the applicability of the results to folded bundle the sound in specific directions. Usually horns. the loudspeakers are placed on a vertical line Convention Paper 7071 and the directivity is mainly in a plane perpendic- ular to that line although the radiation direction 09:30 can be adapted with filter techniques, called beamforming. In this paper we present results P14-2 The Use of Negative Source Impedance with on the applicability of a loudspeaker line array Moving Coil Loudspeaker Drive Units: An where the main directivity is in the direction of Analysis and Review—Michael Turner,1,2 that line, using so-called endfire beamforming, David Wilson1 resulting in a “spotlight” of sound in a preferred 1University of Leeds, W. Yorkshire, UK direction. Optimized beamforming techniques 2Switched Reluctance Drives Ltd., Harrogate, were used, which were developed for the recip- N. Yorkshire, UK rocal problem of directional microphone arrays. The effect of negative source impedance on the Effects of the design parameters of the loud- frequency response and pole-zero pattern of a speaker array system were investigated, and we moving coil loudspeaker drive unit is explored found that the stability factor can be a useful from first principles, and closed-form expres- parameter to control the directional characteris- sions for the transfer function and system poles tics. A prototype constant beamwidth array sys- are developed. Direct control of motor velocity tem was tested by simulation, and measurement via the substantial cancellation of voice coil and the results supported our findings. impedance is discussed. Implementation using Convention Paper 7074 positive current feedback is analyzed, consider- ing loop gain, damping, and stability from a 11:00 control theory perspective. Pole placement tech- P14-5 Mass Nonlinearity and Intrinsic Friction of niques are shown to be effective in controlling the Loudspeaker Membrane—Ivan Djurek,1 theoretical system behavior at high frequencies. Antonio Petosic,2 Danijel Djurek2 Modeled and measured results are presented. A 1University of Zagreb, Zagreb, Croatia selection of previous papers and applications 2AVAC – Alessandro Volta Applied Ceramics, concerned with operation of loudspeakers from Zagreb, Croatia negative source impedances is briefly reviewed. Practical issues and some possible applications Vibration of the loudspeaker’s membrane was are discussed. analyzed in the regime of comparatively low dri- Convention Paper 7072 ving currents (I0 < 100 mA) in terms of mass nonlinearity Meff and intrinsic friction RM. The lat- 10:00 ter contributes to the damping term of the differ- ential equation of motion and depends on the P14-3 Effects of Acoustical Damping on Current- elongation of vibration. RM is the sum of intrinsic Driven Loudspeakers—Rosalfonso Bortoni,1 friction Ri of the membrane and friction Rv com- Sidnei Noceti Filho,2 Homero Sette Silva3 ing from air viscosity on its surface. Independent 1THAT Corporation, Milford, MA, USA measurements of flexural strength of the mem- 2UFSC - Federal University of Santa Catarina, brane were performed and correlated to experi- Santa Catarina, Brazil mental observations of the vibrating system. 3Selenium Loudspeakers, Nova Santa Rita, Brazil Experiments were also performed with mem- Previous works show the benefits of exciting loud- branes additionally reinforced by application of speakers with current sources, but they do not materials with higher Young modules. present a study showing the behavior of this tech- Convention Paper 7075 nique when acoustical damping is applied to the diverse types of loudspeakers. This paper pre- 11:30 sents theoretical and practical analysis of the P14-6 Modeling of an Electrodynamic Loudspeaker frequency response of acoustically damped cur- Using Runge-Kutta ODE Solver—Antonio rent-driven loudspeakers installed in closed box, Petosic,1 Ivan Djurek,1 Danijel Djurek2 vented box, and 4th and 6th order band-pass sys- 1University of Zagreb, Zagreb, Croatia tems. Also, it presents a subjective analysis com- 2AVAC – Alessandro Volta Applied Ceramics, paring a closed box system excited by voltage and Zagreb, Croatia current sources. Convention Paper 7073 The modeling of low frequency (<100Hz) electro- dynamic loudspeaker is presented as one 10:30 degree of freedom nonlinear damped oscillator described by an ordinary differential equation of P14-4 Development of a Highly Directive Endfire motion. The model has been compared to an Loudspeaker Array—Marinus Boone,1 equivalent LRC circuit model, and it was shown Wan-Ho Cho,2 Jeong-Guon Ih2 that differential the equation approach is more 1Delft University of Technology, Delft, The suitable for calculations that include nonlineari- Netherlands ties occurring in an electrodynamic loudspeaker, 2Korea Advanced Institute of Science and as well as couplings of different vibration modes, Technology (KAIST), Daejeon, Korea particularly those coming from vibrating air and

28 Audio Engineering Society 122nd Convention Program, 2007 Spring the loudspeaker itself. The nonlinear differential 09:00 equation of periodically driven anharmonic oscil- lator was solved numerically, and calculated P15-1 A Biologically-Inspired Low-Bit-Rate amplitude frequency dependence and electric Universal Audio Coder—Ramin Pichevar, impedance have been compared to the experi- Hossein Najaf-Zadeh, Louis Thibault, mental data. Calculations included different Communications Research Centre, Ottawa, working regimes of the loudspeaker being oper- Ontario, Canada ated in an evacuated space and air. We propose a new biologically-inspired para- Convention Paper 7076 digm for universal audio coding based on neural spikes. Our proposed approach is based on the 12:00 generation of sparse 2-D representations of audio signals, dubbed as spikegrams. The P14-7 Chaotic State in an Electrodynamic 1 2 spikegrams are generated by projecting the sig- Loudspeaker—Danijel Djurek, Ivan Djurek, nal onto a set of over-complete adaptive gam- Antonio Petosic2 1 machirp (gammatones with additional tuning AVAC-Alessandro Volta Applied Ceramics, parameters) kernels. A masking model is applied Zagreb, Croatia 2 to the spikegrams to remove inaudible spikes University of Zagreb, Zagreb, Croatia and to increase the coding efficiency. The para- An electrodynamic loudspeaker has been oper- digm proposed in this paper is a first step toward ated in a nonlinear regime when the k-factor the implementation of a high-quality audio strongly increases with displacements. For dri- encoder by further processing acoustical events ving AC currents up to 2 A the vibration spec- generated in the spikegrams. Upon necessary trum contains high frequency harmonics of the optimization and fine-tuning our coding system, classic von Kármán type; for currents in the operating at 1 bit/sample for sound sampled at range 2.6 to 4 A doubling of driving period 44.1 kHz, is expected to deliver high quality appears; and for currents in the range from 4 to audio for broadcast applications and other appli- 4.8 A multiple sequences of subharmonic vibra- cations such as archiving and audio recording. tions begin with 1/4f and 3/4f. An application of Convention Paper 7078 currents higher than 4.8 A results in a white noise spectrum, which is a characteristic of 09:30 chaotic state. Convention Paper 7077 P15-2 The Relationship Between Basic Audio Quality and Selected Artifacts in Perceptual Audio Codecs —Part II: Validation Experiment 12:30 —Paulo Marins, Francis Rumsey, Slawomir P14-8 An Improved Electrical Equivalent Circuit Zielinski, University of Surrey, Guildford, Surrey, Model for Dynamic Moving Coil Transducers UK 1 2 —Knud Thorborg, Andrew Unruh, Christopher A pilot study was conducted to investigate the J. Struck3 1 perceptual importance of selected audio coding Tymphany A/S, Taasrup, Denmark artifacts and their relationship with basic audio 2Tymphany Corporation, Cupertino, CA, USA 3 quality. An additional experiment was undertak- Independent Consultant, San Francisco, CA, USA en to validate the results obtained in the pilot A series combination of inductor and resistor is calibration experiment. A listening test was traditionally used to model the blocked electrical designed that required a panel of expert subjects impedance of a dynamic moving coil transducer, to evaluate the selected artifacts used in the ini- such as a loudspeaker driver. In practice, semi- tial study. In this second experiment, however, inductive behavior due to eddy currents and certain experimental parameters were modified; “skin effect” in the pole structure as well as these included different levels of degradation transformer coupling between the voice coil and and program material. The outcomes of the vali- pole piece can be observed, but are not well rep- dation experiment are presented in this paper resented by this simple model. An improved along with a detailed evaluation of the impact of model using only a few additional elements is the chosen experimental artifacts on basic audio introduced to overcome these limitations. This quality assessments for perceptual audio improved model is easily incorporated into exist- codecs. ing equivalent circuit models. The development Convention Paper 7079 of the model is explained and its use is demon- strated. Examples yielding more accurate box 10:00 response simulations are also shown. Convention Paper 7063 P15-3 New Enhancements to Immersive Sound Field Rendition (ISR) System—Chandresh Dubey,1 Raghuram Annadana,1 Deepen Sinha,1 Anibal Ferreira1,2 Session P15 Monday, May 7 09:00 – 13:00 1ATC Labs, Chatham, NJ, USA Room K 2University of Porto, Porto, Portugal LOW BIT-RATE AUDIO CODING Consumer audio applications such as satellite radio broadcasts, multichannel audio streaming, and playback systems coupled with the need to Chair: Karlheinz Brandenburg, Technical meet stringent bandwidth requirements are elicit- University Ilmenau, Ilmenau, Germany ing newer challenges in parametric multichan- ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 29 Technical Program nel audio coding schemes. This paper describes Transmission Using Conjugate Vector the continuation of our research concerning the Quantization—Mylene Kwong, Roch Lefebvre, Immersive Soundfield Rendition (ISR) system Soumaya Cherkaoui, Université de Sherbrooke, and the different enhancements in various algo- Sherbrooke, Quebec, Canada rithmic components. The need to maintain a constant bit rate for many applications requires a This paper explores robustness issues for real- rate control mechanism. The various strategies time audio transmission over perturbed networks utilized in the rate loop mechanism are present- where multiple paths can be considered. Conju- ed. In addition, an innovative phase compensat- gate vector quantization (CVQ), a form of multi- ed downmixing scheme has been incorporated ple description coding, can improve the in the ISR system so as to generate a high quali- resilience to packet losses. This work presents a ty carrier signal. Enhancements have been generalized CVQ structure, where K>2 different made to the blind up-mixing scheme and consid- conjugate codebooks are trained to create the erable gains have been made in terms of best resulting codebook. Experiments show that acoustic diversity. The various enhancements of four-description CVQ performs very closely to the ISR system and its performance are unconstrained VQ in clear channel conditions, detailed. Audio demonstrations are available at while providing significant improvements in lossy http://www.atc-labs.com/isr. channels. We also present a fast search algo- Convention Paper 7080 rithm that allows trade-offs between computa- tional complexity and memory storage at the encoder. This robust quantization scheme can 10:30 encode sensitive information such as spectral P15-4 Aspects of Scalable Audio Coding—Chris coefficients in a speech coder or a perceptual Dunn, Independent Consultant, London, UK audio coder. Convention Paper 7083 Banded weight data is transmitted as side infor- mation within coded audio bit streams in order to 12:00 achieve psychoacoustically-appropriate shaping of quantization noise. Methods of reducing the P15-7 MPEG Surround—The ISO/MPEG Standard information overhead corresponding to weight for Efficient and Compatible Multichannel data are discussed in the context of scalable bit- Audio Coding—Jürgen Herre,1 Kristofer plane coding. Two approaches to coding band Kjörling,2 Jeroen Breebaart,3 Christof Faller,4 weight data are compared in terms of coding effi- Sascha Disch,1 Heiko Purnhagen,2 Jeroen ciency and error resilience. In the first approach, Koppens,5 Johannes Hilpert,1 Jonas Rödén,2 weights are coded as a block of data at the begin- Werner Oomen,5 Karsten Linzmeier,1 Kok Seng ning of each frame, using a predictor and Golomb Chong6 coding of weight prediction residuals to achieve 1Fraunhofer IIS, Erlangen, Germany high coding efficiency. This approach is compared 2Coding Technologies, Stockholm, Sweden to coding weights for bands as they become sig- 3Philips Research, Eindhoven, The Netherlands nificant, with weight data distributed across each 4Agere Systems, Allentown, PA, USA coded bit stream frame. 5Philips Applied Technologies, Eindhoven, Convention Paper 7081 The Netherlands 6Panasonic Singapore Laboratories Pte. Ltd., 11:00 Singappore P15-5 Source-Controlled Variable Bit Rate Extension In 2004 the ISO/MPEG Audio standardization for the AMR-WB+ Audio Codec—Amélie group started a new work item on efficient and Marty, Roch Lefebvre, Université de Sherbrooke, backward compatible coding of high-quality mul- Sherbrooke, Quebec, Canada tichannel sound using parametric coding tech- niques. Finalized in fall of 2006, the resulting This paper presents a source-controlled, vari- MPEG Surround specification allows to carry able bit rate extension to the AMR-WB+ stan- surround sound at bit rates that have been com- dard audio codec. AMR-WB+ allows multirate monly used for coding of mono or stereo sound. operation and, in particular, rate switching at This paper summarizes the results of the stan- every frame. However, the standard does not dardization process by describing the underlying support source-controlled rate determination ideas and providing an overview of the MPEG since it does not include a signal classifier. The Surround technology. The performance of the proposed extension includes a signal classifier scheme is characterized by the results of the and rate mapping function for each signal class. recent verification tests. These tests include sev- Classification is performed at a lower frame rate eral operation modes as they would be used in compared to AMR-WB+, with typically one clas- typical application scenarios to introduce multi- sification decision every second. Significant rate channel audio into existing audio services. savings can be achieved by encoding speech at Convention Paper 7084 lower rates than other signals such as music. Applications include audio broadcasting over 12:30 packet networks and storage of multimedia sig- nals with mixed signals in the audio track. P15-8 Adaptive Design of the Preprocessing Stage Convention Paper 7082 for Stereo Lossless Audio Compression— Florin Ghido, Ioan Tabus, Tampere University of 11:30 Technology, Tampere, Finland P15-6 Multiple Description Coding for Audio We propose a novel lossless audio compression

30 Audio Engineering Society 122nd Convention Program, 2007 Spring scheme, which combines stereo preprocessing Managing workflow processes is an increasingly chal- with stereo prediction. We show that such a lenging task due to the complexity of modern, multime- scheme provides improved asymmetrical com- dia-oriented program departments. Process modeling is pression at almost no complexity increase for a way to graphically draw and analyze workflows, as well decoder (compared with stereo prediction alone), as integrating business parameters from the “Balanced or the same compression for lower decoder com- Score Card” model into those processes. plexity. The stage of stereo prediction is preced- Process modeling with the method ADONIS (used at ed by a rotation-like channel transformation, ORF) and modeling with BPMN and UML in various which improves compression by requiring smaller applications will be explained and demonstrated. inter-channel optimal prediction orders and by obtaining smaller magnitudes for prediction coef- Workshop 15 Monday, May 7 ficients. On a corpus consisting of 84 audio files 09:00 – 10:30 Room 560/561 (in CD-A format), for the OptimFROG-AS (asym- metric) lossless audio compressor using stereo ACOUSTIC CONCEPTS AND TIME MANAGEMENT prediction with orders 8/4, we obtained, on an FOR SURROUND RECORDINGS—CLASSICAL audio corpus (in CD-audio format) of size 51.6 GB, compression improvements up to 5.10 per- cent on average 0.23 percent. Chair: Ulrich Vette, University of Music and Convention Paper 7085 Performing Arts Vienna, Austria Experiences with acoustical surround recordings for the ITU 5.1 loudspeaker setup often lead to the conclusion to combine different ambience microphones to reproduce Workshop 13 Monday, May 7 ambient sound. In this workshop a concept for time 09:00 – 11:30 Room H relations between ambience microphones in different distances to the sound source will be presented and SURROUND RECORDING AND REPRODUCTION discussed. WITH HEIGHT Moreover there will be a comparison between sur- round reproduction using the ITU 5.1 loudspeaker setup Chair: Kimio Hamasaki, NHK Science and and advanced ideas using a second pair of lateral or ele- Technical Research Laboratories, Tokyo, vated rear speakers. Multitrack recordings will be pre- Japan sented from research about different surround micro- phone setups at the University of Music and performing Panelists: Arnaud Laborie, Trinnov Audio, France Arts, Vienna and examples of concert and studio-record- Jeff Levison, DTS, CA, USA ings with more than one pair of microphones for sur- Ville Pulkki, Helsinki University, Finland round reproduction. Wilfried Van Baelen, Galaxy Studio, The Netherlands STUDENT EVENT Helmut Wittek, Schoeps Microphones, Recording Competition—Surround Germany Monday, May 7, 09:00 – 12:00 Room G Wieslaw Woszczyk, McGill University, Canada The Student Recording Competition is a highlight at each convention. A distinguished panel of judges participates Two-dimensional multichannel sound systems such as in critiquing finalists in each of the categories in an inter- 5.1, 6.1, and 7.1 are widely applied to movie theaters, active presentation during the convention. Student mem- home audio, and broadcasting. While new video formats bers can submit stereo and surround recordings in the such as 4k and 8k for the digital cinema is currently dis- categories classical, jazz, folk/world music, and pop/rock. cussed, next-generation multichannel audio systems for Meritorious awards will be presented at the closing Stu- the movie theater such as 10.2 and 22.2 are also dis- dent Delegate Assembly Meeting on Tuesday. cussed. New features on next-generation multichannel audio include the sensation of height and elevation of Schedule: sound sources. A three-dimensional audio system is not 09:00 – Classical Surround only discussed among the sound professionals for movie 10:00 – Non-Classical Surround productions but also among the broadcasters and home 11:00 – Sound For Picture audio developers. This workshop will review the latest Judges: proposals of three-dimensional audio systems and will Classical Surround: Werner Dabringhaus, Josef discuss the surround sound recording and reproduction Schütz, Jürg Jecklin to provide the sensation of height. Non-Classical Surround: Ronald Prent, Akira Fukada, Panelists will give separate demonstrations in another TBA room following this workshop. Details will be announced. Picture: Florian Camerer, Bernhard Maisch, Wilfried van Baelen Workshop 14 Monday, May 7 We would like to acknowledge the generosity of our 09:00 – 11:00 Room P sponsors for the student recording competition: AKG, Audio-Technica, DTS, Magix, PMC, RME, Schoeps, PROCESS MODELING FOR BROADCAST Sennheiser. PROFESSIONALS Monday, May 7 09:00 Room 633 Cochairs: Chris Chambers, BBC Technical Committee Meeting on Human Factors Karl Petermichl, ORF in Audio Systems ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 31 Technical Program Monday, May 7 09:00 Room 142/143 about 40 ms, during which time the direct sound Standards Committee Meeting on SC-03-12 Forensic can be separately perceived. For such a hall the Audio impressions of loudness, clarity, and localization are satisfactory and nearly unchanged over a 6 Session P16 Monday, May 7 09:30 – 11:00 dB range of d/r. As the time period of direct Foyer IK sound dominance decreases, the d/r ratio must be higher for equal subjective clarity. POSTERS: ROOM AND ARCHITECTURAL Convention Paper 7088 ACOUSTICS AND SOUND REINFORCEMENT 09:30

09:30 P16-4 Relation between Correlation Characteristics of Sound Field and Width of Listening Zone P16-1 Acoustic Treatment of the Regional Flight —Elena Prokofieva, Linn Products Ltd., Glasgow, Control Center Hall in Zagreb, Croatia— Scotland, UK Marko Horvat, Hrvoje Domitrovic, Sanja Grubesa, University of Zagreb, Zagreb, Croatia The principal features of an even sound field are large directivity and diffusive nature of the radiat- Acoustic treatment has been realized in the hall ed field. Directivity pattern of stereo loudspeak- of Regional Flight Control Center in Zagreb, ers was analyzed to determine the degree of Croatia, upon complaints made by the flight con- coherence of the sound field in the room. Mea- trol operators working in the mentioned hall. The surements of the sound field in close to a stan- primary complaint was that the operators could dard listening room conditions were conducted hear each other too well across the hall, due to with the loudspeakers placed in ITU-recom- unwanted reflections, so the main task was to mended and specially selected positions. The reduce those reflections. The emphasis was also results showed that the radius of correlation cor- made on reducing the reverberation time of the responds to the size of the sweet spot. Relocat- hall, proven to be too long for the size and ing loudspeakers in the room and taking into intended purpose of the hall, thereby reducing consideration the room environment influence the background noise level in the hall as well. can help to enlarge the listening zone within the Convention Paper 7086 room. These conclusions were confirmed by lis- tening tests and recommendations on sound 09:30 field correlation can be established. Convention Paper 7089 P16-2 Investigating Classroom Acoustics by Means [Paper not presented but Convention Paper of Advanced Reproduction Techniques— 7089 is available for purchase.] Nicola Prodi,1 Andrea Farnetani,1 Yuliya 1,2 3 Smyrnova, Janina Fels 09:30 1University of Ferrara, Ferrara, Italy 2Polish Academy of Sciences, Poland P16-5 On the Implementation of a Room Acoustics 3RWTH Aachen University, Aachen, Germany Modeling Software Using Finite-Differences Time-Domain Method—José Lopez,1 José A research was undertaken to investigate the Escolano,2 Basilio Pueo3 loss of Italian language word intelligibility in 1Technical University of Valencia, Valencia, Spain classrooms caused by low signal to noise ratio 2Technical University of Jaen, Linares (Jaen), and too high reverberation. In the first part of the Spain paper, impulse responses and background nois- 3University of Alicante, Alicante, Spain es were measured in two primary schools using different mono, binaural, and B-format probes. A The Finite-Difference Time-Domain (FDTD) dummy head with child morphology was also approximation method has been introduced into used for the first time in this context. It was thus acoustics in the last years to solve field prob- possible to compare the performance of a child lems numerically. However, the huge amount of head to the conventional adult one. Then the computer power needed to be used in the mod- restitution of the recorded sound fields in a dedi- eling of large rooms has delayed the launch of cated listening room was accomplished, using commercial applications, being the major part stereo dipole and ambisonics technologies. based on ray-tracing. This paper analyzes the Convention Paper 7087 viability of a FDTD implementation for this task in today’s personal computers and presents the 09:30 resulting application. All simulation stages from the architectural model, the generation of the P16-3 Perception of Concert Hall Acoustics in mesh, implementation of the recursion, paral- Seats Where the Reflected Energy Is lelization, and, finally, the result in the form of Stronger than the Direct Energy—David impulse response are discussed. Griesinger, Harman Specialty Group, Bedford, Convention Paper 7090 MA, USA

This paper describes a series of experiments TECHNICAL TOUR 7 into sound perception when the direct/reverber- Vienna Symphonic Library ant ratio (d/r) is low. Sound source localization Monday, May 7, 09:30 – 13:30 and the perception of being adequately close to the musicians are improved when the direct Have a sneak peek into the Silent Stage of the Vienna sound dominates the total reflected energy for Symphonic Library, one of the biggest producers of virtual

32 Audio Engineering Society 122nd Convention Program, 2007 Spring instruments world-wide. Here, recording musicians as well phone setups at the University of Music and performing as sound technicians and musical directors find optimal Arts, Vienna and examples of concert and studio-record- conditions for capturing perfect samples—ultimate control ings with more than one pair of microphones for sur- over the recording conditions and the absolute silence round reproduction. required for sampling sessions—in a studio solely designed for that purpose. See a presentation of the Vien- na Instruments, the most powerful virtual instruments cur- Workshop 17 Monday, May 7 rently on the market, tailor-made for the massive sample 11:00 – 13:00 Room 631/632 database of over 1.5 million samples recorded in the Silent Stage. Meet the creator of MIR, the Multi Impulse Mixing HANDS-ON DEMONSTRATION: SUBJECTIVE and Reverberation Engine which adds space to the alche- EVALUATION BY REFERENCE RECORDINGS my of digital orchestral music production. Based upon the WITH SURROUND MAIN MIKINGS FOR CLASSICAL principle of multi impulse response convolution, the Vienna ORCHESTRA Symphonic Library digitizes (“samples”) the characteristics of spaces like the Great Hall of the Wiener Konzerthaus Presenters: Hideo Irimajiri, Mainichi Broadcasting with unprecedented detail and methodology, taking this Corp. technology to the next level and capturing the great concert Toru Kamekawa, Tokyo National University stages of the world in their full glory—wall to wall, floor to of Fine Arts and Music ceiling, in three dimensions! Masayuki Mimura, Yomiuri Telecasting Corp. Hideaki Nishida, Asahi Broadcasting Corp. Koichi Ono, Kansai Telecasting Corp. Monday, May 7 10:00 Room 633 Technical Committee Meeting on Hearing There are many different setups for surround main and Hearing Loss Prevention microphones for classical music and orchestra. But it is very difficult to research and study them practically and academically under identical conditions and judge their Exhibitor Seminar Monday, May 7 performance. Consequently the AES Japan Surround 10:30 Room 357 Study Project has been organized and put into practice after one year of preparation. It was organized around 10 LEAR CORPORATION/DIRAC RESEARCH broadcasters and 2 universities; 12 manufacturers sup- ported by HBF provided financial support. There were 15 Presenters: Nilo Casimiro Ericsson, Mathias different combinations of main and ambience micro- Johansson, Armin Prommersberger phone setups that were recorded on 96 channels independently in Pro Tools HD at 24 bit / 96-kHz. The “Dirac Live™—A New Approach to Room Correction musical examples were performed by the Osaka Philhar- Algorithms” monic Orchestra on September 24-27, 2006. In this workshop each individual setup will be played A method for improving car audio systems through digi- back. Participants will have the opportunity for feedback tal correction of the impulse/frequency responses of in a listening test environment, and the data will be col- individual loudspeakers is presented. The ability to lected for subjective evaluation. simultaneously optimize different time and frequency domain criteria and flexible spatial averaging are the Workshop 17 will be repeated four times throughout the main benefits of the new approach. The solution is fea- convention: W-17, W-20, W-28, and W-29. tured in the new BMW M5 Individual High End Sound System, recently reviewed by Auto, Motor und Sport Exhibitor Seminar Monday, May 7 (number 8/2007 pp. 146–148) stating: “the most impres- 11:00 Room 441 sive car stereo sound in the world.” An A/B comparison to conventional filtering is available in LEAR’s demo car in our common exhibition booth. SENNHEISER

Presenter: Gregor Zielinsky Workshop 16 Monday, May 7 11:00 – 12:30 Room 560/561 “High Quality Surround Recording with the New MKH 8000 Microphone” ACOUSTIC CONCEPTS AND TIME MANAGEMENT FOR SURROUND RECORDINGS—CLASSICAL In this seminar Sennheiser presents the brand-new MKH 8000 Studio Microphone Series for the first time ever. On the examples of classical symphony and big band multi- Chair: Ulrich Vette, University of Music and track recordings, this “hands-on” seminar will show sev- Performing Arts Vienna, Austria eral surround microphone techniques using the MKH Experiences with acoustical surround recordings for the 8000. The different recordings will be explained and can ITU 5.1 loudspeaker setup often lead to the conclusion to be compared and judged by the listeners. combine different ambience microphones to reproduce ambient sound. In this workshop a concept for time rela- Monday, May 7 11:00 Room 633 tions between ambience microphones in different distances Technical Committee Meeting on Perception to the sound source will be presented and discussed. and Subjective Evaluation of Audio Moreover there will be a comparison between sur- round reproduction using the ITU 5.1 loudspeaker setup and advanced ideas using a second pair of lateral or Monday, May 7 11:00 Room 142/143 elevated rear speakers. Multitrack recordings will be pre- Standards Committee Meeting on SC-03-06 Digital sented from research about different surround micro- Library and Archive Systems ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 33 Technical Program Session P17 Monday, May 7 11:30 – 13:00 room reverberation distorts in time-frequency Foyer IK scales, perceived features in the received signal. These maps are simplified to describe the monaural time-frequency/level distortions and POSTERS: SPATIAL AUDIO PERCEPTION the distortion of the spatial cues (i.e., interchan- AND PROCESSING nel cues and coherence), which are very impor- tant for sound localization in a reverberant 11:30 environment. Such maps are studied here as functions of room parameters (size, acoustics, P17-1 Sound Source Localization and B-Format distance, etc.), as well as due to input signal Enhancement Using Sound Field Microphone properties. Overall perceptual distortion ratings Sets—Charalampos Dimoulas, Kostantinos are produced and reverberation-resilient signal Avdelidis, George Kalliris, George Papanikolaou, features are extracted. Aristotle University of Thessaloniki, Thessaloniki, Convention Paper 7093 Greece The current paper focuses on the implementa- 11:30 tion of sound-field microphone arrays for sound P17-4 A New Structure for Stereo Acoustic Echo source localization purposes and B-format Cancellation Based on Binaural Cue Coding enhancement. There are many applications —Yoomi Hur, Young-Choel Park, Dae Hee where spatial audio information is very impor- Youn, Yonsei University, Seoul, Korea tant, while reverberant sound-field and ambient noise deteriorate the recording conditions. As Most stereo teleconferencing systems involve an examples we may refer to sound recordings dur- acoustic echo canceller to remove undesired ing movie production, virtual reality environ- echoes. However, it is difficult for the stereo ments, teleconference and distance learning echo cancellers to converge to the true echo applications using 3-D audio capabilities. B-for- path since the cross-correlation between the mat components, provided from a single sound stereo input signals is high. To solve the prob- field microphone, are adequate to estimate lem, we propose a new structure that is com- sound source direction of arrival, while the com- bined with Binaural Cue Coding (BCC). BCC is a bination of two sound field microphones allows method for multichannel spatial rendering based estimating the exact source location. In addition, on one down-mixed audio channel and side the eight (or more) available signal components information. Based on the BCC, we propose a can be used to apply delay and sum techniques, new single channel adaptive filter for the stereo enabling SNR improvements and virtual posi- echo cancellation, which takes the down-mixed tioning of a signal B-format microphone to any monaural signal as the reference. Efficient voice desired place. Simplicity, reduced computational conference systems can be implemented by load, and effectiveness are some of the advan- using the proposed structure, since the BCC tages of the proposed methodology, which is scheme to transmit stereo signals as mono sig- evaluated via software simulations. nals with a number of side information, enables Convention Paper 7091 low-bit-rate transmission. Simulation results con- firm that the convergence speed is increased and the misalignment problem is solved. In addi- 11:30 tion, the proposed structure has better tracking P17-2 Research on Widening the Virtual Listening capability. Space in the Automotive Environment— Convention Paper 7094 Jeong-Hun Seo, Lae-Hoon Kim, Hwan Shim, Koeng-Mo Sung, Seoul National University, 11:30 Seoul, Korea P17-5 Efficient Binaural Filtering in QMF Domain for This paper represents the research about a way to BRIR—David Virette,1 Pierrick Philippe,2 widen the virtual space in cars. Generally, the inte- Gregory Pallone,1 Rozenn Nicol,1 Julien Faure,1 rior of cars contains small volume compared to Marc Emerit,2 Alexandre Guerin1 normal listening environments. This makes listen- 1France Telecom, Lannion, France ers feel a little stuffy. Therefore, the way to widen a 2France Telecom, Cesson-Sévigné, France virtual space in cars is needed. One of the most important cues for spaciousness is the lateral The MPEG Surround standard includes two reflections in accordance with room acoustics, so “native” binaural processing modules for repro- we will widen virtual space in cars using artificial ducing 3-D audio content over headphones. In lateral reflections in automotive environments. this paper we present a novel and efficient bin- Convention Paper 7092 aural room impulse response (BRIR) modeling algorithm extending their possibilities. It is based on a parametric decomposition of the BRIR and 11:30 is integrated within the subband domain imple- mentation of the MPEG Surround binaural P17-3 Perceptual Distortion Maps for Room decoder. First we show that for impulse respons- Reverberation—Thomas Zarouchas, John es with room effects, our approach offers a sig- Mourjopoulos, University of Patras, Patras, nificant reduction in terms of computational Greece requirements compared to the native methods. From reverberated audio signals and using as Second, we report results from listening tests reference the input (anechoic) audio, a number comparing different trade-offs between complex- of distortion maps are extracted indicating how ity and quality. We show that using our method,

34 Audio Engineering Society 122nd Convention Program, 2007 Spring the complexity can be reduced by a factor of two 11:30 while preserving the optimum quality. Convention Paper 7095 P17-8 Physical and Filter Pinna Models Based on Anthropometry—Patrick Satarzadeh, V. Ralph Algazi, Richard O. Duda, University of California 11:30 at Davis, Davis, CA, USA P17-6 A Parametric Model of Head-Related Transfer This paper addresses the fundamental problem Functions for Sound Source Localization— of relating the anthropometry of the pinna to the Youngtae Kim, Jungho Kim, Sangchul Ko, localization cues it creates. The HRTFs for iso- Samsung Advanced Institute of Technology, lated pinnae (which are called PRTFs) are ana- Gyeonggi-do, Korea lyzed and modeled for sound sources directly in front of the listener. It is shown that a low-order A simple and effective parametric model of filter model, with parameters suggested by or head-related transfer functions is presented for derived from pinna anthropometry, provides a synthesizing binaural sound for practical 3-D good fit to the data. Methods are reported for ad- sound reproduction systems. The suggested justing the model parameters to fit the PRTF model is based on a simplified time-domain data. It is often possible to estimate the model description of the physics of wave propagation parameters from a few geometrical measure- and diffraction, and the components of the mod- ments of the pinna. However, direct estimation el have a one-to-one correspondence with the from pinna anthropometry in general remains an main characteristics of the measured head-relat- unsolved problem, and the nature of this prob- ed transfer functions such as sound diffraction, lem is discussed. delay, and reflection. Their parameters are Convention Paper 7098 derived from some sets of the measurements, and thus enable the model to fit significant per- 11:30 ceptual impact hidden in head-related transfer functions. Finally simple subjective listening P17-9 A Novel Approach to Robotic Monaural tests verify the perceptual effectiveness of Sound Localization—Fakheredine Keyrouz, the model. This will show some promise of per- Abdallah Bou Saleh, Klaus Diepold, Technische mitting efficient implementation in real-world Universität München, Munich, Germany applications. Convention Paper 7096 This paper presents a novel monaural 3-D sound localization technique that robustly esti- mates a sound source within a 2.5-degree 11:30 azimuth deviation and a 5-degree elevation devi- P17-7 Binaural Response Synthesis from Center-of- ation. The proposed system, an upgrade of Head Position Measurements for Stereo monaural-based localization techniques, uses Applications—Sunil Bharitkar, Chris Kyriakakis, two microphones, one inserted within the ear University of Southern California/Audyssey canal of a humanoid head equipped with an arti- Labs., Los Angeles, CA, USA ficial ear, and the second held outside the ear, 5 cm away from the inner microphone. The outer In two-channel or stereo applications, such as microphone is small enough so that minimal for televisions, automotive infotainment, and reflections that might contribute to localization hi-fi systems, the loudspeakers are typically errors are introduced. The system exploits the placed substantially close to each other. The spectral information of the signals from the two sound field generated from such a setup creates microphones in such a way that a simple corre- an image that is perceived as monophonic while lation mechanism, using a generic set of Head lacking sufficient spatial “presence.” Due to this Related Transfer Functions (HRTFs), is used to limitation, a virtual sound technique may be localize the sound sources. The low computa- utilized to widen the soundstage to give the per- tional requirement provides a basis for robotic ception to listener(s) that sound is originating real-time applications. The technique was tested from a wider angle using head-related-transfer through extensive simulations of a noisy rever- functions (HRTFs). In this paper we present a berant room and further through an experimental method to synthesize responses at a listener’s setup. Both results demonstrated the capability ears given simply two room-response measure- of the monaural system to localize, with high ments, where each measurement is obtained accuracy, sound sources in a three-dimensional between a loudspeaker, in a stereo loudspeaker environment even in presence of strong noise setup, and an assumed center-of-head position and distortion. (where the listener is assumed to be seated). Convention Paper 7099 The binaural response synthesis approach uses head-shadowing models (for inter-aural intensity 11:30 differences) and the Woodworth-Schlosberg delay model. This approach is useful when P17-10 Optimized Binaural Modeling for Immersive dummy heads are not readily available for Audio Applications—Christos Tsakostas,1 HRTF measurements as well as to generalize Andreas Floros2 the approach to reflect measurements that 1Holistiks Engineering Systems, Athens, Greece would have been obtained over a large corpus 2Ionian University, Corfu, Greece of data (viz., human subjects). We also compare the responses obtained from this approach with Recent developments related to immersive measurements done with a dummy-head. audio systems mainly originate from binaural audio processing technology. In this paper a Convention Paper 7097 ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 35 Technical Program novel high-quality binaural modeling engine is Since analog magnetic tape technology is no longer a presented suitable for supporting a wide range of part of the audio production process, specific knowledge applications in the area of virtual reality, mobile is endangered to decline. The workshop discusses the playback, and computer games. Based on a set various problems occurring in the transfer process of his- of optimized algorithms for Head-Related Transfer torical magnetic audio tapes. Basing on a definition of Functions (HRTF) equalization, acoustic environ- historical tape brands and an overview of early magnetic ment modeling, and cross-talk cancellation, it is tape developments, the practical handling of critical shown that the proposed binaural engine can tapes is outlined. achieve significantly improved authenticity for 3-D Starting with the analysis of the physical and chemical audio representation in real-time. A complete bin- preservation status of the individual tape, the choice and aural synthesis application is also presented that adjustment of the replay equipment and parameters are demonstrates the efficiency of the proposed discussed. Problems of carrier handling and physical algorithms. restoration are demonstrated, as well as possible signal Convention Paper 7100 enhancement on the playback process only. The work- shop focuses on a practical demonstration of handling 11:30 and reproduction of original tapes from the early ages of magnetic recording, stored under irregular conditions. P17-11 Head-Related Transfer Function Calculation Using Boundary Element Method—Przemyslaw Tutorial 8 Monday, May 7 Plaskota, Andrzej B. Dobrucki, Wroclaw 11:30 – 13:00 Room P University of Technology, Wroclaw, Poland Measuring the head-related transfer function SYNC AND TIMECODE—BRICKS NOT TRICKS (HRTF) is an efficient method in taking the influ- ence of human body on sound spectrum into Presenter: Chris Woolf, Broadcast Engineering consideration. The database used in reproduc- Systems Ltd. tion of the sound source position is built using the measurement results. The base is individual Synchronization and timecode—intimately connected but for each human, which makes it impossible to not to be confused with each other—form the time-axis make a versatile base for all listeners. In this shells of buildings within which most sound practitioners paper a numerical model of an artificial head is must house their work. Rules-of-thumb, tricks-that-seem- presented. The model allows the determination of to-work, and even blind faith often support rather shaky the value of HRTF without the measurements. structures so this tutorial provides some underpinning: a The model includes both geometrical and acousti- foundation of solid bricks. cal parameters. A method that is often used to The session presumes very little and will be useful to determine the acoustical field parameters is the those with limited experience. However, it will also boundary element method, which was used to appeal to those with gnarled hands and a lot of dust calculate the values of HRTF in this paper. under their fingernails but who harbor secret doubts Convention Paper 7101 about the security of their techniques —dark glasses and a false moustache may be worn. 11:30 Exhibitor Seminar Monday, May 7 P17-12 Binaural Room Synthesis and Binaural Sky— 11:30 Room 357 Flexible Virtual Monitoring for Multichannel Audio also with Height Loudspeakers—Klaus AUDIO EXPORT GEORGE NEUMANN Laumann,1 Jörg Hör,1 Gerd Spikofski,1 Roman Stumpner,1 Günther Theile,1 Helmut Wittek2 1Institut für Rundfunktechnik (IRT), Munich, Presenters: Haimo Godler, ORF; Christoph Keller, Germany Audio Export 2Schoeps Mikrofone, Karlsruhe, Germany “KoKo—The ORF Radio Archive” This is a presentation only; no Convention Paper will be available for purchase. The software for archiving, interfacing, and retrieval was developed especially for the needs of ORF. The solution is a perfect example for a seamless integration between the digital production, the on-air tools based on the Workshop 18 Monday, May 7 automation-system Radiomax, and the sound archive. 11:30 – 13:30 Room H This exhibitor seminar presents the main ideas of the ORF radio archive to the visitors. THE REPLAY OF HISTORICAL MAGNETIC TAPE— MORE THAN PRESSING THE PLAY BUTTON Exhibitor Seminar Monday, May 7 12:30 Room 357 Chair: Dietrich Schüller, Phonogrammarchiv Vienna MINNETONKA

Panelists: Friedrich Engel, Magnetic Recording Historian Presenter: John Schur Alexander Füller, Replay Practitioner, Phonogrammarchiv Vienna Bernhard Graf, Replay Practitioner, “Minnetonka’s Audio Tools Workflow Architecture” Phonogrammarchiv Vienna Nadja Wallaszkovits, Chief Technician Audio, Rapidly increasing numbers of audio files and formats Phonogrammarchiv Vienna represent the largest challenge in media production, rais-

36 Audio Engineering Society 122nd Convention Program, 2007 Spring ing numerous productivity and quality issues. Minneton- Exhibitor Seminar Monday, May 7 ka will present the AudioTools Workflow Architecture and 13:30 Room 357 demonstrate how the first products in the AudioTools family—Batch Pro and Version Pro—will help studios to D.A.V.I.D. be productive, efficient, and organized. Presenter: Klaus Hellmich Monday, May 7 12:30 Room 142/143 Standards Committee Meeting on SC-03-07 Audio “Global Interfaces Optimized for Exchanging Large Metadata Media Files”

Combining video content with radio workflows, increas- Workshop 19 Monday, May 7 ingly large multimedia files have to be exchanged among 13:00 – 15:00 Room G multiple locations. As part of its “Moves Media” product line, D.A.V.I.D. provides the DigaMailbox-IP System, PLATINUM ENGINEERS / 5.1 MIXING TECHNIQUES offering simple, secure and reliable connections over the Internet that are optimized for exchanging extensive Chair: George Massenburg, GML, TN, USA media files between bureaus scattered all over the world.

Panelists: Jeff Wolpert, Desert Fish Inc. Session P18 Monday, May 7 14:00 – 17:00 Ronald Prent, Galaxy Studios Room I

Even with the increasing confusion over the future of sur- MICROPHONES AND LOUDSPEAKERS, PART 2 round music mixes there is an increasing demand for efficiently produced surround products for film and televi- sion, as well as for music-only release. At the same time, Chair: Martin Opitz, AKG Acoustics GmbH, Vienna, it is economically tempting for film and television produc- Austria ers to accept surround mixing simulacrums ("faux 5.1" processing such as Unwrap or Logic 7) even when multi- 14:00 channel originals are available for remix. Methods and techniques will be presented in this work- P18-1 Applications of the Acoustic Center—John shop to demonstrate how good, modern surround mixes Vanderkooy, BW Group, Ltd., Steyning, UK and are being made by successful professional practitioners. University of Waterloo, Waterloo, Ontario, Canada The workshop will cover subjects such as: Different approaches to “spreading out” multichannel sources; This paper focuses on uses for the acoustic cen- Strategies for different media; Use of the center channel; ter concept, which in this paper represents a Bass management; Reverb, delay, and other effects; and particular point for a transducer that acts as the Monitoring. origin of its low-frequency radiation or reception. In addition to the panel presentation, there will be several The concept, although new to loudspeakers, has “break-out” presentations the next day in the same room long been employed for microphones when where each of the panelists will present one or more mixes, accurate acoustic pressure calibration is discuss and present techniques, and interact with the audi- required. A theoretical justification of the concept ence. Seats for these sessions will be limited; tickets will be is presented and several calculation methods available after the workshop. are discussed. We first apply the concept to sub- woofers, for which the acoustic center is essen- tially a cabinet dimension away from the center Monday, May 7 13:00 Room 633 of the cabinet. This has an influence on its radia- Technical Committee Meeting on Semantic Audio tion pattern in a normal room with reflecting Analysis walls. A second application that we consider is the effective position of a microphone, which is necessary if it is to be used for accurate calibra- STUDENT EVENT tion of acoustic pressure. A final application that Education Forum Panel we consider is the effective position of the ears Monday, May 7, 13:30– 15:30 Room P on the head at lower frequencies. Calculations show that the acoustic centers of the ears are Moderator: Jason Corey, University of Michigan well away from the head, and the effective ear Panelists: Thomas Goerne, Detmold University of Music separation is larger than expected. This has Douglas McKinnie, Middle Tennessee State implications for the human localization mecha- University nism. Measurements on a Kemar mannequin Hans Vesterberg, School of Music in Piteå show that the separation is even larger than expected from the calculations, and most of this Educators will discuss pedagogical issues related to can be understood, but the measurements at the teaching audio. Success in field of audio requires an un- lowest frequencies are somewhat uncertain. derstanding of not only audio theory and concepts, but Convention Paper 7102 also practical experience with audio. In fact some con- cepts are best assimilated through practical experience. Conversely some practical experiences require prior 14:30 knowledge of the underlying theory to achieve the best P18-2 Development of a Finite Element Headphone results. The discussion will focus on how educators can Model —Dominik Biba, Martin Opitz, AKG best balance the presentation of theory and practice Acoustics GmbH, Vienna, Austria when teaching audio. For the development of high-end headphones, ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 37 Technical Program numerical simulation of the acoustic behavior is background noise, and thus a high recognition an efficient tool. While lumped-element models rate can be expected in low SNR environments are valid for frequencies up to a few kilohertz, fi- such as an engine room. Since noise is not intro- nite-element models are valid for higher frequen- duced within body-conducted signals that are cies too. A headphone model using finite ele- conducted in solids, even within sites such as ments and boundary elements was built in three engine rooms, which are low SNR environments, phases. In parallel to building the numerical construction of a system with a high recognition model a real-world sample was built and mea- rate can be expected. However, within the con- sured. The validity of the model was verified by struction of such systems, in order to create comparison of both the radiated sound field and models specialized for body-conducted speech, the membrane modal behavior. Agreement be- learning data consisting of sentences that must tween measured and computed amplitude fre- be read in numerous times is required. There- quency responses was achieved. fore, in the present paper we applied a method Convention Paper 7103 in which the specific nature of body-conducted speech is reflected within an existing speech 15:00 recognition system with only small numbers of vocalizations. Simultaneously, the measure by P18-3 Development of Camera Mountable 5.0 pretreatment was also worked on. Surround Microphone and Method of Convention Paper 7105 3-Channel to 5-Channel Signal Recomposing 1 System—Minoru Kobayashi, Setsu 16:00 Komiyama,2 Satoshi Kikkawa,2 Takako Kawashima,2 Takeshi Ishii1 P18-5 Revisiting Proximity Effect Using Broadband 1Sanken Microphone Co., Ltd., Tokyo, Japan Signals—Laurent Millot,1,2 Mohammed Elliq,2 2NHK Japan Broadcasting Corporation, Tokyo, Manuel Lopes,2 Gérard Pelé,1,2 Dominique Japan Lambert1,2 1Université Paris, Paris, France This paper describes the development of small 2ENS Louis-Lumiere, Noisy Le Grand, France 5.0 surround microphone and 3-channel to 5- channel signal re-composing system. The princi- Experiments mainly studying the proximity effect pal reason we started this study was broadcast- are presented. Pink noise and music were used ers’ recent demand for a small, light-weight as stimuli and a combo guitar amplifier as camera-mounted 5.0 surround microphone for source to test several microphones: omnidirec- documentaries, dramas, and sports shooting out- tionnal and directional. We plotted in-axis levels doors. We have produced a 5.0 surround micro- and spectral balances as functions of x, the dis- phone; but if we want to use it as a camera- tance to the source. Proximity effect was found mounted microphone, there is a limitation we for omnidirectionnal microphones. In-axis level could not ignore. The limitation is the number of curves show that 1/x law seems poorly valid. audio tracks. The commonly available HD cam- Spectral balance evolutions depend on micro- eras in the market have only 4 audio tracks. In or- phones and, moreover, on stimuli: bigger der to overcome the limitation, we developed a 3- decreases of low frequencies with pink noise; channel to 5-channel signal re-composing larger increases of other frequencies with music. system. For a naked loudspeaker, we found similar Convention Paper 7104 in-axis level curves under and above the cut-off frequency and propose an explanation. Listening 15:30 to equalized music recordings demonstrates will help to demonstrate proximity effect for tested P18-4 Noise-Robust Recognition System Making microphones. Use of Body-Conducted Speech Microphone Convention Paper 7106 —Shunsuke Ishimitsu,1 Masashi Nakayama,2 3 4 Toshikazu Yoshimi, Hirofumi Yanagawa 16:30 1University of Hyogo, Himeji, Hyogo, Japan 2Toyohashi University of Technology, P18-6 Anechoic Measurements of Particle-Velocity Toyohashi, Aichi, Japan Probes Compared to Pressure Gradient and 3Pioneer Corp., Kawagoe, Saitama, Japan Pressure Microphones—Wieslaw Woszczyk,1 4Chiba Institute of Technology, Chiba, Japan Masakazu Iwaki,2 Takehiro Sugimoto,2 Kazuho Ono,2 Hans-Elias de Bree3 In recent years, speech recognition systems 1McGill University, Montreal, Quebec, Canada have been introduced in a wide variety of envi- 2NHK Science & Technical Research ronments such, as vehicle instrumentation. Laboratories, Setagaya-ku, Tokyo, Japan Speech recognition plays an important role in 3Microflown Technologies, Zevenaar, The ships’ chief engineer systems. In such a system, Netherlands speech recognition supports engine room con- trols, and lower than 0-dB signal-to-noise ratio A number of anechoic measurements of (SNR) operability is required. In such a low SNR Microflown™ particle velocity probes are com- environment, a noise signal can be misjudged as pared to measurements of pressure-gradient speech, dramatically decreasing the recognition and pressure microphones made under identical rate. Hence, speech recognition systems operat- acoustical conditions at varying distances from a ing in low SNR environments have not received point source having a wide frequency range. much attention. Therefore, this paper focuses on Detailed measurements show specific response a recognition system that uses body-conducted changes affected by the distance to the source, signals. Such signals are seldom affected by and focus on the importance of transducer cali-

38 Audio Engineering Society 122nd Convention Program, 2007 Spring bration with respect to distance. algorithm for efficient and high-quality audio Convention Paper 7107 equalization. In this approach, the original full- band audio signal is first down-sampled and Session P19 Monday, May 8 14:00 – 17:00 separated into two or more subband signals Room K using a multirate filterbank, after which the equalization is performed in the down-sampled SIGNAL PROCESSING, SOUND QUALITY DESIGN domains. After the equalization, the signal is up-sampled and combined back to a full-band audio signal. Linear phase FIR filters, designed Chair: Alfred Kraker based on the user-controlled parameters, are used to implement the actual equalization. The 14:00 method presented in this paper helps in design- P19-1 Idle Tone Behavior in Sigma Delta Modulation ing an implementation that results in computa- —Enrique Perez Gonzalez, Josh Reiss, Queen tional savings, while still preserving optimal Mary, University of London, London, UK sound quality with any equalization parameter setting. This paper examines the relationship between Convention Paper 7110 various unwanted phenomena that plague audio engineers in the design of sigma delta modula- 15:30 tors. This work aims to clarify the difference and relationship between DC idle tones, long limit cy- P19-4 Correction of Crossover Phase Distortion cles, short limit cycles, and “periodic” short limit Using Reversed Time All-Pass IIR Filter— cycles, while extending the current knowledge in Veronique Adam, Sebastien Benz, Goldmund idle tone behavior. A relationship between the (Audio Networks SA), Geneva, Switzerland periodic input to the quantizer of a 1-bit delta sig- The purpose of this paper is to describe a correc- ma modulator and the appearance of idle tones tion implementation of group delay distortion aris- is shown. It is shown that for a large range of ing from a two-way loudspeaker system crossover. input signal magnitudes, the fundamental fre- Having determined an IIR all-pass filter having a quency of idle tones is proportional to the DC group delay response corresponding to that of the input. This finding has also been used to exam- system crossover to be corrected, we have vali- ine idle tone aliasing. Numerous simulations are dated under Matlab and implemented in DSP the reported which confirm these findings. time reversal solution proposed by S. A. Azizi,. S. Convention Paper 7108 R. Powell, and P. M. Chau, enabling an IIR filter to be inversed, while retaining stability and causality. 14:30 In addition to theory and calculation validation, we P19-2 Low Distortion Sound Reproduction Using have also carried out preliminary listening tests, 8-Bit uC and ZePoC-Algorithms—Jan supporting the evaluation of timber modification, Wellmann, Olaf Schnick, Wolfgang Mathis, sound clarity, and space localization due to the Leibniz University Hannover, Hannover, Germany group delay distortion correction. Convention Paper 7111 The ZePoC-encoding algorithm for Class-D amplification allows the complete separation of 16:00 the signal-baseband from all higher-frequency switching artifacts. Real-Time-ZePoC-Encoding P19-5 Natural Timbre in Room Correction Systems demands a lot of computational power, but in —Jan Abildgaard Pedersen, Henrik Green applications where recorded signals should be Mortensen, Lyngdorf Audio, Skive, Denmark reproduced, they can be encoded by a software- Room correction systems are often found to pro- defined ZePoC-System in advance. Reproduc- vide a timbre, which is described to be artificial ing this pre-encoded signal has very low hard- or unnatural. This paper presents a new ware requirements: no digital-analog-converter approach to this problem, which is based on the or linear amplifier is needed; the playback device finding that part of the influence of a listening must only contain memory; a counter for forming room is natural to the human ear and should not the rectangular output-signal; and, if higher out- be removed by a room correction system. More put-power is required, an additional switching specifically the smooth increase of level toward power-stage and filter. A simple system made up lower frequencies, also referred to as room gain, of an 8-bit micro-controller at 16-Mhz clock-rate must be preserved after applying a room correc- could reach a signal-to-noise-ratio of 80 dB and tion system. In the described system this is done a usable frequency range of up to 15 kHz. A test- as an integral part of the automatic target calcu- system made up of an 8-bit RISC-Processor, lator, which also takes into account the main external memory, and a single-transistor, single characteristics of the used loudspeaker, e.g., power-supply switching-stage will be presented. lower cut-off frequency and directivity index. Convention Paper 7109 Convention Paper 7112

15:00 16:30 P19-3 Efficient, High-Quality Equalization Using a P19-6 Multi Core/Multi Thread Processing in Object- Multirate Filterbank and FIR Filters—Riitta Based Real Time Audio Rendering: Approach- Väänänen, Jarmo Hiipakka, Nokia Research es and Solutions for an Optimization Problem Center, Helsinki, Finland —Ulrich Reiter, Andreas Partzsch, Technische This paper presents a digital signal processing Universität Ilmenau, Ilmenau, Germany ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 39 Technical Program This paper presents considerations, approaches, special ear training, most subjects made great and solutions to the problem of optimization of progress with a nearly 85 percent average cor- thread distribution for multi core processing in rectness rate. real time audio rendering environments. It Convention Paper 7114 explains some basic problems, describes the constraints, and finally suggests an approach 14:00 based on solving an optimization problem by analyzing a directed graph representing the sig- P20-3 The Training and Analysis on Listening nal processing flow. The suggested approach Discrimination of Pure Tone Frequency—Zhi can handle an arbitrary number of CPU cores Liu, Fan Wu, Qing Yang, Beijing Union University, and is therefore well primed for future processor Beijing, China developments. After special and scientific ear training, more Convention Paper 7159 than 90 percent of people will get great progress on the discrimination of pure tone frequency. This has been proven by long-term ear training Session P20 Monday, May 7 14:00 – 15:30 for two groups of subjects. Several pure tones Foyer IK were selected in an octave step for the training. Fifty-seven subjects took part in the training. The POSTERS: PSYCHOACOUSTICS, PERCEPTION, training was conducted once a week, for a total AND LISTENING TESTS of 15 weeks for one group. The average correct- ness rate increased from around 60 percent to above 90 percent. The test results also show 14:00 that human ears have poor discrimination with P20-1 Detection and Lateralization of Sinusoidal middle frequencies, while strong ability with high Signals in the Presence of Dichotic Pink frequency and low frequencies. The relationship Noise—Tuomo Raitio, Heidi-Maria Lehtonen, between the improvement and training time indi- Petteri Laine, Ville Pulkki, Helsinki University of cates that the training has the similar effect as Technology, Espoo, Finland the physical training. Convention Paper 7115 This paper investigates the ability to lateralize low-frequency sound in the presence of interfer- 14:00 ing dichotic noise. This is addressed by measur- ing the detection and lateralization thresholds of P20-4 Virtual Hearing Aid—A Computer Application four sinusoidal signals (62.5, 125, 250, and 500 for Simulating Hearing Aid Performance— Hz) in the presence of uncorrelated pink noise in Andrzej Czyzewski,1 Bozena Kostek,1,2 headphone listening. In lateralization tests the Lukasz Kosikowski1 signals were positioned to left or right by delay- 1Gdansk University of Technology, Gdansk, Poland ing either of the headphone channels by 0.5 ms. 2Institute of Physiology and Pathology of The results show that the lateralization threshold Hearing, Warsaw, Poland does not depart from detection threshold at fre- quencies 250 and 500 Hz. Interestingly, below The virtual hearing aid is a computer application 250 Hz the lateralization threshold rises fast, and allowing an approximate simulation of hearing at 62.5 Hz, the signal has to be amplified 18 dB aid performance. The computer application im- from the detection level before being lateralized plements algorithms simulating band-pass filters, correctly. This suggests that low-frequency ITD compressors, and the perceptual masking decoding mechanisms are easily distracted by strategies for audio signal processing. Individual random changes in a signal phase. This persons' hearing characteristics were taken into explains, at least partly, why the direction of a account for this purpose. The experimental part subwoofer cannot be detected easily in surround comprises verification of engineered algorithms sound listening of broad-band signals. implemented to virtual hearing prosthesis. The Convention Paper 7113 paper also contains results of examinations of patients aimed at verifying the applicability of the proposed signal processing strategy to the 14:00 domain of hearing prostheses. P20-2 The Practice and Study of Ear Training on Convention Paper 7116 Discrimination of Sound Attributes—Zhi Liu, Fan Wu, Qing Yang, Beijing Union University, 14:00 Beijing, China P20-5 Training Versus Practice in Spatial Audio In order to improve subjects’ discrimination of Attribute Evaluation Tasks—Rafael Kassier, sound attributes, an ear training course has Tim Brookes, Francis Rumsey, University of been designed. The training includes: discrimi- Surrey, Guildford, Surrey, UK nation of a pure tone’s frequency, the frequency changes, sound level changes, musical instru- Listener training in published studies has tended ment timbre, irregularity of frequency response. to focus on simple repetitive practice of experi- etc. All the items were carried on in interlaced mental tasks without feedback. Time savings in order to avoid listening fatigue. Meanwhile, listening panel selection and training could be some explanation of the psychoacoustic princi- accomplished if a more general training system ples and some tests were also conducted. In all, could be used and applied to a variety of tasks. 57 subjects, divided into 2 groups, took part in In order for a training system for spatial audio lis- the training course for about 15 weeks. After the tening skills to prove effective, it must demon-

40 Audio Engineering Society 122nd Convention Program, 2007 Spring strate that learned skills are transferable, and it musical sound signal. We considered the cause must compare favorably with repetitive practice of these differences and then tried to connect on specific tasks. A novel study to compare a the sound impression to an analysis result. WT training system with repetitive practice has been of the sound of music was carried out to evalu- extended to include a total of 48 subjects. Trans- ate two amplifiers. The sound quality of amplifier fer is assessed and practice and training are A related to the esthetic factor of the “depth” and compared against a control group for tasks has been analyzed as the high region of WT, involving transfer of spatial audio training. and that of amplifier B related to the force factor Convention Paper 7117 and has been analyzed as the low region of WT. Thus, we were able to visualize the impression 14:00 of listening to the music by correlation of the auditory experiment and wavelet analysis. P20-6 Subjective Assessment of Quality of Multimedia Convention Paper 7120 Signals by Means of A-B Test—Stefan Brachmanski, Wroclaw University of Technology, Wroclaw, Poland Tutorial 9 Monday, May 7 In this paper an automated method of subjective 14:00 – 16:00 Room H assessment of speech, music, image, and video quality has been described. In the method the BALANCED INTERFACES sound, image or video samples were random- ized and paired in A-B sets and then presented Presenter: Bill Whitlock, Jensen Transformers, Inc., to a group of listeners. On the base A-B results a Chatsworth, CA, USA preference matrix was calculated. The conver- sion from the preference matrix to a numerical One goal in the design of audio equipment is to maintain scale was performed with accordance to Thur- a high signal-to-noise ratio. But audio equipment most stone's V model of paired comparisons. The often operates on utility AC power, which, even under method was applied to evaluate an influence of ideal conditions, normally creates ground voltage differ- various coding techniques on a quality of video ences, magnetic fields, and electric fields. RF energy is signals and natural speech. increasingly omnipresent, too. Balanced interfaces are Convention Paper 7118 capable of conveying wide dynamic range analog audio signals while giving them unrivaled immunity to interfer- 14:00 ence. Realizing this full capability in real-world, mass- produced equipment is not necessarily costly but P20-7 Influence of Visual Stimuli on the Sound requires some understanding of several common mis- Quality Evaluation of Loudspeaker Systems takes made by equipment designers. The telephone —Alex Karandreas, Flemming Christensen, company pioneered the widespread use of balanced Aalborg University, Aalborg, Denmark lines and for 50 years virtually all audio equipment used Product sound quality evaluation aims to identify transformers at its balanced inputs and outputs—their relevant attributes and assess their influence on high noise rejection was taken for granted. the overall auditory impression. Extending this When solid-state differential amplifiers began replac- sound-specific rationale, the present paper eval- ing transformers, most designers failed to recognize the uates overall impression in relation to hearing importance of common-mode impedances—which are and vision, specifically for loudspeakers. In order solely responsible for noise rejection. Instead, most to quantify the bias that the image of a loud- believed that “balance” meant equal and opposite signal speaker has on the sound quality evaluation of a swings—which is a myth. As a result, most modern naive listening panel, loudspeaker sounds of audio equipment has poor noise rejection when operat- varied degradation are coupled with positively or ing in real-world systems, even though it may have negatively biasing visual input of actual loud- impressive rejection in a laboratory test. The IEC recog- speakers, and in a separate experiment by pic- nized this dichotomy when they revised their CMRR test tures of the same loudspeakers. standards in 2000 (at the urging of this author). A new Convention Paper 7119 IC uses bootstrap techniques to raise its common-mode impedances, and real-world noise rejection, to levels comparable to the finest transformers. 14:00 The three basic types of balanced output circuits, each P20-8 The Study of Audio Equipment Evaluations with a peculiar set of trade-offs, must be accommodated Using the Sound of Music—Shunsuke by balanced input circuits. Further, certain cable con- Ishimitsu,1 Koji Sakamoto,1 Keitaro Sugawara,2 structions and shield connections can degrade noise Toshikazu Yoshimi,2 Atusushi Makino,2 rejection of an otherwise perfect interface. A very com- Katsuhiro Sasaki,3 Hirofumi Yanagawa4 mon equipment design error, the “pin 1 problem,” causes 1University of Hyogo, Himeji, Hyogo, Japan shield connections to behave as low-impedance audio 2Pioneer Corp., Kawagoe, Saitama, Japan inputs, allowing power-line noise and RF interference to 3Tohoku Pioneer Corp., Tendo, Yamagata, Japan enter the signal path. 4Chiba Institute of Technology, Narashino, Chiba, Japan TECHNICAL TOUR 8 Mediathek In this paper we considered audio equipment Monday, May 7, 14:00 – 17:00 evaluation using musical sounds. Audio ampli- fiers were set up as the evaluation targets, and The Österreichische Mediathek in Vienna is the national sound quality differences between them were archive for audio and video recordings. Over 1.2 million visualized by a wavelet analysis using an actual audio recordings are stored on various carriers like tape, ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 41 Technical Program records, 78s, CDs, DATs, or audio cassettes. Since the Exhibitor Seminar Monday, May 7 year 2000 our staff developed and implemented half auto- 15:00 Room 441 mated workflows for digitizing all these different formats. Meanwhile there exists a complete system for digitizing SENNHEISER over cataloging to long term preservation in a mass storage system. The tour starts with a short presentation of this Presenter: Gregor Zielinsky concept, followed by a visit of the storage vault and various premises. Also interesting is the digital presentation plat- form of the content: the catalog and a growing collection of “High Quality Surround Recording with the New MKH galleries on the Web. 8000 Microphone”

Monday, May 7 14:00 Room 633 In this seminar Sennheiser presents the brand-new MKH Technical Committee Meeting on Audio Recording 8000 Studio Microphone Series for the first time ever. On and Mastering Systems the examples of classical symphony and big band multi- track recordings, this “hands-on” seminar will show sev- eral surround microphone techniques using the MKH Exhibitor Seminar Monday, May 7 8000. The different recordings will be explained and can 14:30 Room 357 be compared and judged by the listeners.

ODEON Monday, May 7 15:00 Room 633 Technical Committee Meeting on Network Audio Presenters: Claus Lynge Christensen, Jens Holger Systems Rindel Workshop 21 Monday, May 7 “ODEON Room Acoustic Simulations” 15:30 – 18:00 Room G ODEON is the state-of-the-art software for room acoustic simulations and auralizations, used in major concert hall RADIO MUSIC RECORDING and theater projects. The seminar will show examples of modeling including CAD-import, source directivity, array Panelists: Martin Leitner, ORF modeling, tools for reflection analysis, and ODEON’s Anton Reininger, ORF multichannel high fidelity auralization. Josef Schütz, ORF Ulrike Schwarz, BR Monday, May 7 14:30 Room 142/143 This workshop will present three senior sound balancing Standards Committee Meeting on SC-02-12 Audio engineers from ORF Radio, sharing their own individual Applications of IEEE 1394 working procedures toward music recordings in 5.1 and stereo for live and deferred broadcasting. The musical Workshop 20 Monday, May 7 genres will encompass classical, jazz, and world music. 15:00 – 17:00 Room 631/632 A method used for surround pickup, the “Karreth-Disc,” will be introduced by the BR engineer. HANDS-ON DEMONSTRATION SUBJECTIVE EVALUATION BY REFERENCE RECORDINGS Exhibitor Seminar Monday, May 7 WITH SURROUND MAIN MIKINGS FOR CLASSICAL 15:30 Room 357 ORCHESTRA SCHOEPS Presenters: Hideo Irimajiri, Mainichi Broadcasting Corp. Toru Kamekawa, Tokyo National University Presenter: Helmut Wittek of Fine Arts and Music Masayuki Mimura, Yomiuri Telecasting Corp. Hideaki Nishida, Asahi Broadcasting Corp. “Double M/S Recording in Stereo and Surround” Koichi Ono, Kansai Telecasting Corp. “Double M/S” is a technique for two-channel or multi- There are many different setups for surround main channel stereo recording. It uses a compact microphone microphones for classical music and orchestra. But it is arrangement that requires only three recorder channels, very difficult to research and study them practically and while allowing full postproduction processing. During the academically under identical conditions and judge their seminar, recorded examples from a range of scenarios performance. Consequently the AES Japan Surround will be presented. The new Schoeps VST Plug-In will Study Project has been organized and put into practice also be introduced and its capabilities demonstrated. after one year of preparation. It was organized around 10 broadcasters and 2 universities; 12 manufacturers sup- Workshop 22 Monday, May 7 ported by HBF provided financial support. There were 15 16:00 – 18:00 Room H different combinations of main and ambience microphone setups that were recorded on 96 channels independently CURRENT ISSUES IN BINAURAL LOUDNESS in Pro Tools HD at 24 bit / 96-kHz. The musical examples ASSESSMENT were performed by the Osaka Philharmonic Orchestra on September 24–27, 2006. Chair: Ville P. Sivonen, University Hospital of Oulu, In this workshop each individual setup will be played Oulun Yliopisto, Finland back. Participants will have the opportunity for feedback in a listening test environment, and the data will be col- Panelists: Pavel Zahorik, University of Louisville, lected for subjective evaluation. Lousiville, KY, USATBA

42 Audio Engineering Society 122nd Convention Program, 2007 Spring The most commonly-used standardized model for pre- Exhibitor Seminar Monday, May 7 dicting the loudness of sound fields (ISO-532B, 1975) 17:30 Room 357 relies on the use of monophonic acoustical measure- ments and does not take into account the signals that FLEX ACOUSTICS actually reach a listener’s ears. That is, the acoustical input to the model is obtained in the absence of a listen- Presenter: Niels Werner Adelman-Larsen er, using a microphone placed in the position where the center of the listener’s head would be. Binaural sound fields encountered in typical measurements are more “Acoustics for Rock—The Flexible Bass Absorber” complex than those assumed by the standard model, Flex Acoustics made what seems to be the first ever sci- most often consisting of a combination of spatially local- entific investigation into the recommended room ized sound source(s) and spatially distributed indirect acoustics for rock and pop concert halls. The results sound. Workshop panelists will discuss current issues in showed the necessity of additional bass absorption and binaural loudness modeling and report the results of led to the invention of a flexible bass absorber. The recent studies on directional loudness perception. investigation and the absorber will be presented.

TECHNICAL TOUR 9 SPECIAL EVENT Organ Concert Recording Organ Recital by Graham Blyth Monday, May 7, 16:00 – 21:00 Monday, May 7, 20:00 – 22:00 This tour will center on the setup and recording of Gra- Jesuitenkirche ham Blyth’s organ concert at the Jesuitenkirche. This will Dr. Ignaz Seipel-Platz, Vienna be a hands-on session with direct involvement of any- Graham Blythe’s traditional organ concert will be given at body attending. The concert will be recorded in 2+2+2+2 the Jesuitenkirche (Jesuit Church), also known as the surround with height and in conventional 5.1 surround. Universitätskirche (University Church), an ornate church The following aspects of the recording process will be on Dr. Ignaz Seipel-Platz, immediately adjacent to the old explained during the setup: microphone selection and University of Vienna buildings. It was built between 1623 placement; equipment configuration, recording chain; and 1627 on the site of an earlier chapel. In 1703, Brother and on-location monitoring via headphones. Andrea Pozzo, SJ, an architect, painter and sculptor, and a master in the quadratura, was invited by the emperor Monday, May 7 16:00 Room 633 Leopold I to redecorate the church. He added twin towers Technical Committee Meeting on Microphones and reworked the facade in an early Baroque style with and Applications narrow horizontal and vertical sections. Despite its rela- tively austere exterior, the interior is remarkably opulent Workshop 23 Monday, May 7 with ersatz marble pillars, gilding, and a number of alle- 16:30 – 17:30 Room P gorical ceiling frescoes. Immediately adjacent is the Aula, where Haydn’s oratorio The Creation had its premiere, as BC 9001: QUALITY MANAGEMENT SYSTEMS did Beethoven’s Seventh Symphony. FOR BROADCASTERS The new organ of the Jesuitenkirche is one of the most famous organs in Vienna. Its control elements are rebuilt Chair: Guillaume Cheneviere, CERTIMEDIA from a typical Cavaillé-Coll console. The program features works by Bach, Franck (Three A group of international experts has compiled a compre- Pieces), Vierne, and Widor (selections from 6th Symphony). hensive certification program that is compatible with ISO Graham Blyth was born in 1948, began playing the 9001:2000. This program is targeted directly to the needs piano aged 4 and received his early musical training as a and organizational procedures of national public and pri- Junior Exhibitioner at Trinity College of Music in London, vate radio and television broadcasters and should be of England. Subsequently, at Bristol University, he took up high interest to anyone looking for a framework to analyze conducting, performing Bach’s St. Matthew Passion be- and certify the practical operation of their stations. fore he was 21. He holds diplomas in Organ Perfor- mance from the Royal College of Organists, The Royal Exhibitor Seminar Monday, May 7 College of Music and Trinity College of Music. In the late 16:30 Room 357 1980s he renewed his studies with Sulemita Aronowsky for piano and Robert Munns for organ. He gives numer- MICROTECH GEFELL ous concerts each year, principally as organist and pianist, but also as a conductor and harpsichord player. Presenter: Ulrich Apel He made his international debut with an organ recital at St. Thomas Church, New York in 1993 and since then “Surround Microphones Not Only for Concert Halls” has played in San Francisco (Grace Cathedral), Los Angeles (Cathedral of Our Lady of Los Angeles), Ams- Microtech-Gefell presents microphones with special rain- terdam, Copenhagen, Munich (Liebfrauen Dom), Paris proof windscreens that are applicable not only in closed (Madeleine and St. Etienne du Mont) and Berlin. He has premises but for recordings of atmos on locations; even lived in Wantage, Oxfordshire, since 1984 where he is for film and television. Herewith Microtech-Gefell translat- currently Artistic Director of Wantage Chamber Concerts ed their outdoor-and environmental experiences into and Director of the Wantage Festival of Arts. their studio-microphones. An application of this wind-and He divides his time between being a designer of pro- rainshield and an operation of new switchable micro- fessional audio equipment (he is a co-founder and Tech- phone UM930 will be shown. nical Director of Soundcraft) and organ related activities. In 2006 he was elected a Fellow of the Royal Society of Monday, May 7 17:00 Room 633 Arts in recognition of his work in product design relating Technical Committee Meeting Advisory Group on to the performing arts. Regulations ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 43 Technical Program Session P21 Tuesday, May 8 09:00 – 12:30 ic musical pitch from an artifact is demonstrated. Room K Convention Paper 7123

INSTRUMENTATION AND MEASUREMENT 10:30 P21-4 Spatial Distribution Meter: A New Method to Chair: Heinrich Pichler, Technical University Vienna, Display Spatial Impression Over Time—Joerg Austria Bitzer, University of Applied Science Oldenburg, Oldenburg, Germany

09:00 A wide-spread method for visualizing the spatial behavior of stereo signals is the vector-scope or P21-1 Advancements in Impulse Response goniometer, which shows the relation between Measurements by Sine Sweeps—Angelo the left and right channel. Well-trained eyes can Farina, University of Parma, Parma, Italy see misbalance and mono compatibility prob- Sine sweeps have been employed for long time lems in these very fast changing figures. Howev- for audio and acoustics measurements, but in er, this analysis tool contains no information over recent years (2000 and later) their usage be- time and no details can be seen. In this paper came much larger, thanks to the computational we introduce a new method for analyzing stereo capabilities of modern computers. Recent signals, which is based on the vector-scope but research results now allow for a further step in shows the behavior over time. The final graph sine sweep measurements, particularly when looks like a spectrogram/sonogram, where the dealing with the problem of measuring impulse axes are time and angle. Useful applications of responses, distortion, and when working with this new spatial distribution meter are the analy- systems that are neither time invariant nor linear. sis of stereo impulse responses, the shift of The paper present some of these advancements stereo base over time, and the typical applica- and provide experimental results aimed to quan- tions of a vector-scope. tify the improvement in signal-to-noise ratio, the Convention Paper 7124 suppression of pre-ringing, and the techniques employed for performing these measurements 11:00 cheaply using a standard PC and a good-quality P21-5 Low Level Audio Signal Transfer Through sound interface, and currently available loud- Transformer Conflicts with Permeability speakers and microphones. Behavior Inside Their Cores—Menno van der Convention Paper 7121 Veen, ir. bureau Vanderveen bv, Zwolle, The Netherlands 09:30 At the threshold of audibility, the signal and flux P21-2 The Challenges of MP3 Player Testing— density levels in an amplifier with audio trans- Steve Temme, Pascal Brunet, Zachary formers are very small. At those levels the Rimkunas, Listen, Inc., Boston, MA, USA relative magnetic permeability of the iron trans- MP3 player audio performance is discussed former core collapses and the inductance of the including measurements of frequency response, transformer becomes very small. The imped- phase response, crosstalk, distortion, sampling ances connected to the transformer plus its sig- rate errors, jitter, and maximum sound pressure nal level and frequency-dependent inductance level with headphones. In order to make these behave as a high pass filter that corner frequen- measurements, several measurement techniques cies slip into the audio bandwidth, resulting in a and algorithms are presented to overcome some nonlinear signal transfer through the trans- of the challenges of testing MP3 players. We dis- former. This paper explains deviations in the cuss test equipment requirements, selection of reproduction of micro details at the threshold of test signals, and the effects of the encoding on audibility. these test signals. A new method for measuring Convention Paper 7125 noncoherent distortion using any test signal including music is also presented. 11:30 Convention Paper 7122 P21-6 New Techniques for Measuring Speech Privacy and Efficiency of Sound Masking 10:00 Systems—Peter Mapp, Peter Mapp + P21-3 Tracking Harmonics and Artifacts in Spectra Associates, Colchester, Essex, UK Using Sinusoidal and Spiral Maps—Palmyra Speech privacy is becoming an increasingly im- Catravas, Union College, Schenectady, NY, portant aspect for many workplace and security USA environments as well as hospital and medical A technique for tracking a harmonic series in a centers where patient confidentially is of critical spectrum using a combination of sinusoidal and importance. Traditionally, speech privacy has spiral maps is described. The spiral map been measured by means of the Articulation enhances patterns that appear when a sinusoid Index, transposed to rate privacy rather than in- is sampled near the Nyquist rate. The corre- telligibility (PI=1-AI). However, this is an indirect spondence between the maps facilitates deriva- and cumbersome method that usually requires a tion of properties and motivates the use of spreadsheet calculation to yield the Privacy curves that cut across the sinusoid or spiral. As Index rating. The paper discusses the potential an application, the spatial separation of a specif- use of STI and STIPa as direct measures of

44 Audio Engineering Society 122nd Convention Program, 2007 Spring speech privacy. The benefits and limitations of 09:00 the methods are highlighted together with the P22-2 Managing the Leap from Synchronous to IP results from a number of case studies. It is con- for Radio Broadcasters: A Look at Equipment, cluded that while the method has potential merit, Network, and Compression Considerations— a number of the limiting factors require further Simon Daniels, APT, Belfast, Ireland, UK research. Convention Paper 7126 Increasingly radio broadcasters are looking at making the leap from synchronous networks to IP networks for their distribution and contribution 12:00 links. The advantages of migrating away from synchronous networks to IP networks are numer- P21-7 Onset Detection Method in Piano Music: ous but often tempered by a number of concerns Sensibility to Threshold Values—Luis regarding the IP transport mechanism including Ortiz-Berenguer, Javier Casajus-Quiros, latency, lost packets, packet size, protocol selec- Marison Torres-Guijarro, Jon Beraceochea, tion, jitter and algorithm selection. This paper will Technical University of Madrid, Madrid, Spain address concerns such as which IP Protocol is most suitable for real time audio delivery, which Piano music transcription requires a stage of algorithm is most suited to IP to reduce the af- onset detection. Every time a new note or chord fects of the inherent latency on an IP network, is played, a new analysis of the note or chord is how to protect against packet loss and how to needed. It is a critical issue to correctly detect if deal with the inherent delay involved in packetiz- a new note or chord has been played. Onset ing audio for delivery over an IP network. detection should have a simple solution, but sev- Convention Paper 7129 eral problems arise when attempting to perform it. This paper present a study of the sensibility of a detection method depending on the adjustable parameters. It also compares some results with 09:00 a simpler method based on the analysis of the P22-3 An XML-Based Approach to Audio Connection derivative of the energy envelope. The methods Management—Richard Foss, Brad Klinkradt, have been tested with six piano compositions. Rhodes University, Grahamstown, South Africa As a conclusion, accurate automatic onset detection in piano music is not a simple task An XML-based approach to firewire audio con- even in the case of notes played alone. nection management has been developed that Convention Paper 7127 allows for the creation of connection manage- ment applications using a range of implementa- tion tools. The XML connection management re- quests flow between a client and server, where Session P22 Tuesday, May 8 09:00 – 10:30 the client and server can reside on the same or Foyer IK separate workstations. The server maintains the state of the firewire audio device configuration as well as information about potential users. XML is POSTERS: AUDIO NETWORKING also used to control user access of devices. Convention Paper 7130

09:00 P22-1 Benefits of Using SIP for Audio Broadcasting 09:00 Applications—Serge de Jaham, AETA Audio P22-4 Rhythm Based Error Correction Approach for Systems, Le Plessis Robinson, France Scalable Audio Streaming Over the Internet— J. C. Cuevas-Martinez, P. Vera-Candeas, N. The SIP protocol has gained popularity for setting Ruiz-Reyes, University of Jaén, Linares, Jaén, up temporary audio links over IP networks for Spain broadcast applications. This paper briefly describes SIP and discusses its distinctive advantages, espe- Multimedia is nowadays the most important kind cially in comparison with proprietary systems. The of information over the internet due to the im- main and obvious benefit is standardization, which pressive growth of the Web and streaming tech- opens the way to interoperation between different nologies. Although there are faster lines, the makes of codecs. SIP, as a signaling protocol, amount of potential users can exceed the actual readily provides efficient methods for link setups, available band width. In that way, scalable audio while preserving ease of use. Now a mature tech- streaming makes it possible. However, error cor- nology in the VoIP field, it is supported by a wide rection is left in a second level of importance for range of network devices and includes provision for multimedia, using in some cases TCP, FEC (for- specific issues like firewall or NAT traversal. As a ward error correction) that are useless in low bit result, SIP should be the key to the transition from rate coders or even nothing. Therefore, in this ISDN to IP networks, while providing at least as paper a rhythm-based error correction approach flexible operating modes. is presented. This solution can avoid important Convention Paper 7128 redundant information, leaving almost all the er- ror processing at the decoder side, without any feedback to the sender. Convention Paper 7131

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Audio Engineering Society 122nd Convention Program, 2007 Spring 45 Technical Program Workshop 24 Tuesday, May 8 Chair: Gerhard Graber, Technical University Graz, 09:00 – 11:00 Room H Austria

DIGITAL STORAGE CARRIERS FOR ARCHIVES 09:30 Chair: Stefani Renner, Ingenieurbuero Renner P23-1 Sound-Transformation and Remixing in Real- Time—Hannes Raffaseder, St. Pölten University of Applied Sciences, St. Pölten, Austria Panelists: Wolfgang Draese, Hitachi Data Systems Magnus Widmer, IBM Switzerland, Storage Starting with a short overview on some very Systems Group basic principles of sound perception, this paper acts on the assumption that recording, storage, Nearly all sectors of the professional audio industry, in editing, and reproduction of audio signals have particular large radio stations, face the same problem: compensated for at least some of these princi- masses of analog archives, waiting to be digitized. ples and, therefore, have significantly changed Because digitalization pervades all levels of production human listening habits. Reflecting on these today, converting from analog archives to digital media changes, the idea of sound transformation and has become a necessity, not only to improve quality and remixing in real-time is suggested as part of the preservation but also to make those archives compatible performance and composition process. Some with current production methods. So, it should be no sur- techniques are explored and a number of imple- prise that digital archiving has become a major field of mentations are introduced. interest in broadcasting. Convention Paper 7132 This workshop will focus on identifying the best stor- age carrier for archives. Since one storage carrier no longer fits every situation—as it did back when analog 10:00 tape predominated—each archive should be able to uti- P23-2 Hybrid Time-Scale Modification of Audio— lize the mass storage system that best meets its needs. Patrick-André Savard, Philippe Gournay, Roch Today’s archives generally use one of three types of Lefebvre, Université de Sherbrooke, Sherbrooke, mass storage carrier: tape-based, hard disk, and optical. Quebec, Canada In this workshop we will present an overview of these three types of carrier along with their specific features This paper presents a novel technique for time- and conclude with an in-depth discussion on how to scale modification (TSM), which integrates time-do- select the best storage system for your archiving needs. main and frequency-domain processing. The Aspects such as the future-proof of the storage and the method relies on frame-by-frame classification to total cost of ownership are examined as well as ques- choose between different techniques adapted to tions about the behavior in high humidity, in dusty envi- different signal types. Provisions are taken to ronments or at high temperatures. In addition, real-life seamlessly switch between techniques. The result experiences with archive installations in broadcasting will is a more universal TSM algorithm that yields con- be shared. tinuous high quality results on a wider range of audio signals. The method is tested on mixed-con- STUDENT EVENT tent signals and formal listening tests results are Student Delegate Assembly Meeting—Part 2 discussed. Tuesday, May 8, 09:00 – 10:30 Room P Convention Paper 7133

At this meeting the SDA will elect a new vice chair. One 10:30 vote will be cast by the designated representative from each recognized AES student section in the European and Inter- P23-3 New Audio Editor Functionality Using national Regions. Judges’ comments and awards will be Harmonic Sinusoids—Wen Xue, Mark Sandler, presented for the Design and Recording Competitions. Queen Mary, University of London, London, UK Plans for future student activities at local, regional, and This paper introduces the application of harmon- international levels will be summarized. ic sinusoid model in an audio editor. The har- monic sinusoid model is a parametric model for TECHNICAL TOUR 10 representing pitched audio events, allowing Boesendorfer Grand Piano Factory amplitude and pitch evolution. While standard Tuesday, May 8, 09:00 – 13:00 audio editors enable the user to select a time or Watch the birth of a famous Boesendorfer Grand Piano! frequency range to edit, with the harmonic sinu- See the initial preprocessing stages of the wooden and soidal parameters estimated in phase, we are iron raw material, the treatment of the iron frame, the able to select a pitched event and edit it as if it sound board, pin blocks, key blocks, the frame installa- were separated from the background. The user tion, string production, voicing, assembly, etc. You can interface is designed as a simple one-click get more information of the process at www. selection, while the user is given further options boesendorfer.com. for better results. Convention Paper 7134 Tuesday, May 8 09:00 Room 142/143 Standards Committee Meeting AESSC Plenary 11:00 P23-4 Measurement and Optimization of Acoustic Session P23 Tuesday, May 8 09:30 – 12:00 Feedback of Control Elements in Cars— Room I Alexander Treiber, Gerhard Gruhler, Heilbronn University, Heilbronn, Germany ANALYSIS AND SYNTHESIS OF SOUND Acoustical quality of control elements in cars is

46 Audio Engineering Society 122nd Convention Program, 2007 Spring increasingly important for manufacturers in order ual loudspeakers is presented. The ability to simultane- to improve the quality, appearance, and security ously optimize different time and frequency domain crite- of their products. This paper presents methods ria and flexible spatial averaging are the main benefits of and tools used in an ongoing research project. the new approach. The solution is featured in the new The project’s goal is to support the industry with BMW M5 Individual High End Sound System, recently the definition of suitable parameters and limits reviewed by Auto, Motor und Sport (number 8/2007 pp. as well as to develop realizable proposals for 146–148) stating: “the most impressive car stereo sound measuring equipment. Jury tests are hereby in the world.” An A/B comparison to conventional filtering used to create the scientific basis for the hear- is available in LEAR’s demo car in our common exhibi- ing-related benchmarking of signals. tion booth. Convention Paper 7135 Workshop 26 Tuesday, May 8 11:30 11:00 – 12:30 Room 560/561 P23-5 On the Training of Multilayer Perceptrons for Speech/Nonspeech Classification in Hearing ACOUSTIC CONCEPTS AND TIME MANAGEMENT Aids—Lorena Álvarez, Enrique Alexandre, FOR SURROUND RECORDINGS—ORGAN Lucas Cuadra, Manuel Rosa-Zurera, Universidad de Alcalá, Alcalá de Henares, Spain Cochairs: Werner Dabringhaus, MDG Ulrich Vette, University of Music and This paper explores the application of multilayer Performing Arts, Vienna perceptrons (MLP) to the problem of speech/ nonspeech classification in digital hearing aids. This presentation concentrates on the comparison of dif- When properly designed and trained, MLPs are ferent surround concepts with the outlook for upcoming able to generate an arbitrary classification fron- technical possibilties. The organ recording from the tier with a relatively low computational complexi- recital of Graham Blyth in the Jesuitenkirche (see Tech- ty. The paper will focus on studying the key influ- nical Tour 9, page 15) will be reproduced in 2+2+2+2, ence of the training process on the performance ITU 5.1 and further loudspeaker setups. of the system. An appropriate election of the training algorithm will help to provide better clas- Exhibitor Seminar Tuesday, May 8 sification with a lower number of neurons in the 11:00 Room 411 network, which leads to a lower computational complexity. The results obtained will be compared SENNHEISER with those obtained from two reference algorithms (the Fisher linear discriminant and the k-Nearest Presenter: Gregor Zielinsky Neighbour), along with some comments regarding the computational complexity. Convention Paper 7136 “High Quality Surround Recording with the New MKH 8000 Microphone” In this seminar Sennheiser presents the brand-new MKH Workshop 25 Tuesday, May 8 8000 Studio Microphone Series for the first time ever. On 09:30 – 11:30 Room 631/632 the examples of classical symphony and big band multi- track recordings, this "hands-on" seminar will show sev- “WANNA FEEL MY LFE?” AND 5.1 OTHER eral surround microphone techniques using the MKH QUESTIONS TO CONFUSE YOUR GIRLFRIEND 8000. The different recordings will be explained and can AND SCARE YOUR GRANDMA be compared and judged by the listeners.

Cochairs: Florian Camerer, Austrian TV Tutorial 10 Tuesday, May 8 Bosse Ternstrom, Swedish Radio 11:30 – 13:30 Room H This radical multichannel battle features two notorious GROUNDING AND SHIELDING surround-sound-nerds with an armoury to die for. From the loudest to the softest, the most brain-melting to the Presenter: Bill Whitlock, Jensen Transformers, Inc., downright boring, a variety of mixes will provide the basis Chatsworth, CA, USA of a discussion of esthetics, philosophies and the quest for the ultimate place in town for a cold beer. Many designers and installers of audio/video systems think of grounding and interfacing as a “black art.” Do signal cables really “pick up” noise, presumably from the Exhibitor Seminar Tuesday, May 8 air like a radio receiver? Equipment manufacturers, 10:30 Room 357 installers, and users rarely understand the real sources of system noise and ground loop problems, routinely LEAR CORPORATION/DIRAC RESEARCH overlooking or ignoring basic laws of physics. Although myth and misinformation are epidemic, this tutorial brings Presenters: Nilo Casimiro Ericsson, Mathias insight and knowledge to the subject. Signals accumu- Johansson, Armin Prommersberger late noise and interference as they flow through system equipment and cables. Both balanced and unbalanced “Dirac Live™—A New Approach to Room Correction interfaces transport signals but are also vulnerable to Algorithms” coupling of interference from the power line and other sources. The realities of ac power distribution and safety A method for improving car audio systems through digital are such that some widely used noise reduction strate- correction of the impulse/frequency responses of individ- gies are both illegal and dangerous. Properly wired, ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 47 Technical Program fully code-compliant systems always exhibit small but Mixing Techniques” will do a hands-on demonstration of significant residual voltages between pieces of equip- several of their mixes with audience interactivity. The mix- ment as well as tiny leakage currents that flow in signal es are presented with the original multitrack sessions, so cables. The unbalanced interface has an intrinsic prob- they can be broken down into the individual elements, and lem, common-impedance coupling, making it very vulner- mixing techniques can be experienced very intuitively. able to noise problems. The balanced interface, because of a property called common-mode rejection, can theo- Schedule: retically nullify noise problems. Balanced interfaces are 12:00 – George Massenburg widely misunderstood and their common-mode rejections 13:00 – Ronald Prent suffer severe degradation in most real-world systems. 14:00 – Jeff Wolpert Many pieces of equipment, because of an innocent 15:00 – George Massenburg design error, have a built-in noise coupling mechanism 16:00 – Ronald Prent dubbed the “pin 1 problem” by Neil Muncy. A simple trou- 17:00 – Jeff Wolpert bleshooting method that uses no test equipment will be described. It can pinpoint the exact location and cause of Workshop 27 Tuesday, May 8 system noise. Most often, devices known as ground iso- 13:00 – 14:30 Room 560/561 lators are the best way to eliminate noise coupling. Sig- nal quality and other practical issues are discussed as ACOUSTIC CONCEPTS AND TIME MANAGEMENT well as how to properly connect unbalanced and bal- FOR SURROUND RECORDINGS—ORGAN anced interfaces to each other. While immunity to RF interference is a part of good equipment design, it must often be provided externally. Finally, power line treat- Cochairs: Werner Dabringhaus, MDG ments such as technical power, balanced power, power Ulrich Vette, University of Music and isolation transformers, and surge suppression are Performing Arts, Vienna discussed. This presentation concentrates on the comparison of dif- ferent surround concepts with the outlook for upcoming Tutorial 11 Tuesday, May 8 technical possibilties. The organ recording from the 11:30 – 13:30 Room P recital of Graham Blyth in the Jesuitenkirche (see Tech- nical Tour 9, page 15) will be reproduced in 2+2+2+2, AUDIO CONTRIBUTION WITH IP OVER ITU 5.1 and further loudspeaker setups. WAN NETWORKS Workshop 28 Tuesday, May 8 Presenters: Mathias Coinchon, EBU 13:00 – 15:00 Room 631/632 Lars Jonsson, EBU/Swedish Radio Gregory Massey, APT Ltd., Ireland HANDS-ON DEMONSTRATION: SUBJECTIVE EVALUATION BY REFERENCE RECORDINGS Audio over IP end units have become common in radio WITH SURROUND MAIN MIKINGS FOR CLASSICAL and tv operations for streaming of audio over IP net- ORCHESTRA works, from remote sites or local offices into main studio centers. ISDN is gradually replaced by IP-circuits. The IP networks used can be well-managed private Presenters: Hideo Irimajiri, Mainichi Broadcasting Corp. networks with controlled quality of service. The Internet Toru Kamekawa, Tokyo National University is increasingly also used for various cases of radio con- of Fine Arts and Music tribution, especially over longer distances. Radio corre- Masayuki Mimura, Yomiuri Telecasting Corp. spondents will have the choice in their equipment to use Hideaki Nishida, Asahi Broadcasting Corp. either ISDN or Internet via ADSL to deliver their reports. Koichi Ono, Kansai Telecasting Corp. In France even distribution to FM transmitters via IP over well managed MPLS-networks is planned to replace old- There are many different setups for surround main er circuits. microphones for classical music and orchestra. But it is More than 15 manufacturers provide equipment for very difficult to research and study them practically and these applications. academically under identical conditions and judge their With almost no exceptions, end units coming from one performance. Consequently the AES Japan Surround manufacturer today are not compatible with another Study Project has been organized and put into practice company's unit. Based on an initiative coming from Ger- after one year of preparation. It was organized around man vendors and broadcasters, the European Broad- 10 broadcasters and 2 universities; 12 manufacturers casting Union, EBU, has started a project group, N/ACIP, supported by HBF provided financial support. There Audio Contribution over IP, to suggest a method of cre- were 15 different combinations of main and ambience ate interoperability for audio over IP. A draft standard microphone setups that were recorded on 96 channels has been proposed by the EBU. Some manufacturers independently in Pro Tools HD at 24 bit / 96-kHz. The already have begun implementation as a minimum inter- musical examples were performed by the Osaka Phil- operability option. harmonic Orchestra on September 24-27, 2006. The tutorial will cover the standardization process and In this workshop each individual setup will be played give a basic overview of audio over IP. back. Participants will have the opportunity for feedback in a listening test environment, and the data will be col- lected for subjective evaluation. Tuesday, May 8 12:00 – 18:00 Room G Session P24 Tuesday, May 8 13:30 – 15:00 PLATINUM BREAKOUT SESSIONS Room I

The panelists of the workshop “Platinum Engineers – 5.1 AUDIO-VIDEO SYSTEMS

48 Audio Engineering Society 122nd Convention Program, 2007 Spring Chair: Gregor Widholm, Musikuniversität Wien, 14:30 Austria P24-3 A New Method for Measuring Time Code Quality —Michael Beckinger, René Rodigast, 13:30 Florian Müller-Kähler, Fraunhofer Institut IDMT, P24-1 Production and Live Transmission of 22.2 Ilmenau, Germany Multichannel Sound with Ultra-High A high-quality time code and word clock syn- Definition TV —Toshiyuki Nishiguchi,1 chronization is essential to prevent audio drop Yasushige Nakayama,1 Reiko Okumura,1 outs and flutters in sound studios. A bad adjust- Takehiro Sugimoto,1 Atsushi Imai,1 Masakazu ment of time code generators, respectively word Iwaki,1 Kimio Hamasaki,1 Akio Ando,1 Shoji clock synchronizers, requires extensive error Kitajima,2 Yutaka Otsuka,2 Satoko Shimaoka2 checks in synchronization networks. For this rea- 1NHK Science & Technical Research son, a new measurement method is presented Laboratories, Tokyo, Japan that enables sound engineers to measure longi- 2NHK Broadcast Engineering Department, tudinal time code jitter and to check time Tokyo, Japan code/word clock synchronization. A 22.2 multichannel sound system was devel- Convention Paper 7139 oped for ultra-high definition TV. The improve- ment in the spatial quality created by this system as compared to that of two-dimensional sound Session P25 Tuesday, May 8 13:30 – 17:30 was evaluated and reported in previous papers. Room K The first experiment on large-scale live produc- tion and transmission of 22.2 multichannel ROOM AND ARCHITECTURAL ACOUSTICS sound with ultra-high definition video was carried AND SOUND REINFORCEMENT out to show the possible application of this sys- tem to next-generation broadcasting. In Tokyo, Chair: Franz Lechleitner, Austrain Academy of 22.2 multichannel sound was live mixed and Sciences, Vienna, Austria transmitted to Osaka using an IP optical network. This paper describes in detail this live production and its transmission using the 22.2 13:30 multichannel sound system. It also discusses P25-1 50 Years of Sound Control Room Design— various issues of sound design, capturing, and Jan Voetmann, DELTA Acoustics, Hørsholm, mixing for three-dimensional sound. Denmark Convention Paper 7137 Sound control room design is an interesting cor- ner of small room acoustics and represents most 14:00 of the problems found here: frequency balanced P24-2 Automated Audio Detection, Segmentation, reverberation time, proper distribution of room and Indexing with Application to modes, low frequency reproduction, sound Postproduction Editing—Charalampos source and receiver positioning, etc. The func- Dimoulas, Christos Vegiris, Kostantinos tion of the control room is twofold, which is often Avdelidis, George Kalliris, George Papanikolaou, overlooked. On one hand, the control room Aristotle University of Thessaloniki, Thessaloniki, together with the monitor loudspeakers should Greece reproduce as faithfully as possible the efforts of the sound engineer and the producer in creating The current paper deals with audio event detec- a new recording. On the other hand, the control tion, segmentation, and characterization in order room should mimic the perceived acoustics of an to be further utilized in postproduction. Browsing, average living room when checking the final selection, and characterization of audio-visual result of the recording. Simply because most content is a tiresome task, especially in audio/ musical productions are aimed at the listening video editing applications, where an enormous environment of a living room. amount of recordings with different characteris- Convention Paper 7140 tics is usually involved. Automated detection, segmentation, and general audio classification 14:00 are essential to deploy flexible and effective au- dio-visual content management. A multi-resolu- P25-2 Acoustics in Rock and Pop Music Halls— tion scanning procedure, based mainly in Niels W. Adelman-Larsen,1 Eric R. Thompson,2 wavelet processing, is currently proposed, where Anders Christian Gade2 various energy-based comparators and signal- 1Flex Acoustics, Lyngby, Denmark complexity metrics have been tested for detec- 2Technical University of Denmark, Lyngby, tion purposes. A variety of audio features, in- Denmark cluding many MPEG-7 audio low level The existing body of literature regarding the descriptors, have been considered for events’ acoustic design of concert halls has focused al- characterization and indexing purposes. Extrac- most exclusively on classical music, although tion of the detection/ characterization results via there are many more performances of rhythmic MPEG-7 description schemes or similar indexing music, including rock and pop. Objective mea- files are considered. surements were made of the acoustics of twenty Convention Paper 7138 rock music venues in Denmark and a question- naire was used in a subjective assessment of ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 49 Technical Program those venues with professional rock musicians The correlation properties of a directly carrier- and sound engineers. Correlations between the modulated code sequence modulation signal are measurements and the questionnaire answers exploited to investigate sound propagation in tur- lead, among others, to a recommendation for bulent air. An experiment is described in which reverberation time as a function of hall volume. the correlation properties of the spread spectrum Since the bass frequency sounds are typically signal are demonstrated and are used to calculate highly amplified, they play an important role in accurate times of flight that compare well with the subjective ratings, and the 63 Hz band must sonic anemometer measurements of speed of be included in objective measurements and rec- sound. The results illustrate that a directly carrier- ommendations. modulated code sequence modulation system Convention Paper 7141 can provide significantly improved ways of mea- suring sound propagation outdoors. Moreover, the 14:30 technique directly measures wind speed. This can be used to compensate the time of flight thus P25-3 The Flexible Bass Absorber—Niels W. allowing the measurement of acoustic impulse Adelman-Larsen,1 Eric R. Thompson,2 responses in nonstationary media, for example Anders Christian Gade2 outdoors, where reliable measurements have pre- 1Flex Acoustics, Lyngby, Denmark viously been difficult to obtain. 2Technical University of Denmark, Lyngby, Convention Paper 7144 Denmark Multipurpose concert halls face a dilemma. They 16:00 host different performance types that require sig- P25-6 Holographic Sound Field Analysis with a nificantly different acoustic conditions in order to Scalable Spherical Microphone-Array—Anton provide the best sound quality to both perform- Schlesinger,1 Giovanni Del Galdo,2 Jörg Lotze,2 ers, sound engineers, and audience. Pop and Stephan Husung,2 Bernhard Albrecht2 rock music often contain high levels of bass 1Delft University of Technology, Delft, The sound energy but still require high definition for Netherlands good sound quality. The mid- and high-frequen- 2Technical University of Ilmenau, Ilmenau, Germany cy absorption is easily regulated, but adjusting the low-frequency absorption has typically been Room acoustic parameters vary greatly with the too expensive or requires too much space to be position of the receiver and of the source, so that practical for multipurpose halls. A practical solu- we cannot extract exhaustive information on the tion to the dilemma has been developed. Mea- room acoustics from independent single-point surements were made on a variable and mobile measurements. Using array measurements per- low-frequency absorber. The paper presents the mits the prediction of the sound field with a high results of prototype sound absorption measure- spatial resolution and leads to a more precise ments as well as elements of the design. assessment of the room acoustic properties. We Convention Paper 7142 propose an array technique to investigate room acoustics by reconstructing the volumetric sound 15:00 field from measurements taken on the surface of a sphere, by means of the methods of nearfield P25-4 Improvements to Binary Amplitude Diffusers acoustical holography (NAH). A virtual spherical —Elizabeth Payne-Johnson,1 Gillian Gehring,1 single-microphone-array was constructed and Jamie Angus2 successfully tested in room acoustical modal 1University of Sheffield, Sheffield, South analysis. Yorkshire, UK Convention Paper 7145 2University of Salford, Salford, Greater Manchester, UK 16:30 Improved forms of diffusion structures based on P25-7 A Comparison of Modeling Techniques for absorption reflection gratings are presented. The Small Acoustic Spaces such as Car Cabins— theory, design, advantages, and limitations of Neil Harris, New Transducers Ltd. (NXT), these structures are discussed and their perfor- Huntingdon, Cambridgeshire, UK mance presented. Two methods of improving performance are suggested. The first structure is This paper results from a case study comparing based on diffusion limited aggregation, which the relative cost effectiveness of three modeling models non-regular fractal growth. The second techniques applied to a small acoustic space structure was a panel with square absorption such as a car cabin. The techniques considered patches of a variable size that was determined by are finite element analysis, analytical solutions, an m-sequence. Of the two, the second structure and the quasi-analytical ray-trace or image performed best. This paper demonstrates that method. A simple test-case is used to compare improved diffusing structures that take up less solution times and accuracy. space than phase reflecting ones are possible. Convention Paper 7146 Convention Paper 7143 17:00 15:30 P25-8 Acoustical and Musical Design of the Sea P25-5 An MLS Method for Nonstationary and Organ in Zadar—Ivan Stamac, Stims d.o.o., Outdoor Acoustic Paths—Jamie Angus, David Zagreb, Croatia Waddington, University of Salford, Salford, Greater Manchester, UK The Sea Organ in Zadar, Croatia, is an awarded

50 Audio Engineering Society 122nd Convention Program, 2007 Spring urban architectural installation using sea wave that a stereo DAC can introduce unexpected random kinetic energy to produce quasi-musical nonlinear effects. These results suggest that a sounds. It contains 35 flue pipes built into subter- future revision of the standards should include a ranean tunnels having outward-bound apertures measure of interchannel nonlinear cross talk in for the sound to emanate. Each flue pipe is blown the stereo DAC. Results are presented and an by a column of air pushed in turn by a column of intermodulation distortion (IMD) loop test pro- moving water entering an immersed tube. The posed to enable this measurement to be made pipes are tuned to 9 tones of the diatonic major with precision. chords G and C6. The series of excited tones is a Convention Paper 7149 statistical function of time- and space-distributed wave energy to particular pipes. In this paper the 13:30 acoustical and musical design propositions and solutions, as parts of the multidiscipline design P26-3 A Low-Distortion Fast-Settling Audio process, will be presented. Oscillator: A Tribute to the Late Peter J. Convention Paper 7147 Baxandall, Analog Audio Expert—John Vanderkooy, University of Waterloo, Waterloo, Ontario, Canada Session P26 Tuesday, May 8 13:30 – 15:00 This paper is dedicated to the memory of Peter Foyer IK Baxandall, well-known for his work in audio and electronics. It is an exposition and analysis of a POSTERS: INSTRUMENTATION AND MEASUREMENT low-distortion fast-settling audio oscillator that he designed and built. Normal oscillators are shown to suffer from amplitude instability when the ther- 13:30 mally-variable controlling resistance has a long P26-1 A Wireless PDA-Based Acoustics time constant. The genius of the present two-in- Measurement Platform—Petros Alexandridis, tegrator design is that it derives its amplitude Nicolas-Alexander Tatlas, Panos Hatziantoniou, stability from the cancellation of two square John Mourjopoulos, University of Patras, Patras, wave signals, of which one is fixed in amplitude, Greece the other proportional to the oscillator output, with a threshold. A detailed analysis of the oscil- The proposed platform allows acoustic measure- lator is presented. The result is an oscillator with ments via a flexible, portable system, based on a distortion below 0.01 percent and settling commercially available hardware, such as a per- times of approximately 1 oscillation period. It is sonal digital assistant (PDA) equipped with a particularly useful in automated test equipment. wireless adapter and a digital audio capture card Convention Paper 7150 as well as a personal computer (PC) intercon- nected with off-the-shelf wireless networking 13:30 hardware. Using this hardware, three software applications were implemented: (i) a device dri- P26-4 Direct Current Offset and Balance for Audio ver that handles the communication of the digital Transformers Used with Paralleled Tubes or audio capture card with the PDA; (ii) a PDA ap- Solid State Devices—Aristide Polisois, Pierre plication that realizes the WiFi connection with Touzelet, S.E.R.E.M.E. S.A., Bondoufle Cedex, the personal computer, also incorporating a France recording function that captures data and pre- sents the user with their analysis; and (iii) the In May 2005 (118th AES Convention, Barcelona, personal computer application that initiates the Spain, Paper 6346), I described a self compen- playback sequence as dictated by the connected sated transformer for SE audio amplifiers, PDA. The system can assist the fast measure- designed as SC-OPT and based on the principle ment of large spaces. Room Impulse Response that an auxiliary winding (tertiary), crossed by (RIR) measurement tests were conducted in a the same current as the primary winding, oppos- laboratory room, in order to evaluate the effec- es a magnetic flux that reduces the overall flux, tiveness and functionality of the measurement produced by the direct current, to almost zero, system. thus leaving the whole magnetic headroom in Convention Paper 7148 the core, for alternating current purposes. The opposed alternating current built in the tertiary was short-circuited with a suitable capacitor. 13:30 Subsequently, a dedicated auxiliary magnetic P26-2 Nonlinear Cross Talk in Personal Computer- core was added to the tertiary, acting as a flux Based Audio Systems—R. Allan Belcher, escape, to reduce the antagonism of the tertiary Jonathon Chambers, Cardiff University, Cardiff, on the primary, which is responsible for the loss Wales, UK of primary inductance. An improved layout to obtain the same result was achieved with the International Electrotechnical Commission (IEC) SC-SCC-SET (Split Core-Stereo Common Cir- and AES standards provide comprehensive tests cuit-Single Ended Transformer), invented by for the performance of audio analog to digital Polisois and Mariani (120th AES Convention, (ADC) and digital to analog (DAC) converters for Paris, France, Paper 6831). This model allows both consumer and professional applications. It significantly improving the bass range. However, is usually assumed that the ADC is more likely to to achieve the DC generated flux cancellation, it degrade audio sound quality than the DAC. needs an external balancing device of the DC Tests on two samples of a professional quality flowing in the two primaries (left and right chan- PC-based audio system are presented that show nel), situated on the same magnetic core. The ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 51 Technical Program transformer described hereafter named 4x4 SC- Tutorial 12 Tuesday, May 8 SCC-SET (4x4 Self Compensated-Single Com- 14:00 – 17:00 Room H mon Circuit-Single Ended Transformer), over- comes this requirement. It also has many novel LOUDSPEAKER NONLINEARITIES— features, proceeding from the adopted arrange- CAUSES, PARAMETERS, SYMPTOMS ment of the windings. Convention Paper 7151 Presenter: Wolfgang Klippel, Klippel GmbH, Dresden, Germany

13:30 This tutorial addresses the relationship between nonlin- ear distortion measurements and nonlinearities that are P26-5 Acoustical Issues and Proposed Improvements the physical causes for signal distortion in loudspeakers, for NASA Spacesuits—Durand R. Begault, headphones, micro-speakers, and other transducers. James L. Hieronymus, NASA Ames Research Using simulation techniques characteristic symptoms are Center, Moffett Field, CA, USA identified for each nonlinearity and presented systemati- cally in a guide for loudspeaker diagnostics. This infor- This paper reviews current acoustical issues rel- mation is important for understanding the implications of evant to the design of future NASA spacesuits, nonlinear parameters and for performing measurements based on measurements conducted in the cur- that describe the loudspeaker more comprehensively. rent Mark III advanced prototype surface suit The practical application of the new techniques are and proposes solutions for improving voice com- demonstrated on practical examples. munications. Methods for mitigating problems including noise from the air supply, structure- borne noise from the suit, and detrimental Tutorial 13 Tuesday, May 8 acoustical reflections are reviewed. 14:00 – 16:00 Room P Convention Paper 7152 AUDIO QUALITY IN NEW LOW BIT-RATE DISTRIBUTION SYSTEMS 13:30 P26-6 Design, Construction, and Qualification of Presenters: Lars Jonsson, EBU/Swedish Radio the New Anechoic Chamber at Laboratorio Gregory Massey, APT Ltd., Ireland, UK de Sonido, Universidad Politécnica de Madrid Gerhard Stoll, IRT Munich, Germany —Juan José Gómez-Alfageme, José Luis Audio quality in new low bit-rate distribution systems— Sánchez-Bote, Elena Blanco-Martin, Universidad with a broadcasters cascade perspective with many Politécnica de Madrid, Madrid, Spain re-encodings at too low bit rates. What is the solution to The year 2005 has seen the design and con- this problem? struction of a new anechoic chamber at Labora- In all new digital distribution systems broadcasters are torio de Sonido of the Universidad Politécnica de faced with the problem of using perceptual coding sys- Madrid. This new chamber has a free volume of tems in all stages of the broadcasting chain, from early 70 cubic meters and is built with rock wool capturing to contribution over editing and on-air. Systems wedges covered with a porous cotton cloth. The in the home are now also using recording with digital me- chamber cutoff frequency is 150 Hz. The cham- dia with low bit rate systems. The resulting overall quality ber has been qualified according to that estab- is often degraded by the cascading artifacts in more than lished in the ISO 3745 standard for the determi- five steps of coding and re-encoding. nation of the maximum distance between the This tutorial discusses state of the art methods and lis- sound source and the measurement position tening test results within the EBU to overcome these where the inverse square law is observed, within problems. some tolerance. For the qualification, different types of excitation signals have been used as TECHNICAL TOUR 12 pure tones, broadband noise, narrow band Vienna State Opera noise, and pseudorandom sequences MLS. Tuesday, May 8, 14:00 – 16:00 Convention Paper 7153 The Vienna State Opera is one of the most famous and important opera houses in the world. With about 50 operas and 20 ballet performances every season it is a TECHNICAL TOUR 11 true repertoire theater with a special bonus: the orches- Music Studio Thomas Rabitsch tra consists of members of the world-renowned Vienna Tuesday, May 8, 13:30 – 16:30 Philharmonic! The tour through this prestigeous house will include a behind-the-scenes look of the stage and Studio Rabitsch is the home of producer/arranger/musi- the most prominent mechanical installations and will con- cian Thomas Rabitsch who started as a keyboard player tinue to various well-known halls within the State Opera in the Viennese Underground Music scene, later palying like the “Gobelin-Saal,” the “Marmorsaal,” and the with Austria’s most popular and successful popstar, Fal- “Schwindfoyer.” There is also a fully-featured organ to be co. Among various pop-music projects for a vast number seen in the “Orgelsaal”before the visitors will get an of Austrian artists, he works for TV and radio companies, in-depth view of the elaborate electro-acoustical possibil- as a music supervisor and composer as well as arranger. ities like the tailor-made TOA ix-3000 theater-console, The main studio features an AMS/Neve Capricorn con- the computer-controlled amplifier- and sequence switch- sole with 5.1 Dynaudio Air20 Monitoring and an exten- ing as well as the enormous loudspeaker installation. sive ProTools and Nuendo rig. Visitors will be able to get Professor Wolfgang Fritz, the head of the electroacousti- a first impression of a current surround sound remix pro- cal departement, will be the guide on this exciting tour ject of a Falco-Live-Concert. through one of the premier sights of Vienna and Austria.

52 Audio Engineering Society 122nd Convention Program, 2007 Spring Session P27 Tuesday, May 8 15:30 – 17:00 Ewa Lukasik, Poznan University of Technology, Foyer IK Poznan, Poland The paper presents an open source Java library POSTERS: ANALYSIS AND SYNTHESIS OF SOUND intended for analysis and classification of musi- cal instrument sounds. It consists of two main 15:30 parts: one devoted for feature extraction and the second performing musical instruments recogni- P27-1 Time Signature Detection by Using tion and similarity assessment. The project’s a Multiresolution Audio Similarity Matrix— plug-in based structure enables further Mikel Gainza, Eugene Coyle, Dublin Institute extendibility of both modules. In the current ver- of Technology, Dublin, Ireland sion two separate sound modeling algorithms A method that estimates the time signature of a have been implemented: k-means and Gaussian piece of music is presented. The approach Mixture Models. The software has been created exploits the repetitive structure of most of the for the purpose of recognition of different exem- music, where the same musical bar is repeated in plars of the same type of instruments and vali- different parts of a piece. The method utilizes a dated for electric guitars, guitar-amplifiers, and multiresolution audio similarity matrix approach, violins. The Java project follows the latest trends which allows comparisons between longer audio in software engineering. It enables the developer segments (bars) by combining comparisons of to easily create highly usable, reliable, and shorter segments (fraction of a note). The time extendable programs. The entire software dis- signature method only depends on musical struc- cussed here is open source. ture, and does not depend on the presence of Convention Paper 7157 percussive instruments or strong musical accents. Convention Paper 7154 15:30 P27-5 Extraction of Long-Term Rhythmic 15:30 Structures Using the Empirical Mode P27-2 Signal Processing Parameters for Tonality Decomposition—Peyman Heydarian, Joshua Estimation—Katy Noland, Mark Sandler, Queen D. Reiss, Queen Mary, University of London, Mary, University of London, London, UK London, UK All musical audio feature extraction techniques Long-term musical structures provide informa- require some form of signal processing as a first tion concerning rhythm, melody, and the compo- step. However, the choice of low level parame- sition. Although highly musically relevant, these ters such as window sizes is often disregarded, structures are difficult to determine using stan- and arbitrary values are chosen. We present an dard signal processing techniques. In this paper investigation into the effects of low level parame- a new technique based on the time-domain ter choice on different tonality estimation algo- empirical mode decomposition is explained. It rithms and show that the low level parameters decomposes a given signal into its constituent can make a significant difference to the results. oscillations that can be modified to produce a We also show that the choice of parameters is new version of the signal. It enables us to ana- algorithm specific, so optimization is required for lyze the long-term metrical structures in musical each different method. signals and provides insight into perceived Convention Paper 7155 rhythms and their relationship to the signal. The technique is explained, and results are reported 15:30 and discussed. Convention Paper 7158 P27-3 Audio Effects for Real-Time Performance Using Beat Tracking—A. M. Stark, M. D. Plumbley, M. E. P. Davies, Queen Mary, Session P28 Tuesday, May 8 15:30 – 16:30 University of London, London, UK Room I We present a new class of digital audio effects that can automatically relate parameter values to the AUDIO IN COMPUTERS (GAMES, INTERNET, tempo of a musical input in real-time. Using a beat AND DESKTOP COMPUTER AUDIO) tracking system as the front end, we demonstrate a tempo-dependent delay effect and a set of beat- synchronous low frequency oscillator (LFO) effects Chair: Gregor Widholm, Musikuniversität Wien, including auto-wah, tremolo, and vibrato. The Austria effects show better performance than might be expected as they are blind to certain beat tracker 15:30 errors. All effects are implemented as VST-plug- P28-1 A Distributed Real-Time Virtual Acoustic ins that operate in real-time, enabling their use Rendering System for Dynamic Geometries— both in live musical performance and the off-line Raine Kajastila,1 Samuel Siltanen,1 Peter modification of studio recordings. Lundén,2 Tapio Lokki,1 Lauri Savioja1 Convention Paper 7156 1Helsinki University of Technology, Espoo, Finland 15:30 2Interactive Institute, Stockholm, Sweden P27-4 JAVA Library for Automatic Musical A novel room acoustic simulation system capa- Instruments Recognition—Piotr Aniola, ble of producing interactive sound environ- ¯

Audio Engineering Society 122nd Convention Program, 2007 Spring 53 Technical Program ments in dynamic and complex 3-D geometries need to remember that there should be available is introduced. The system is distributed to sever- a standardized vario 3.5” jack for stereo and al modules that share the same 3-D geometry. more channel signal supply. This paper presents All changes made by one module are immedi- the basics of how to realize this spatial auditory ately updated in all other modules in real time. event via 4-channel headphones plus the mode The auralization tools of the system include a of a direct audio signal supply that is loudspeak- geometry reduction tool, a beam tracing algo- er compatible. rithm, and a sound rendering application. The Convention Paper 7161 geometry reduction simplifies 3-D models for a beam tracing module that forwards direct sound and early reflection paths for sound rendering. Workshop 29 Tuesday, May 8 The sound rendering application contains an 16:00 – 18:00 Room 631/632 automatic estimation of late reverberation parameters, based on early reflections. Convention Paper 7160 HANDS-ON DEMONSTRATION: SUBJECTIVE EVALUATION BY REFERENCE RECORDINGS WITH SURROUND MAIN MIKINGS FOR CLASSICAL 16:00 ORCHESTRA P28-2 To Create Spatial Auditory Events via More Channel Headphones Related on Portable 5.1 Presenters:Hideo Irimajiri, Mainichi Broadcasting Corp. / 5.0 Surround Reproductions of Sound— Toru Kamekawa, Tokyo National University Florian Koenig, ULTRASONE AG, Tutzing, of Fine Arts and Music Germany Masayuki Mimura, Yomiuri Telecasting Corp. Hideaki Nishida, Asahi Broadcasting Corp. In the future portable surround devices will be Koichi Ono, Kansai Telecasting Corp. the successor of stereo applications in games, cell/mobile phone applications or mp3-players. This portable technique evolution needs world- There are many different setups for surround main wide compatible electro-acoustic headphones microphones for classical music and orchestra. But it is without any “mean” HRTF (head-related transfer very difficult to research and study them practically and function) or binaural DSP (digital signal process- academically under identical conditions and judge their ing). Such headphones offer natural and realistic performance. Consequently the AES Japan Surround 3-D images of sound with a minimum of eleva- Study Project has been organized and put into practice tion effects and a virtual distance perception after one year of preparation. It was organized around 10 front and back. The individualized HRTF, for broadcasters and 2 universities; 12 manufacturers sup- instance, made by a near-field offset/de-centric ported by HBF provided financial support. There were 15 headphone loudspeaker placement at the ear- different combinations of main and ambience micro- cups offers measurable advantages in regards phone setups that were recorded on 96 channels inde- to a “mean” HRTF via DSP. Past AES confer- pendently in Pro Tools HD at 24 bit / 96-kHz. The musi- ences presented papers stating further problems cal examples were performed by the Osaka such as the compatible downmix of 5.1 to 2.0 Philharmonic Orchestra on September 24-27, 2006. signals mainly due to DSP based headphones, In this workshop each individual setup will be played but also to remind to the 4.0 downmix seeing back. Participants will have the opportunity for feedback 4-channel surround sound headphones. As well in a listening test environment, and the data will be col- discussing the connections of headphones we lected for subjective evaluation.

122nd Convention Papers and CD-ROM Presen ted at t 2007 M he 122n ay 5 – d Conve May 8 • ntion Vienna, Convention Papers of many of the presentations given at the Austria 122nd Convention and a CD-ROM containing the 122nd Convention Papers are available from the Audio Engineering Society. Prices follow:

122nd CONVENTION PAPERS (single copy) Member: $ 5. (USA) Nonmember: $ 20. (USA)

122nd CD-ROM (172 papers) Member: $160. (USA) 122nd Convention Nonmember: $200. (USA) 2007 May 5– May 8 Vienna, Austria

54 Audio Engineering Society 122nd Convention Program, 2007 Spring