AES Warsaw 2015
Paper Session Details

P1 - (Lecture) Spatial Audio—Part 1

Thursday, May 7, 10:00 — 12:30 (Room: Belweder)

Chair:
Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada

P1-1 Subjective Loudness of 22.2 Multichannel Programs—Tomoyasu Komori, NHK Science and Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Waseda University - Shinjuku-ku, Tokyo, Japan; Satoshi Oode, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Kazuho Ono, NHK Engineering System Inc. - Setagaya-ku, Tokyo, Japan; Kensuke Irie, Science & Technology Research Laboratories, Japan Broadcasting Corp. - Setagaya, Tokyo, Japan; Yo Sasaki, NHK Science & Technology Research Laboratories - Kinuta, Setagaya-ku, Tokyo, Japan; Tomomi Hasegawa, NHK Science & Technology Research Laboratories - Kinuta, Setagaya-ku, Tokyo, Japan; Ikuko Sawaya, Science & Technology Research Laboratories, Japan Broadcasting Corp (NHK). - Setagaya, Tokyo, Japan
NHK is planning 8K Super Hi-Vision (SHV) broadcasting with 22.2 multichannel sound as a new broadcasting service. The current loudness measurement algorithm, however, are only standardized up to 5.1 channels in Recommendation ITU-R BS.1770. To extend the algorithm beyond 5.1 ch, we conducted a subjective loudness evaluation of various program materials and formats. The results showed that different formats differed only slightly. Furthermore, we measured objective loudness values on the basis of an algorithm compatible with the current algorithm and found that the objective loudness values had a good correlation with the subjective loudness values.
Convention Paper 9219 (Purchase now)

P1-2 MPEG-D Spatial Audio Object Coding for Dialogue Enhancement (SAOC-DE)—Jouni Paulus, Fraunhofer IIS - Erlangen, Germany; International Audio Laboratories Erlangen - Erlangen, Germany; Jürgen Herre, International Audio Laboratories Erlangen - Erlangen, Germany; Fraunhofer IIS - Erlangen, Germany; Adrian Murtaza, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Leon Terentiv, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Harald Fuchs, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; Sascha Disch, International Audio Laboratories Erlangen - Erlangen, Germany; Falko Ridderbusch, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
The topic of Dialogue Enhancement and personalization of audio has recently received increased attention. Both hearing-impaired and normal-hearing audience benefit, for example, from the possibility of boosting the commentator speech to minimize listening effort, or to attenuate the speech in favor of sports stadium atmosphere in order to enhance the feeling of being there. In late 2014, the ISO/MPEG standardization group made available a new specification, Spatial Audio Object Coding for Dialogue Enhancement (SAOC-DE), which was closely derived from the well-known MPEG-D Spatial Audio Object Coding (SAOC). This paper describes the architecture and features of the new system. The envisioned applications will be discussed and the performance of the new technology is demonstrated in subjective listening tests.
Convention Paper 9220 (Purchase now)

P1-3 Multichannel Systems: Listeners Choose Separate Reproduction of Direct and Reflected Sounds—Piotr Kleczkowski, AGH University of Science and Technology - Krakow, Poland; Aleksandra Król, AGH University of Science and Technology - Krakow, Poland; Pawel Malecki, AGH University of Science and Technology - Krakow, Poland
Arguments can be put forward for the separation of direct and reflected components of the sound field and reproducing them through appropriate transducers, but there is no definite opinion about that. In this work the perceptual effect of separation in commonly used 5.0 and 7.0 multichannel systems was investigated. Four listening experiments were performed involving several schemes of separation and a variety of experimental conditions. The listeners consistently preferred some schemes involving separation to schemes without separation.
Convention Paper 9221 (Purchase now)

P1-4 On the Influence of Headphone Quality in the Spatial Immersion Produced by Binaural Recordings—Pablo Gutierrez-Parera, Universitat de Valencia - Valencia, Spain; Jose J. Lopez, Universidad Politecnica de Valencia - Valencia, Spain; Emanuel Aguilera, Universidad Politecnica de Valencia - Valencia, Spain
The binaural recordings obtained using an acoustic manikin produce a realistic sound immersion played through high quality headphones. However, most people commonly use headphones of inferior quality as the ones provided with smartphones or music players. Factors such as frequency response, distortion, and the disparity between the left-right transducers could be some of the degrading factors. This work lays the foundation for a strategy for studying what are the factors that affect the end result and what level do. A first experiment focuses on the analysis of how the disparity in levels between the two transducers affects the final result. A second test studies the influence of the frequency response. A third test analyzes the effects of distortion using a Volterra kernels scheme for the simulation of the distortion using convolutions. The results of this work reveal how disparity between both transducers can affect the perception of direction. In the case of frequency response the results are more difficult to quantify and further work will be necessary. Finally the study reveals that the distortion produced by the range of headphones tested does not affect to the perception of binaural sound.
Convention Paper 9222 (Purchase now)

P1-5 Binaural Audio with Relative and Pseudo Head Tracking—Christof Faller, Illusonic GmbH - Zurich, Switzerland; EPFL - Lausanne, Switzerland; Fritz Menzer, Technische Universität München - Munich, Germany; Christophe Tournery, Illusonic GmbH - Lausanne, Switzerland
While it has been known for years that head-tracking can significantly improve binaural rendering, it has not been widely used in consumer applications. The goal of the proposed techniques is to leverage head tracking, while making it more usable for mobile applications, where the sound image shall not have an absolute position in space. Relative head tracking keeps the sound image in front, while reducing the effect of head movements to only small fluctuations. Relative head tracking can be implemented with only a gyrometer; there is no need for absolute direction. An even more economical technique with the goal to improve binaural rendering is pseudo head tracking. It generates small head movements using a random process without resorting to a gyroscope. The results of a subjective test indicate that both relative and pseudo head tracking can contribute to spaciousness and front/back differentiation.
Convention Paper 9223 (Purchase now)

 
 

P2 - (Poster) Education and Perception

Thursday, May 7, 10:30 — 12:30 (Foyer)

P2-1 Effects of Ear Training on Education on Sound Quality of Digital Audio for Non-Technical Undergraduates—Akira Nishimura, Tokyo Univeristy Information Sciences - Chiba-shi, Japan
This paper demonstrates the effectiveness of ear training in lectures on audio processing conducted over the 2013 and 2014 academic terms. Student understanding of the lecture content was assessed by comparing scores of written tests that covered the sound quality of perceptual audio codecs and other topics, which were administered after lectures with and without ear training on identifying bit rates of sound files. The same approach was applied to a lecture on audio digitizing and ear training on identifying sampling frequencies. The test scores of assessments that focused on the sound quality of perceptual audio codecs were significantly higher among students who had participated in ear training compared to those who had not participated in such training. In contrast, no significant difference was found in the group scores of participants tested after ear training on identifying sampling frequency. The effectiveness of ear training being limited to perceptual codings was investigated in terms of prior knowledge of the technical terms.
Convention Paper 9224 (Purchase now)

P2-2 Evaluation of the Low-Delay Coding of Applause and Hand-Clapping Sounds Caused by Music Appreciation—Kazuhiko Kawahara, Kyushu University - Fukuoka, Japan; Yutaka Kamamoto, NTT Communication Science Laboratories - Kanagawa, Japan; Akira Omoto, Kyushu University - Fukuoka, Japan; Takehiro Moriya, NTT Communicatin Science Labs - Atsugi-shi, Kanagawa-ken, Japan
Recently, the improvement of network resources enables us to distribute the contents in real-time. This paper presents the low-delay coding of applause sound and hand-clapping sound with less parameters by means of synthesizing these sounds at the receiver site. We found that number of people clapping their hands were corresponding to a sound volume of applause. In other words, no one considers who is clapping. Additionally, on the hand-clapping sound, the time interval of clapping also should be important. Based on such information, preliminary experiments confirm that our approach, which synthesize applause and hand-clapping sound from a few parameters, successfully generates natural applause and hand-clapping sounds.
Convention Paper 9225 (Purchase now)

P2-3 Subjective Evaluation of High Resolution Audio under In-Car Listening Environments—Mitsunori Mizumachi, Kyushu Institute of Technology - Kitakyushu, Fukuoka, Japan; Ryuta Yamamoto, Digifusion Japan Co., Ltd. - Hiroshima, Japan; Katsuyuki Niyada, Hiroshima Cosmopolitan University - Hiroshima, Japan
High resolution audio (HRA) becomes increasingly popular both for music production and the consumers. It enables to record a music performance in a wide-band and precise digital audio format. It is, however, unclear in its perceptual advantage under some listening environments. In this study listening tests were carried out inside cars where 34 participants listened to the same music in four different audio formats. The participants chose an audio format with better quality in paired comparison among 192 kHz/24 bits PCM, 48 kHz/16 bits PCM, and two kinds of lossy-compressed MPEG audio formats. The participants, who are familiar with HRA and live music performance, could significantly discriminate among the audio formats.
Convention Paper 9226 (Purchase now)

P2-4 Investigating Factors that Guitar Players to Perceive Depending on Amount of Distortion in Timbre—Koji Tsumoto, Tokyo University of the Arts - Adachi-ku, Tokyo, Japan; Atsushi Marui, Tokyo University of the Arts - Tokyo, Japan; Toru Kamekawa, Tokyo University of the Arts - Adachi-ku, Tokyo, Japan
Typical electric guitar timbre could be classified into three classes according to amount of distortion. Timbre with less distortion is called "Clean" and heavily distorted timbre is called "Distorted." Timbre between "Clean" and "Distorted" is called "Crunch." To investigate the factors that guitar players perceive depending on amount of distortion, semantic differential analysis using eight bipolar adjective scales was employed. Twenty guitar players including six professionals played their instruments through a guitar amp with nine different distortion level settings. Two factors were found in factor analysis, and "Clean" and "Distorted" were located opposite to each other. "Crunch" was located in the middle of latent factors and each anchoring adjectives used in the evaluation. Also the result of regression analysis indicated "Activeness Factor" was the reliable factor corresponding to the amount of distortion.
Convention Paper 9227 (Purchase now)

P2-5 Perception of Timbre Changes vs. Temporary Threshold Shift—Bartlomiej Kruk, Wroclaw University of Technology - Wroclaw, Poland; Maurycy Kin, Wroclaw University of Technology - Wroclaw, Poland
The paper presents results of research on an influence of Temporary Threshold Shift (TTS) on the detection of changes in timbre of musical samples. The experiment was carried out with conditions that normally exist in a studio when sound material is recorded and mixed. The level of sound exposure that represents the noise signal is 90 dB, and this is an average value of sound level existing in control room. This musical material may be treated as a noise so TTS phenomenon may occur after several time durations: 60, 90, and 120 minutes. Ten subjects participated in the main part of the experiment and all of them have the normal hearing thresholds. The stimuli contained the musical material with introduced changes in timbre up to +/–6 dB in low (100 Hz), middle (1 kHz), and high frequency (10 kHz) regions. It turned out that listening to the music with an exposure of 90 dB for 1 hour influences the hearing thresholds for middle frequency region (about 1–2 kHz); and this has been reflected in a perception of timbre changes: after 1 hour listening the changes of spectrum in middle-frequencies region are perceived with a threshold of 3 dB while the changes of low and high ranges of spectrum were perceived with the thresholds of 1.8 and 1.5 dB, respectively. After the longer exposure, the thresholds shifted up to 3.5 dB for the all investigated stimuli.
Convention Paper 9228 (Purchase now)

P2-6 Hybrid Multiresolution Analysis of “Punch” in Musical Signals—Steven Fenton, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK; Jonathan Wakefield, University of Huddersfield - Huddersfield, UK
This paper presents a hybrid multi-resolution technique for the extraction and measurement of attributes contained within a musical signal. Decomposing music into simpler percussive, harmonic, and noise components is useful when detailed extraction of signal attributes is required. The key parameter of interest in this paper is that of punch. A methodology is explored that decomposes the musical signal using a critically sampled constant-Q filterbank of quadrature mirror filters (QMF) before adaptive windowed short term Fourier transforms (STFT). The proposed hybrid method offers accuracy in both the time and frequency domains. Following the decomposition transform process, attributes are analyzed. It is shown that analysis of these components may yield parameters that would be of use in both mixing/mastering and also audio transcription and retrieval.
Convention Paper 9229 (Purchase now)

P2-7 Five Aspects of Maximizing Objectivity from Perceptual Evaluations of Loudspeakers: A Literature Study—Christer Volk, DELTA SenseLab - Hørsholm, Denmark; Aalborg University, Department of Electronic Systems - Aalborg East, Denmark; Søren Bech, Bang & Olufsen a/s - Struer, Denmark; Aalborg University - Aalborg, Denmark; Torben H. Pedersen, DELTA SenseLab - Hørsholm, Denmark; Flemming Christensen, Aalborg University - Aalborg, Denmark
A literature study was conducted focusing on maximizing objectivity of results from listening evaluations aimed at establishing the relationship between physical and perceptual measurements of loudspeakers. The purpose of this study was to identify and examine factors influencing the objectivity of data from the listening evaluations. This paper addresses the following subset of aspects for increasing the objectivity of data from listening tests: The choice of perceptual attributes, relevance of perceptual attributes, choice of loudness equalization strategy, optimum listening room specifications, as well as loudspeaker listening in-situ vs. listening to recordings of loudspeakers over headphones.
Convention Paper 9230 (Purchase now)

P2-8 Modding Game Audio for Education—Ricardo Bragança, United Arab Emirates University - Al Ain, Abu Dhabi, UAE
Worldwide there is no formal curriculum for game audio. This paper will dwell on what can be done to change the current status quo. We intend to shed some light on possible solutions and guidelines that can be used by schools in order to achieve a higher awareness on how to implement game audio successfully in a university’s curriculum. We believe that due to its interdisciplinary nature, cross faculty cooperation and corporate partnerships are advised and will promote a better understanding on how to tackle the topic. Constructivist teaching methods and a student centric inquiry based learning approach is suggested to enhance the learning experience and insure adequate content absorption.
Convention Paper 9231 (Purchase now)

P2-9 The Acoustic Properties of Different Types of Earplug Used by Sound Engineers—Bartlomiej Kruk, Wroclaw University of Technology - Wroclaw, Poland; Michal Luczynski, Wroclaw University of Technology - Wroclaw, Poland
The main aim of this paper is to test various types of earplugs used by sound engineers. At live events, when sound engineers need to use earplugs for health reasons, it is very important that they maintain correct hearing perception abilities. The linear frequency response allows to avoid mistakes when working with sound. Earplugs were tested for attenuation depending on frequency. The authors tested earplugs in the different methods: subjectively using pure tone audiometry and objectively using the designed and created ear canal model. Research allowed to choose the appropriate earplugs for sound engineering purposes.
Convention Paper 9233 (Purchase now)

P2-10 Psychoacoustic Annoyance Monitoring with WASN for Assessment in Urban Areas—Jaume Segura-Garcia, Universitat de Valencia - Burjassot, Valencia, Spain; Polytechnic University of Valencia; Santiago Felici, Universitat de Valencia - Burjassot, Spain; Maximo Cobos, Universitat de Valencia - Burjassot, Spain; Ana Torres, Polytechic University School of Cuenca - Cuenca, Spain; Juan M. Navarro, Universidad Católica San Antonio - Murcia - Guadalupe (Murcia), Spain
The assessment of the subjective annoyance caused by noise pollution in cities is a matter of major importance as its influence is growing-up in urban areas. Different methods and techniques have been used to model this annoyance in terms of several psychoacoustic parameters, which define different aspects of the acoustic affection from noise pollution in the human behavior. In this paper we describe a monitoring system based on a wireless acoustic sensor network that measures and computes the psychoacoustic metrics following the Zwicker's annoyance model, in a distributed way and at different points simultaneously in urban areas. The nodes of this network run complex algorithms to find out these metrics. These nodes are Single-Board Computer platforms, in particular Raspberry Pi.
Convention Paper 9234 (Purchase now)

P2-11 The Advanced Sound System Listening Room at Dolby—Sunil G. Bharitkar, Dolby Laboratories - San Francisco, CA, USA
A listening room at Dolby has been designed to test the spatial and timbre performance of next generation audio formats recommended in the new ITU-R BS.2051-0 (Advanced sound system for program production). The room has been best designed to conform to the new ITU-R BS.1116-2 (Methods for the subjective assessment of small impairments in audio systems) specification for testing the performance of next-generation audio codecs. Detailed physical and acoustical measurements have been conducted using international standards that demonstrate satisfying elements in both these international recommendations and that are presented in the paper. Subjective testing is ongoing and some preliminary feedback is included as well.
Convention Paper 9337 (Purchase now)

 
 

P3 - (Lecture) Recording and Production

Thursday, May 7, 15:00 — 17:30 (Room: Belweder)

P3-1 Perceptual Evaluation of Music Mixing Practices—Brecht De Man, Queen Mary University of London - London, UK; Matthew Boerum, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT); Brett Leonard, University of Nebraska at Omaha - Omaha, NE, USA; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; George Massenburg, Schulich School of Music, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada; Joshua D. Reiss, Queen Mary University of London - London, UK
The relation of music production practices to preference is still poorly understood. Due to the highly complex process of mixing music, few studies have been able to reliably investigate mixing engineering, as investigating one process parameter or feature without considering the correlation with other parameters inevitably oversimplifies the problem. In this paper we present an experiment where different mixes of different songs, obtained with a representative set of audio engineering tools, are rated by experienced subjects. The relation between the perceived mix quality and sonic features extracted from the mixes is investigated, and we find that a number of features correlate with quality.
Convention Paper 9235 (Purchase now)

P3-2 Automated Equalization of Mobile Device’s Microphones—Przemek Maziewski, Intel Technology Poland - Gdansk, Poland
To achieve high and uniform audio quality in mobile devices their microphones must be equalized. The equalization is typically done manually, requiring lab time, costly equipment, and experienced engineers. This paper presents an automated equalization procedure. It is done using a reference microphone and an external loudspeaker. Each internal microphone is tuned to match the reference microphone’s response to the excitations generated via the external loudspeaker. Additionally, each internal microphone’s equalization is amended with the inverse equalization characteristic of the reference microphone calculated against the chosen reference, e.g., specific certification requirements. This way the final equalization includes both the internal vs reference microphone delta and the correction required for the reference microphone to pass the chosen certification.
Convention Paper 9236 (Purchase now)

P3-3 Use of Audio Editors in Radio Production—Chris Baume, BBC Research and Development - London, UK; University of Surrey - Guildford, Surrey, UK; Mark D. Plumbley, University of Surrey - Guildford, Surrey, UK; Janko Calic, University of Surrey - Guildford, Surrey, UK
Audio editing is performed at scale in the production of radio, but often the tools used are poorly targeted toward the task at hand. There are a number of audio analysis techniques that have the potential to aid radio producers, but without a detailed understanding of their process and requirements, it can be difficult to apply these methods. To aid this understanding, a study of radio production practice was conducted on three varied case studies—a news bulletin, drama, and documentary. It examined the audio/metadata workflow, the roles and motivations of the producers, and environmental factors. The study found that producers prefer to interact with higher-level representations of audio content like transcripts and enjoy working on paper. The study also identified opportunities to improve the work flow with tools that link audio to text, highlight repetitions, compare takes, and segment speakers.
Convention Paper 9237 (Purchase now)

P3-4 Cross-Adaptive Polarity Switching Strategies for Optimization of Audio Mixes—Pedro Duarte Pestana, Catholic University of Oporto - CITAR - Oporto, Portugal; Universidade Lusíada de Lisboa - Lisbon, Portugal; Joshua D. Reiss, Queen Mary University of London - London, UK; Alvaro Barbosa, Catholic University of Oporto - CITAR - Oporto, Portugal
Crest factor is an often overlooked part of audio production, yet it acts as an important limit to overall loudness. We propose a technique to optimize relative polarities in order to yield the lowest possible peak value. We suggest this is a way of addressing loudness maximization that is more sonically transparent than peak limiting or compression. We also explore additional uses that polarity analysis may have in the context of mixing audio. Results show this is a fairly effective strategy, with average crest factor reductions of 3 dB, resulting in equivalent values for loudness enhancement. While still not comparable to the amount of reduction peak limiters are typically used for, the approach is seen as more transparent via subjective evaluation, through a multi-stimulus test.
Convention Paper 9238 (Purchase now)

P3-5 Adaptation and Varying Acoustical Condition and the Resulting Effect on Consistency of High Frequency Preference—Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Brett Leonard, University of Nebraska at Omaha - Omaha, NE, USA; Stuart Bremner, McGill University - Montreal, Quebec, Canada; The Centre for Interdiciplinary Research in music Media & Technology - Montreal, QC, Canada; Grzegorz Sikora, Bang & Olufsen Deutschland GmbH - Pullach, Germany
The ability to consistently evaluate the frequency response of a music program in a listening room is one of the most fundamental tasks required of an audio engineer. This study requires expert listeners to adjust the high frequency content of audio program under the influence of three different acoustic conditions. The length of exposure is varied to test the role of adaptation on such a task. Results show that there is not a significant difference in the variance of participants’ results when exposed to one condition for a longer period of time. However, some individual subjects exhibit adaptive tendencies within the temporal range tested.
Convention Paper 9239 (Purchase now)

 
 

P4 - (Poster) Spatial Audio

Thursday, May 7, 16:00 — 18:00 (Foyer)

P4-1 Variation of Interaural Time Difference Caused by Head Offset Relative to Coordinate Origin—Guangzheng Yu, South China University of Technology - Guangzhou, Guangdong, China; Yuye Wu, South China University of Technology - Guangzhou, China
Interaural time difference (ITD) is related with spatial position (distance and direction) of the sound source and head size. Assuming the sound source and the coordinate system are fixed, the position relationship between the sound source and head center will be influenced by the head offset relative to the coordinate origin, which may lead to the spatial distribution distortion of ITD in measuring head-related transfer functions (HRTFs). In this paper the variation of ITD caused by head offset is analyzed using the conventional Woodworth ITD model consisting of a spherical head and a point sound source. Results show that the forward (or backward) offsets of head result in small variation of ITDs, however, the spatial distribution distortion of ITDs introduced by the rightward (or leftward) offset of head is unacceptable.
Convention Paper 9240 (Purchase now)

P4-2 Functional Representation for Efficient Interpolations of Head Related Transfer Functions in Mobile Headphone Listening—Joseph Sinker, University of Salford - Salford, UK; Jamie Angus, University of Salford - Salford, Greater Manchester, UK
In this paper two common methods of HRTF/HRIR dataset interpolation, that is simple linear interpolation in the time and frequency domain, are assessed using a Normalized Mean Square Error metric. Frequency domain linear interpolation is shown to be the superior of the two methods, but both suffer from poor behavior and inconsistency over interpolated regions. An alternative interpolation approach based upon the Principal Component Analysis of the dataset is offered; the method uses a novel application of the Discrete Cosine Transform to obtain a functional representation of the PCA weight vectors that may be queried for any angle on a continuous scale. The PCA/DCT method is shown to perform favorably to the simple time domain method, even when applied to a dataset that has been heavily compressed during both the PCA and DCT analysis.
Convention Paper 9241 (Purchase now)

P4-3 Binaural Hearing Aids with Wireless Microphone Systems including Speaker Localization and Spatialization—Gilles Courtois, Swiss Federal Institute of Technology (EPFL) - Lausanne, Switzerland; Patrick Marmaroli, Swiss Federal Institute of Technology (EPFL) - Lausanne, Switzerland; Hervé Lissek, Swiss Federal Institute of Technology (EPFL) - Lausanne, Switzerland; Yves Oesch, Phonak Communciations AG - Murten, Switzerland; William Balande, Phonak Communciations AG - Murten, Switzerland
The digital wireless microphones systems for hearing aids have been developed to provide a clean and intelligible speech signal to hearing-impaired listeners for, e.g., school or teleconference applications. In this technology, the voice of the speaker is picked up by a body-worn microphone, wirelessly transmitted to the hearing aids and rendered in a diotic way (same signal at both ears), preventing any speaker localization clues from being provided. The reported algorithm performs a real-time binaural localization and tracking of the talker so that the clean speech signal can then be spatialized, according to its estimated position relative to the aided listener. This feature is supposed to increase comfort, sense of immersion, and intelligibility for the users of such wireless microphone systems.
Convention Paper 9242 (Purchase now)

P4-4 On the Development of a Matlab-Based Tool for Real-Time Spatial Audio Rendering—Gabriel Moreno, Universitat de Valencia - Burjassot, Spain; spat; Maximo Cobos, Universitat de Valencia - Burjassot, Spain; Jesus Lopez-Ballester, Universitat de Valencia - Burjassot, Spain; Pablo Gutierrez-Parera, Universitat de Valencia - Valencia, Spain; Jaume Segura-Garcia, Universitat de Valencia - Burjassot, Valencia, Spain; Polytechnic University of Valencia; Ana Torres, Polytechic University School of Cuenca - Cuenca, Spain
Spatial audio has been a topic of intensive research in the last decades. Although there are many tools available for developing real-time spatial sound systems, most of them work under audio-oriented frameworks. However, despite a significant number of signal processing researchers and engineers who develop their algorithms in MATLAB, there is not currently any MATLAB-based tool for rapid spatial audio system prototyping and algorithm testing. This paper presents a tool for spatial audio research and education under this framework. The presented tool provides the user with a friendly graphical user interface (GUI) that allows to move freely a number of sound sources in 3D and to develop specific functions to be used during their reproduction.
Convention Paper 9243 (Purchase now)

P4-5 Psychoacoustic Investigation on the Auralization of Spherical Microphone Array Processing with Wave Field Synthesis—Gyan Vardhan Singh, Technische Universität Ilmenau - Ilmenau, Germany
In the present work we have investigated the perceptual effects induced by various errors and artifacts that arise when spherical microphone arrays are used on the recording side. For spatial audio it is very important to characterize the acoustic scene in three-dimensional space. In order to achieve this three dimensional characterization of a sonic scene, spherical microphone arrays are employed. The use of these spherical arrays has some inherent issues because of some errors and by virtue of mathematics involved the in processing. In this paper we analyzed these issues on recording side (spherical microphone array) that plague the audio quality on the rendering side and did a psychoacoustic investigation to access the extent to which the errors and artifacts produce a perceivable affect during auralization when the acoustic scene is reproduced using wave field synthesis.
Convention Paper 9244 (Purchase now)

P4-6 Evaluation of a Frequency-Domain Source Position Estimator for VBAP-Panned Recordings—Alexander Adami, International Audio Laboratories Erlangen - Erlangen, Germany; Jürgen Herre, International Audio Laboratories Erlangen - Erlangen, Germany; Fraunhofer IIS - Erlangen, Germany
A frequency-domain source position estimator is presented that extracts the position of a VBAP-panned directional source by means of a direct-ambience signal decomposition. The directional signal components are used to derive an estimate of the panning gains that can be used to derive the estimated source position. We evaluated the mean estimated source positions as a function of the ideal source position as well as of different ambience energy levels using simulations. Additionally, we analyzed the influence of a second directional source to the estimated source positions.
Convention Paper 9245 (Purchase now)

P4-7 A Listener Position Adaptive Stereo System for Object-Based Reproduction—Marcos F. Simón Gálvez, University of Southampton - Southampton, UK; Dylan Menzies, University of Southampton - Southampton, UK; Filippo Maria Fazi, University of Southampton - Southampton, Hampshire, UK; Teofilo de Campos, University of Surrey - Guildford, Surrey, UK; Adrian Hilton, University of Surrey - Guildford, Surrey, UK
Stereo reproduction of spatial audio allows the creation of stable acoustic images when the listener is placed in the sweet spot, a small region in the vicinity of the axis of symmetry between both loudspeakers. If the listener moves slightly towards one of the sources, however, the images collapse to the loudspeaker the listener is leaning to. In order to overcome such limitation, a stereo reproduction technique that adapts the sweet spot to the listener position is presented here. This strategy introduces a new approach that maximizes listener immersion by rendering object-based audio, in which several audio objects or sources are placed at virtual locations between the stereo span. By using a video tracking device, the listener is allowed to move freely between the loudspeaker span, while loudspeaker outputs are compensated using conventional panning algorithms so that the position of the different audio objects is kept independent from that of the listener.
Convention Paper 9246 (Purchase now)

P4-8 Optimization of Reproduced Wave Surface for Three-Dimensional Panning—Akio Ando, University of Toyama - Toyama, Japan; Hiro Furuya, University of Toyama - Toyama, Japan; Masafumi Fujii, University of Toyama - Toyama, Japan; Minoru Tahara, University of Toyama - Toyama, Japan
Three-dimensional panning is an essential tool for production of 3D sound material. The typical method is an amplitude panning. The amplitude panning generates the weighting coefficients on the basis of the direction of virtual sound source (desired direction) and the directions of loudspeakers, or the distances between the virtual source and each loudspeaker. It then distributes the weighted signal of the corresponding sound into loudspeakers. The amplitude panning sometimes brings blurred image and deteriorates the timbre of sound. In this paper we propose the new method that optimizes the shape of the wave surface synthesized by multiple loudspeakers. The computer simulation with the frontal six-loudspeaker system showed that the new method achieved the improvement of the reproduced wave surface of sound and its frequency response.
Convention Paper 9247 (Purchase now)

P4-9 Estimation of the Radiation Pattern of a Violin During the Performance Using Plenacoustic Methods—Antonio Canclini, Politecnico di Milano - Milan, Italy; Luca Mucci, Politecnico di Milano - Milan, Italy; Fabio Antonacci, Politecnico di Milano - Milan, Italy; Augusto Sarti, Politecnico di Milano - Milan, Italy; Stefano Tubaro, Politecnico di Milano - Milan, Italy
We propose a method for estimating the 3D radiation pattern of violins during the performance of a musician. A rectangular array of 32 microphones is adopted for measuring the energy radiated by the violin in the observed directions. In order to gather measurements from all the 3D angular directions, the musician is free to move and rotate in front of the array. The position and orientation of the violin is estimated through a tracking system. As the adopted hardware is very compact and non-invasive, the musician plays in a natural fashion, thus replicating the radiation conditions of a real scenario. The experimental results prove the accuracy and the effectiveness of the method.
Convention Paper 9248 (Purchase now)

P4-10 An Evaluation of the IDHOA Ambisonics Decoder in Irregular Planar Layouts—Davide Scaini, Universitat Pompeu Fabra - Barcelona, Spain; Dolby Iberia S.L. - Barcelona, Spain; Daniel Arteaga, Dolby Iberia S.L. - Barcelona, Spain; Universitat Pompeu Fabra - Barcelona, Spain
In previous papers we presented an algorithm for decoding higher order Ambisonics for irregular real-world 3D loudspeaker arrays, implemented in the form of IDHOA, an open source project. IDHOA has many features tailored for the reproduction of Ambisonics in real audio venues. In order to benchmark the performance of the decoder against other decoding solutions, we restrict the decoder to 2D layouts, and in particular to the well studied 5.1 and 7.1 surround layouts and in particular to the well studied stereo, 5.1,and 7.1 surrounds. We report on the results of the objective evaluation of the IDHOA decoder in these layouts and of the subjective evaluation in 5.1 by benchmarking IDHOA against different decoding solutions.
Convention Paper 9249 (Purchase now)

P4-11 A General Purpose Modular Microphone Array for Spatial Audio Acquisition—Jesus Lopez-Ballester, Universitat de Valencia - Burjassot, Spain; Maximo Cobos, Universitat de Valencia - Burjassot, Spain; Juan J. Perez-Solano, Universitat de Valencia - Burjassot, Spain; Gabriel Moreno, Universitat de Valencia - Burjassot, Spain; spat; Jaume Segura-Garcia, Universitat de Valencia - Burjassot, Valencia, Spain; Polytechnic University of Valencia
Sound acquisition for spatial audio applications usually requires the use of microphone arrays. Surround recording and advanced reproduction techniques such as Ambisonics or Wave-Field Synthesis usually require the use of multi-capsule microphones. In this context, a proper sound acquisition system is necessary for achieving the desired effect. Besides spatial audio reproduction, other applications such as source localization, speech enhancement or acoustic monitoring using distributed microphone arrays are becoming increasingly important. In this paper we present the design of a general-purpose modular microphone array to be used in the above application contexts. The presented system allows performing multichannel recordings using multiple capsules arranged in different 2D and 3D geometries.
Convention Paper 9250 (Purchase now)

P4-12 Immersive Content in Three Dimensional Recording Techniques for Single Instruments in Popular Music—Bryan Martin, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, QC, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Brett Leonard, University of Nebraska at Omaha - Omaha, NE, USA; David Benson, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Will Howie, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada
“3D Audio” has become a popular topic in recent years. A great deal of research is underway in spatial sound reproduction through computer modeling and signal processing, while less focus is being placed on actual recording practice. This study is a preliminary test in establishing effective levels of height-channel information based on the results of a listening test. In this case, an acoustic guitar was used as the source. Eight discrete channels of height information were combined with an eight-channel surround sound mix reproduced at the listener’s ear height. Data from the resulting listening test suggests that while substantial levels of height channel information increase the effect of immersion, more subtle levels fail to provide increased immersion over the conventional multichannel mix.
Convention Paper 9251 (Purchase now)

 
 

P5 - (Lecture) Audio Signal Processing

Friday, May 8, 09:00 — 11:00 (Room: Belweder)

Chair:
Christoph M. Musialik, Sennheiser Audio Labs - Waldshut-Tiengen, Germany

P5-1 Multi-Rate System for Arbitrary Audio Processing—Daekyoung Noh, DTS, Inc. - Santa Ana, CA, USA
An efficient multi-rate system for arbitrary audio processing is proposed. In order to minimize computational complexity, high sampling rate signals are decimated and split into two subbands. The process can be repeated in the low band to obtain a maximally decimated system. Then, only the lowest band is being processed with arbitrary audio processing. Amplitude and group delay compensation are performed to the rest of the bands to minimize aliasing noise and amplitude distortion that can be caused when the bands are recombined due to arbitrary audio processing performed in the low bands. The Goertzel algorithm transition band addition/subtraction method is introduced for group delay correction in real-time processing. Once arbitrary processing is done in the lowest band the subbands are then up-sampled and recombined. Finally, test results and computational advantages are discussed.
Convention Paper 9252 (Purchase now)

P5-2 A Short-Term Analysis of a Digital Sigma-Delta Modulator with Nonstationary Audio Signals—Marcin Lewandowski, Warsaw University of Technology - Warsaw, Poland
Signal conversion quality of sigma-delta (S?) digital-to-analog audio converters (DACs) mainly depends on the S? modulator's parameters. Conventional quality examination of S? audio DACs has been performed in the frequency domain and can be considered indicative of quality only in the case of linear and stationary systems. However, highly nonlinear and nonstationary S? modulators create errors that depend on the input signal. In this study, a method for evaluating S? modulators in the time-domain is presented. Simulations and analysis were performed with the use of music signals. Results showed that the short-term performance of digital S? modulators is highly correlated with the variation of the input signal. This is particularly important as S? modulators are commonly used in DACs of both consumer and professional audio equipment.
Convention Paper 9253 (Purchase now)

P5-3 Application of Sinusoidal Coding for Enhanced Bandwidth Extension in MPEG-H USAC—Tomasz Zernicki, Zylia sp. z o.o. - Poznan, Poland; Maciej Bartkowiak, Poznan University of Technology - Poznan, Poland; Lukasz Januszkiewicz, Zylia sp. z.o.o. - Poznan, Poland; Marcin Chryszczanowicz, Zylia sp. z.o.o. - Poznan, Poland
A new audio coding technique applicable to very low bit rates is proposed. The existing MPEG-D standard of Unified Speech and Audio Coding (USAC) is enhanced by a new High Frequency Sinusoidal Coder (HFSC), employed for improving the subjective quality of high frequency spectral content. The paper gives an overview of the new technique as well as it offers some insight into the operation modes and delay issues. A statistically evidenced significant improvement of the audio quality resulting from applying this technique is demonstrated.
Convention Paper 9254 (Purchase now)

P5-4 Practical Considerations of Time-Varying Feedback Delay Networks—Sebastian J. Schlecht, International Audio Laboratories - Erlangen, Germany; Emanuël A. P. Habets, International Audio Laboratories Erlangen - Erlangen, Germany
Feedback delay networks (FDNs) can be efficiently used to generate parametric artificial reverberation. Recently, the authors proposed a novel approach to time-varying FDNs by introducing a time-varying feedback matrix. The formulation of the time-varying feedback matrix was given in the complex eigenvalue domain, whereas this contribution specifies the requirements for real valued time-domain processing. In addition, the computational costs of different time-varying feedback matrices, which depend on the matrix type and modulation function, are discussed. In a performance evaluation, the proposed orthogonal matrix modulation is compared to a direct interpolation of the matrix entries.
Convention Paper 9255 (Purchase now)

 
 

P6 - (Lecture) Room Acoustics

Friday, May 8, 09:00 — 11:30 (Room: Królewski)

Chair:
Lauri Savioja, Aalto University - Aalto, Finland

P6-1 Radio Studio Acoustics Part 1: Subjective Evaluation—Ian Dash, Australian Broadcasting Corporation - Sydney, NSW, Australia; Mark Bowry, Australian Broadcasting Corporation - Sydney, NSW, Australia
A subjective evaluation exercise to determine the perceived acoustical quality of 12 small acoustic spaces used for radio production and presentation was conducted using two recording methods. One set of recordings used multiple microphones and the other set used a single microphone. A common listening test was run using both sets of recordings. The novel approach using multiple microphones was found to be a more sensitive and reliable method, and leads to different conclusions on acoustic performance criteria compared with a more conventional assessment. Group differences due to the reader, listener gender, listener location, and listener skill level were performed using ANOVA.
Convention Paper 9256 (Purchase now)

P6-2 Radio Studio Acoustics Part 2: Correlation of Objective Measurements with Subjective Assessment—Ian Dash, Australian Broadcasting Corporation - Sydney, NSW, Australia; Mark Bowry, Australian Broadcasting Corporation - Sydney, NSW, Australia
A subjective evaluation was made of the acoustic quality of 12 small- to medium-sized acoustic spaces used for radio production and presentation. Correlation analysis was used to relate the results from that evaluation to measured objective acoustical parameters of those rooms. The results suggest that low early reverberation time at lower frequencies is of high importance to listeners, and that listeners prefer consistent reverberation time in all bands from 125 Hz upwards.
Convention Paper 9257 (Purchase now)

P6-3 Estimation of Room Reflection Parameters for a Reverberant Spatial Audio Object—Luca Remaggi, University of Surrey - Guildford, Surrey, UK; Philip Jackson, University of Surrey - Guildford, Surrey, UK; Philip Coleman, University of Surrey - Guildford, Surrey, UK
Estimating and parameterizing the early and late reflections of an enclosed space is an interesting topic in acoustics. With a suitable set of parameters, the current concept of a spatial audio object (SAO), which is typically limited to either direct (dry) sound or diffuse field components, could be extended to afford an editable spatial description of the room acoustics. In this paper we present an analysis/synthesis method for parameterizing a set of measured room impulse responses (RIRs). RIRs were recorded in a medium-sized auditorium, using a uniform circular array of microphones representing the perspective of a listener in the front row. During the analysis process, these RIRs were decomposed, in time, into three parts: the direct sound, the early reflections, and the late reflections. From the direct sound and early reflections, parameters were extracted for the length, amplitude, and direction of arrival (DOA) of the propagation paths by exploiting the dynamic programming projected phase-slope algorithm (DYPSA) and classical delay-and-sum beamformer (DSB). Their spectral envelope was calculated using linear predictive coding (LPC). Late reflections were modeled by frequency-dependent decays excited by band-limited Gaussian noise. The combination of these parameters for a given source position and the direct source signal represents the reverberant or “wet” spatial audio object. RIRs synthesized for a specified rendering and reproduction arrangement were convolved with dry sources to form reverberant components of the sound scene. The resulting signals demonstrated potential for these techniques, e.g., in SAO reproduction over a 22.2 surround sound system.
Convention Paper 9258 (Purchase now)

P6-4 Sound Radiation Control for Reducing the Effect of Strong Reflections—Jiho Chang, Samsung Electronics Co. Ltd. - Suwon, Gyeonggi-do, Korea; Jaeyoun Cho, Samsung Electronics Co. Ltd. - Suwon-si, Gyeonggi-do, Korea; Yoonjae Lee, Samsung Electronics Co. Ltd. - Suwon, Gyeonggi-do, Korea
This paper is concerned with sound radiation control of a loudspeaker that has omni-directional radiation in free field condition. When the loudspeaker is placed very close to flat surfaces, the reflections from the surfaces are as strong as the direct sounds and deteriorate the omni-directionality. This study assumes a proximate wall and attempts to analyze the effect of a reflection from the wall in terms of the directivity and to improve the omni-directionality by using a circular array of loudspeakers. For a given distance, weights for loudspeakers are calculated that make the sound radiation omni-directional in spite of the wall.
Convention Paper 9259 (Purchase now)

P6-5 Acoustical Measurements of Warsaw’s Chamber Opera House Using Two Types of Sound Sources for Subsequent Auralization—Wieslaw Woszczyk, McGill University - Montreal, QC, Canada; Tadeusz Fidecki, F. Chopin University of Music - Warszawa, Poland; Jung Wook (Jonathan) Hong, McGill University - Montreal, QC, Canada; GKL Audio Inc. - Montreal, QC, Canada; Tomasz Rudzki, Frederic Chopin University of Music - Warszawa, Poland; Coherent Audio Systems; David Benson, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
Impulse response measurements using log sine sweeps were made in the Warsaw’s Chamber Opera House in eight microphone locations on the floor area and at two locations on the balcony, with two microphone elevations, using two types of sound sources having different directional radiation characteristics. The Opera House having only 159 seats originates from 1775 and is renowned for its excellent acoustics fitting for Mozart operas. The measurements show how within this relatively small venue, an opera director can create a wide range of acoustic perspectives for voices and instruments, and achieve a desired dramatic effect. In a subsequent multichannel auralization, anechoic instrumental and vocal sounds were placed virtually in the opera house, and a listening panel compared the renderings. The experiment underlines the importance of choosing directional characteristics of sound sources used in the measurements of room impulse responses intended for subsequent applications
Convention Paper 9260 (Purchase now)

 
 

P7 - (Poster) Audio Signal Processing

Friday, May 8, 11:00 — 13:00 (Foyer)

P7-1 Feature Learning for Classifying Drum Components from Nonnegative Matrix Factorization—Matthias Leimeister, Native Instruments GmbH - Berlin, Germany
This paper explores automatic feature learning methods to classify percussive components in nonnegative matrix factorization (NMF). To circumvent the necessity of designing appropriate spectral and temporal features for component clustering, as usually used in NMF-based transcription systems, multilayer perceptrons and deep belief networks are trained directly on the factorization of a large number of isolated samples of kick and snare drums. The learned features are then used to assign components resulting from the analysis of polyphonic music to the different drum classes and retrieve the temporal activation curves. The evaluation on a set of 145 excerpts of polyphonic music shows that the algorithms can efficiently classify drum components and compare favorably to a classic “bag-of-features” approach using support vector machines and spectral mid-level features.
Convention Paper 9261 (Purchase now)

P7-2 Blind Bandwidth Extension System Utilizing Advanced Spectral Envelope Predictor—Kihyun Choo, Samsung Electronics Co., Ltd. - Suwon, Korea; Anton Porov, Samsung R&D Institute Russia - Moscow, Russia; ITMO University - Saint-Petersburg, Russia; Eunmi Oh, Samsung Electronics Co., Ltd. - Suwon, Korea
We propose a blind bandwidth extension (BWE) technique that improves the quality of a narrow-band speech signal using time domain extension and spectral envelope prediction in the frequency domain. In the time domain, we use a spectral double shifting method. Further, a new spectral envelope predictor is introduced in the frequency domain. We observe less distortion when the attribute is transferred from low to high frequency, instead of reflecting the original high band. The proposed blind BWE system is applied to the decoded output of an adaptive multi-rate (AMR) codec at 12.2 kbps to generate a high-frequency spectrum from 4 to 8 kHz. The blind BWE was objectively evaluated with the AMR and AMR wideband codecs and subjectively evaluated by comparing it with the AMR.
Convention Paper 9262 (Purchase now)

P7-3 Time Domain Extrapolative Packet Loss Concealment for MDCT Based Voice Codec—Shen Huang, Dolby Laboratories - Beijing, China; Xuejing Sun, Dolby Laboratories - Beijing, China
A novel low latency packet loss concealment technique for transform-based codecs is proposed. The algorithm combines signals from Inverse Modulated Discrete Cosine Transform (IMDCT) domain and the previous reconstructed signal from time domain with aligned phase, with which a pitch-synchronized concealment is performed. This minimizes aliasing artifacts that occur in MDCT domain concealment for voiced speech signals. For unvoiced speech, speech-shaped comfort noise is inserted. When there is a burst loss, a position-dependent concealment process is performed for different stages of packet losses. Subjective listening tests using both naïve and expert listeners suggest that the proposed algorithm generates fewer artifacts and offers significantly better performance against legacy packet repetition based approaches.
Convention Paper 9263 (Purchase now)

P7-4 Scalable Parametric Audio Coder Using Sparse Approximation with Frame-to-Frame Perceptually Optimized Wavelet Packet Based Dictionary—Alexey Petrovsky, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Vadzim Herasimovich, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Alexander Petrovsky, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus
This paper is devoted to the development of a scalable parametric audio coder based on a matching pursuit algorithm with a frame-based psychoacoustic optimized wavelet packet dictionary. The main idea is to parameterize audio signal with a minimum number of non-negative elements. This can be done by applying sparse approximation such as matching pursuit algorithm. In contrast with current approaches in audio coding based on sparse approximation we introduce a model of dynamic dictionary forming for each frame of input audio signal individually based on wavelet packet decomposition and dynamic wavelet packet tree transformation with psychoacoustic model. Experimental results of developed encoder and comparison with modern popular audio encoders are provided.
Convention Paper 9264 (Purchase now)

P7-5 General-Purpose Listening Enhancement Based on Subband Non-Linear Amplification with Psychoacoustic Criterion—Elias Azarov, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Maxim Vashkevich, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Vadzim Herasimovich, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Alexander Petrovsky, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus
Near end listening enhancement is an effective approach for speech intelligibility improvement in noisy conditions that is applied mainly for telecommunications. However potential application field of the concept of near end listening enhancement is much wider and can be extended for listening of any audio content (including music and other sounds) in quiet and noisy conditions. This paper proposes an algorithm for near end listening enhancement designed for processing both speech and music that, considering subjective listening tests, significantly improves the listening experience. The algorithm is based on subband non-linear amplification of the audio signal in accordance with noise spectral characteristics and personal hearing thresholds of the listener. The algorithm is experimentally implemented as an application for smartphones.
Convention Paper 9265 (Purchase now)

P7-6 Poster Moved to Session P16—N/A


P7-7 Speech Analysis Based on Sinusoidal Model with Time-Varying Parameters—Elias Azarov, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Maxim Vashkevich, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Alexander Petrovsky, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus
Extracting speech-specific characteristics from a signal such as spectral envelope and pitch is essential for parametrical speech processing. These characteristics are used in many speech applications including coding, parametrical text-to-speech synthesis, voice morphing, and others. This paper presents some original estimation techniques that extract these characteristics using a sinusoidal model of speech with instantaneous parameters. The analysis scheme consists of two steps: first the parameters of sinusoidal model are extracted from the signal, and then these parameters are transformed to required characteristics. Some evaluations of the presented techniques are carried out on synthetic and natural speech signals to show potential of the presented approach.
Convention Paper 9267 (Purchase now)

P7-8 A Low-Delay Algorithm for Instantaneous Pitch Estimation—Elias Azarov, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Maxim Vashkevich, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; D. Likhachov, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus; Alexander Petrovsky, Belarusian State University of Informatics and Radioelectronics - Minsk, Belarus
Estimation of instantaneous pitch provides high accuracy for frequency-modulated pitches and can be beneficial compared to conventional pitch extraction techniques for unsteady voiced sounds. However, applying an estimator of instantaneous pitch to a practical real-time speech processing application is a hard problem because of high computational cost and high inherent delay. The paper presents an algorithm for instantaneous pitch estimation specifically designed for real-time applications. The analysis scheme is based on the robust algorithm for instantaneous pitch tracking (IRAPT) featuring an efficient processing scheme and low inherent delay. The paper presents some evaluation results using synthesized and natural speech signals that illustrate actual performance of the algorithm.
Convention Paper 9268 (Purchase now)

P7-9 Content-Based Music Structure Analysis Using Vector Quantization—Nikolaos Tsipas, Aristotle University of Thessaloniki - Thessaloniki, Greece; Lazaros Vrysis, Aristotle University of Thessaloniki - Thessaloniki, Greece; Playcompass Entertainment; Charalampos A. Dimoulas, Aristotle University of Thessaloniki - Thessaloniki, Greece; George Papanikolaou, Aristotle University of Thessaloniki - Thessaloniki, Greece
Music structure analysis has been one of the challenging problems in the field of music information retrieval during the last decade. Past years advances in the field have contributed toward the establishment and standardization of a framework covering repetition, homogeneity, and novelty based approaches. With this paper an optimized fusion algorithm for transition points detection in musical pieces is proposed, as an extension to existing state-of-the-art techniques. Vector-Quantization is introduced as an adaptive filtering mechanism for time-lag matrices while a structure-features based self-similarity matrix is proposed for novelty detection. The method is evaluated against 124 pop songs from the INRIA Eurovision dataset and performance results are presented in comparison with existing state-of-the-art implementations for music structure analysis.
Convention Paper 9269 (Purchase now)

P7-10 Clock Skew Compensation by Adaptive Resampling for Audio Networking—Leonardo Gabrielli, Universitá Politecnica delle Marche - Ancona, Italy; Michele Bussolotto, Universitá Politecnica della Marche - Ancona, Italy; Stefano Squartini, Università Politecnica delle Marche - Ancona, Italy; Fons Adriaensen, Huawei European Research Center - Munich, Germany
Wired Audio Networking is an established practice since years based on both proprietary solutions or open hardware and protocols. One of the most cost-effective solutions is the use of a general purpose IEEE 802.3 infrastructure and personal computers running IP based protocols. One obvious shortcoming of such setups is the lack of synchronization at the audio level and the presence of a network delay affected by jitter. Two approaches to sustain a continuous audio flow are described, implemented by the authors in open source projects based on a relative and absolute time adaptive resampling. A description of the mechanisms is provided along with simulated and measured results, which show the validity of both approaches.
Convention Paper 9270 (Purchase now)

P7-11 Analysis of Onset Detection with a Maximum Filter in Recordings of Bowed Instruments—Bartlomiej Stasiak, Lodz University of Technology - Lodz, Poland; Jedrzej Monko, Lódz University of Technology - Lódz, Poland
This work presents a new approach to assessment of the quality of onset detection functions on the example of bowed instruments recordings. Using this method, we test a vibrato suppression technique based on a maximum filter. The results, obtained with the aid of a specially constructed database of audio recordings, reveal problems connected with certain qualities of the sound signal generated by a bowed instrument and with the effectiveness of the onset detection process.
Convention Paper 9271 (Purchase now)

P7-12 An FPGA-Based Virtual Reality Audio System—Wolfgang Fohl, Hamburg University of Applied Sciences - Hamburg, Germany; David Hemmer, Hamburg University of Applied Sciences - Hamburg, Germany
A distributed system for mobile virtual reality audio is presented. The system consists of an audio server running on a PC or Mac, a remote control app for an iOS6 device, and the mobile renderer running on a system-on-chip (SoC) with a CPU core and signal processing hardware. The server communicates with the renderer via WLAN. It sends audio streams over a self-defined lightweight protocol and exchanges status and control data as OSC (Open Sound Control) messages. On the mobile renderer, HRTF filters are applied to each audio signal according to the relative positions of the source and the listener’s head. The complete audio signal processing chain has been designed in Simulink. The VHDL code for the SoC’s FPGA hardware has been automatically generated by Xilinx’s System Generator. The system is capable of rendering up to eight independent virtual sources.
Convention Paper 9328 (Purchase now)

 
 

P8 - (Lecture) Transducers—Part 1

Friday, May 8, 14:00 — 16:00 (Room: Królewski)

Chair:
Finn T. Agerkvist, Technical University of Denmark - Kgs. Lyngby, Denmark

P8-1 Assessing Influence of a Headphone Type on Individualized Ear Training—Sungyoung Kim, Rochester Institute of Technology - Rochester, NY, USA; Sean Olive, Harman International - Northridge, CA, USA
Technical ear training has been provided for two groups of engineering students. The treatment group received and conducted the training using a professional-level headphone and the control group did same training with their own consumer-level earphone or headphone. To investigate a possible influence of a headphone type, both groups took two standardized matching tests before and after 15-week technical ear training. The comparison of two test results shows that the headphone type significantly differentiated trainees' matching performance of the treatment group from the control group.
Convention Paper 9272 (Purchase now)

P8-2 GaN Power Stage for Switch Mode Audio Amplification—Rasmus Overgaard Ploug, Technical University of Denmark - Kongens Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Søren Bang Poulsen, Texas Instruments Denmark A/S - Kongens Lyngby, Denmark
Gallium Nitride (GaN) based power transistors are gaining more and more attention since the introduction of the enhancement mode eGaN Field Effect Transistor (FET), which makes an adaptation from Metal-Oxide Semiconductor (MOSFET) to eGaN based technology less complex than by using depletion mode GaN FETs. This project seeks to investigate the possibilities of using eGaN FETs as the power switching device in a full bridge power stage intended for switch mode audio amplification. A 50 W 1 MHz power stage was built and provided promising audio performance. Future work includes optimization of dead time and investigation of switching frequency versus audio performance.
Convention Paper 9273 (Purchase now)

P8-3 Characterizing the Frequency Response of Headphones—A New Paradigm—Ulrich Horbach, Harman Advanced Technology Group - Northridge, CA, USA
Traditional headphone measurements suffer from large variations if carried out on human subjects with probe microphones, and standardized couplers introduce additional biases, as concluded in a recent paper. Beyond that, there is no clear indication in literature about what the actual perceived frequency response of a headphone might be. This paper explores new measurement methods that avoid the human body as much as possible by measuring the headphone directly, in an attempt to overcome these restrictions and gain more accuracy. Design principles are described in the second part. A novel, DSP controlled, high-quality headphone is introduced that offers the ability to auto-calibrate its frequency response to the individual who is wearing it.
Convention Paper 9274 (Purchase now)

P8-4 Improved Measurement of Leakage Effects for Circum-Aural and Supra-Aural Headphones—Todd Welti, Harman International Inc. - Northridge, CA, USA
Headphone leakage effects can have a profound effect on low frequency performance of headphones. A large survey, including over 2000 individual headphone measurements, was undertaken in order to compare leakage effects on test subjects and leakage effects of the same headphones measured on a test fixture. Ten different commercially available headphones were used, each measured on eight different test subjects and a test fixture with several sets of pinnae. Modifications to the pinnae were investigated to see if the leakage effects measured on the test fixture could be made to better match the real word leakage effects measured on human test subjects.
Convention Paper 9275 (Purchase now)

 
 

P9 - (Lecture) Perception—Part 1

Friday, May 8, 14:30 — 18:00 (Room: Belweder)

Chair:
Jürgen Herre, International Audio Laboratories Erlangen - Erlangen, Germany; Fraunhofer IIS - Erlangen, Germany

P9-1 Exposure of Music Students to Sound in Large Music Ensembles—Maciej Jasinski, Warsaw University of Technology - Warsaw, Poland; Agnieszka Pietrzak, Warsaw University of Technology - Warsaw, Poland; Jun Ho Shin, Warsaw University of Technology - Warsaw, Poland; Kyungpook National University - Daegu, Korea; Jan Zera, Warsaw Institute of Technology - Warsaw, Poland
Exposure of musicians to sounds on stage has been a topic of numerous studies over the past 50 years. Nevertheless, the problem is still being researched as inconsistent conclusions have been obtained as to the risk of hearing loss among musicians. In this study exposure of music students to sound was measured during their activity as members of large ensembles: a student symphony orchestra, a wind orchestra, and a big-band. The measurements showed that critical conditions exceeding the permissible daily sound exposure level of 85 dB (A) occurred in the case of musicians playing brass, woodwinds, and percussion instruments with a high exposure of the neighboring groups of musicians directly exposed to the sound thereof.
Convention Paper 9276 (Purchase now)

P9-2 Effects of Psychoacoustical Factors on the Perception of Musical Signals in the Context of Environmental Soundscape—Zhiyong Deng, Capital Normal University - Beijing, China; University of Sheffield - Sheffield, UK; Jian Kang, University of Sheffield - Sheffield, South Yorkshire, UK; Aili Liu, Capital Normal University - Beijing, China
Increasing attention is paid to the benefits of music or music-like signals in soundscape and the soundscape design projects. The perception and awareness of musical signals in the context of environmental soundscape has been suggested with a number of psychoacoustical factors involved. In this paper sound pressure level, noisiness, listeners’ hearing training background, and interaction of the sound sources have been found to influence the perception of consonance, extraction, and awareness of the musical signals in the context of environmental soundscape. This paper also gives a brief discussion on the theoretic definition of music perception and consonance or pleasance.
Convention Paper 9277 (Purchase now)

P9-3 Directional Bands Revisited—Rory Wallis, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK
Listening tests were undertaken as part of a comprehensive analysis of directional bands. The effects of frequency, loudspeaker position, signal duration, and bandwidth were all considered. The results confirmed the existence of directional bands for 1, 4, and 8 kHz 1/3-octave band bursts. A relationship between pitch and height was also observed, leading to the suggestion that the pitch-height effect and directional bands are part of the same localizational mechanism. Bandwidth was found to have a variable effect on localization, depending on frequency, indicating that the spectral cues used in vertical localization are not of equal bandwidth. Loudspeaker position and signal duration also had some influence on localization judgments although this was found to be somewhat erratic.
Convention Paper 9278 (Purchase now)

P9-4 The Effect of Dynamic Range Compression on Perceived Loudness for Octave Bands of Pink Noise in Relation to Crest Factor—Mark Wendl, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK
A listening test was performed to find the changes in perceived loudness for differing crest factors of octave bands of pink noise as a result of limiting. Each octave band had a continuous and a transient sample of which both had five samples ranging from an uncompressed to a compressed with a difference of 4 dB FS crest factor calculation with increments of 1 dB FS. Two playback levels of 50 dB SPL and 70 dB SPL were used. The perceived loudness followed the RMS change within the pink noise; however certain octave bands appeared to have a non-linear relationship between loudness perception and crest factor changes.
Convention Paper 9279 (Purchase now)

P9-5 Interaction of Perceived Distance and Depth Comparing Audio Playback System and Musical Context—Toru Kamekawa, Tokyo University of the Arts - Adachi-ku, Tokyo, Japan; Atsushi Marui, Tokyo University of the Arts - Tokyo, Japan
The effects of the audio playback system and musical context were studied focusing on the perceived distance and the spatial depth. The experiments were carried out using a method of magnitude estimation comparing how near or far the combination of perceived visual and auditory event is, between two performers where one is fixed and another is moved back and forward. In the first experiment, the participants compared the difference among seven distances. In the second experiment, the participants compared direct-to-reverberant ratio (DR ratio) and differences of sound pressure level (SPL) among several musical context such as melody and accompaniment, precedence and chase of almost identical phrases, and non musical stimulus (pulse sound). The results showed that the perceived distance and depth were affected by the existence of image, DR ratio, and SPL. Furthermore these effects are different from musical context and playback system such as the existence of center, rear, and height loudspeakers.
Convention Paper 9280 (Purchase now)

P9-6 Auditory Adaptation in Spatial Listening Tasks—Florian Klein, Technische Universität Ilmenau - Ilmenau, Germany; Stephan Werner, Technische Universität Ilmenau - Ilmenau, Germany
This paper investigates auditory adaptation processes in spatial listening tasks for normal hearing people. The auditory adaptation process to altered auditory cues of thirteen participants is monitored and compared to their normal hearing listening performance. Binaural room impulse responses are measured for each participant and for an artificial head. Listeners are trained to artificial binaural room impulse responses in an audio-visual training task. Nine out of thirteen listeners could increase their elevation perception significantly and two of these listeners performed better with trained artificial binaural room impulse responses than with their individual measured room impulse responses regarding elevation error in the median plane. The listening test is supported by an interview that asks for externality.
Convention Paper 9281 (Purchase now)

P9-7 Discrimination of Formant Amplitude in Noise—Tomira Rogala, Fryderyk Chopin University of Music - Warsaw, Poland; Piotr Sliwka, Stefan Cardinal Wyszynski University - Warsaw, Poland
The paper reports the results of an experiment carried out to determine the just noticeable difference in timbre of noise. The variations of timbre were obtained through modification of the spectrum envelope of a pink noise—the formant amplitude was increased. The listeners were asked to indicate which one of three noise bursts in a trial sounded different than the remaining two. The results of listeners without musical experience were only a little worse than those obtained from tonmeister students. The skill of detecting slight changes in noise is easy to train and the jnd for formant amplitude change is very low.
Convention Paper 9282 (Purchase now)

 
 

P10 - (Poster) Transducers

Friday, May 8, 16:00 — 18:00 (Foyer)

P10-1 Loudspeaker Systems by Linear Motion Type Piezoelectric Ultrasonic Actuators—Daichi Nagaoka, Tokyo University of Technology - Hachioji-shi, Tokyo, Japan; Juro Ohga, Shibauro Institute of Technology / Mix Corporation - Kanagawa, Japan; Hirokazu Negishi, Mix Corporation - Kanagawa, Japan; Ikuo Oohira, Consultant - Kanagawa-ken, Japan; Kazuaki Maeda, TOA Corporation - Hyogo, Japan; Kunio Oishi, Tokyo University of Technology - Tokyo, Japan
The research group of authors have been developing completely new loudspeaker constructions that are driven by piezoelectric ultrasonic motors. This paper proposes two sorts of applications of piezoelectric linear actuators to both direct radiator and horn loudspeakers. A direct-radiator loudspeaker with a cone radiator driver by piezoelectric actuators shows smooth frequency characteristics in low frequency region because its radiating motion includes no significant resonance in the working frequency region. Therefore, it is useful for radiation of the lowest frequency part of audio signal. A horn loudspeaker by the same actuators works in rather moderate frequency region that is higher than the cut-off frequency of hones of ordinary size.
Convention Paper 9283 (Purchase now)

P10-2 Low Frequency Nonlinear Model for Loudspeaker Transducers—Shaolin Wei, Guo Guang Electric Corporation (GGEC) - Guangzhou, China; Tony Xie, Sr., Guo Guang Electric Corporation (GGEC) - Guangzhou, China; Hunter Huang, Sr., Guo Guang Electric Corporation (GGEC) - Guangzhou, China
In this paper a nonlinear loudspeaker transducer model and its solution are presented. A simple and effective iteration procedure to obtain the solution of the nonlinear equation is proposed. This procedure is a powerful tool for determination of a periodic solution of a non-linear equation of motion. Further, the sound pressure of fundamental, second order, and third order harmonic distortion are also calculated. The solutions obtained using the present iteration method can give the directions to how to lower the second and third harmonics. Due to unavoidable circumstances this poster will not be presented.
Convention Paper 9284 (Purchase now)

P10-3 Active Control of a String Instrument Bridge Using the Posicast Technique—Liam B. Donovan, Queen Mary University of London - London, UK; Andrew McPherson, Queen Mary University of London - London, UK
This paper presents an active bridge allowing for precise audio-rate manipulation of a string’s termination for the purposes of modifying string instrument timbre. The design of the bridge actuator and height sensor is discussed, and the benefits of using feedforward posicast control over a feedback compensator for controlling the dynamics of the severely underdamped bridge actuator system are established.
Convention Paper 9285 (Purchase now)

P10-4 Efficiency Optimization in Class-D Audio Amplifiers—Akira Yamauchi, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Ivan H. H. Jørgensen, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
This paper presents a new power efficiency optimization routine for designing Class-D audio amplifiers. The proposed optimization procedure finds design parameters for the power stage and the output filter, and the optimum switching frequency such that the weighted power losses are minimized under the given constraints. The optimization routine is applied to minimize the power losses in a 130 W class-D audio amplifier based on consumer behavior investigations, where the amplifier operates at idle and low power levels most of the time. Experimental results demonstrate that the optimization method can lead to around 30% of efficiency improvement at 1.3 W output power without significant effects both on the audio performances and on the efficiency at high power levels.
Convention Paper 9286 (Purchase now)

P10-5 Investigation of Energy Consumption and Sound Quality for Class-D Audio Amplifiers Using Tracking Power Supplies—Akira Yamauchi, Technical University of Denmark - Kgs. Lyngby, Denmark; Henrik Schneider, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Ivan H. H. Jørgensen, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
The main advantage of Class-D audio amplifiers is high efficiency that is often stated to be more than 90% but, at idle or low power levels the efficiency is much lower. The waste energy is an environmental concern, a concern in mobile applications where long battery operation is required and a concern in other applications where multiple amplifier channels are generating heat problems. It is found that power losses at low power levels account for close to 78% of energy consumption based on typical consumer behavior investigations. This paper investigates the theoretical limits of ideal stepless power supply tracking and its influence on power losses, audio performance, and environmental impact for a 130 W class-D amplifier. Both modeled and experimental results verify that a large improvement of efficiency can be achieved with a new challenge for a self-oscillating controller to keep the audio quality in such a system. The energy consumption may be reduced by up to 72%. The investigation is extended to a commercialized class-D amplifier as well.
Convention Paper 9287 (Purchase now)

P10-6 Soundbar System with Embedded Multichannel Digital Amplifier SoC—Jeongil Seo, Electronics & Telecom. Research Institute (ETRI) - Daejeon, Korea; Jae-Hyoun Yoo, Electronics and Telecommunications Research Institute (ETRI) - Daejeon, Korea; Taejin Park, Electronics and Telecommunications Research Institute (ETRI) - Daejeon, Korea; Taejin Lee, Electronics and Telecommunications Research Institute (ETRI) - Daejeon, Korea; Myunggeun Yoo, NeoFidelity - Seongnam-si, Gyeonggi-do, Korea; Geunho Jang, NeoFidelity - Seongnam-si, Gyeonggi-do, Korea; Jae-Hee Won, NeoFidelity - Seongnam-si, Gyeonggi-do, Korea; Yeongha Choi, NeoFidelity, Inc. - Seongnamsi, Kyunggido, Korea
This paper presents collaboration approach results between an audio signal processing algorithm and a digital amplifier structure for efficient implementation of soundbar applications. If we want to provide a virtual surround sound image with a linear loudspeaker array, high performance DSP and multichannel digital amplifiers are required. The required performance for DSP is depended on the algorithm complexity for creating a virtual surround sound image. However the final computation before generating the output loudspeaker signals (e.g., 16 channels) is generally composed of simple delay and sum computations from an input audio signal (e.g., 5.1 channels) to output multichannel loudspeaker signals. Therefore we redesigned the audio signal processing software by two-block processing, and the second block for simple delay and sum computation is implemented at the multichannel digital amplifier, which has an independent DSP core. Through computational simulation and hardware implementation, the proposed system showed equivalent performance with the conventional one block processing system.
Convention Paper 9288 (Purchase now)

P10-7 Loudspeaker Impedance Emulator for Multi Resonant Systems—Niels Elkjær Iversen, Technical University of Denmark - Kongens Lyngby, Denmark; Danmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark
Specifying the performance of audio amplifiers is typically done by playing sine waves into a pure ohmic load. However real loudspeaker impedances are not purely ohmic but characterized by its electrical, mechanical, and acoustical properties. Therefore a loudspeaker emulator capable of adjusting its impedance to that of a given loudspeaker is desired for measurement purposes. An adjustable RLC-based emulator is implemented with switch controlled capacitors, air gap controlled inductors, and potentiometers. Calculations and experimental results are compared and show that it is possible to emulate the loudspeaker impedance infinite baffle-, closed box-, and the multi resonant vented box-loudspeaker by tuning the component values in the proposed circuit. Future work is outlined and encouraged that the proposed impedance emulator is used as part of a control circuit in a switch-mode based impedance emulator.
Convention Paper 9289 (Purchase now)

P10-8 How "Green" Is My Amp?—Jamie Angus, University of Salford - Salford, Greater Manchester, UK
This paper examines the potential threat of power restriction legislation on audio power amplifier design. By considering the interaction between the amplitude distribution of audio signals with the efficiency characteristics of the different amplifier classes it shows that some of the linear classes can perform well as regards energy consumption and thus can compete with switching class D systems. Furthermore, it discusses the possibility of optimizing some of the linear amplifier classes in conjunction with the amplitude probabilities of real audio signals to effect a further reduction in average power consumption. Thus resulting in the "greenest possible" amplifier for a given class of power amplification.
Convention Paper 9290 (Purchase now)

P10-9 An Investigation into Utilizing Opto-Sensors to Function as Parts of MIDI Controllers—Richard Corke, Southampton Solent University - Southampton, UK; Andrew J. Horsburgh, Southampton Solent University - Southampton, UK
The presented research focuses on the application of MIDI controllers utilizing opto-sensors to read and translate physical contact into controllable MIDI information. The described technology was found to provide improved interaction and degree of movement translation with other MIDI-capable devices. To demonstrate this, a projection controller using existing infrared technology will be used in conjunction with a microcontroller allowing for communication between analog and digital control signals.
Convention Paper 9291 (Purchase now)

P10-10 Investigation of Current Driven Loudspeakers—Henrik Schneider, Technical University of Denmark - Kgs. Lyngby, Denmark; Finn T. Agerkvist, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
Current driven loudspeakers have previously been investigated but the literature is limited and the advantages and disadvantages are yet to be fully identified. This paper makes use of a non-linear loudspeaker model to analyze loudspeakers with distinct non-linear characteristics under voltage and current drive. A multi tone test signal is used in the evaluation of the driving schemes since it resembles audio signals to a higher degree than the signals used in total harmonic distortion and intermodulation distortion test methods. It is found that current drive is superior over voltage drive in a 5" woofer where a copper ring in the pole piece has not been implemented to compensate for eddy currents. However the drive method seems to be irrelevant for a 5" woofer where the compliance, force factor as well as the voice coil inductance has been optimized for linearity.
Convention Paper 9292 (Purchase now)

P10-11 Design and Evaluation of Accelerometer-Based Motional Feedback—Henrik Schneider, Technical University of Denmark - Kgs. Lyngby, Denmark; Emilio Pranjic, Technical University of Denmark - Kongens Lyngby, Denmark; Finn T. Agerkvist, Technical University of Denmark - Kgs. Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
The electro dynamic loudspeaker is often referred to as the weakest link in the audio chain due to low efficiency and high distortion levels at low frequencies and high diaphragm excursion. Compensating for loudspeaker non-linearities using feedback or feedforward methods can improve the distortion and enable radical design changes in the loudspeaker that can lead to efficiency improvements. In combination this has motivated a revisit of the accelerometer-based motional feedback technique. Experimental results on an 8-inch subwoofer show that the total harmonic distortion can be significantly reduced at low frequencies and large displacements.
Convention Paper 9293 (Purchase now)

 
 

P11 - (Lecture) Sound Localization and Separation

Saturday, May 9, 09:00 — 12:00 (Room: Belweder)

Chair:
Christof Faller, Illusonic GmbH - Zurich, Switzerland; EPFL - Lausanne, Switzerland

P11-1 Classification of Spatial Audio Location and Content Using Convolutional Neural Networks—Toni Hirvonen, Dolby Laboratories - Stockholm, Sweden
This paper investigates the use of Convolutional Neural Networks for spatial audio classification. In contrast to traditional methods that use hand-engineered features and algorithms, we show that a Convolutional Network in combination with generic preprocessing can give good results and allows for specialization to challenging conditions. The method can adapt to e.g. different source distances and microphone arrays, as well as estimate both spatial location and audio content type jointly. For example, with typical single-source material in a simulated reverberant room, we can achieve cross-validation accuracy of 94.3% for 40-ms frames across 16 classes (eight spatial directions, content type speech vs. music).
Convention Paper 9294 (Purchase now)

P11-2 A Theoretical Analysis of Sound Localization, with Application to Amplitude Panning—Dylan Menzies, University of Southampton - Southampton, UK; Filippo Maria Fazi, University of Southampton - Southampton, Hampshire, UK
Below 700 Hz sound fields can be approximated well over a region of space that encloses the human head, using the acoustic pressure and gradient. With this representation convenient expressions are found for the resulting Interaural Time Difference (ITD) and Interaural Level Difference (ILD). This formulation facilitates the investigation of various head-related phenomena of natural and synthesized fields. As an example, perceived image direction is related to head direction and the sound field description. This result is then applied to a general amplitude panning system and can be used to create images that are stable with respect to head direction.
Convention Paper 9295 (Purchase now)

P11-3 Audio Object Separation Using Microphone Array Beamforming—Philip Coleman, University of Surrey - Guildford, Surrey, UK; Philip Jackson, University of Surrey - Guildford, Surrey, UK; Jon Francombe, University of Surrey - Guildford, Surrey, UK
Audio production is moving toward an object-based approach, where content is represented as audio together with metadata that describe the sound scene. From current object definitions, it would usually be expected that the audio portion of the object is free from interfering sources. This poses a potential problem for object-based capture, if microphones cannot be placed close to a source. This paper investigates the application of microphone array beamforming to separate a mixture into distinct audio objects. Real mixtures recorded by a 48-channel microphone array in reflective rooms were separated, and the results were evaluated using perceptual models in addition to physical measures based on the beam pattern. The effect of interfering objects was reduced by applying the beamforming techniques.
Convention Paper 9296 (Purchase now)

P11-4 Limits of Speech Source Localization in Acoustic Wireless Sensor Networks—David Ayllón, University of Alcalá - Alcalá de Henares, Spain; Roberto Gil-Pita, University of Alcalá - Alcalá de Henares, Madrid, Spain; Manuel Rosa-Zurera, University of Alcalá - Alcalá de Henares, Madrid, Spain; Guillermo Ramos-Auñón, University of Alcalá - Alcalá de Henares, Madrid, Spain
Acoustic Wireless Sensor Networks (AWSN) have become very popular in the last years due to the drastic increment in the number of wireless nodes with microphones and computational capability. In such networks accurate knowledge of sensor node locations is often not available, but this information is crucial to process the collected data by means of array processing techniques. In this paper we consider the error in the estimation of the position of the nodes as a traditional microphone mismatch with large values, and we perform a detailed study of the effect that a large microphone mismatch has on the accuracy of TDOA- based source localization techniques.
Convention Paper 9297 (Purchase now)

P11-5 Improving Speech Mixture Synchronization in Blind Source Separation Problems—Cosme Llerena-Aguilar, Sr., University of Alcalá - Alcala de Henares (Madrid), Spain; Guillermo Ramos-Auñón, University of Alcalá - Alcalá de Henares, Madrid, Spain; Francisco J. Llerena-Aguilar, University of Alcalá - Alcalá de Henares, Madrid, Spain; Héctor A. Sánchez-Hevia, University of Alcala - Alcalá de Henares, Madrid, Spain; Manuel Rosa-Zurera, University of Alcalá - Alcalá de Henares, Madrid, Spain
The use of wireless acoustic sensor networks carry many advantages in the speech separation framework. Since nodes are separated by greater distances than a few centimeters, they can cover rooms completely, although these new distances involve certain problems to be solved. For instance, important time differences of arrival between the speech mixtures captured at the different microphones can appear, affecting the performance of classical sound separation algorithms. One solution consists in synchronizing the speech mixtures captured at the microphones. Following with this idea, we put forward in this paper a new time delay estimation method that outperforms classical methods in order to synchronize speech mixtures. The results obtained show the feasibility of using our proposal aiming at synchronizing speech mixtures.
Convention Paper 9298 (Purchase now)

P11-6 Direction of Arrival Estimation of Multiple Sound Sources Based on Frequency-Domain Minimum Variance Distortionless Response Beamforming—Seung Woo Yu, Gwangju Institute of Science and Technology - Gwangju, Korea; Kwang Myung Jeon, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Dong Yun Lee, Gwangju Institute of Science and Technology (GIST) - Gwangju, Korea; Hong Kook Kim, Gwangju Institute of Science and Tech (GIST) - Gwangju, Korea; City University of New York - New York, NY, USA
In this paper a method for estimating the direction-of-arrivals (DOAs) of multiple non-stationary sound sources is proposed on the basis of a frequency-domain minimum variance distortionless response (FD-MVDR) beamformer. First, an FD-MVDR beamformer is applied to multiple sound sources, where the beamformer weights are updated according to the surrounding environments for the reduction of the sidelobe effect of the beamformer. Then, multistage DOA estimation is performed to reduce computational complexity regarding the beam search. Finally, a median filter is applied to improve the DOA estimation accuracy. It is demonstrated that the average DOA estimation error of the proposed method is smaller than those of the methods based on conventional GCC-PHAT, MVDR-PHAT, and FD-MVDR, with lower computational complexity than that of the conventional FD-MVDR-based DOA estimation method.
Convention Paper 9299 (Purchase now)

 
 

P12 - (Lecture) Applications in Audio

Saturday, May 9, 09:00 — 12:30 (Room: Królewski)

Chair:
Wieslaw Woszczyk, McGill University - Montreal, QC, Canada

P12-1 Reconstruction of Mechanically Recorded Audio Signals Using White-Light Interferometry—Khac Phuc Hung Thai, INSA Centre Val de Loire - Blois, France; Université de Sherbrooke - Sherbrooke, QC, Canada; Philippe Gournay, Université de Sherbrooke - Sherbrooke, QC, Canada; Voiceage - Montreal, Quebec, Canada; Roch Lefebvre, Universite de Sherbrooke - Sherbrooke, QC, Canada; Serge Charlebois, Université de Sherbrooke - Sherbrooke, QC, Canada
This paper presents a method to reconstruct a digital audio signal from a physical and analog sound-recording medium such as the Edison cylinder. A non-contact 3D optical profilometer based on white-light interferometry provides topographic information about small overlapping sections of the recording medium. For each of these sections, the natural curvature of the medium is compensated, grooves on the surface are detected, and short depth trajectories representing the audio signal are extracted. These trajectories are then concatenated to produce a digital audio signal that is post-processed to simulate the distinctive frequency response of an actual mechanical player. The effectiveness of this method is demonstrated on both tonal and vocal recordings using time, frequency, and time-frequency features as well as informal listening.
Convention Paper 9300 (Purchase now)

P12-2 Recognition of Hazardous Acoustic Events Employing Parallel Processing on a Supercomputing Cluster—Kuba Lopatka, Gdansk University of Technology - Gdansk, Poland; Andrzej Czyzewski, Gdansk University of Technology - Gdansk, Poland
A method for automatic recognition of hazardous acoustic events operating on a super computing cluster is introduced. The methods employed for detecting and classifying the acoustic events are outlined. The evaluation of the recognition engine is provided: both on the training set and using real-life signals. The algorithms yield sufficient performance in practical conditions to be employed in security surveillance systems. The specialized framework for parallel processing of multimedia data streams KASKADA, in which the methods are implemented, is briefly introduced. An experiment intended to assess outcomes of parallel processing of audio data on a supercomputing cluster is featured. It is shown that by employing supercomputing services the time needed to analyze the data is greatly reduced.
Convention Paper 9301 (Purchase now)

P12-3 Perceptual Evaluation of an Audio Film for Visually Impaired Audiences—Mariana Lopez, Anglia Ruskin University - Cambridge, UK
This paper explores a format of sonic art referred to as audio film that was developed to study different ways in which film sound production and postproduction techniques could be applied to the enhancement of Audio Description (AD) for visually impaired film and television audiences. A prototype of this format was tested with a group of nine volunteers with sight loss in order to test its effectiveness. The perceptual evaluation demonstrated the potential of this format for conveying a clear narrative as well as providing an entertaining experience. Future work will include the investigation of conventions to indicate scene changes within audio-only formats as well as studying the impact of object-based mixing on audio films.
Convention Paper 9302 (Purchase now)

P12-4 Reproduction of Realistic Background Noise for Testing Telecommunications Devices—Juan David Gil Corrales, Technical University of Denmark - Lyngby, Denmark; Wookeun Song, Brüel & Kjær Sound and Vibration Measurement A/S - Nærum, Denmark; Ewen MacDonald, Technical University of Denmark - Lyngby, Denmark
A method for reproduction of sound, based on crosstalk cancellation using inverse filters, was implemented in the context of testing telecommunications devices. The effect of the regularization parameter, number of loudspeakers, type of background noise, and a technique to attenuate audible artifacts, were investigated. The quality of the reproduced sound was evaluated both objectively and subjectively with respect to the reference sounds, at points where telecommunications devices would be potentially placed around the head. The highest regularization value gave the best results, the performance was equally good when using eight or four loudspeakers, and the reproduction method was shown to be robust for different program materials. The proposed technique to reduce audible artifacts increased the perceived similarity.
Convention Paper 9303 (Purchase now)

P12-5 Simulation of Parameters of Tube Audio Circuits Using Web Browsers—Grzegorz Makarewicz, Warsaw University of Technology - Warsaw, Poland
The paper describes the program/simulator for computer-aided design of audio amplifiers using electron tubes. It was developed in JavaScript scripting language and thanks to its embedding in a web browser it does not require installation on the user’s computer. The simulator can be used for the design and education purposes, without any limitations, by multiple users simultaneously. It is based on mathematical models of the triode and pentode and allows parameters of electron tubes as well as the most important parameters of tube amplifiers to be simulated for both unbalanced (single-ended) and balanced (push-pull) configuration.
Convention Paper 9304 (Purchase now)

P12-6 Improvements and User Preferences in Auralization for Multi-Party Teleconferencing Systems Using Binaural Audio—Emanuel Aguilera, Universidad Politecnica de Valencia - Valencia, Spain; Jose J. Lopez, Universidad Politecnica de Valencia - Valencia, Spain; Pablo Gutierrez-Parera, Universitat de Valencia - Valencia, Spain
The introduction of spatial audio in multi-party teleconferencing systems create realistic communication environments with increased immersion compared to monaural systems. Moreover, the introduction of auralization effects can increase even more the immersion but at the expenses of a reduced intelligibility. In this paper we study the influence of some specific auralization processing details for a trade-off between realism and intelligibility. Our own spatial multi-party teleconferencing software running on smartphones and tablets has been developed for carrying out different subjective experiments. By means of subjective testing with a jury, it has been evaluated the influence in immersion, intelligibility, and user preferences in relation with early echoes, late reverberation, and the introduction of simple near-field HRTF processing when audio sources are very close to the user. Results provide interesting guidelines for developing teleconference systems with more subtle auralization and HRTF effects.
Convention Paper 9305 (Purchase now)

P12-7 Subjective Assessment of Commercial Sound Enhancement System—Krzysztof Brawata, AGH University of Science and Technology - Krakow, Poland; Pawel Malecki, AGH University of Science and Technology - Krakow, Poland; Adam Pilch, AGH University of Science and Technology - Krakow, Poland; Tadeusz Kamisinski, AGH University of Science and Technology - Krakow, Poland
Sound enhancement systems are becoming more and more popular even in very sophisticated concert halls. Especially in places with some acoustics deficiencies, musicians and concert-goers have accepted that kind of solution as very natural sounding and of great possibilities to easily obtain variable acoustics in rooms. On the market, there are some sound enhancement commercial systems, with similar acoustical parameters. Choosing the best one, for defined application, is possible only on the basis of properly designed listening tests. In the paper subjective listening tests of two sound enhancement systems installed in the same room are presented. On the basis of listeners’ evaluation, the quality of stage acoustics, naturalness, and spaciousness of sound created by systems were analyzed.
Convention Paper 9306 (Purchase now)

 
 

P13 - (Lecture) Perception—Part 2

Saturday, May 9, 14:30 — 18:00 (Room: Belweder)

Chair:
Hyunkook Lee, University of Huddersfield - Huddersfield, UK

P13-1 Elicitation of the Differences between Real and Reproduced Audio—Jon Francombe, University of Surrey - Guildford, Surrey, UK; Tim Brookes, University of Surrey - Guildford, Surrey, UK; Russell Mason, University of Surrey - Guildford, Surrey, UK
To improve the experience of listening to reproduced audio, it is beneficial to determine the differences between listening to a live performance and a recording. An experiment was performed in which three live performances (a jazz duet, a jazz-rock quintet, and a brass quintet) were captured and simultaneously replayed over a nine-channel with-height surround sound system. Experienced and inexperienced listeners moved freely between the live performance and the reproduction and described the difference in listening experience. In subsequent group discussions, the experienced listeners produced twenty-nine categories using some terms that are not commonly found in the current spatial audio literature. The inexperienced listeners produced five categories that overlapped with the experienced group terms but that were not as detailed.
Convention Paper 9307 (Purchase now)

P13-2 Towards Unification of Methods for Speech, Audio, Picture, and Multimedia Quality Assessment—Slawomir Zielinski, Bialystok University of Technology - Bialystok, Poland; Francis Rumsey, Logophon Ltd. - Oxfordshire, UK; Søren Bech, Bang & Olufsen a/s - Struer, Denmark; Aalborg University - Aalborg, Denmark
The paper addresses the need to develop unified methods for subjective and objective quality assessment across speech, audio, picture, and multimedia applications. Commonalities and differences between the currently used standards are overviewed. Examples of the already undertaken research attempting to “bridge the gap” between the quality assessment methods used in various disciplines are indicated. Prospective challenges faced by researchers in the unification process are outlined. They include development of unified scales, defining unified anchors, integration of objective models, maintaining “backward comparability,” and undertaking joint standardization efforts across industry sectors.
Convention Paper 9308 (Purchase now)

P13-3 An Investigation of the Relationship between Listener Envelopment and Room Acoustic Parameters–The Influence of Varied Direct Sound Levels and Onset Times of Late Reverberation on Listener Envelopment—Mai Ishida, Tokyo University of the Arts - Adachi-ku, Tokyo, Japan; Toru Kamekawa, Tokyo University of the Arts - Adachi-ku, Tokyo, Japan; Atsushi Marui, Tokyo University of the Arts - Tokyo, Japan
It is proposed that the late reflection energy from 80 ms after the direct sound contributes to listener envelopment (LEV). According to previous research, 80 ms is not necessarily the most suitable onset-time of late reverberation for predicting LEV. In addition to this, LEV tends to increase if C80 decreases. However, it is possible for LEV to be low for very low early energy with relatively higher late energy. In this study the LEV for stimuli of varied direct sound levels and onset-times of late reverberation was investigated. As a result, it is suggested that as the direct sound level increases, the LEV increases. Additionally, we confirmed that LEV increased as C10 increased.
Convention Paper 9309 (Purchase now)

P13-4 The Development of a Sound Wheel for Reproduced Sound—Torben H. Pedersen, DELTA SenseLab - Hørsholm, Denmark; Nick Zacharov, DELTA SenseLab - Hørsholm, Denmark
Sound quality is an important aspect in many sound reproduction applications. In recent years sensory evaluation techniques have been gaining popularity for the detailed perceptual assessment of device sound characteristics. From the literature, hundreds of descriptors can be found to describe the nature of sound quality and this often becomes the focus of debate among researchers, rather than the product development itself. In an effort to shift the focus back to the areas of importance, i.e., the product, this study seeks to define a common terminology, a lexicon, for the characterization of sound quality in loudspeakers, headphones, or other sound reproduction systems. The study summarized the gathering of descriptors for sound character from the literature and then experimental leading to a structure protocol of perceptual sound quality attributes for this domain of application. A structured sound wheel is presented comprising of different layers of attributes. For each attribute, definitions have been developed with associated sound samples for training. The paper presents the on-going development process, including validation of attributes and their definitions.
Convention Paper 9310 (Purchase now)

P13-5 How Much Is the Use of a Rating Scale by a Listener Influenced by Anchors and by the Listener's Experience?—Nadja Schinkel-Bielefeld, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany; International Audio Laboratories - Erlangen, Germany; Anna Katharina Leschanowsky, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
It has been postulated that anchors in multi-stimulus listening tests for audio quality evaluation should have an item-independent quality, as listeners will likely shift their rating scale if the quality of the anchor varies. However, expert listeners have a very stable internal rating scale, which can be seen from the repeatability of their results when performing the same test multiple times. So they may stick to their usual scale even if the anchor varies. We find that listeners do not shift their rating scale by the full amount the anchor is shifted but only up to 60% of that. Nevertheless this makes quantitative comparisons between different test results difficult even if the anchor varies only by 5 Mushra points.
Convention Paper 9311 (Purchase now)

P13-6 Quantifying Auditory Perception: Dimensions of Pleasantness and Unpleasantness—Judith Liebetrau, Fraunhofer IDMT - Ilmenau, Germany; Thomas Sporer, Fraunhofer Institute for Digital Media Technology IDMT - Ilmenau, Germany; Marius Becker, Technische Universität Ilmenau - Ilmenau, Germany; Thanh Phong Duong, Technische Universität Ilmenau - Ilmenau, Germany; Andreas Ebert, Technische Universität Ilmenau - Ilmenau, Germany; Martin Härtig, Technische Universität Ilmenau - Ilmenau, Germany; Phillip Heidrich, Technische Universität Ilmenau - Ilmenau, Germany; Jakob Kirner, Technische Universität Ilmenau - Ilmenau, Germany; Oliver Rehling, Technische Universität Ilmenau - Ilmenau, Germany; Dominik Vöst, Technische Universität Ilmenau - Ilmenau, Germany; Roberto Walter, Technische Universität Ilmenau - Ilmenau, Germany; Michael Zierenner, Technische Universität Ilmenau - Ilmenau, Germany; Tobian Clauß, Fraunhofer IDMT - Ilmenau, Germany
Psychoacoustic attributes like roughness, sharpness, tonality, and fluctuation strength are often used to explain and calculate major properties of sound. While in general these properties are well understood, the concept of “pleasantness” depends on several factors. Investigating the underlying dimensions of pleasantness was the goal of the presented studies. The perception of psychoacoustic attributes was assessed for 22 different audio stimuli by more than 15 listeners. In addition the perception of pleasantness and unpleasantness for these items was evaluated. All tests were conducted in laboratory as well as home environment. The results showed that a link between psychoacoustic attributes and concept of pleasantness could be established. Surprisingly, the relation between the single attributes and pleasantness changed dependent on the applied analysis method.
Convention Paper 9312 (Purchase now)

P13-7 Audio Quality Moderates Localization Accuracy: Two Distinct Perceptual Effects?—PerMagnus Lindborg, Nanyang Technological University - Singapore; Nicholas A. Kwan, Nanyang Technological University - Singapore
Audio quality is known to cross-modally influence reaction speed, sense of presence, and visual quality. We designed an experiment to test the effect of audio quality on source localization. Stimuli with different MP3 compression rates, as a proxy for audio quality, were generated from drum samples. Participants (n = 18) estimated the position of a snare drum target while compression rate, masker, and target position were systematically manipulated in a full-factorial repeated-measures experiment design. Analysis of variance revealed that location accuracy was better in wide target positions than in narrow, with a medium effect size; and that the effect of target position was moderated by compression rate in different directions for wide and narrow targets. The results suggest that there might be two perceptual effects at play: one, whereby increased audio quality causes a widening of the soundstage, possibly via a SMARC-like mechanism, and two, whereby it enables higher localization accuracy. In the narrow target positions in this experiment, the two effects acted in opposite directions and largely cancelled each other out. In the wide target presentations, their effects were compounded and led to significant correlations between compression rate and localization error.
Convention Paper 9313 (Purchase now)

 
 

P14 - (Lecture) Transducers—Part 2

Saturday, May 9, 14:30 — 17:30 (Room: Królewski)

Chair:
Aki Mäkivirta, Genelec - Lapinlahti, Finland

P14-1 High Power Efficiency and Broad Flat Radiation Bandwidth of a Parametric Array PMUT Loudspeaker—Kyounghun Been, POSTECH - Gyeongsangbuk-do, Korea; Younghwan Hwang, POSTECH - Gyeongsangbuk-do, Korea; Yub Je, Agency for Defense Development - Gyeongsangnam-do, Korea; Haksue Lee, Agency for Defense Development - Gyeongsangnam-do, Korea; Wonkyu Moon, POSTECH - Gyeongsangbuk-do, Korea
Parametric Array loudspeakers can generate a sound beam using nonlinear acoustic interactions, widely known as “Parametric Array,” that can enable private listening in a public area. Parametric array loudspeakers can be applied to many applications, such as information technology devices, that require a high power efficiency and wide bandwidth. In a previous study, a piezoelectric micro-machined ultrasonic transducer (PMUT) is shown to be an efficient unit for a parametric array loudspeaker. In this paper we will describe realization of a parametric array loudspeaker with high power efficiency (up to 71%) and wide flat radiation bandwidth (19.5 kHz, difference frequency wave with equalization), which consists of an array of PMUTs with two resonance frequencies (f1 = 100 kHz, f2 = 110 kHz) and use of “out-of-phase” driving techniques.
Convention Paper 9314 (Purchase now)

P14-2 Slit-Firing Sound Plate design with Slim Elliptical Speaker—Gyeong-Tae Lee, DMC R&D Center, Samsung Electronics Co. - Suwon-si, Gyeonggi-do, Korea; Jong-Bae Kim, Samsung Electronics - Suwon-si, Gyeonggi-do, Korea; Seong-Ha Son, Samsung Electronics Co. Ltd. - Suwon, Gyeonggi-do, Korea
Slim design has emerged recently as a new form factor for the speaker system of an electronic device. However, this form factor is disadvantageous to sound performance because of the narrow space for a speaker unit. In this paper, to overcome this drawback, we designed Slit-Firing Sound Plate, which is a slim speaker system using diffraction through a slit, and developed a slim elliptical speaker that is optimized for the boundary conditions for the slit. After building and tuning a prototype, the sound performance of the prototype was assessed by measuring and examining frequency response, transient response, and directivity beam pattern. As a result, proposed novel design shows high performance that makes it suitable for the application of slim electronic devices.
Convention Paper 9315 (Purchase now)

P14-3 Improvements in Elimination of Loudspeaker Distortion in Acoustic Measurements—Finn T. Agerkvist, Technical University of Denmark - Kgs. Lyngby, Denmark; Antoni Torras-Rosell, Danish National Metrology Institute - Lyngby, Denmark; Richard McWalter, Technical University of Denmark - Lyngby, Denmark
This paper investigates the influence of nonlinear components that contaminate the linear response of acoustic transducers and presents improved methods for eliminating the influence of nonlinearities in acoustic measurements. The method is evaluated with pure sinusoidal signals as well as swept sine signals and is tested on models of memoryless nonlinear systems as well as nonlinear loudspeakers. The method is shown to give a clear benefit over existing methods. Two techniques that improve the signal to noise ratio are demonstrated: the first uses more measurement level than the number of orders to be separated, whereas the other one is based on standard Tikhonov regularization. Both methods are shown to significantly improve the signal to noise ratio.
Convention Paper 9316 (Purchase now)

P14-4 Flux Modulation in the Electrodynamic Loudspeaker—Morten Halvorsen, PointSource Acoustics - Roskilde, Denmark; Carsten Tinggaard, PointSource Acoustics - Roskilde, Denmark; Finn T. Agerkvist, Technical University of Denmark - Kgs. Lyngby, Denmark
This paper discusses the effect of flux modulation in the electrodynamic loudspeaker with main focus on the effect on the force factor. A measurement setup to measure the AC flux modulation with static voice coil is explained and the measurements show good consistency with FEA simulations. Measurements of the generated AC flux modulation shows that eddy currents are the main source to magnetic losses in form of phase lag and amplitude changes. Use of a copper cap shows a decrease in flux modulation amplitude at the expense of increased power losses. Finally, simulations show that there is a high dependency between the generated AC flux modulation from the voice coil and the AC force factor change.
Convention Paper 9317 (Purchase now)

P14-5 Validation of Power Requirement Model for Active Loudspeakers—Henrik Schneider, Technical University of Denmark - Kgs. Lyngby, Denmark; Anders N. Madsen, Technical University of Denmark - Kongens Lyngby, Denmark; Ruben Bjerregaard, Technical University of Denmark - Kongens Lyngby, Denmark; Arnold Knott, Technical University of Denmark - Kgs. Lyngby, Denmark; Michael A. E. Andersen, Technical University of Denmark - Kgs. Lyngby, Denmark
The actual power requirement of an active loudspeaker during playback of music has not received much attention in the literature. This is probably because no single and simple solution exists and because a complete system knowledge from input voltage to output sound pressure level is required. There are, however, many advantages that could be harvested from such knowledge like size, cost, and efficiency improvements. In this paper a recently proposed power requirement model for active loudspeakers is experimentally validated and the model is expanded to include the closed and vented type enclosures in addition to the main loudspeaker non-linearities.
Convention Paper 9318 (Purchase now)

P14-6 Subwoofers in Rooms: Stereophonic Reproduction—Juha Backman, Microsoft - Espoo, Finland
A study based on computational model of interaural level and time differences at the lowest audio frequencies, often reproduced through subwoofers, is presented. This work studies whether interaural differences can exist, and if they do, what kind of relationship there is between the loudspeaker direction and the interaural differences when monophonic and stereophonic subwoofer arrangements are considered. The calculations are made for both simple amplitude panned signals and for simulated microphone signals. The results indicate that strong narrow-band differences can exist, especially near room eigenfrequencies when the listener is close to nodes of the room modes and that the modes of the recording room can have an effect on the sound field of the listening room. In addition to the computational results an analysis of interchannel level differences in recordings is presented, confirming the computational model.
Convention Paper 9319 (Purchase now)

 
 

P15 - (Lecture) Spatial Audio—Part 2

Sunday, May 10, 09:00 — 12:30 (Room: Belweder)

Chair:
Ville Pulkki, Aalto University - Espoo, Finland

P15-1 Analysis on the Timbre Coloration of Wave Field Synthesis Using a Binaural Loudness Model—Bosun Xie, South China University of Technology - Guangzhou, China; Haiming Mai, South China University of Technology - Guangzhou, China; Yang Liu, South China University of Technology - Guangzhou, China; Xiaoli Zhong, South China University of Technology - Guangzhou, China
Wave field synthesis (WFS) aims to reconstruct a target sound field within an extending region. An ideal WFS system requires continuous loudspeakers array. Discrete loudspeaker array in practical WFS causes spatial aliasing errors above the Nyquist frequency limit, and thus results in timbre coloration. The present work analyzes the timbre in WFS using Moore’s modified binaural loudness model, in which the binaural loudness level spectra is used as a criterion to evaluate the timbre coloration. The results prove that timbre coloration reduces with the increasing distance between field point and active loudspeakers; and reducing the space between adjacent loudspeakers reduces perceivable timbre coloration. A psychoacoustic experiment yields consistent results with those of analysis, and therefore validates the proposed method.
Convention Paper 9320 (Purchase now)

P15-2 Physical Properties of Local Wave Field Synthesis Using Linear Loudspeaker Arrays—Fiete Winter, University of Rostock - Rostock, Germany; Sascha Spors, University of Rostock - Rostock, Germany
Wave Field Synthesis aims at a physically accurate synthesis of a desired sound field inside an extended listening area. Due to limitation of practical loudspeaker setups, the accuracy of this sound field synthesis technique over the entire listening area is limited. Local Wave Field Synthesis narrows the spatial extent down to a local listening area in order to improve the reproduction accuracy inside this limited region. Recently a method has been published, which utilizes focused sources as a distribution of more densely placed virtual secondary sources around the local area. Within this paper an analytical framework is established to analyze the physical properties of this approach for linear loudspeaker setups.
Convention Paper 9321 (Purchase now)

P15-3 Pressure-Matching Beamforming Method for Loudspeaker Arrays with Frequency Dependent Selection of Control Points—Ferdinando Olivieri, University of Southampton - Southampton, Hampshire, UK; Filippo Maria Fazi, University of Southampton - Southampton, Hampshire, UK; Mincheol Shin, ISVR, University of Southampton - Southampton, Hampshire, UK; Philip Nelson, ISVR, University of Southampton - Southampton, UK
The Pressure-Matching Method (PMM) is a signal processing technique used to generate the digital filters required by a loudspeaker array to reproduce a desired sound field. System performance depends on the choice of a number of parameters of the numerical algorithm, such as the target field and the regularization factor. If a target sound field is chosen with large amplitude variation between the so-called control points, performance might also depend on the relative distance between these points in relation to a the wavelength of the sound to be reproduced. If this distance is too small, the accuracy of the reproduced field may be reduced at the listener location. A strategy is proposed to improve the PMM that is based on a frequency-dependent selection of the control points that contribute to the PMM cost function. By means of numerical simulations and experiments in anechoic environment, it is shown that the proposed method allows for accurate control of the response of the reproduced field at the listener location.
Convention Paper 9322 (Purchase now)

P15-4 Discussion of the Wavefront Sculpture Technology Criteria for Straight Line Arrays—Frank Schultz, University of Rostock / Institute of Communications Engineering - Rostock, Germany; Florian Straube, TU Berlin - Berlin, Germany; Sascha Spors, University of Rostock - Rostock, Germany
Wavefront Sculpture Technology introduced line source arrays for large scale sound reinforcement, aiming at the synthesis of highly spatial-aliasing free sound fields for full audio bandwidth. The paper revisits this technology and its criteria for straight arrays using a signal processing model from sound field synthesis. Since the latest array designs exhibit very small driver distances, the sampling condition for grating lobe free electronic beam forming regains special interest. Furthermore, a discussion that extends the initial derivations of the spatial lowpass characteristics of circular and line pistons and line pistons with wavefront curvature applied in subarrays is given.
Convention Paper 9323 (Purchase now)

P15-5 Sound Field Synthesis of Virtual Cylindrical Waves Using Circular and Spherical Loudspeaker Arrays—Nara Hahn, University of Rostock - Rostock, Germany; Sascha Spors, University of Rostock - Rostock, Germany
In sound field synthesis, like near-field compensated higher-order Ambisonics or Wave Field Synthesis, various virtual source models are used to describe a virtual sound scene. In near-field compensated higher-order Ambisonics, the virtual sound field has to be expanded into spherical harmonics. Unlike plane waves or spherical waves, cylindrical waves are not conveniently represented in the spherical harmonics domain. In this paper we tackle this problem and derive closed form driving functions for virtual cylindrical waves. The physical properties of synthesized sound fields are investigated through numerical simulations, where the results are compared with virtual cylindrical waves in wave field synthesis.
Convention Paper 9324 (Purchase now)

P15-6 Perceptual Band Allocation (PBA) for the Rendering of Vertical Image Spread with a Vertical 2D Loudspeaker Array—Hyunkook Lee, University of Huddersfield - Huddersfield, UK
A series of subjective experiments were conducted to investigate a novel vertical image rendering method named “Perceptual Band Allocation (PBA),” using octave bands of pink noise with a vertical 2D reproduction setup with main and height loudspeaker pairs. The perceived height of each octave band was first measured for the main and height loudspeakers individually. Results suggested a significant difference between monophonic and stereophonic images in the perceived relationship between frequency and height. Six different test conditions have been created aiming for various degrees of vertical image spread, in such a way that each frequency band was mapped to either the main or height loudspeaker layer based on the results from the localization experiment. Multiple comparison tests were conducted to grade the perceived magnitude of vertical image spread. It was generally found that various degrees of vertical image spread could be rendered using different PBA schemes, but the perceived results did not fully match predicted results based on the localization results. Differences between the main and height loudspeaker layers in the spectral weightings of ear-input signal at certain frequencies was identified as one of the factors that influenced this result.
Convention Paper 9325 (Purchase now)

P15-7 Synthesis of Moving Reverberation Using Active Acoustics—Preliminary Report—Jung Wook (Jonathan) Hong, McGill University - Montreal, QC, Canada; GKL Audio Inc. - Montreal, QC, Canada; Wieslaw Woszczyk, McGill University - Montreal, QC, Canada; Durand R. Begault, NASA Ames Research Center - Moffett Field, CA, USA; David Benson, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
An ambient sound field created artificially using active acoustics (virtual acoustics) attempts to resurrect the perceived naturalness of the original architectural space and the distinct responsiveness to musical sound sources. Moving reverberation is typically associated with coupled volumes contained within a larger architectural space where each volume is activated by a sound source at a different moment in time due to propagation delay. The produced energy has a diverse characteristic rate of decay with which its energy is mixed within the common space causing multiple slopes on the decay. This causes a sensation of a decaying diffused sound that is not homogenized but distinctly appearing in different zones of the space as a shifting acoustic energy. In order to reconstruct the moving reverberation, an active acoustics system was used to render an ambient sound field from measured impulse responses of large architectural space, the Grace Cathedral in San Francisco.
Convention Paper 9326 (Purchase now)

 
 

P16 - (Poster) Applications in Audio

Sunday, May 10, 10:00 — 12:00 (Foyer)

P16-1 Dubbing Studio for 22.2 Multichannel Sound System in NHK Broadcasting Center—Ikuko Sawaya, Science & Technology Research Laboratories, Japan Broadcasting Corp (NHK). - Setagaya, Tokyo, Japan; Kengo Sasaki, Japan Broadcasting Corporation - Shibuya, Tokyo, Japan; Shinji Mikami, Japan Broadcasting Corporation - Shibuya, Tokyo, Japan; Hiroyuki Okubo, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Kazuho Ono, NHK Engineering System Inc. - Setagaya-ku, Tokyo, Japan
8K Super Hi-Vision is planned to be on test broadcasting in 2016 and to launch a full broadcasting service in 2018 in Japan. NHK has developed a program production system for 22.2 multichannel sound for 8K Super Hi-Vision. As part of the development NHK completed the construction of a 22.2 ch dubbing studio in the NHK Broadcasting Center in July 2014. This is the first 22.2 ch dubbing studio in the production field in the world with a loudspeaker configuration that meets the standard Recommendation ITU-R BS.2051. In this paper we discuss the 22.2 ch production system, including its sound mixing system, loudspeaker system for monitoring, and perforated screen for 8K resolution, as well as the room design and the characteristics of the room acoustics in the studio.
Convention Paper 9327 (Purchase now)

P16-2 A Floor Acoustic Sensor for Fall Classification—Emanuele Principi, Università Politecnica delle Marche - Ancona, Italy; Paolo Olivetti, Scientific Direction, Italian National Institute of Health and Science on Aging (INRCA) - Ancona, Italy; Stefano Squartini, Università Politecnica delle Marche - Ancona, Italy; Roberto Bonfigli, Universita Politecnica delle Marche - Ancona, Italy; Francesco Piazza, Universitá Politecnica della Marche - Ancona (AN), Italy
The interest in assistive technologies for supporting people at home is constantly increasing, both in academia and industry. In this context the authors propose a fall classification system based on an innovative acoustic sensor that operates similarly to stethoscopes and captures the acoustic waves transmitted through the floor. The sensor is designed to minimize the impact of aerial sounds in recordings, thus allowing a more focused acoustic description of fall events. In this preliminary work, the audio signals acquired by means of the sensor are processed by a fall recognition algorithm based on Mel-Frequency Cepstral Coefficients, Supervectors, and Support Vector Machines to discriminate among different types of fall events. The performance of the algorithm has been evaluated against a specific audio corpus comprising falls of persons and of common objects. The results show the effectiveness of the approach.
Convention Paper 9329 (Purchase now)

P16-3 Active Field Control in the Teatr Wielki—Opera Narodowa—Takayuki Watanabe, Yamaha Corp. - Hamamatsu, Shizuoka, Japan; Hideo Miyazaki, Yamaha Corp. - Hamamatsu, Shizuoka, Japan; Shinichi Sawara, Yamaha Corp. - Hamamatsu, Shizuoka, Japan; Masahiro Ikeda, Yamaha Corporation - Hamamatsu, Shizuoka, Japan; Ron Bakker, Yamaha Commercial Audio Systems Europe - Rellingen, Germany
This opera house of 1,828 seats boasts one of Europe's largest stages and is highly reputed for its repertoire and acoustics. However, it presented a number of issues including poor communication between the singers and the orchestra pit, insufficient loudness of the upstage singers for the audience, a lack of reverberation when the house was occupied, and insufficient loudness at the seats under the balconies. For these reasons Active Field Control System (AFC) was adopted as a means to improve the acoustics while preserving the historic architecture of the opera house. This paper presents an overview of that system and the benefits achieved by its introduction.
Convention Paper 9330 (Purchase now)

P16-4 An Environment for Submillisecond-Latency Audio and Sensor Processing on BeagleBone Black—Andrew McPherson, Queen Mary University of London - London, UK; Victor Zappi, University of British Columbia - Vancouver, BC, Canada
This paper presents a new environment for ultra-low-latency processing of audio and sensor data on embedded hardware. The platform, which is targeted at digital musical instruments and audio effects, is based on the low-cost BeagleBone Black single-board computer. A custom expansion board features stereo audio and 8 channels each of 16-bit ADC and 16-bit DAC for sensors and actuators. In contrast to typical embedded Linux approaches, the platform uses the Xenomai real-time kernel extensions to achieve latency as low as 80 microseconds, making the platform suitable for the most demanding of low-latency audio tasks. The paper presents the hardware, software, evaluation, and applications of the system.
Convention Paper 9331 (Purchase now)

P16-5 Commonwealth Games 2014 Host Broadcaster Training Initiative–A Game Changer?—Patrick Quinn, Glasgow Caledonian University - Glasgow, Scotland, UK; David Moore, Glasgow Caledonian University - Glasgow, Lanarkshire, UK
Glasgow Commonwealth Games 2014 provided an ideal platform for over 200 students to gain work experience in sports broadcasting as part of the Host Broadcaster Training Initiative. Organized by SVGTV and Creative Loop with the intention of attracting students into this growing area of broadcasting, the successful initiative has encouraged many of the students involved, including Audio Technology students from Glasgow Caledonian University, to consider and subsequently pursue careers in broadcasting. In addition as a legacy from the initiative a new course is planned at Glasgow Caledonian University in Broadcasting Technology.
Convention Paper 9332 (Purchase now)

P16-6 Influence of Noise on the Effectiveness of Speaker Identification in the Acoustics of Crime—Tomasz Smutnicki, Wroclaw University of Technology - Wroclaw, Poland; Stefan Brachmanski, Wroclaw University of Technology - Wroclaw, Poland
One of the main elements of the research in acoustics of crime is to compare the evidential recording with the comparative adequate pattern. Unfortunately, the evidential recording usually has poor quality and contains relatively high level of noise, which results from the way of its acquiring, namely eavesdropping or record of automatic monitoring. The signal quality and the noise to signal ratio have an impact on the value of the extracted voice metrics. In this paper we analyze factors that may have an impact on formants value in the human voice. Based on Six Sigma methodology we also designed and performed an experiment that allowed us to determine the extent in which various factors influence on the resulting parameters.
Convention Paper 9333 (Purchase now)

P16-7 Acoustic Profile of Identified Speaker in Forensics—Krystian Kapala, Wroclaw University of Technology - Wroclaw, Poland; Stefan Brachmanski, Wroclaw University of Technology - Wroclaw, Poland
Speaker identification is deemed to be one of the basic tasks in audio forensics. Delivering a categorical opinion is often difficult due to insufficient quality of the recorded material, simulation or modulation of speaker’s voice. Hence, a wide-ranging approach to the identification process is used, including both subjective and objective methods. With their help, it becomes possible to obtain a broad spectrum of speech characteristics ranging from low-level features relating to physical construction of the vocal tract to advanced ones concerning various ways of expressing oneself and articulation, acquired during socialization process. This article describes an experiment undertaken to create acoustic profiles of a chosen group of speakers based on the features mentioned above.
Convention Paper 9334 (Purchase now)

P16-8 An Implementation Of Beamforming Algorithm On FPGA Platform with Digital Microphone Array—Iva Salom, Institute Mihajlo Pupin, University of Belgrade - Belgrade, Serbia; Vladimir Celebic, Institute Mihailo Pupin, University of Belgrade - Belgrade, Serbia; Milan Milanovic, Institute Mihailo Pupin, University of Belgrade - Belgrade, Serbia; Dejan Todorovic, Dirigent Acoustics Ltd. - Belgrade, Serbia; Jurij Prezelj, University of Ljubljana - Ljubljana, Slovenia
The goal of the project described in this paper was to design an acoustic system for localization of the dominant noise source by implementation of the conventional delay-and-sum beamforming algorithm on FPGA platform with a sound receiver system based on digital MEMS microphone array. The system consists of a platform for acoustic signal acquisition and data processing (microphone array, interface, and central block), and a platform for monitoring and control (a computer with a user application). Such configuration provides the execution of the beamforming algorithm in real time. Additionally, FPGAs are bringing many benefits in terms of safety, reliability, rapidity, and power consumption. The platform was tested and verified with various microphone array configurations and results are presented in the paper.
Convention Paper 9335 (Purchase now)

P16-9 Measuring and Analyzing Audio Levels in Film, Commercials, and Movie Trailers Using L_eq(A) Values and the LUFS Loudness Model—Bozena Kostek, Gdansk University of Technology - Gdansk, Poland; Audio Acoustics Lab.; Kamila Milarska, Gdansk University of Technology - Gdansk, Poland; Aleksander Zakrzewski, Gdansk University of Technology - Gdansk, Poland
The purpose of this paper is to describe the measurement of loudness levels in movies, movie trailers, and commercials displayed before feature films at movie theaters. In the initial section, the paper discusses the issues related to measurement of loudness levels, provides recommendations regarding permissible loudness levels during movie screenings, and mentions the applied units of measurement. The following section of the paper describes the actual measurements, measuring equipment, as well as analysis of the results of the measurements. The summary provides conclusions about the measured loudness levels at movie theaters, for DVD and Blu-ray discs, and for YouTube videos.
Convention Paper 9336 (Purchase now)

P16-10 Arbitrary Trajectory Estimation of a Moving Acoustic Source—Sai Gunaranjan Pelluri, Indian Institute of Science - Bangalore, India; Thippur V. Sreenivas, Indian Institute of Science - Bangalore, India
The state-of-the-art passive methods for estimating the trajectory of a moving acoustic source involve computing the cross-correlation function either directly or indirectly (as in the case of the Beam Forming Approach) between pairs of microphones. Also there have been several Active Acoustic techniques such as SONAR, RADAR, etc., which have been used to estimate the source parameters such as velocity, trajectory, etc. They involve pinging the source with a known signal. In this paper, given the fact that the moving source itself generates a signal, we propose a technique by which we estimate the source trajectory using only the signal captured at the receiver thereby avoiding the need to ping the source and without computing the cross-correlation function.
Convention Paper 9266 (Purchase now)

 
 

P17 - (Poster) Recording and Production

Sunday, May 10, 13:00 — 15:00 (Foyer)

P17-1 Tom–Tom Drumheads Miking Analysis—Andrés Felipe Quiroga, Universidad de San Buenaventura - Bogotá, Colombia; Juan David Garcia, Universidad de San Buenaventura - Bogota, Colombia; Dario Páez, Universidad de San Buenaventura - Bogotá, Colombia
Different drum recording techniques have been developed through time, from stereo to close miking techniques. This is relevant, since the techniques and the characteristics of the instrument will define its sound within the final mix. A study was designed that gives experienced and non-experienced recording engineers tools and specific characteristics of tom–tom close miking techniques with different drumheads, microphones, and capture positions. Results indicate the behaviors of the different drumheads and capture positions with the different microphones. The first frequency band of resonance (attack) shows the highest decay level compared to the second band of resonance (tone), and the edge position presented the lower decay level on the second band of resonance, showing its resonant behavior on the envelope.
Convention Paper 9338 (Purchase now)

P17-2 Concept of Film Sound Restoration by Adapting to Contemporary Cinema Theatre—Joanna Napieralska, Frederic Chopin University of Music - Warsaw, Poland
This paper presents an individual approach to restoration of Polish film sound based on the author’s own works. It answers the following question: under what conditions, and by the use of which techniques, may the restoration of archive film sound provide the viewer with a cleaner reproduction of the original sound while maintaining the standard expected by a modern cinema-going audience. At its basic level, the sound restoration routine comprises the following: transfer from the magnetic tape, syncing, cleaning of low/high frequency noises, repair of material impairments and reprinting effect, and mastering for broadcasting, cinema, DVD/Blu-ray, and internet formats. The reconstruction discussed modifies the sound quality and sometimes the contents. However, it can be performed only under certain legal restrictions.
Convention Paper 9339 (Purchase now)

P17-3 Deep Sound Design: Procedural Implementations Based on General Audiovisual Production Pipeline Integration—José Roberto Cabezas Hernández, Universidad Nacional Autónoma de México - Mexico City, Mexico
This work is an exploration for the integration of data available on the visual post-production pipeline for the development of procedural sound design and composition techniques, by implementing different methods to allow access of different file formats for scene and shot reconstruction. The main purpose in an audiovisual creation context is to investigate stronger and inmost image-sound cognitive perceptions and relationships generated by data usage and analysis; also, reducing automation time by directly linking data to parameters for developing a creative editing, mixing, design, and compositional workflow based on a shot by shot manipulation.
Convention Paper 9340 (Purchase now)

P17-4 An Investigation into the Efficacy of Methods Commonly Employed by Mix Engineers to Reduce Frequency Masking in the Mixing of Multitrack Musical Recordings—Jonathan Wakefield, University of Huddersfield - Huddersfield, UK; Christopher Dewey, University of Huddersfield - Huddersfield, UK
Studio engineers use a variety of techniques to reduce frequency masking between instruments when mixing multitrack musical recordings. This study evaluates the efficacy of three techniques, namely mirrored equalization, frequency spectrum sharing, and stereo panning against their variations to confirm the veracity of accepted practice. Mirrored equalization involves boosting one instrument and cutting the other at the same frequency. Frequency spectrum sharing involves low pass filtering one instrument and high pass filtering the other. Panning involves placing two competing instruments at different pan positions. Test subjects used eight tools comprising a single unlabeled slider to reduce frequency masking in several two instrument scenarios. Satisfaction values were recorded. Results indicate subjects preferred using tools that panned both audio tracks.
Convention Paper 9341 (Purchase now)

P17-5 An Interactive Multimedia Experience: A Case Study—Andrew J. Horsburgh, Southampton Solent University - Southampton, UK
Accurate representation of three dimensional spaces, both real and virtual, within an environment is a matter of concern for researchers and content producers in the media industry; it is expected that truly immersive experiences will become more desirable outside of research labs and bespoke facilities. This paper presents a case study examining the implementation between visual and audible elements to form a singular experience of immersion, AIME, at Solent University. The computer-based system uses a time-code generator that allows for seamless integration between audio workstations, visual playback, and external lighting. The prototype system uses 2nd order ambisonic audio reproduction, three large panel displays for vision, and an external lighting rig running from time code.
Convention Paper 9342 (Purchase now)

P17-6 Evaluation of an Algorithm for the Automatic Detection of Salient Frequencies in Individual Tracks of Multitrack Musical Recordings—Jonathan Wakefield, University of Huddersfield - Huddersfield, UK; Christopher Dewey, University of Huddersfield - Huddersfield, UK
This paper evaluates the performance of a salient frequency detection algorithm. The algorithm calculates each FFT bin maximum as the maximum value of that bin across an audio region and identifies the FFT bin maximum peaks with the highest five deemed to be the most salient frequencies. To determine the algorithm’s efficacy test subjects were asked to identify the salient frequencies in eighteen audio tracks. These results were compared against the algorithm’s results. The algorithm was successful with electric guitars but struggled with other instruments and in detecting secondary salient frequencies. In a second experiment subjects equalised the same audio tracks using the detected peaks as fixed centre frequencies. Subjects were more satisfied than expected when using these frequencies.
Convention Paper 9343 (Purchase now)

P17-7 The Sonic Vernacular: Considering Communicative Timbral Gestures in Modern Music Production—Leah Kardos, Kingston University London - Kingston Upon Thames, Surrey, UK
Over the course of audio recording history, we have seen the activity of sound recording widen in scope “from a technical matter to a conceptual and artistic one” (Moorefield 2010) and the producer’s role evolving from technician to “auteur.” For recording practitioners engaged in artistic and commercial industry and discourse, fluency in contemporary and historic sound languages is advantageous This paper seeks to find the best, most practically useful method to describe these characteristics in practice, identify a clear and suitable way to talk about and analyze these uses of communicative timbral gestures, as heard in modern music productions.”
Convention Paper 9344 (Purchase now)

P17-8 Auto Panning In-Ear Monitors for Live Performers—Tom Webb, Southampton Solent University - Southampton, UK; Andrew J. Horsburgh, Southampton Solent University - Southampton, UK
In a live musical performance, accurate stage monitoring is a vital element to achieve the optimal performance. Current stage monitoring uses traditional musician-facing loudspeakers. Problems can be surmised as excessive Sound Pressure Level (SPL), the inability to hear themselves, acoustic feedback, and general stage untidiness/space requirements. In-ear monitors (IEM’s) can offer a solution to these problems when the IEM system has been properly designed [7]. One crucial issue with IEMs is the sense of isolation and disconnection from stage noise and crowd. To overcome this issue, an auto-panning system that adjusts spatial placement of audio channels within the performers stage mix has been designed and built.
Convention Paper 9345 (Purchase now)

P17-9 An Investigation into Plausibility in the Mixing of Foley Sounds in Film and Television—Braham Hughes, University of Huddersfield - Huddersfield, UK; Jonathan Wakefield, University of Huddersfield - Huddersfield, UK
This paper describes an experiment that tested the plausibility of a selection of post-production audio mixes of Foley for a short film. The mixes differed in the implementation of four primary audio mixing parameters: panning, level, equalization, and the control of reverberation effects. The experiments presented test subjects with mixes in which one of the four primary parameters was altered while the rest remained at levels deemed to conform to an “industry standard” reference mix that had been verified by an expert industry practitioner. Results show that there is a statistically significant affect on plausibility of using even slight dynamic variation of pan, level, and equalization control to enhance the perception of realism of Foley that move in a scene.
Convention Paper 9346 (Purchase now)

P17-10 A Semantically Motivated Gestural Interface for the Control of a Dynamic Range Compressor—Thomas Wilson, University of Huddersfield - Huddersfield, West Yorkshire, UK; Steven Fenton, University of Huddersfield - Huddersfield, West Yorkshire, UK; Name Withheld, Removed at the request of the presenter.
This paper presents a simplified 2D gesture based approach to modifying dynamics within a musical signal. Despite the growth in gesture-controlled audio seen over recent years, it has primarily been limited to the upper workflow/navigation level. This has been compounded by the Skeuomorphic design approaches of graphical user interfaces (GUI). This design approach, although representative of the original piece of audio equipment, often lowers workflow and hinders the simultaneous control of parameters. Following a large scale gesture elicitation exercise utilizing a common 2D touch pad and analysis of semantic audio control parameters, a set of reduced multi-modal parameters are proposed that offers both workflow efficiency and a much simplified method of control for dynamic range compression.
Convention Paper 9347 (Purchase now)

P17-11 Natural Sound Recording of an Orchestra with Three-Dimensional Sound—Kimio Hamasaki, ARTSRIDGE LLC - Chiba, Japan; Wilfried Van Baelen, Auro Technologies N.V. - Mol, Belgium
This paper introduces the microphone techniques for recording an orchestra with three-dimensional multichannel sound and discusses the spatial impression provided by the recorded sound of an orchestra. Listeners in a concert hall simultaneously hear both a direct sound arriving from each musical instrument and an indirect sound reflected from the walls and the ceiling. Concerning a direct sound, existing microphone techniques can be used for three-dimensional multichannel sound with necessary modification, but new microphone techniques should be developed for an indirect sound. This paper will propose the microphone technique consisting of a main microphone array and an ambience microphone array, which will enable us to control spatial impressions easily and realize the stable sound source localization.
Convention Paper 9348 (Purchase now)

 
 

P18 - (Lecture) Semantic Audio

Sunday, May 10, 14:00 — 17:00 (Room: Belweder)

Chair:
Pedro Duarte Pestana, Catholic University of Oporto - CITAR - Oporto, Portugal; Universidade Lusíada de Lisboa - Lisbon, Portugal

P18-1 Music Onset Detection Using a Bidirectional Mismatch Procedure Based on Smoothly Varying-Q Transform—Li Luo, University of Duisburg-Essen - Duisburg, Germany; Guido H. Bruck, University of Duisburg-Essen - Duisburg, Germany; Peter Jung, University of Duisburg-Essen - Duisburg, Germany
This paper describes a novel onset detector for music signal based on the smoothly varying-Q transform, where the Q-factors vary following a linear function of the center frequencies. The smoothly varying-Q factors allow the time-frequency representation to coincide with the auditory critical-band scale. As the analysis basis of the input signal, the time-frequency image generated by smoothly varying-Q transform indicates the frequency evolution. On the detection stage, a bidirectional mismatch procedure is designed to estimate the discrepancies of frequency partials between currently processed frame and its bidirectional neighboring frames. An onset strength signal is obtained by measuring the mismatch error between the neighboring frames. The evaluation of the proposed algorithm is performed on a fully onset annotated music database and the results show that the proposed algorithm can achieve high detection accuracy and satisfied results.
Convention Paper 9349 (Purchase now)

P18-2 A Real-Time System for Measuring Sound Goodness in Instrumental Sounds—Oriol Romani Picas, Universitat Pompeu Fabra - Barcelona, Spain; Hector Parra Rodriguez, Universitat Pompeu Fabra - Barcelona, Spain; Dara Dabiri, Universitat Pompeu Fabra - Barcelona, Spain; Hiroshi Tokuda, KORG Inc. - Tokyo, Japan; Wataru Hariya, KORG Inc. - Tokyo, Japan; Koji Oishi, KORG Inc. - Tokyo, Japan; Xavier Serra, Universitat Pompeu Fabra - Barcelona, Spain
This paper presents a system that complements the tuner functionality by evaluating the sound quality of a music performer in real-time. It consists of a software tool that computes a score of how well single notes are played with respect to a collection of reference sounds. To develop such a tool we first record a collection of single notes played by professional performers. Then, the collection is annotated by music teachers in terms of the performance quality of each individual sample. From the recorded samples, several audio features are extracted and a machine learning method is used to find the features that best described performance quality according to musician's annotations. An evaluation is carried out to assess the correlation between systems’ predictions and musicians’ criteria. Results show that the system can reasonably predict musicians’ annotations of performance quality.
Convention Paper 9350 (Purchase now)

P18-3 Timbre Solfege: Development of Auditory Cues for the Identification of Spectral Characteristics of Sound—Teresa Rosciszewska, Fryderyk Chopin University of Music - Warsaw, Poland; Andrzej Miskiewicz, Fryderyk Chopin University of Music - Warsaw, Poland
This paper is concerned with listening exercises conducted during a technical ear training course called Timbre Solfege, taught to the students of sound engineering at the Fryderyk Chopin University of Music in Warsaw. Discussed are auditory cues used for identification of the characteristics of timbre produced by varying the sound frequency bandwidth and by boosting of selective frequency bands with the use of a spectrum equalizer. The students’ ability of identifying those modifications of the spectrum envelope has been assessed in a variety of progress tests. Results of the tests show that systematic training during the Timbre Solfege course considerably improves memory for timbre and develops the ability of associating the perceived characteristics of timbre with the spectral properties of sounds.
Convention Paper 9351 (Purchase now)

P18-4 Automatic Vocal Percussion Transcription Aimed at Mobile Music Production—Héctor A. Sánchez-Hevia, University of Alcala - Alcalá de Henares, Madrid, Spain; Cosme Llerena-Aguilar, Sr., University of Alcalá - Alcala de Henares (Madrid), Spain; Guillermo Ramos-Auñón, University of Alcalá - Alcalá de Henares, Madrid, Spain; Roberto Gil-Pita, University of Alcalá - Alcalá de Henares, Madrid, Spain
In this paper we present an automatic vocal percussion transcription system aimed to be an alternative to touchscreen input for drum and percussion programming. The objective of the system is to simplify the workflow of the user by letting him create percussive tracks made up of different samples triggered by his own voice without the need of any demanding skill by creating a system tailored to his specific needs. The system consists of four stages: event detection, feature extraction, and classification. We are employing small user-generated databases to adapt to particular vocalizations while avoiding overfitting and maintaining computational complexity as low as possible.
Convention Paper 9352 (Purchase now)

P18-5 Training-Based Semantic Descriptors Modeling for Violin Quality Sound Characterization—Massimiliano Zanoni, Politecnico di Milano - Milan, Italy; Francesco Setragno, Politecnico di Milano - Milan, Italy; Fabio Antonacci, Politecnico di Milano - Milan, Italy; Augusto Sarti, Politecnico di Milano - Milan, Italy; György Fazekas, Queen Mary University of London - London, UK; Mark B. Sandler, Queen Mary University of London - London, UK
Violin makers and musicians describe the timbral qualities of violins using semantic terms coming from natural language. In this study we use regression techniques of machine intelligence and audio features to model in a training-based fashion a set of high-level (semantic) descriptors for the automatic annotation of musical instruments. The most relevant semantic descriptors are collected through interviews to violin makers. These descriptors are then correlated with objective features extracted from a set of violins from the historical and contemporary collections of the Museo del Violino and of the International School of Luthiery both in Cremona. As sound description can vary throughout a performance, our approach also enables the modeling of time-varying (evolutive) semantic annotations.
Convention Paper 9353 (Purchase now)

P18-6 Audibility of Lossy Compressed Musical Instrument Tones—Agata Rogowska, Warsaw University of Technology - Warsaw, Poland
The aim of the conducted study was to evaluate differences in the audibility of different instruments by three commonly used lossy codecs. Seven instrument tones were compressed using MP3-LAME, Vorbis, and Opus to determine how the detection of compressed sounds varies with bit rate, instrument, and compression formats. Audibility of lossy compression was examined on six naïve subjects during 60 hours of listening. At the bit rate of 32 kbps the compressed signals were easily discriminable with significant differences between subjects. With magnifying the bit rate audibility decreased, the signal becoming inaudible at 64–96 kbps. Discrimination varied significantly from instrument to instrument.
Convention Paper 9232 (Purchase now)

 
 

P17-2 Demo

Sunday, May 10, 15:15 — 15:45 (Room: Saski)

 
 

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