The AES 43rd International Conference, Audio for WirelesslyNetworked Personal Devices, was held at the POSCO Interna-tional Center at Pohang University of Science and Technology

(POSTECH), Pohang, Republic of Korea, September 29 to October 1,2011. Participants from around the world listened to 18 paper presen-tations, including the keynote lecture, four invited lectures, and aclosing panel discussion.

The conference focused on state-of-the-art audio technologies forwirelessly networked portable devices. It brought together experts inboth audio and systems engineering to discuss recent developmentsin this area. Very broadly, the conference paper topics covered audiocompression, immersive audio, audio over networks, and systems-level issues in networked portable devices.

The conference was held at the POSCO International Center,which is on the campus of Pohang University of Science andTechnology (POSTECH). Pohang became a major industrial centerafter steelmaker POSCO choose to locate its first mills in Pohang in1972. Subsequently, growth of heavy industry in the city brought tothe local economy to a blend of iron, steel, shipbuilding, and fish-eries by the end of the 20th century. POSTECH has strong programsin electrical engineering and communications, including theEducational Institute of Future Information Technology, and hasthe unique Graduate Institute of Ferrous Technology, the only grad-uate program for the study of steel science and technology.

DAY 1The conference cochairs, John Oh of Pulsus and KyungwhoonCheun of POSTECH, opened the conference and welcomed theattendees. John Oh noted that this is the third in a series of confer-ences on this topic. Over this time span there has been significantchange in the landscape of portable devices, which are now morecomputationally capable and have relatively high-bandwidth wire-less channels.

The technical program began with the keynote lecture, “MPEGUnified Speech and Audio Coding,” delivered by SchuylerQuackenbush of Audio Research Labs. Unified speech and audiocoding (USAC) is the newest MPEG audio standard, published inlate 2011. It is able to achieve consistently state-of-the-artcompression performance for any mix of speech and music content.Quackenbush noted that USAC exploits both models of soundperception and models of speech production in order to achieve avery high level of compression. He gave an overview of the archi-tecture of the USAC algorithm and how the various compressiontools operate in response to the instantaneous statistics of arbitrarymixed-content signals. The architecture combines transform-coding tools from MPEG Advanced Audio Coding (AAC), transform-coded-excitation (TCX) tools, analysis-by-synthesis speech codingtools, MPEG spectral band replication (SBR), and MPEG Surroundspatial audio coding tools. He briefly described the tools that give

the greatest compression performance. He presentedthe results of a comprehensive subjective listeningtest showing that the new standard performed better

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876 J. Audio Eng. Soc., Vol. 59, No. 11, 2011 November

Audio for Wirelessly NetworkedPersonal DevicesAES 43rd International ConferenceSeptember 29 to October 1, 2011

Conference cochairs John Oh (left) and Kyungwhoon Cheun Schuyler Quackenbush, keynote speaker

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J. Audio Eng. Soc., Vol. 59, No. 11, 2011 November 877

than either of the state-of-the-art benchmarkcoders—HE-AAC v2 andAMR-WB+—over the entirerange of tested bit rates of 8kb/s for mono to 96 kbps forstereo. In the USAC “sweetspot” of 16 to 32 kbps stereo,this performance advantagewas large. This superiorperformance was maintainedregardless of the type ofcontent coded, e.g., speech,music, or a mix of speechand music.

Following the keynote, thefirst paper session was oninteractive audio.

The first paper in the session was the invited paper “Living withNet Lag,” presented by Chris Chafe, cofounder of MusicianLink anddirector of Stanford’s CCRMA. This talk was delivered via videoconference, with the presenter at Stanford University. His talkopened with a video of an example application in which a “garageband” used laptops to collaborate from geographically disparatelocations via networked audio and video links. He then showed a“jamLink” hardware module that can compress up to four channelsof audio and connect directly to the user’s network router, therebyachieving the lowest possible latency in the audio link. Chafeemphasized that a high-quality experience imposed several require-ments on the networked audio: that the total one-way audio latencybe less than 25 ms, that the network throughput be greater than1 Mbps, that the audio compression scheme (if any) have lowlatency, and that the delivery of packets over the network have lowjitter. The system supports both audio and video, but does notattempt to synchronize audio and video streams. Rather it deliversaudio with the lowest possible latency and video with much higherlatency. Several examples of collaborative music sessions weregiven. Finally, Chafe noted that while tightly-synchronized“jamming” does require less than 25 ms latency, other types ofcollaborative music may be amenable to much higher one-waylatency, permitting a much wider geographic area for collaboration.

The next paper was, “Symphony Orchestra Recording forInteractive Opera Performances,” presented by Lars Hallberg ofLulea University of Technology. The presenter gave a brief overviewof music studies at Lulea University. One recent project at theuniversity was the creation of an opera; the music of the opera,performed as an interactive opera or iOpera, was discussed in thepaper. The concept of iOpera is that the sound from each section ofsymphonic instruments and the vocalists can have its level adjustedinteractively according to personal preference. In order to createthe audio signals to support such interactivity, the symphony wasrecorded using relatively few microphones with an emphasis oncapturing both the instrument sound and the associated hall rever-beration. This was done by recording each section of instrumentsin the symphony (e.g. violin, viola, cello, woodwinds) separatelyand thus having the component signals available to be manipulatedinteractivity. The result was a rich set of symphonic signals thatcould be the basis for many Internet-based interactive presenta-tions.

The second session in the conference “Next-Generation AudioCoding ,” was the first of two sessions on this topic.

The first paper was, “Enhanced Stereo Algorithms in Unified

Speech and Audio Coding,” presented by Eunmi Oh, Samsung. Thisbuilt on the Keynote presentation by describing the stereo codingcomponents present in Unified Speech and Audio Coding (USAC).In USAC, stereo coding tools are found in the frequency-domain(FD) coding tools and again in the MPS212 sound-stage codingtool. The presenter noted that, by her estimation, nearly one-thirdof the core-experiment activity in USAC related to improvements instereo coding. The result is that USAC has significant improve-ments in coding performance for the stereo signal. In the lowestrange of 16 to 24 kbps MPS212, with its parametrically codedstereo phase information can deliver high stereo quality in a verybit-efficient manner. In the mid range of 48 to 64 kbps, a newunified stereo scheme predicts the residual signal from the down-mix (i.e. sum) signal. This reduces the energy in the residual andhence the bits needed to code it. Finally, at higher bitrates, wherethe core coder is operating at the full sampling rate (i.e. no SBR)with a full-band residual, complex stereo prediction in the MDCTdomain can be used with a novel real-to-imaginary transform. Thisis in addition to the conventional Mid/Side stereo coding. Over abroad range of bit rates, the newly developed methods in the USACgive bit-efficient stereo coding with little additional complexityresulting in excellent quality for any audio content.

The second paper was, “LPD Single Mode MPEG-D USACTechnology,” presented by Keunwoo Choi, ETRI. The presenterreviewed the goals of the MPEG Unified Speech and Audio Coding(USAC) work and noted that the USAC Frequency-Domain (FD)coding mode is best for music-like signals and that the LPD modeis best for speech-like signals. Since the USAC TCX windows with“flat tops” have higher spectral side-lobes as compared to a sinewindow, the paper proposes a modified architecture using only sinewindows in the TCX tool. This modification does not alter the over-all one-way latency and does not increase the computationalcomplexity of the proposed system as compared to that of the USACstandard. Subjective quality tests indicate that the quality of theproposed scheme is not different from that of the USAC standard.

The third paper was, “Discrimination Module for Voice/AudioSignals Based on Wavelet Ridges Analysis,” presented by DanielSauceda, Metropolitan Autonomous University, Mexico City. The

Conference committee and staff in front of POSCO

Invited speaker Chris Chafeaddressed the attendees via ateleconference link.

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878 J. Audio Eng. Soc., Vol. 59, No. 11, 2011 November

paper presented a method for categorizing the type of input signalfor the purpose of configuring the set of encoder tools in UnifiedSpeech and Audio Coding (USAC), i.e. for selecting the TCX orACELP coding modes. The method is based on the continuouswavelet transform using a Morlet complex wavelet. Only the lowband of the input signal (up to 1200 Hz) was considered. Thediscrimination criterion involved identifying “wavelet ridges” in thetransform domain, and a signal is classified as “speech” if the ridgeshave more than a minimum time duration, less than a maximumfrequency and less than a maximum standard deviation overfrequency. When analyzing a large set of signals, the algorithm wasable to correctly classify a signal as speech in 60% of the cases.

In the next session the companies that have brought “IndustrialSolution” demonstrations gave a brief introduction to their technol-ogy.

John Richards, Oxford Digital, gave an overview of two Oxford prod-ucts and applications. The first, Sonic Tuning Solutions, is used toremove loudspeaker and cabinet resonances, extending bass frequencyresponse, dynamic range control, and increasing signal loudnessunder the constraints of low power consumption. The second, AudioDSP Core and Toolset, address product production workflow andpermits changes in algorithms on programmable processors even asproducts enter the production phase. The DSP can be programmedusing various methods, ranging from source code (e.g. assembly or C-code) to using optimized graphical block diagram “linkers.”

The day was concluded with the session “Immersive Audio,”which was the first of two sessions on this topic. The first paper was“Acoustic Measure of Causality Artifacts for Generating FocusedSource,” presented by Min-Ho Song, Korea Advanced Institute ofScience and Technology (KAIST). He presented the concept of afocused source, which is a virtual sound source located inside alistening area bounded by loudspeakers. The focused source canonly be perceived by a listener located in the “diverging region”which is beyond the location of the virtual source with respect tothe generating loudspeakers. Causality artifacts are undesiredwavefronts that arrive at the listener’s ear before those of thefocused source. The paper proposes an acoustic measure that canbe used to quantify the amount of causality artifacts for a givenvirtual source and listener position. Audio rendering could be viahigher order ambisonics (HOA), wavefield synthesis (WFS) orAcoustic Focusing. The acoustic measure uses a “threshold ofarrival time difference” which is the difference in wavefront arrivaltimes between the virtual source and an artifact wavefront. Theacoustic measure of causality artifacts, Q, is defined as the ratio ofenergy of the virtual source wave to the artifact wave. Simulationswere presented that gave the value of Q for virtual sources gener-ated by linear arrays of 11 and 21 loudspeakers and circular arraysof 70 loudspeakers.

The second paper was “Two-Band Approximation of VirtualSound Imaging Systems,” presented by Young-cheol Park, YonseiUniversity. Virtual sound imaging is simply how to localize an arbi-trary virtual sound in a virtual space. One method to localize avirtual sound is to use pair-wise panning with stereo loudspeakers,where the amplitude difference between loudspeakers is equivalentto inter-aural time differences (ITD). A second method is to usehead-related transfer function (HRTF) based panning. The paperproposes a trade-off between pair-wise panning and HRTF panningby treating the panning problem in frequency bands: in the lowestband, (below 700 Hz) use pair-wise amplitude panning and thehighest band (1000 to 5000 Hz) use new method based on an inter-aural level differences (ILD) and ITD model. The intermediate banduses a “cross-faded” linear combination of the two methods. It wasfound that, for localization, the proposed model was better thanamplitude panning and comparable to the HRTF-based method,

Daniel Sauceda (left) and JohnRichards engage the audience onDay 2.

A large group of delegates at the 43rd International Conference

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J. Audio Eng. Soc., Vol. 59, No. 11, 2011 November 879

however, in terms of sound quality, theproposed model was much better than HRTF-based panning. The proposed method uses onlya few parameters, which leads to a low-complex-ity implementation.

The final paper in the session was, “AdaptiveCrosstalk Cancellation Using CommonAcoustical Pole and Zero (CAPZ) Model,”presented by Hanwook Chung, Seoul NationalUniversity. The author presented the crosstalkcancellation problem in the context of stereoloudspeakers and the conventional LMS solu-tion to the problem. Next, the CommonAcoustical Pole and Zero (CAPZ) model was presented. In the CAPZmodel for HRTF, the zeros describe the acoustical propagation pathand common poles describe the characteristics of the ear’s auditorysystem. Based on simulations, the proposed model provides bettercrosstalk cancellation and shows lower contralateral response. Theproposed model also provides a linear phase response.

DAY 2The first session of the day was devoted to immersive audio. Theopening paper was an invited paper, “Spatial Equalizer—Designand Implementation,” presented by Yang-Hann Kim of KoreaAdvanced Institute of Science and Technology (KAIST). The presen-ter began by proposing that a spatial equalizer is a device to manip-ulate sound in space and time and that “realistic sound” is repro-duced sound that can not be distinguished from the original livesound. He gave several examples of manipulating sound in space,for example creating a very focused “private” audio listening zone.Several methods of controlling a set of loudspeakers for the pur-pose of sound manipulation were reviewed. Wavefield synthesis(WFS) is one means, and the general WFS case is amenable to sim-plification from a surface of loudspeakers to a line of loudspeakers.However, WFS is subject to several drawbacks due to the simplifica-tions, such as frequency aliasing and effects due to array lengthtruncation. Another means is higher-order ambisonics (HOA). Thismethod reproduces the pressure at a point in space as a sum ofspherical harmonic basis functions. The greater the number ofHOA modes, the larger the “sweet spot” in the reproduction sys-

tem. A third method is “acousticbrightness and contrast control.”This is inherently an ill-posedproblem, but can have a uniquesolution if additional constraintsare imposed. Each method and thedesired manipulation can lead topreferred loudspeaker arrange-ments, for example, linear array,circular array, or arbitrary loca-tions. Given that there are a num-ber of methods, the next questionis: how does one evaluate whichmethod is “best,” where evaluationcould be some objective measure-ments or a subjective preference. Ifone goes the route of subjectivepreference, then it may be appro-priate to impose on the sound gen-eration method a “spatial equal-izer” that is tuned to personalpreferences. Investigating such a

spatial equalizer is Kim’s research problem. He concluded with anoverview of his laboratory for investigating these issues.

The second paper in the session was another invited paper, “The22.2 Multichannel Sounds and its Reproduction at Home andPersonal Environment,” presented by Kimio Hamasaki of the NHKScience and Technology Laboratory. This presentation was via videoconference with the presenter in London. NHK is keenly interestedin a more immersive presence in audiovisual presentations. Forvideo, immersion is directly related to the viewing angle subtendedby the visual display. A good impression of visual immersion resultsfrom at least a 120-degree display field of view. For audio, a three-dimensional soundfield is essential, with stable and precise soundlocalization in all directions. Hamasaki noted that a stereo pair ofloudspeakers can create a virtual center sound source, but as theseparation between speakers becomes greater, the virtual sourcebecomes less stable. Adding a physical center speaker can stabilizethe sound image. This argues for a large number of loudspeakers ina 3-dimensional arrangement. NHK has conducted subjective teststhat suggest that in the horizontal plane at least 8 loudspeakers arerequired for good image stability across all frequencies. Tests ofvertical sound localization suggest that the lateral direction is mostdiscriminating. Front and back discrimination is much less. Suchexperiments led NHK to propose a 22.2 channel loudspeakerconfiguration as optimal for its Super-High Vision (SHV) system.Hamasaki showed an NHK “speaker array frame” video display inwhich there are a large number of loudspeakers incorporated intothe peripheral of the visual display, which permits a large numberof loudspeakers to easily be accommodated into the home environ-ment. Finally, Hamasaki discussed the problem of downmixingfrom M to N channels (e.g. from 22.2 to 5.1 channels). He proposesto use binaural reproduction via headphones for personal devices,including the possibility of a listener’s exact head-related transferfunctions (HRTF).

The next session was on implementations. The first paper was“Approximation of a Virtual-Reverberation Filter for HandheldDevices,” presented by Kwang Myung Jeon of the Gwangju Instituteof Science and Technology (GIST). The motivation for the paper isto reproduce the reverberation present in real acoustic spacesusing a low-complexity method appropriate to hand-held devices.Complexity is reduced by means of an approximation of the rever-beration filter and via indexed-based convolution. A combination ofa constant value and a decaying exponential is used as a threshold,and impulse response values below the threshold are reassigned tozero. A table of indices associated with nonzero values is used toimplement the convolution. In general, the fixed threshold resultsin retention of early reflections while the subsequent decayingexponential selects components of the reverberant sound. Jeongave an audio demonstration of the reverberation with variouslevels of simplification. The paper gives results of a MUSHRA

Invited speaker Yang-HannKim discusses spatial equalizerdesign.

Lars Hallberg asks a question during one ofthe conference workshops.

Young-Cheol Park delivers his paper on avirtual sound imaging system.

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880 J. Audio Eng. Soc., Vol. 59, No. 11, 2011 November

subjective quality evaluation and complexity evaluation for thevarious levels of simplification. The subjectively preferred simpli-fied method reduced the complexity by several orders of magnituderelative to the full impulse response convolution.

The second paper in the session was “System Approach to AvoidAudio Amplifier Oversizing in Mobile Phone Applications,”presented by Eric Sturtzer of the Lyon Institute of Nanotechnology(INL). He noted that conversion from digital signal to acousticenergy in a hand-held device requires two groups of experts: analogelectronics experts and acoustic experts. Typical mobile phonedesigns achieve no more than 0.01% power use efficiency, andamplifier power efficiency is not optimized at nominal outputpower levels. The paper proposes a process to find a globally opti-mal solution to the audio output problem that maximizes qualityand minimizes power consumption. The acoustic conversion effi-ciency is typically no more than 0.0006% for headphones and0.01% for hands-free loudspeakers. The electrical conversion effi-ciency is approximately 70% for class A/B linear power amplifiers(which are most appropriate for headphones) and more than 90%for class D power amplifiers (which are most appropriate for loud-speakers). By using the proposed global approach to system designand by being less strict on some electrical specifications, such asTHD+N level or the frequency range of the audio amplifier, it ispossible to improve the nominal system efficiency without signifi-cant reduction in audio quality.

The final paper in the session was “Audio and Control:Simulation to Embedded in Seconds,” presented by John Richardsof Oxford Digital. He gave an overview of “graphical design” tools inwhich the graphical building blocks are all “prevalidated.” Theimplication of this is that validation is at the system level (i.e. of thesystem-level connections between blocks) and can be done in thesimulation environment. No subsequent validation is needed at therealization level (e.g. at the source code or hardware level). Thispermits rapid workflow from product specification, design, re-design, to realization.

The afternoon of Day 2 attendees took a break from the technicalsessions for an excursion followed by the conference social event.For the excursion, the bus first took the group to the BulguksaTemple in the foothills of Toham-san (Toham Mountain). Thetemple is considered a masterpiece of the golden age of Buddhistart in the Silla kingdom and dates from eighth century, withconstruction completed by Kim Daeseong. Most notable are its twostone pagodas (Dabotap and Seokgatap), stone bridge staircases(Cheongun-gyo), and two gilt-bronze statues of Buddha. Next was avisit to Seokguram Grotto, a man-made grotto containing a stoneBuddha statue. Legend has it that Bulguksa Temple was dedicatedto Kim’s parents in his present life while the Seokguram Grottowas dedicated to Kim’s parents from a previous life. It is very popu-lar to view sunrise over the sea from the grotto entrance. Bulguksaand Seokguram Grotto are UNESCO World Heritage sites. The nextstop was a visit to the Cheonmachong Tomb, one of several verylarge earthen tomb mounds from the fifth or sixth century. Thistomb was excavated, permitting visitors to see the wooden tombchamber and the various precious artifacts that were buried withthe king. As sunset approached, the group visited Anapji, part of thepalace complex of ancient Silla. It was constructed in the seventhcentury. The site consists of several ornate pavilions on the shore ofa beautifully landscaped lake. As the sun set, the group was able tosnap photos of the pavilions perfectly reflected in the lake.

The banquet, held at the Gyeongju Hilton, featured a delicioustraditional Korean meal and a performance of traditional musicand dance.

DAY 3The first session on Day 3 was on interactive Audio. The sessionstarted with the invited paper “Portable and Networked Devices forMusical Creativity,” presented by Juha Backman of Nokia. He notedthat musical performance as both an individual or a group activityis highly valued in any human society. Technological advancementsthat facilitate cooperative music creation in which artists are notphysically colocated can open new avenues of collaborationbetween different musical cultures. The key challenge of distrib-uted performances over any network, long-distance or local, islatency. Studies of performance in acoustic spaces indicate thatsound from fellow musicians must reach a performer within 30 msto provide support for ensemble playing. Longer delays cause con-fusion, even for experienced performers in acoustical spaces. Evenwhen latency issues are addressed, music collaboration platformsshould be accessible to persons having a range of musical experi-ence. New consumer platforms such as tablets and smartphonesoffer great promise in that they have built-in network connectivityand can offer intuitive interfaces.

The second paper in the session was “Gaussian Mixture Model forSinging Voice Separation from Stereophonic Music,” presented byKeun-woo Choi of ETRI. The paper proposed an adaptive predictionmethod for binaural cues such as interchannel level difference(ILD) and interchannel phase difference (IPD) for signals havingcentrally positioned vocals. Based on the assumption that thetarget source has a specific position in the stereophonic soundfield,such as centrally positioned vocals, binaural cues of input mixturesignals were clustered using GMM. Experimental results oncommercial music showed improvement in separation performanceas compared to ordinary hard-decision methods.

The next session was on next-generation audio coding. The firstpaper was “Enhanced Interchannel Correlation (ICC) Synthesis forSpatial Audio Coding,” presented by Dong-il Hyun of YonseiUniversity. He noted that in spatial audio coding, interchannelcorrelation (ICC) may be estimated as the real part of the normal-ized crosscorrelation coefficient between two channels and thuscan result in a negative value. Conventional methods assume thatambient components mixed to two output channels are in anti-phase, while the primary signals are assumed to be in phase. Whena negative-valued ICC is encountered, this assumption can causeexcessive ambient mixing. The paper proposes to solve this problemwith a new ICC synthesis method based on an assumption that theprimary signals are in anti-phase when negative ICCs are indicated.The solution uses an upmix matrix that satisfies the assumption forthe primary componentsin a negative ICC environ-ment. The effectiveness ofthe proposed method wasverified by computer simu-lations and subjectivelistening tests.

The second paper of thesession was “A UnifiedCoding Approach forWireless Audio StreamingBetween Networked Per-sonal Devices,” presentedby David Trainor ofCambridge Silicon Radio.This paper discussed theuse of dynamic composi-tion and adaptation of a

Invited speaker Juha Backmanintroduces the topic of musical creativityon portable networked devices.

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J. Audio Eng. Soc., Vol. 59, No. 11, 2011 November 881

base library of signal-processing functions as a means to provide ascalable and unified audio codec for real-time wireless audiostreaming that can be practically implemented on networkedpersonal devices. For example, a scalable codec could entaildynamic and joint optimization of the following audio codecperformance characteristics based on the performance prioritiesand constraints supplied by the wider system: coded audio quality,

coded audio bit rate, audio coding delay, codec computationalcomplexity, and coded audio robustness to network errors or otherdata loss.

The final event of the conference was the workshop “Audio inFuture Networked Personal Devices.” All the invited speakers andthe attendees participated in the discussion. John Oh addressedend-user benefits such as new experience, high quality, and easyaccessibility that should be provided by future technologies. Healso stated the importance of experiential innovation and interoper-ability of future mobile devices. Juha Backman addressed the possi-ble use of networked mobile terminals for collecting andrepresenting sound scenes. Young-cheol Park remarked that thetechnologies we have now might be enough to find innovativeapplications, and engineering creativity would be more importantthan technological development. Lars Hallberg mentioned idealenvironments for creation of interactive opera, and other atten-dance shared their thoughts on future of immersive audio environ-ments at home and office.

The success of this third AES conference on the topic of wirelesspersonal devices and the explosive growth in the use of thesedevices worldwide make it a good bet that there will be a fourthconference in the not-to-distant future.

Delegates visit the Bulguksa Temple in the foothills of Toham-san.

Editor’s note:The CD-ROM of conference papers can bepurchased at www.aes.org/publications/conferences. Individualpapers can be purchased at www.aes.org/e-lib.

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