Saturday, May 20, 15:00 — 18:00 (Gallery Window Area 2)
EB01-01 Source Separation in Action: Demixing the Beatles at the Hollywood Bowl
James Clarke (Presenting Author)
Except for unamplified dramatic performances or concerts where the audience is either used to sitting quietly or can be persuaded to listen quietly, high levels of amplifications are required so that the speech and music can still be heard above the self-noise generated by the crowd. Crowd noise can reach surprisingly high sound levels when several thousand people are shouting at each other, all at the same time. In this paper a method is presented that was used to isolate the crowd noise non destructively, so the raw instrumentation can be targeted in isolation. These isolated sources now become available for re-mixing and balancing.
Engineering Brief 307
EB01-02 Feature Selection for Real-Time Acoustic Drone Detection Using Genetic Algorithms
Marta Bautista-Durán (Presenting Author), Joaquin García-Gomez (Author), Roberto Gil-Pita (Author), Manuel Rosa-Zurera (Author)
Drones are taking off in a big way, but people sometimes use them in order to invade the privacy of others or to bypass the security systems, making their detection an actual issue. The objective of the proposed system is to design real-time acoustic drone detectors, able to distinguish them from objects that can be acoustically similar. A set of features related to the propeller sounds have been extracted, and genetic algorithms have been used to select the best subset. The classification error achieved with 30 features is below 13%, making feasible the real-time implementation of the proposed system.
Engineering Brief 308
Sunday, May 21, 09:30 — 12:30 (Gallery Window)
EB02-01 AKtools—An Open Software Toolbox for Signal Acquisition, Processing, and Inspection in Acoustics
Fabian Brinkmann (Presenting Author), Stefan Weinzierl (Author)
The acquisition, processing, and inspection of audio data plays a central role in the everyday practice of acousticians. However, these steps are commonly distributed among different and often closed software packages making it difficult to document this work. AKtools includes Matlab methods for audio playback and recording, as well as a versatile plotting tool for inspection of single/multichannel data acquired on spherical, and arbitrary spatial sampling grids. Functional blocks cover test signal generation (e.g., pulses, noise, and sweeps), spectral deconvolution, transfer function inversion using frequency dependent regularization, spherical harmonics transform and interpolation among others. Well documented demo scripts show the exemplary use of the main parts, with more detailed information in the description of each method. To foster reproducible research, AKtools is available under the open software European Union Public Licence (EUPL) allowing everyone to use, change, and redistribute it for any purpose: www.ak.tu-berlin.de/aktools.
Engineering Brief 309
EB02-02 Disagreement between STI and STIPA Measurements Due to High Level, Discrete Reflections
Ross Hammond (Presenting Author), Adam Hill (Author), Peter Mapp (Author)
Objective measures of intelligibility, speech transmission index (STI), and speech transmission index for public address systems (STIPA) often form the basis for sound system verification. The reported work challenges the accuracy of both measures when encountering high level, discrete reflections. Tests were carried out in an anechoic environment with artificial reflections added between 0 and 500 ms. Discrepancies were found to occur above 80 ms due to synchronization between modulation frequencies and reflection arrival times. Differences between STI and STIPA of up to 0.1 were found to occur for the same delay condition. Results suggest STIPA should be avoided in acoustic environments where high level, discrete reflections occur after 80 ms and STI should only be used alongside other verification methods.
Engineering Brief 310
EB02-03 Soundscape Recording: Review of Approaches
Katarzyna Sochaczewska (Presenting Author), Dorota Czopek (Author), Pawel Malecki (Author), Jerzy Wiciak (Author)
In the soundscape analysis the collected data is crucial for possible relationships between the results of measurements of acoustic, psychoacoustic research results, and characteristics of the respondents. Such analysis shall verify that the physical characteristics of sound affect the subjective assessment. The article shows a review of commonly used approaches in soundscape recording both for analysis and archive purposes. Discussion on recording from one or several spots in the middle of the sound sources versus moving with the microphone towards or inside the acoustic environment is provided. Also, special attention is paid on traditional microphone techniques in sound engineering, binaural recordings, and sound field synthesis with spherical harmonics.
Engineering Brief 311
EB02-04 Ambience Recording for 3D Audio
Marco Hanelt (Presenting Author), Andreas Ehret (Author)
3D audio is emerging as a production format to enable immersive consumer audio experiences, including the sensation of height. This eBrief focuses on the use of 3D microphone arrays for recording 3D ambiences. Multiple microphone arrays were evaluated, both in theory as well as in practice. The arrays were assessed for their perceptual performance such as spatial envelopment, location accuracy, and timbre. Furthermore, the practical usability of these recordings in a real world movie project and the handling in a common post-production environment have been tested. With the information gathered, a Best Practice guide for different use cases has been developed.
Engineering Brief 312
EB02-05 Do Microphone Angles Result in Audible Differences When Recording a Guitar Amplifier?
Ellen Culloo (Presenting Author), Malachy Ronan (Author)
Objective measurements using a sinusoidal sweep show that microphone angle has little effect on the frequency response of a guitar amplifier recording . However, anecdotal evidence suggests that alterations to the microphone angle hold merit when recording ecologically valid sound sources. An ABX listening experiment was conducted with 20 participants to investigate whether microphone angles of 0, 30, and 60 degrees were audibly different to this cohort. Both dynamic and ribbon microphones were used and the loudness normalized guitar recordings were presented in solo and within a music mix. The experimental results suggest that microphone angles did not generate any perceivable changes to this cohort on this program material.
Engineering Brief 313
EB02-06 The Mixing Glove and Leap Motion Controller: Exploratory Research and Development of Gesture Controllers for Audio Mixing
Jack Kelly (Presenting Author), Diego Quiroz (Author)
Digital musical instruments (DMIs) have been evolving over the past several decades, and much research has been done on the subject of capturing the gestures of performers in an effort to re-map them to digital instruments. The practice of audio mixing is no different from a musical performance in this respect. The gestures used by engineers are expressive and have complex metaphorical significance. In this paper two approaches to gesture-based mixing tools for audio engineers, the Mixing Glove and the Leap Motion Controller are explored. Both systems are designed to control volume, panning, solo/mute, and reverb, using hand gestures alone.
Engineering Brief 314
EB02-07 Discussion on Subjective Characteristics of High Resolution Audio
Mitsunori Mizumachi (Presenting Author), Katsuyuki Niyada (Author), Ryuta Yamamoto (Author)
High resolution audio gains in popularity on both audio production and consumer markets. It is necessary to characterize the advantage of high resolution audio over legacy audio formats. The authors have already reported perceptual discrimination rates for high resolution audio, that is, 192 kHz/24 bits PCM format, against a 48 kHz/16 bits format and its compressed versions under a listening room and in-car environments, respectively. In this paper we also focus on perceptual discrimination concerning bit depth in between 24 and 16 bits. Participants were asked to judge either the same or not for each paired stimulus of 48 kHz/24 bits and 48 kHz/16 bits formats. The discrimination rates depend on the reproduction environments, although those subjects could discriminate the difference in between 192 kHz/24 bits and 48 kHz/16 bits formats. It is supposed that high resolution audio benefits more from the wide frequency range than from the bit depth.
Engineering Brief 315
EB02-08 Accurate Extraction of Dominant Reflections from Measured Sound Intensity Responses in a Room
Masataka Nakahara (Presenting Author), Yasuhiko Nagatomo (Author), Akira Omoto (Author)
The intensity responses in three orthogonal directions, which are calculated from measured impulse responses, include information of dominant reflections in 4-pi space. This information is important for restoration of a sound field, e.g., a spatial reverberator. A method of extracting significant reflections from instantaneous intensities or their envelopes is, therefore, the key application element. To improve the accuracy of detecting reflections, the authors have introduced a new strategy, named “speed detection,” which calculates the moving speed of instantaneous intensities at every sampled time. If the speed is lower than an assumed threshold, these intensities indicate reflections. On the contrary, faster speeds indicate the residual transient components of intensities. This “speed detection” is verified with the measured data of several experiments.
Engineering Brief 316
EB02-09 Investigation of Interchangeability of Audio Objects' Spatial Sound Direction between 3D Audio Rendering Systems and Rooms by VSV (Virtual Source Visualizer)
Takashi Mikami (Presenting Author), Masataka Nakahara (Author), Akira Omoto (Author)
Recently, some kinds of 3D sound rendering systems (such as Dolby Atmos, DTS:X, 22.2ch. etc.) are proposed and commercialized in audio industries. Similarity / difference in 3D sound localizations were examined by using sound intensity. Sound intensities are measured on different rendering systems in the same room, and also on the same rendering systems in different rooms. Subjective study to evaluate 3D sound rendering systems requires much time and labor. Evaluation by sound intensities, measured physically, is very useful. The session discusses interchangeability / difference of sound direction between rendering systems and between rooms obtained by analyzing the visual images and numerical data of sound localization.
Engineering Brief 317
EB02-10 Loudness Management in the Blu-ray Disc Ecosystem in the Context of Today’s Playback Environments
Andreas Ehret (Presenting Author), Sripal Mehta (Author), Mike Ward (Author)
Loudness management within the Blu-ray Disc ecosystem has historically been less of a priority than in other media playback ecosystems. Instead, the industry has focused on delivering the highest fidelity and full dynamic range audio. As a result, the measured loudness of the content on Blu-ray Disc is generally not accurately indicated in the audio bitstreams carried on Blu-ray discs. However, as more use-cases emerge to connect Blu-ray Disc players to playback environments with limited dynamic range reproduction capabilities (such as TVs or Sound bars), loudness management is becoming more important to ensure optimal playback for these new device types. This brief explains the value of loudness management in the Blu-ray Disc ecosystem to address new playback environments and gives example workflows for correctly setting loudness values in audio bitstreams delivered on Blu-ray Disc.
Engineering Brief 318
Sunday, May 21, 14:00 — 14:45 (Salon 2+3 Rome)
Engineering Brief 319
EB03-02 High Frequency—Ultra Audio Band Mode Voice Coil Temperature Measurement
Isao G. Anazawa (Presenting Author)
As the power driving mobile devices loudspeaker increases for a better audio experience, an accurate measurement of the voice coil temperature becomes necessary in order to protect the loudspeaker from over-heating. Electrical solutions have been developed in the past to measure the temperature indirectly from the voice coil resistance using a low frequency probe tone or using main audio contents. This paper explains an ultra-audio band high frequency probe to measure the resistance. The test results show good accuracy without the known side effects that exist with current methods.
Engineering Brief 320
EB03-03 U 87—Microphone Development in the 1960s
Martin Schneider (Presenting Author)
The U 87 was introduced in 1967 as part of the first generation of transistorized condenser microphones. In the 1960s many aspects like dynamic range, powering, and the seemingly simple question of connectors had to be reconsidered differently from the preceding generations of tube microphones, and for the new recording environments of the time. A detailed look at this microphone and its many variants (for different countries, and different powering schemes) give an insight into the spectrum of development topics and recording technology of the 1960s and 1970s.
Engineering Brief 321
Sunday, May 21, 14:30 — 15:00 (Salon 1 Moscow)
Christoph Pörschmann (Chair)
EB04-01 A Spherical Near-Field HRTF Set for Auralization and Psychoacoustic Research
Christoph Pörschmann (Presenting Author), Johannes M. Arend (Author), Annika Neidhardt (Author)
Head-related transfer functions (HRTFs) describe the directional filtering caused by the head, pinna, and torso and are an essential component of binaural synthesis systems. Currently most of these systems are based on far-field HRTFs and thus do not consider acoustical specifics of nearby sound sources. One reason might be that full spherical near-field HRTF sets are rarely available. In this paper we present an HRTF set of a Neumann KU100 dummy head and a technical evaluation of the set. The set is freely available for download and contains post-processed impulse responses, captured on a circular and full spherical grid at distances between 0.25 m and 1.50 m. It can be used for psychoacoustic research and for applications where nearby virtual sound sources shall be auralized.
Engineering Brief 322
EB04-02 Binaural Recording System and Sound Map of Malaga
Carmen Rosas (Presenting Author), Salvador Luna-Ramirez (Author)
This Engineering Brief describes part of the results obtained in the Master Thesis ‘”Binaural Recording System and Sound Map” for the Masters in Acoustic Engineering at the University of Malaga. The aim of this project is the construction and the characterization of a pair of in-ear binaural microphones with high-quality capsules. A set of HRTF measurements was obtained and applied to different audio signals for the realization of a psychoacoustic experiment to assess the spatiality provided by the system. For the system assessment, another set of audio samples was generated from the MIT’s HRTFs, and both results have been compared. Additionally, different soundscapes have been recorded with the binaural system, and a binaural sound map of Malaga has been developed, which aims to create an archive to collect and conserve the most distinctive sounds of the city using an immersive technology.
Engineering Brief 323
Sunday, May 21, 15:00 — 18:00 (Gallery Window)
EB05-01 Practical Method to Evaluate Noise Generation Systems
Roksana Kostyk (Presenting Author), Przemyslaw Maziewski (Author), Dominik Stanczak (Author)
The paper presents the method used to test the accuracy of a fully calibrated noise generation system. The method is based on a frequency response comparison of binaural recordings done in real and simulated environments. The paper will give examples coming from two different audio laboratories. It will also illustrate the influence of the head and torso unit’s miss-position on the noise reproduction accuracy.
Engineering Brief 324
EB05-02 Free Database of Low Frequency Corrected Head-Related Transfer Functions and Headphone Compensation Filters
Vera Erbes (Presenting Author), Matthias Geier (Author), Sascha Spors (Author), Hagen Wierstorf (Author)
A database of publicly available head-related transfer functions (HRTFs) of a KEMAR manikin together with headphone compensation filters for various headphone types is presented. The HRTFs are based on previously published data from Wierstorf et al. (2011) that have additionally been corrected for low frequencies. This compensates for missing information due to low excitation energy in this frequency range during the measurement and allows for shorter impulse responses. A further benefit is demonstrated by the interpolation of HRTFs via magnitude and phase that is only possible with consistent phase information. Both the low-frequency correction as well as the generation of the headphone compensation filters are accompanied by Matlab code to document the processing.
Engineering Brief 325
EB05-03 Dataset of In-the-Ear and Behind-the-Ear Binaural Room Impulse Responses Used for Spatial Listening with Hearing Implants
Stephan Werner (Presenting Author), Anja Chilian (Author), Maria Gadyuchko (Author), Florian Klein (Author)
The contribution presents a dataset of binaural room impulse responses (BRIRs) using a KEMAR head and torso simulator. Sixteen positions around the head are recorded in three rooms with differing room acoustics. The rooms represent a standardized listening lab, a room for rehabilitation of hearing diseases, and a large sized room. Additionally to the in-the-ear recordings, behind-the-ear BRIRs are recorded to simulate the microphone positions of hearing aid devices. The dataset is used in a research project to develop innovative methods and technologies for spatial listening and speech intelligibility using cochlear implants and bone conduction hearing aids. The dataset enables binaural resynthesis of different directions and rooms for research and rehabilitation.
Engineering Brief 326
EB05-04 Virtual Source Width in Binaural Synthesis with Frequency-Dependent Directions
Hengwei Su (Presenting Author), Toru Kamekawa (Author), Atsushi Marui (Author)
To control the perceived source width rendering by headphone, a method to distribute different frequency bands of a sound source to different directions by binaural synthesis was investigated. Three types of signals including two anechoic musical recordings and white noise were filtered and split into 1/3 octave bands, and each band was convolved with HRTFs from different directions within the intended source width range. Subjective listening tests were conducted to evaluate the performance of this process. There are no evident results showing that this method can successfully synthesis extended sound images. However, it suggests that the distribution of bands according to spectral characteristics of signals is necessary to synthesize sound image without displacement of localization.
Engineering Brief 327
EB05-05 Testing Babble Noise Reduction Performance of Headset Microphones
Antti Kelloniemi (Presenting Author), Ergo Esken (Author)
Babble noise is a typical and specific problem in open offices and call centers, which is why workers in these environments use headsets. Babble noise causes disturbance in these spaces, and it easily leaks through typical send noise suppression processing to far end in telecommunication. To improve the send direction signal-to-noise ratio, headsets are equipped with microphone booms with acoustic noise cancelling microphones or microphone arrays. A method to evaluate their efficiency in reducing babble noise is described in this paper.
Engineering Brief 328
EB05-06 Binaural Spatialization Methods for Indoor Navigation
Sylvain Ferrand (Presenting Author), François Alouges (Author), Matthieu Aussal (Author)
The visually impaired people are able to follow sound sources with a remarkable accuracy. They often use this ability to follow a guide in everyday activities or for practicing sports, like running or cycling. On the same principle, it is possible to guide people with spatialized sound. We have thus developed a navigation device to guide with sounds using binaural synthesis techniques. In this device we are using both localization information provided by a precise and low latency positioning system and heading data computed from an Inertial Measurement Unit. These positioning data are feeding an HRTF based binaural engine, producing spatialized sound in real-time and guiding the user along a way. The user follows the sound, quite naturally and without initial training. Experiments show that it is possible to guide a walker with enough precision.
Engineering Brief 329
EB05-07 The Two!Ears Database
Fiete Winter (Presenting Author), Alexander Raake (Author), Sascha Spors (Author), Hagen Wierstorf (Author)
TWO !EARS was an EU-funded project for binaural auditory modelling with ten international partners involved. Its main goal was to provide a computational framework for the modelling of active exploratory listening that assigns meaning to auditory scenes. As one outcome of the project, a database including data acquired by the involved partners as well as third-party measurements has been published. Among others, a large collection of Head Related Impulse Responses and Binaural Room Impulse Responses is part of the database. Further, results from psychoacoustic experiments conducted within TWO !EARS to validate the developed auditory model were added. For the usage of the database together with the TWO !EARS model, a software interface was developed to download the data from the database on demand.
Engineering Brief 330
EB05-08 Personalized HRTF Measurement and 3D Audio Rendering for AR/VR Headsets
Woon-Seng Gan (Presenting Author), Nitesh Kumar Chaudhary (Author), Nguyen Duy Hai (Author), JianJun He (Author), Santi Peksi (Author), Rishabh Ranjan (Author)
This e-Brief describes our recent work in acquiring a fast, personalized head related transfer function (HRTF) and a personalized 3D audio rendering headsets for augmented and virtual reality (AVR) headsets. Binaural signal acquisition and rendering are important tasks in capturing the idiosyncratic acoustics of the pinnae, head and torso, and playback via headphones to the left and right ears. We will highlight a personalized HRTF binaural acquisition cum 3D audio headphone playback system that can take advantage of our individual ear-head anthropometry information in 3D sound acquisition and rendering.
Engineering Brief 331
EB05-09 Effect of a Known Environment on the Estimation of Sound Source Distance
Shashank Aswathanarayana (Presenting Author)
The estimation of sound source distance has been a topic of research interest for a number of decades now. Humans are known to be good at localizing sound in the azimuth and elevation but are poor at estimating the sound source distance. This project looks at examining the effect of a known environment on the estimation of sound source distance. The project aims at initially testing the subjects perception of sound source in an unknown environment and then examining the effect of training the subject to the environment to see if training/learning the acoustics of the environment improves the estimation of the source distance.
Engineering Brief 332
EB05-10 The Effects of Decreasing the Magnitude of Elevation-Dependent Notches in HRTFs on Median Plane Localization
Jade Raine Clarke (Presenting Author), Hyunkook Lee (Author)
A binaural experiment was conducted to investigate whether a necessary magnitude of pinna related spectral notches in HRTFs exist. Individual HRIRs were measured at 0°, 30°, and 60° in the median plane for three subjects. The original HRTFs were manipulated so that dominant spectral notches between 5 and 10 kHz were filled in two different degrees. Localization tests were carried out with each subject judging each stimulus condition 15 times in a randomized order. It was found that for the 30° and 60° sources, two subjects tended to perceive the image to move upwards as pinna related notches were reduced. For 0°, however, an increase in front-back confusion occurred as a result of notch magnitude manipulation.
Engineering Brief 333
EB05-11 An Impulse Response Dataset for Dynamic Data-Based Auralization of Advanced Sound Systems
Chris Pike (Presenting Author), Michael Romanov (Author)
This engineering brief presents a freely-available binaural room impulse response (BRIR) dataset measured on a multichannel loudspeaker system. The 32-loudspeaker array includes all loudspeaker layouts specified in Recommendation ITU-R BS.2051. Measurements were carried out in an ITU-R BS.1116-compliant listening room using a Neumann KU100 dummy head microphone. BRIRs were measured at 2° steps of rotation of the dummy head. The dataset can be used for dynamic data-based auralization of multichannel loudspeaker signals, such as those generated by the so-called advanced sound systems described in ITU-R BS.2051, i.e., systems that can render surround sound with height signals from channel-based, object-based, and/or scene-based content representations. The dataset is made freely-available in the SOFA file format.
Engineering Brief 334
EB05-12 Equipment for Fast Measurement of Head-Related Transfer Functions
Jose J. Lopez (Presenting Author), Pablo Gutierrez-Parera (Author)
Binaural audio can become the future of spatial sound systems as more and more music is consumed on mobile devices through headphones. For a better experience, the binaural sound must be individualized for each subject through the use of their personal Head-Related Transfer Function (HRTF). The most straightforward way of personalization is to measure in-situ the HRTF. However, installations and set-ups for that purpose require anechoic chambers and complex motorized positioning systems. In this brief, we present an installation deployed in a non-anechoic room with multiple loudspeakers that provide a way of measuring the HRTF with an excellent resolution in the azimuthal plane and a sufficient resolution on elevation for common purposes.
Engineering Brief 335
Monday, May 22, 13:30 — 14:45 (Salon 2+3 Rome)
Alfred Svobodnik (Chair)
EB06-01 The DFA Fader: Exploring the Power of Suggestion in Loudness Judgments
Jack Haigh (Presenting Author), Malachy Ronan (Author)
Anecdotal evidence suggests that when performers request loudness increases in their on-stage monitoring device, feedback regarding task completion is sometimes sufficient for the performer to perceive a loudness change. This is colloquially known as a DFA fader. Given the dearth of empirical evidence, qualitative interviews were conducted with live sound engineers to investigate the type of feedback required to successfully deliver a suggestion of a loudness change. Following this, 22 participants completed a paired comparison listening experiment to determine whether verbal suggestions produce perceived loudness changes. The experimental results demonstrate a significant difference between participants receiving a verbal suggestion and those that did not in 12 out of 20 presentations. These results support the use of verbal suggestions to convey loudness increases in live sound contexts.
Engineering Brief 336
EB06-02 Quantization Noise of Warped and Parallel Filters Using Floating Point Arithmetic
Balázs Bank (Presenting Author), Kristóf Horváth (Author)
For audio filter and equalizer design it is desirable to take into account the frequency resolution of hearing. Therefore, various specialized filter design methodologies have been developed, from which warped and parallel filters are particularly appealing options due to their simple design and good approximation properties. This paper compares the quantization noise of two different warped IIR implementations with that of fixed-pole parallel filters in single-precision floating point arithmetic. It is shown by simulations that the parallel filter provides the best compromise between quantization noise and computational complexity, since it significantly outperforms the series second-order warped IIR implementation in terms of noise performance, while requires less computational resources compared to the original warped IIR structure.
Engineering Brief 337
EB06-03 Warped Implementation of Parallel Second-Order Filters with Optimized Quantization Noise Performance
Balázs Bank (Presenting Author), Kristóf Horváth (Author)
Fixed-pole second-order parallel filters provide an efficient way of implementing IIR filters with a logarithmic frequency resolution. However, the fine frequency resolution needed at low frequencies can only be achieved by poles near the unit circle. This may lead to large roundoff noise at low frequencies when the filters are implemented using bit-depths of 24 bits or lower in fixed-point arithmetic. This paper investigates the performance improvement when the parallel second-order sections are implemented as warped IIR filters. In addition, an analytical expression is given for computing the warping parameter as a function of the pole location of the original second-order section so that the quantization noise power is minimized.
Engineering Brief 338
EB06-04 Power Out of Thin Air: The Harvesting of Acoustic Energy
Charalampos Papadokos (Presenting Author), John Mourjopoulos (Author)
Recent evolution in Acoustic Energy Harvesting (AEH) indicate that beyond its communication function, sound can be a potential energy resource for powering contemporary and future applications operating in the range nW - mW. Acoustic energy can be either ambient or produced via speech and music reproduction, portable and mobile devices, jet and automobile engines, means of transport, electroacoustic transducers, etc. This work provides a short review of relevant studies in the art and focuses on AEH inside closed-box loudspeaker enclosures.
Engineering Brief 339
EB06-05 Fully Digital Development of Automotive Audio Systems
Alfred Svobodnik (Presenting Author), Christof Faller (Author), Marc Levasseur (Author)
This paper describes the building blocks of a fully digital development environment for automotive audio systems. The whole development process, including all major engineering disciplines, has been virtualized—up to the realistic audibility of the sound systems by means of auralizations. All building blocks are based on simulations, and thus fully digital prototypes can be used already in the early concept phase. Hence, product quality, i.e., reproduced sound performance, can be assessed, and improved, long before any hardware exists.
Engineering Brief 340
Monday, May 22, 15:00 — 18:00 (Gallery Window Area 2)
EB07-01 Establishing the Performance of a DIY Tapped Horn Subwoofer
Andy Wardle (Presenting Author)
A DIY Tapped Horn subwoofer was constructed and driven using modestly priced hardware. The position of the acoustic center, frequency and polar responses were established under pseudo-free field conditions. Polar responses and results were compared to a commercially available device. The DIY product performed comparatively poorly with respect to amplitude and frequency response but displayed comparable polar response. Very low frequencies (<80 Hz) displayed no directionality, with effective pattern control beginning at 100 Hz in line with the cutoff frequency predicted by its mouth area. It was established that if processed appropriately, multiple units would provide additional SPL and lower the frequency of pattern control providing a viable alternative to more expensive products for small to medium scale outdoor events.
Engineering Brief 341
EB07-02 The Influence of the Passive Electronic Components Quality on the Electroacoustic Parameters of the Audio Devices
Maciej Sabiniok (Presenting Author)
A large group of young and inexperienced electronics engineers interested in building audio devices asked how the quality of the passive components such as resistors and capacitors affected on the electroacoustic parameters of designed circuits. This group also includes students who are the members of the Polish Student Section of the Audio Engineering Society at Wroclaw University of Science and Technology willing to work with audio electronics and obtain the best possible quality of the constructed equipment. The aim of this paper is to investigate the impact of the varying passive components quality into the audio circuits performance. The results will allow students to know the limitations related to the choice of passive components.
Engineering Brief 342
EB07-03 Design of a Digitally Controlled Graphic Equalizer
Marcelo Herrera Martinez (Presenting Author), Carlos Mauricio Betancur Vargas (Author), Jonnathan Montenegro Niño (Author), Dario Alfonso Páez Soto (Author), Vladimir Trujillo Olaya (Author)
This article deals with the design of a digitally audio controller for use in general applications. The goal is to create a 10-band graphic equalizer of which the signal gain or attenuation in every octave band is controllable by a smartphone /tablet application. The application provides a user interface to enhance perceptive audio quality intuitively. Making the equalizer digitally controllable by an app eliminates the necessity of manually adjusting the equalizer faders, thus the need of the presence of a musician/engineer at the location of the equalizer is removed. Preset configurations are easily activated in the equalizer hardware with only one touch within the app. Further testing and optimization efforts are required for the validation of the system.
Engineering Brief 343
EB07-04 Design of an Algorithm for VST Audio Mixing Based on Gibson Diagrams
Marcelo Herrera Martinez (Presenting Author), Belman Jahir Rodriguez Nino (Author)
This project consists on the creation of a plugin on the Ableton Live platform, with the aim of providing visually the audio mixing process in real-time. The software programming is developed on Max for Live–a program to establish the link between Max Msp and Ableton Live. The plugin is assigned for each channel with the aim of visualizing the correspondent sound to a “sphere“ object on a 3D window and there to observe the variations in real time of loudness, panning, and frequency analysis based on David Gibson´s interpretation in his book The Art of Mixing.
Engineering Brief 344
Monday, May 22, 17:00 — 18:00 (Salon 1 Moscow)
Juha Backman (Chair)
EB08-01 Motion-to-Sound Latency Measurement Procedure for VR Sound Reproduction
Jorgos Estrella (Presenting Author), Jan Plogsties (Author)
In the last couple of years, virtual auditory displays have finally reached the consumer market as part of emerging VR technologies. One of the challenges VR technology providers have to face is to reach affordable low motion-to-sound latency. Low latency is a very important factor while aiming towards immersive spatial sound reproduction. In this e-Brief a motion-to-sound latency measurement approach is proposed. This method employs a simplified parallel system running externally as reference. Here, a second head-orientation sensor is used to modulate a signal generator. Correlation analysis between the generated signal and the output signal of the device under test are used to assess latency.
Engineering Brief 345
EB08-02 Flexible Python Tool for Dynamic Binaural Synthesis Applications
Annika Neidhardt (Presenting Author), Thomas Köllmer (Author), Florian Klein (Author), Niklas Knoop (Author)
In this report, we present an open source tool for real-time dynamic binaural synthesis implemented in Python on top of PyAudio. The core is an efficient implementation of the uniformly partitioned convolution with the overlap-save approach. The dynamic and interactive reproduction of spatial audio scenes has become a common requirement in science and industry. Use cases are various, reaching from listening tests considering head rotation to complex reproduction scenarios for augmented or virtual reality, e.g., in combination with head mounted displays. With Python as a flexible and easy to learn programming language, PyBinSim offers great value in research and teaching of binaural synthesis. Source code, examples and documentation are available online.
Engineering Brief 346
EB08-03 A Self-Calibrating Earphone
Juha Backman (Presenting Author), Tom Campbell (Author), Marko Hiipakka (Author), Jari Kleimola (Author)
A self-calibrating system estimates the acoustical transfer function from sound pressures at the entrance of the ear canal to sound pressures at the eardrum: An earphone plays a broadband sound into the auditory meatus and an in-ear microphone then receives the sound at the entrance of the ear canal. Parenthetically, assessing calibration, results showed that spectral analysis of recordings of this signal is replicable to within 3 dB from 0.5 to 22 kHz for each given ear. A digital signal processing unit calculates an individualized filter from that signal. The calculated filter neutralizes the transfer function via software, which controls the digital signal processing unit’s output into the earphones whilst playing media.
Engineering Brief 347
EB08-04 End-To-End Process for HRTF Personalization
Tomi Huttunen (Presenting Author), Antti Vanne (Author)
The personalization of the head-related transfer functions (HRTFs) improves externalization and spatialization in headphone listening. The accurate measurement of an individual HRTF is time-consuming and complicated that has led to increased interest towards simulation based HRTF acquisition. The main challenge for simulations has been the lack of the fast and simple method to generate the three-dimensional (3D) geometry of the head and pinnae. On the other hand, a numerical solution of the 3D wave equation that characterizes the HRTF has been considered computationally demanding. We introduce an end-to-end process from the acquisition of the geometry to use of the personalized HRTFs in several applications. Results from the preliminary listening tests and future improvements are also discussed.
Engineering Brief 348
Tuesday, May 23, 14:30 — 15:45 (Salon 2+3 Rome)
Scott G. Norcross (Chair)
EB09-01 Object-Based Audio in Large Scale Live Sound Reinforcement Controlled by Motion Tracking
Jakob Bergner (Presenting Author), Mario Seideneck (Author), Christoph Sladeczek (Author)
This work shows a detailed application of an optical tracking system to control the positioning of sound sources in an object-based audio reproduction system for live sound reinforcement. This need is brought up by live performances with moving actors like operas, musicals or spoken theater. With state-of-the-art object-based audio reproduction systems it is possible to distribute virtual sound sources for improved sound localization within the audience area. To cope with applications of high complexity automated auxiliary systems like motion tracking provide valuable control data and thus enhance the usability of such systems. The presented approach shows a solution with focus on interfaces between systems and devices.
Engineering Brief 349
EB09-02 A Basic Study of the Upmix Method for 22.2 Multichannel Sound
Toru Kamekawa (Presenting Author), Atsushi Marui (Author)
The upmix technique to 22.2 multichannel sound from 9 channel, 5 channel, and 3 channel IRs (impulse responses) were studied. The two upmix techniques were used. One is the IRs made from the original IRs converted to phase randomized signals with the same time envelope to original signal and the other is the IRs obtained by simply adding signals of adjacent channels. The experimental stimuli were obtained from these impulse responses convolved to the sound of a xylophone and a female voice recorded in an anechoic room. From the results, there is a tendency of different impression between these two methods, and it is suggested that the phase randomized method may be effective with the case using from less channels.
Engineering Brief 350
EB09-03 The Dawn of Audiophile Quality Audio on Your Smartphone
Stefan Gustavsson (Presenting Author)
The mobile phone has become the primary device for personal music and multimedia consumption. This increases the focus on audio quality, especially when listening through headphones, providing impetus for a transition from mobile audio being a low quality, best effort music player to one that can be compared to dedicated high audio quality playback systems. The task of delivering audiophile quality music to mobile phone users provides a unique challenge requiring extremely low power. The latest, highly integrated audio solutions for smartphone chipsets use technologies, architectures, and algorithms that can deliver HIFI audio while still providing attractive power consumption and cost. With this fast improvement in mobile phone audio the system level design and testing methodologies need to keep up.
Engineering Brief 351
EB09-04 Practical Loudness Measurement and Management for Immersive Audio
Scott G. Norcross (Presenting Author), Sachin Nanda (Author), Marvin Pribadi (Author)
Loudness management is an essential and often mandatory aspect for content providers and broadcasters. Regional requirements/guidelines based on Recommendation ITU-R BS.1770 form the basis for the loudness practice in broadcasting. It has recently been revised to support new immersive-channel formats, but not explicitly for object-based audio formats. Object-based audio is currently being delivered over-the-top (OTT) and loudness management must be addressed to meet requirements and provide a good user experience. These new audio formats allow the content to be played back over a larger range of playback configurations, which has the potential for loudness variations. This brief describes loudness measurement and management for currently deployed object-based delivery and shows how legacy playback of this content meets the current broadcasting recommendations.
Engineering Brief 352
EB09-05 An Analog Audio Sensor Board for Microcontrollers
Colin Zyskowski (Presenting Author), Mauricio de Oliveira (Author)
This paper presents work on the Audio Sensor Board (ASB), a circuit board designed to serve as an analog interface between audio signals and the digital/analog inputs of common micro-controllers. The ASB allows for low-level manipulation of audio signals so that those signals can be easily used as control parameters, in essence creating a two-channel analog/digital sensor for sound sources. In this paper we layout the various functions of the board, its design, and describe practical purposes for which it has successfully been used.
Engineering Brief 353