AES Warsaw 2015
Engineering Brief Details
EB-1 - e-Brief Posters
Saturday, May 9, 16:00 — 18:00 (Foyer)
EB-1-1 Preferred Sound Level for Concert Listeners and Correlations between Sound Quality Dimensions—Avo-Rein Tereping, Tallinn University, Institute of Psychology - Tallinn, Estonia
Increasing loudness at concert performances is not caused by listeners’ preferences, but by the opinions of sound engineers and/or producers. The average sound level at public concerts ranges up to 100–105 decibels. Loudness preferences have been examined for listening with earphones but not in the open air. This e-brief describes research on preferred sound quality dimensions at Nordea Concert Hall in Tallinn with live music samples. The experiments revealed that preferred loudness don’t differ across age groups or between women and men. The preferred loudness were found 85–87 dB. Fidelity was the most important sound quality parameter influencing to overall pleasantness. No correlation were found between loudness and overall pleasantness.
Engineering Brief 183 (Download now)
EB-1-2 Evaluation of a Novel Approach to Virtual Bass Synthesis Strategy—Piotr Hoffmann, Gdansk University of Technology - Gdansk, Poland; Bozena Kostek, Gdansk University of Technology - Gdansk, Poland; Audio Acoustics Lab.
The aim of this paper is to present a novel approach to the Virtual Bass Synthesis (VBS) strategy applied to portable computers. The developed algorithms involve intelligent, rule-based settings of bass synthesis parameters with regard to music genre of an audio excerpt and the type of a portable device in use. The Smart VBS algorithm performs the synthesis based on a nonlinear device (NLD) with artificial controlling synthesis system according to music genre. Classification of musical genres is performed using the k-Nearest Neighbor algorithm and the extracted MPEG 7-based feature vectors optimized by the Principal Component Analysis method. To confirm the relationship between the presented excerpt of music from a variety of music genres and the listener’s preferences, subjective tests using the Mushra method are performed. On the basis of the listeners’ opinions statistical tests are carried out and show that listeners in most cases prefer the SVBS strategy developed by the authors in comparison to either an audio excerpt with the bass boost algorithm applied and unprocessed audio file. Furthermore, the listeners indicated that perception of the proposed SVBS strategy is similar for different types of portable devices.
Engineering Brief 184 (Download now)
EB-1-3 Analogue Hearts, Digital Minds? An Investigation into Perceptions of the Audio Quality of Vinyl—Michael Uwins, De Montfort University - Leicester, UK; University of Huddersfield - Huddersfield, UK
This study investigates the vinyl revival, with particular focus given to the listener’s perception of audio quality. A new album was produced using known source material. Subjects then participated in a series of double-blind listening tests, comparing vinyl to established digital formats. Subsequent usability tests required subjects not only to re-appraise the audio, but also to interact with the physical media and playback equipment. Digital vinyl systems were used in order to investigate the influence of non-auditory factors on their perception of sound quality. Both qualitative and quantitative data was also gathered from subjects of the usability tests, with the correlation (or contradiction) between the results being analyzed. The study concludes that sound quality is not the sole defining factor and that listener preferences are profoundly influenced by other, non-auditory attributes and that such factors are as much a part of the vinyl experience as the music etched into the grooves.
Engineering Brief 185 (Download now)
EB-1-4 Toward the Development of a Universal Listening Test Interface Generator in Max—Christopher Gribben, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK
This engineering brief describes HULTI-GEN (Huddersfield Universal Listening Test Interface Generator), a Cycling ‘74 Max-based tool. HULTI-GEN is a user-customizable environment that takes user-defined parameters (e.g., the number of trials, stimuli, and scale settings) and automatically constructs an interface for comparing auditory stimuli, while also randomizing the stimuli and trial order. To assist the user, templates based on ITU-R recommended methods have been included. As the recommended methods are often adjusted for different test requirements, HULTI-GEN also supports flexible editing of these presets. Furthermore, some existing techniques have been summarized within this brief, including their restrictions and how they might be altered through using HULTI-GEN. A finalized version of HULTI-GEN is to be made freely available online at: http://www.hud.ac.uk/research/researchcentres/mtprg/projects/apl/
Engineering Brief 187 (Download now)
EB-1-5 Multidimensional Ability Evaluation of Participants of Listening Tests: Part II—Tomasz Dziedzic, AGH University of Science and Technology - Krakow, Poland; Piotr Kleczkowski, AGH University of Science and Technology - Krakow, Poland
The problem of selecting appropriate participants for a listening test was addressed by the authors' e-brief presented during the 136th AES Convention. After the Convention, the project was continued and the application was further developed. The next stage consisted in the comparison of resolutions in the time and frequency domains for particular listeners. The resolution in time was investigated by a task for gap detection in noise. The resolution in frequency was tested by two tasks: a two-tone detection and distinguishing pitches of two tones. The results showed that there was no correlation between these resolutions (neither negative nor positive), but considerable frequency dependent differences were found. A new version of the test application was developed for this stage.
Engineering Brief 188 (Download now)
EB-1-6 Database of Single-Channel and Binaural Room Impulse Responses of a 64-Channel Loudspeaker Array—Vera Erbes, University of Rostock - Rostock, Germany; Matthias Geier, University of Rostock - Rostock, Germany; Stefan Weinzierl, Technical University of Berlin - Berlin, Germany; Audio Communication Group; Sascha Spors, University of Rostock - Rostock, Germany
A freely available database of measured single-channel and binaural room impulse responses (RIRs and BRIRs) of a 64-channel loudspeaker array of rectangular shape under varying room acoustical conditions is presented. The RIRs have been measured at three receiver positions for four different absorber configurations. Corresponding BRIRs for head-orientations in the range of ±80° in 2° steps with a KEMAR manikin have been captured for a subset of seven combinations of position and absorber configurations. The data is provided in the Spatially Oriented Format for Acoustics (SOFA). It can be used to study the influence of the listening room on multichannel audio reproduction. As an application RIRs for the synthesis of a sound field by Wave Field Synthesis are shown.
Engineering Brief 189 (Download now)
EB-1-7 HAART: A New Impulse Response Toolbox for Spatial Audio Research—Dale Johnson, The University of Huddersfield - Huddersfield, UK; Alex Harker, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK
This engineering brief describes a new, open source code library named HAART (Huddersfield Acoustical Analysis Research Toolbox). HAART simplifies the measurement and analysis of multi-channel impulse responses (IRs). For the purposes of this engineering brief the code library is compiled as a set of Max objects that form a prototype program in Max. This program is able to perform the acquisition, manipulation and analysis of IRs using subjective and objective measures described in acoustics literature. HAART is also able to convolve IRs with audio material and, most importantly, able to binaurally synthesize virtual, multichannel speaker arrays over headphones, negating the need for multichannel setups when out in the field. The code library is freely available from: http://www.hud.ac.uk/research/researchcentres/mtprg/projects/apl/
Engineering Brief 190 (Download now)
EB-2 - e-Brief Lectures
Sunday, May 10, 09:00 — 12:00 (Room: Królewski)
Chair:
Dylan Menzies, University of Southampton - Southampton, UK
EB-2-1 Comparing Room Acoustics for the Performance of Wagners‘ Lohengrin—Winfried Lachenmayr, Mueller-BBM - Munich, Germany; Gunter Engel, Mueller-BBM - Munich, Germany
Wagner is the example of a composer caring for the entire “production” chain of his works. A specific venue, the Festspielhaus in Bayreuth, was dedicated and built for a handful of his own compositions. But through their popularity Wagner’s operas are nowadays played in theaters and opera houses all over the world with different acoustic conditions. How does this affect the performance and musical perception and what can be observed in recordings or measurements? As an example, the acoustic recordings of performances of “Lohengrin” in (1) a typical smaller opera house in Germany and (2) in the Festspielhaus Bayreuth are compared and analyzed. Differences regarding running- and decay reverberation are addressed.
Engineering Brief 191 (Download now)
EB-2-2 Progress in Power Amplifiers: Thermal Distortion—Douglas Self, The Signal Transfer Company - London, UK
You sometimes read of "thermal distortion" in power amplifiers supposed to be due to cyclic changes in device parameters due to varying heating over that cycle. Until recently I was unimpressed by the likelihood of its existence, reasoning that it must cause rising distortion with falling frequency, and this was not observable in measurements to 0.001% THD. (Changes in distortion due to long-term thermal changes in the output stage quiescent conditions are another matter and clearly do exist.) Improvements in test gear allowing 0.0002% measurements and the use of higher powers discloses that THD can rise with falling frequency. The component concerned is identified, and methods given for both confirming the presence of thermal distortion and reducing it.
Engineering Brief 192 (Download now)
EB-2-3 Immersive Sound Design Using Particle Systems—Nuno Fonseca, ESTG/CIIC, Polytechnic Institute of Leiria - Leiria, Portugal
With the release of major immersive audio formats for cinema, including Auro-3D and Dolby Atmos, a new sound dimension is getting the attention of sound professionals—height. Although traditional panning techniques are still possible, more interesting approaches could be used to better explore space. Particles systems, which are widely used on computer graphics, could present themselves as a very interesting sound design approach. With immersive virtual microphones, capable of supporting many different setups, perfect sound and space coherence can be obtained. By controlling the particle system, instead of individual sound sources, a high number of sounds can be rendered in 3D. A particle system software was created, capable of running highly complex situations with up to several millions of sound sources, which is currently under testing by major Hollywood studios.
Engineering Brief 193 (Download now)
EB-2-4 Examining Influence of Distance to Microphone on Accuracy of Speech Recognition—Piotr Bratoszewski, Gdansk University of Technology - Gdansk, Poland; Marcin Szykulski, Gdansk University of Technology - Gdansk, Poland; Andrzej Czyzewski, Gdansk University of Technology - Gdansk, Poland
The problem of controlling a machine by the distant-talking speaker without a necessity of handheld or body-worn equipment usage is considered. A laboratory setup is introduced for examination of performance of the developed automatic speech recognition system fed by direct and by distant speech acquired by microphones placed at three different distances from the speaker (0.5 m to 1.5 m). For feature extraction from the voice signal the Mel- Frequency Cepstral Coefficients (MFCC) are used. The experiments are conducted employing the HTK engine (Hidden Markov Toolkit) for the Automatic Speech Recognition (ASR) task. The dictionary of 184 words was employed and WER (Word Error Rate), correctness and accuracy measures were calculated in order to verify and to compare obtained results of speech recognition.
Engineering Brief 194 (Download now)
EB-2-5 Investigating the Sound Quality Lexicon of Analogue Compression Using Category Analysis—Malachy Ronan, University of Limerick - Limerick, Ireland; Nicholas Ward, University of Limerick - Limerick, Ireland; Robert Sazdov, University of Limerick - Limerick, Ireland
This study investigates the lexicon used to describe analogue compression. Extant documents comprising 51 reviews of analogue compressors over 15 years are coded using category analysis. A total of 160 adjectives emerge and are further refined to nine inductive categories: transparency; frequency related; signal distortion; energy; transient shaping; glue; association; spatial dimensions; and character. A final abstraction reveals two primary attributes governing the perceived sound quality of analogue compression: “character” and “transient shaping.” Transparent dynamic range compression is found to be less important. This investigation clarifies the lexicon used to describe the sound quality attributes of analogue compression.
Engineering Brief 195 (Download now)
EB-2-6 Exploratory Microphone Techniques for Three-Dimensional Classical Music Recording—Will Howie, McGill University - Montreal, QC, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
At McGill University's Redpath Hall, a conventional stereo recording array was augmented with additional microphones in both the horizontal and vertical planes, yielding a 14-channel surround sound recording featuring seven discrete height channels. Based on existing multichannel recording models, microphone placement was designed to prioritize listener envelopment. Preliminary evaluations of the recordings by the authors and fellow researchers from the Graduate Program in Sound Recording at McGill University found that these 3D recordings have an increased sense of envelopment and realism as compared to traditional 5.1 surround sound. The authors have identified several areas to be further investigated through future recordings and listening tests.
Engineering Brief 196 (Download now)
EB-2-7 Automated (Microcontroller-Based) Impedance Tube—Alex Faron, University of Miami - Coral Gables, FL, USA; Connor McCullough, University of Miami - Coral Gables, FL, USA; Diego Ugaz, University of Miami - Miami, FL, USA
The acquisition of acoustic properties such as absorption and transmission-loss coefficients is necessary for the analysis and synthesis of acoustic materials. Because of its accuracy, the impedance tube is the preferred method of measurement. However, current implementations require a strong technical background in acoustics and considerable time to produce results. The system described in this work will use a microcontroller to automate the measurement process and expedite the delivery of information to a non-expert end-user.
Engineering Brief 197 (Download now)
EB-2-8 Analysis of DML Sound Reinforcement Systems Behavior in Large Concert Halls—Dragan Novkovic, College of Electrical Engineering and Computer Sciences - Belgrade, Serbia; Stefan Dimitrijevic, College of Electrical Engineering and Computer Sciences - Belgrade, Serbia; Faculty of Electrical Engineering, University of Belgrade
Sound reinforcement systems based on DML technology, as a sound sources with unique signal generation and radiation characteristics, provides some particular features when it comes to audience coverage and speech intelligibility. Although this technology exists for over a quarter of century, not so many sound reinforcement systems based on this technology exists and, therefore, it is possible to perceive a lack of data on the behavior of this type of system in real-life conditions. This paper has a goal to present the results of impulse response measurements conducted in big concert venue that was alternately excited with conventional and DML sound reinforcement systems.
Engineering Brief 198 (Download now)
EB-2-9 Production and Reproduction of Program Material for a Variety of Spatial Audio Formats—Jon Francombe, University of Surrey - Guildford, Surrey, UK; Tim Brookes, University of Surrey - Guildford, Surrey, UK; Russell Mason, University of Surrey - Guildford, Surrey, UK; Rupert Flindt, University of Surrey - Guildford, Surrey, UK; Philip Coleman, University of Surrey - Guildford, Surrey, UK; Qingju Liu, University of Surrey - Guildford, Surrey, UK; Philip Jackson, University of Surrey - Guildford, Surrey, UK
For subjective experimentation on 3D audio systems, suitable program material is needed. A large-scale recording session was performed in which four ensembles were recorded with a range of existing microphone techniques (aimed at mono, stereo, 5.0, 9.0, 22.0, ambisonic, and headphone reproduction) and a novel 48-channel circular microphone array. Further material was produced by remixing and augmenting pre-existing multichannel content. To mix and monitor the program items (that included classical, jazz, pop, and experimental music, and excerpts from a sports broadcast and a film soundtrack), a flexible 3D audio reproduction environment was created. Solutions to the following challenges were required: level calibration for different reproduction formats; bass management; and adaptable signal routing from different software and file formats.
Engineering Brief 199 (Download now)
EB-2-10 Low Frequency Performance of Circular Loudspeaker Arrays—Filippo Maria Fazi, University of Southampton - Southampton, Hampshire, UK; Mincheol Shin, ISVR, University of Southampton - Southampton, Hampshire, UK; Ferdinando Olivieri, University of Southampton - Southampton, Hampshire, UK; Simone Fontana, Huawei European Research Center - Munich, Germany
Compact loudspeaker arrays are widely used for the localized delivery of audio messages and beamforming applications. The optimal directivity performance of these devices is limited to a given frequency limit, whose upper bound is defined by the occurrence of spatial aliasing. The lower bound of this frequency range is caused by the limited capability of the array to generate a directional sound beam when the wavelength of the sound to be reproduced is large in comparison to the size of the array. In this work a theoretical and experimental study is presented of the directivity limitations of circular loudspeaker arrays at low frequencies. The frequency at which the array directivity pattern starts to diverge from the desired one is calculated analytically and put into relation with the dynamic range of the transducers and with the regularization scheme used when designing the beamforming digital filters.
Engineering Brief 200 (Download now)
EB-2-11 An Exploratory Evaluation of User Interfaces for 3D Audio Mixing—Steven Gelineck, Aalborg University Copenhagen - Copenhagen, Denmark; Dannie Korsgaard, Aalborg University - Copenhagen, Denmark
The paper presents an exploratory evaluation comparing different versions of a mid-air gesture based interface for mixing 3D audio exploring: (1) how such an interface generally compares to a more traditional physical interface, (2) methods for grabbing/releasing audio channels in mid-air, and (3) representation of sources in separate 3D views vs. in one shared 3D view. Results suggest that while the traditional physical interface is generally intuitive and easy to use, the 3D gesture interface provides an improved understanding of the 3D space and provides a better control of especially moving sources. The shared view provides a better overview and workflow than the separated view and grabbing sounds using hand-gestures causes difficulties.
Engineering Brief 201 (Download now)
EB-2-12 Differences between Recorded and Emulated Guitar Sounds—Maciej Majewski, Akademia Górniczo-Hutnicza - Krakow, Poland; Pawel Malecki, AGH University of Science and Technology - Krakow, Poland
Is it possible to create an emulated guitar sound similar to the recorded one? Why not! First of all, the direct signal from the guitar is prepared. After that, using a “reamping” technique the desired sound is recorded. Subsequently the whole audio track is emulated using device called “Kemper.” Then the listening tests among people who work in the music industry were performed for subjective comparison of prepared sounds. The comparison of numerical audio parameters is provided using Matlab scripts. The results analysis show the performance of modern emulation techniques in compare to the traditional multitrack recording. Major benefits and losses are discussed.
Engineering Brief 186 (Download now)
