AES Paris 2016
Engineering Brief EB1
EB1 - eBriefs 1: Posters
Saturday, June 4, 09:00 — 10:15 (Foyer)
EB1-1 Sound Pressure Analysis For Closed-Box Loudspeaker Enclosures—Charalampos Papadakos, University of Patras - Rio, Greece; Gavriil Kamaris, University of Patras - Rion Campus, Greece; John Mourjopoulos, University of Patras - Patras, Greece
This study employs a physical modeling method to explore the pressure distribution within typical closed-box loudspeaker enclosures of different shape and inner volume. The simulation results are compared to measurements in such enclosures. The results indicate that sound pressure within such enclosures often exceeds levels of 130 dB. The pressure profile is usually constant at lower frequencies and displays some strong resonances at higher frequencies due to normal modes. Such levels traditionally challenge enclosure air-tightness, box rigidity, but they can also provide useful acoustic energy for harvesting.
Engineering Brief 236 (Download now)
EB1-2 Mobile Platform Acoustical Noise Identification Using Internal and Reference Microphones—Przemek Maziewski, Intel Technology Poland - Gdansk, Poland
Proposed paper addresses the problem of microphone noise. The performance of built-in microphones in laptops and other mobile devices can suffer in the presence of noise. Identifying noisy components and separating the internal from external origins allows for the noise sources to be root caused and eliminated. This capability is crucial when developing new platforms. The proposed method employs a series of recordings, conducted using both built-in and reference microphones. The recordings are obtained under different operating conditions of the device, e.g., AC vs battery power. The recordings are then analyzed to identify different characteristics resulting from use of the internal versus the external microphone. Based on these results noise components can be separated and the potential noise source identified.
Engineering Brief 237 (Download now)
EB1-3 Automatically Generating VST Plugins from MATLAB Code—Charlie DeVane, MathWorks - Natick, MA, USA; UMass Lowell - Lowell, MA, USA; Gabriele Bunkheila, MathWorks - Cambridge, UK
We describe the automatic generation of VST audio plugins from MATLAB code using the Audio System Toolbox from MathWorks. We provide MATLAB code for three complete example plugins, discuss problems that may be encountered, and describe a workflow to generate VST plugins as quickly and easily as possible.
Engineering Brief 238 (Download now)
EB1-4 Echo Thresholds for a 3-D Loudspeaker Configuration—Lee Davis, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK
Echo thresholds were examined with differing stimuli, lag sound directions, and decision criteria in a 3D loudspeaker reproduction environment. Two tests were undertaken to examine two different criteria: echo threshold with fusion and echo threshold with complete separation, each with three stimuli (orchestral, pink noise burst, and speech) and six lag sound directions in total. An adapted method of adjustment was used by subjects to control the delay between the lead and lag loudspeakers. Results showed that there were significant differences in echo threshold when the decision criteria differed. The orchestral stimulus was found to be significantly different from the pink noise burst and speech in both criteria. Few significant differences were noted between angles. In general, echo thresholds were higher with lag sources located in the median plane.
Engineering Brief 239 (Download now)
EB1-5 A New Response Method for Auditory Localization and Spread Tests—Hyunkook Lee, University of Huddersfield - Huddersfield, UK; Dale Johnson, The University of Huddersfield - Huddersfield, UK; Manchester, UK; Maksims Mironovs, University of Huddersfield - Huddersfield, West Yorkshire, UK
This Engineering Brief presents a new response method developed for auditory localization and spread tests. The proposed method uses a flexible strip with a series of LEDs, which are powered by a microcontroller, for eliciting subjective responses. For the localization test, the position of an active LED is controlled and recorded in Max using a dial. For the spread test, multiple LEDs can be positioned on the strip to visually describe the lower and upper boundaries of the perceived image. The required system is easy to build and relatively inexpensive. Vertical stereophonic localization tests were conducted to compare between the LED method and a visual marker method. Results showed that the proposed method was more accurate, consistent, and time-efficient than the marker method.
Engineering Brief 240 (Download now)
EB1-6 Setting Up and Making an AES67 Network Coexist with Standard Network Traffic—Mickaël Henry, UVHC - Valenciennes, France; Digigram - Montbonnot, France; Lucas Rémond, CNSMDP - Paris, France; Nicolas Sturmel, Digigram S.A. - Montbonnot, France
In this paper we will show how an AES67 network can coexist within a standard non-audio network. We will detail the difficulties usually encountered when setting up and using AES67 networks. We will analyze the utility of the network protocols required by AES67: (i) IGMP and its impact on devices features, (ii) PTP and the clock recovery performance when using PTP enabled switches, and (iii) QoS and the impact of non-audio traffic such as web and corporate traffic. We will use a set-up of 10 different AES67 compliant devices from many manufacturers and supporting various AoIP protocols all compliant to AES67. We will provide recommendations in order to provide proper quality of experience while making networks coexist.
Engineering Brief 241 (Download now)
EB1-7 Implementation of Faster than Real Time Audio Analysis for Use with Web Audio API: An FFT Case Study—Luis Joglar-Ongay, University of Huddersfield - Huddersfield, West Yorkshire, UK; Christopher Dewey, University of Huddersfield - Huddersfield, UK; Jonathan Wakefield, University of Huddersfield - Huddersfield, UK
There is significant interest in the audio community in developing web-based applications using HTML5 and Web Audio API. Whilst this newly emerging API goes some way to provide offline audio analysis in the web browser it is limited to a relatively basic FFT with fixed Blackman windowing and no overlap facility. Most previously documented solutions to this issue operate in real time. This paper demonstrates how to perform more sophisticated, faster than real time FFT analysis for use within Web Audio applications. It makes use of the Web Audio API and the dsp.js library. Academics and researchers can use this paper as a tutorial to develop similar solutions within their own web based audio applications.
Engineering Brief 242 (Download now)
EB1-8 Block-Sparse Fast Recursive Approximated Memory Improved Proportionate Affine Projection Algorithm—Felix Albu, Valahia University of Targoviste - Targoviste, Romania
A new approximated memory improved proportionate affine projection algorithm for block sparse echo cancellation is proposed. This contribution presents a fast recursive implementation combined with the use of dichotomous coordinate descent iterations. It is shown that the proposed algorithm has superior convergence speed and tracking abilities for echo path changes in the context of acoustic and network echo cancellation applications. Also it is proved that these achievements are obtained while having a reduced numerical complexity than competing algorithms.
Engineering Brief 243 (Download now)
EB1-9 The Effect of Loop Length and Musical Material on Discrimination Between MP3 and WAV Files—Denis Martin, McGill University - Montreal, QC, Canada; CIRMMT - Montreal, QC, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
Listening test results generally show that for bit-rates higher than 128 kbps listeners can rarely distinguish between MP3 and WAV files with any statistical significance while many audio professionals agree it is possible. This project attempts to explain some of the reasons why typical AB and ABX tests often fail by looking at the effects of loop length and music choice on listener success. An informal take-home AB listening test was used with varying musical material and music looping at different lengths. The results show that performance drops significantly with short loop lengths (<2sec, p = .02) and that the participants were able to discriminate between these two different file formats with great significance (p < .001).
Engineering Brief 244 (Download now)
EB1-10 Considerations When Calibrating Program Material Stimuli Using LUFS—Malachy Ronan, University of Limerick - Limerick, Ireland; Nicholas Ward, University of Limerick - Limerick, Ireland; Robert Sazdov, University of Limerick - Limerick, Ireland
While the LUFS standard was originally developed for broadcast applications, it offers a convenient means of calibrating program material stimuli to an equal loudness level, while remaining in a multichannel format. However, this calibration is based on an absolute sound pressure level of 60 dBA, the preferred listening level when watching television. Levels used in analytical listening and perceptual experiments tend to be significantly higher. This disparity may affect the accuracy of the Leq(RLB) weighting filter employed in LUFS meters. To address this issue, the development of the LUFS standard is examined to assess its suitability for the task. The findings suggest that a compromise between analytical listening and loudness matching in perceptual experiments requires careful consideration of experimental variables.
Engineering Brief 245 (Download now)
EB1-11 Selective Mixing Improves Reproduction Quaity with Portable Loudspeakers—Piotr Kleczkowski, AGH University of Science and Technology - Krakow, Poland; Tomasz Dziedzic, AGH University of Science and Technology - Krakow, Poland
Selective mixing of sounds is an experimental method of mixing, first proposed in [1]. Further developments and listening experiments confirmed that inexperienced listeners more often than not prefer this type of processing over direct mixing, while it is the other way round with mixing engineers. It has been found lately, that besides the extent of the effect, there is another independent variable associated with this method—quality of the reproduction system. Experiments have shown, that the percentage of listeners choosing selective mixing versions is higher when the music is reproduced over small loudspeakers of portable devices, like notebook computers.
Engineering Brief 246 (Download now)
