AES New York 2013
Engineering Brief Details
EB1 - Audio Processing
Thursday, October 17, 5:30 pm — 7:00 pm (Room 1E09)
Chair:
David Benson, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
EB1-1 Modeling the Korg35 Lowpass and Highpass Filters—Will Pirkle, University of Miami - Coral Gables, FL, USA
The Korg35 Filter is a voltage controlled Sallen-Key topology capable of producing both lowpass (LPF) and highpass (HPF) filter responses. It is known for its ability to self-oscillate as well as its saturated or distorted output as the damping factor of the filter approaches zero where self-oscillation occurs. Both LPF and HPF are second order but the highpass version features a 6 dB/octave roll off instead of the usual 12 dB/octave giving it a distinctive sound with more bass response. The analog Sallen-Key topology uses a delay-less positive feedback loop to implement the Q control of the filter. The saturation circuit is also inside this loop. Rather than use a typical biquad or state variable filter, we choose to use Virtual Analog (VA) filters [1] as building blocks to replicate the analog filter topology directly from its signal flow graph, including the delay-less loop as well as the saturation circuit. Both loaded and unloaded (lossy) versions of the Sallen-Key topology were designed and implemented in C++ with the point of self-oscillation exactly matching both analog transfer functions. Sample code and extra documentation are available at www.willpirkle.com.
Engineering Brief 103 (Download now)
EB1-2 Modeling the Harmonic Exciter—Priyanka Shekar, Stanford University - Stanford, CA, USA; Julius O. Smith, III, Stanford University - Stanford, CA, USA
A harmonic exciter is an audio effects signal processor applied to enhance the brightness and clarity of a sound, particularly used for vocals. This is achieved by inducing a measured amount of high-frequency distortion. In this paper an exciter is digitally modeled and implemented as a standalone application (or plugin) using the FAUST (Functional AUdio STream) programming language for real-time audio. The model is based on the Aural Exciter by Aphex Ltd., an analog hardware unit. Technical specifications of the Aural Exciter are drawn from the original 1979 patent. The digital model performs as expected, recreating a “intage" style audio effect.
Engineering Brief 104 (Download now)
EB1-3 Fourier Transforms, Audio Engineering, and the Quantum Nature of Reality—Scott Hawley, Dept of Chemistry & Physics, Belmont University - Nashville, TN, USA
A short interdisciplinary, educational survey is presented illustrating ways in which audio spectral analysis and quantum physics are intimately related. A basic conceptual understanding of Fourier transforms and their applications in audio engineering is sufficient to grasp aspects of quantum wave packets, the Heisenberg Uncertainty Principle, and more. Similarly, concepts from quantum mechanics can inform the understanding of audio effects such as aliasing, convolutions and wavelet transforms. The presenter is a computational physicist who authored a computer audio analysis suite for audio engineering students, noting several interdisciplinary connections in the process.
Engineering Brief 105 (Download now)
EB1-4 Experiments with Dither in Level-Calibrated Floating Point Audio Processing—Douglas Rollow, Sennheiser Technology and Innovation - San Francisco, CA, USA
The use of dither to decorrelate quantization error in fixed point signal processing systems is a well-established practice in professional audio. Floating point computation, however, is quite common due to the ease of use and ubiquity of high performance platforms, among other reasons. Dither is (anecdotally) less frequently found in floating point audio systems, until the final mapping to fixed point representation, but quantization error occurs in the rounding operation during intermediate calculations. Widrow and others have provided detailed treatment of the quantization error in floating point audio calculations, and in the present work experiments using dither during the internal rounding operation in a floating point unit are compared to the external addition of noise when the signal levels are known to be calibrated from the original analog source.
Engineering Brief 106 (Download now)
EB1-5 Virtual Development of Audio Systems—Application of CAE Methods—Alfred Svobodnik, Konzept-X - Karlsruhe, Austria
This paper will give an overview of state-of-the-art CAE methods for virtual product development of audio systems, especially focusing on automotive audio and small transducers. Matrix based CAE methods will be discussed to be used for multi-physical modeling of transducers, acoustic enclosures (e.g., doors or rear shelves of automobiles, or acoustic systems for phones) and listening spaces, especially automotive car cabins. Ultimately a process model allowing the simulation of complete systems, representing a fully virtualized product development environment, will be shown.
Engineering Brief 107 (Download now)
EB1-6 A New DSP Tool for Drum Leakage Suppression—Elias Kokkinis, accusonus - Patras, Greece; Alexandros Tsilfidis, accusonus, Patras Innovation Hub - Patras, Greece; Thanos Kostis, accusonus, Patras Innovation Hub - Patras, Greece; Kostas Karamitas, accusonus, Patras Innovation Hub - Patras, Greece
Microphone leakage is problem that sound engineers face every day. Leakage complicates audio editing, processing, and mixing, and it is a well known problem in drum recordings. To this day, sound engineers have only a limited amount of options available in order to address this problem. These mostly consist of simple and empirical methods. A novel technology that addresses the problem of microphone leakage in multichannel drum recordings is presented here. In addition we discuss the problem definition as deduced from the specific properties of drum recordings, as well as the resulting signal processing framework.
Engineering Brief 108 (Download now)
EB2 - Recording & Production
Friday, October 18, 12:00 pm — 12:45 pm (Room 1E07)
Chair:
Scott Levine, Skywalker Sound - San Francisco, CA, USA; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada
EB2-1 Controlling Drum Bleed with Laser Vibrometry—Andrew Greenwood, Sennheiser Electronic - San Francisco, CA, USA; Sebastian Chafe, Sennheiser Technology and Innovation - San Francisco, CA, USA
Using multiple microphones to capture the sound of multiple drums on a drum kit is common practice. As well, the bleed captured by such a setup is a common problem for sound engineers. Gating is often used in an attempt to manage drum and instrument bleed into individual drum channels. However, the overlap in amplitude and frequency content of different drums makes gating based solely on the microphone audio difficult and unwanted triggering of the gate is a common problem. By measuring the physical vibration of the drum head using simple laser vibrometry and using this signal to run the sidechain of a gate, the dynamic range of the gate’s signal follower is increased and false triggering is easier to avoid. This allows for more precise control over each drum channel’s tone and dynamics
Engineering Brief 109 (Download now)
EB2-2 The Urban Mix Engineer—Paul "Willie Green" Womack, Willie Green Music - Brooklyn, NY, USA
Although hip-hop is over 40 years old and influences the sound of everyone from Stevie Wonder to Taylor Swift, the term "hip-hop engineer" often evokes visions of kids with cracked software and distorted records. This presentation explores arguments often referenced in this ongoing debate of skill, including mixing "real vs. synthetic" instruments, the loop based nature of hip-hop, and mixing samples. It also illustrates a time when engineers in other genres once overcame similar criticisms. Providing an honest, first-hand look at what’s involved in mixing urban records, and the hurdles that exist in and out of the studio, this presentation sheds light on the importance of the mixing engineer, in any genre, as a vital part of the arrangement process.
Engineering Brief 110 (Download now)
EB2-3 Subjective Comparison of Surround Microphone Recording Techniques Presented With and Without Video—Luiz Fernando Kruszielski, Globo TV Network - Rio de Janeiro, Brazil; Tokyo University of the Arts - Tokyo, Japan
The comparison of different setups of microphones for surround recordings of music is a topic that has a large interest in the audio community. Although it is known that image has a strong effect on sound perception, particularly in spatial aspects, very little research had been done aiming the surround sound recordings with accompanied video. To compare possible influences in this perception, a test was created using five different surround recording techniques that was done simultaneously at a Brazilian carnival parade. The subjects were presented to the sound with and without video and asked to rate four different aspects: localization, deepness, immersion, and preference. The results show that there is a difference in perception depending on the presence or absence of video.
Engineering Brief 111 (Download now)
EB3 - E-Brief Posters—Part 1
Friday, October 18, 5:00 pm — 6:30 pm (1EFoyer)
EB3-1 A Special Room for 3D Audio and Ultra High Definition Video for Quality Assessment of Future TV—Matthieu Parmentier, francetélévisions - Paris, France
francetélévisions, the French public broadcaster, is involved in collaborative projects that aim to embrace the future of television. Ultra High Definition video in conjunction with 3D sound are today explored within the range of content production, techniques, costs, and quality of experience for consumer applications. With its new-dedicated room, the innovations & developments department of francetélévisions has built a necessary tool to drive its strategy for facing the upcoming challenges.
Engineering Brief 112 (Download now)
EB3-2 Real-Time Head-Related Impulse Response Filtering with Distance Control—Julian Villegas, University of Aizu - Aizu Wakamatsu, Fukushima, Japan; Michael Cohen, University of Aizu - Aizu Wakamatsu-shi, Fukashima-ken, Japan
We present a new software application based on a recently collected HRIR database comprising measurements at different distances. The new application, programmed in Pure-data, is capable of directionalizing sound objects at any azimuth, at elevations between -40 degrees and 90 degrees, and at distances 20-160 cm. This truly 3D spatialization is done by pre-calculating the minimum-phase version of the HRIRs and computing the interpolation of a maximum of four HRIR measurements, depending upon the virtual location. In the same way, interaural time differences are computed and applied to the convolved signal. For demanding real-time constraints, the number of taps used for the convolution can be adjusted, up to a maximum of 1024.
Engineering Brief 113 (Download now)
EB3-3 Results on Automated Tuning of a Voice Quality Enhancement System Using Objective Quality Measures—Daniele Giacobello, Beats By Dr. Dre - Santa Monica, CA, USA; Joshua Atkins, Beats Electronics, LLC - Santa Monica, CA, USA; Jason Wung, Beats Electronics, LLC - Santa Monica, CA, USA; Raghavendra Prabhu, Beats by Dr. Dre - Santa Monica, CA, USA
In this work we present a formal procedure for automating the tuning of the various parameters comprising a voice quality enhancer. First, we formalize the problem of tuning as a large-scale nonlinear programming problem. Second, we evaluate the performance of perceptual objective quality measures as optimization criteria for our tuning problem. We then perform a subjective quality assessment to compare the output of a voice enhancer obtained with parameters calculated with these different criteria and also with those obtained through a conventional approach of tuning by expert listening. The results show that using this automated methodology performs well in finding reasonable solutions for the tuning problem, potentially saving time and resources over manual evaluation and tuning.
Engineering Brief 114 (Download now)
EB3-4 Influence of Loudspeaker Systems on Acquisition of Head-Related Transfer Functions—Shouichi Takane, Akita Prefectural University - Yurihonjo, Akita, Japan; Koji Abe, Akita Prefecture University - Yurihonjo, Akita, Japan; Kanji Watanabe, Akita Prefecture University - Yurihonjo, Akita, Japan; Sojun Sato, Akita Prefecture University - Yurihonjo, Akita, Japan
HHead-Related Transfer Function (HRTF) is defined by the ratio of sound pressure at the center of the head without listener and the one at his/her ear. Frequency characteristics of a sound source ought to be cancelled in its acquisition based on this definition, but they are not when the sound sources are spatially distributed such as conventional multiple-driver loudspeaker systems. In this e-brief such influence was investigated by using the HRTFs acquired with various types of loudspeaker systems. As a result, it was found that the HRTFs acquired with four types of loudspeakers roughly agreed when the distance from the sound source is 1.5 m and farther.
Engineering Brief 115 (Download now)
EB3-5 Application of Audio Engineering and Psychoacoustic Principles to Audible Medical Alarms—Christopher Bennett, University of Miami - Coral Gables, FL, USA; Oygo Sound LLC - Miami, FL, USA; Colby N. Leider, University of Miami - Coral Gables, FL, USA; Richard McNeer, University of Miami - Miami, FL, USA
Audible medical alarms standards have recently undergone extensive review by regulatory and safety organizations due to reported ineffectiveness of alarms and the role of “alarm fatigue” in contributing to morbidity and mortality among patients. Many of the problems associated with alarm fatigue stem from an improper application of psychoacoustic and audio engineering principles and naive design of auditory streams that lead to poor segregation, confusion among clinicians, and ultimately fatigue. The audio engineer has a clear role in defining solutions to problems arising in hospital units, some of which have previously been addressed in sound production, sound design, and auditory scene analysis. The roles of sonification, psychoacoustics, and sound perception are discussed as they apply to audible medical alarms.
Engineering Brief 116 (Download now)
EB3-6 Revisiting the Space—Applying 5.1 Surround Sound—Mike Godwin, University of Western Ontario - London, ON, Canada
Origins of this project grew from requests from Faculty and Performers wishing there was a way to better experience the live acoustic again while listening to our archival recordings. As such, my objective was to research an approach to record one of our early music choirs performing in an ambient venue utilizing 5.1 surround techniques. Then through listening tests, obtain subject preferences for the stereo vs. 5.1 versions with comment categories for each. For this project the goal was to use the simplest possible setup as we are most often in a live concert environment, and where setup time is also a consideration. Initially I did recordings testing both cardioid and omni microphones to decide on the best patterns, and placements.
Engineering Brief 117 (Download now)
EB4 - Applications in Audio
Saturday, October 19, 11:30 am — 1:15 pm (Room 1E07)
Chair:
David Romblom, McGill University - Montreal, Quebec, Canada; Centre for Interdisciplinary Research in Music Media and Technology (CIRMMT) - Montreal, Quebec, Canada
EB4-1 SyncAV—Workflow Tool for File-Based Video Shootings—Andreas Fitza, University of Applied Science Mainz - Mainz, Germany
The Sync-AV workflow tool eases the sorting and synchronization of video and audio footage without the need for expensive special hardware. It supports pre-production, shooting and post-production. It consists of these elements: a script-information and metadata-gathering web app that’s connected to a server database; a local import client that manages the footage ingest and sorts the files together; the client also takes care of the synchronization of the video that contains audio and separately recorded audio files and it renames the files and implements the metadata; and the client uploads this synchronized preview files to our server so they can be shown at our web app. This e-Brief shows the current development and some specific solutions of Sync-AV.
Engineering Brief 118 (Download now)
EB4-2 Inconsistencies in the Practical Design and Measurement of Sound Systems in Reverberant Spaces Requiring a Minimum STI Standard—David McNutt, The McNutt Group - Chicago, IL, USA; Columbia College Chicago - Chicago, IL, USA
Minimum Speech Transmission Index measurement is now a requirement for Emergency Communication Systems as set forth in NFPA 72 2013 code. Professional audio design engineers have the greatest effect on potential intelligibility through their choice of the type, number, and distribution of loudspeakers and the power at which they are driven. Design Engineers often model sound systems for STI using EASE. Using this STI modeling approach can lead to varying results especially in reverberant sound fields. This brief discusses the conflicting results of three design/build projects in highly reverberant spaces in the Federal Plaza in Chicago.
Engineering Brief 119 (Download now)
EB4-3 The Advantages of Using Active Crossovers in High-End Wireless Speakers—David Jones, CSR Limited - Manchester, UK
With the availability of standardized wireless interfaces and high performance codecs, wireless speakers can be designed that suit the consumer demands of compactness and ease of use. This paper will examine the performance benefits of using active crossovers and digital equalization in an amplification subsystem based on a high performance digital input switching amplifier. Measurements of distortion and damping factors will be compared in an example signal chain and the influence these parameters have on the perceived audio quality of the
speaker system will be discussed.
Engineering Brief 85 (Download now)
EB4-4 Low Latency Replacement of ISDN and 4-Wire for Remote Broadcasts—Anthony Faust, Atlantic Post Production - Toronto, ON, Canada; Netmondi
Integrated Services Digital Network (ISDN) lines are being replaced by other forms of Internet Protocol (IP) connectivity for high-quality remote broadcasts. In particular, the use of bonding diversity (diversity) over multiple public Internet networks for remote broadcasts has been proven in challenging environments with excellent results. This high-performance of this approach makes it likely to become the standard for remote broadcasts.
Engineering Brief 120 (Download now)
EB4-5 Design and Construction of the Stringer: A Polyphonic Signal Switcher for 13-pin DIN M—Michael Palumbo, Concordia University - Montreal, Quebec, Canada; Pouya Hamidi, McGill University - Montreal, QC, Canada; Richard King, McGill University - Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology - Montreal, Quebec, Canada; Donald Pavlasek, McGill University - Montreal, Quebec, Canada
The Stringer is a polyphonic signal switcher for use with 13-pin DIN MIDI guitar pickups. Used as an intermediary between a guitar and a synthesizer pedal, the purpose of the device is to isolate a single string for monophony, such as bass note accompaniment. A height-and-depth-adjustable fulcrum bar supports the performer’s feet, and brings them closer to the foot switches, allowing for smoother and faster string switching. The current model can isolate strings 6, 5, and 4; mute all strings; and can run in bypass mode to pass all string signals through for standard operation. The circuit is powered by a 9V DC external adapter, and housed in a custom aluminum chassis.
Engineering Brief 121 (Download now)
EB4-6 Design of a Sound Reinforcement System for Koerner Hall—Jeffery Bamford, Engineering Harmonics Inc. - Toronto, ON, Canada
Built over three years, the 1,135-seat Koerner Hall is the jewel of the new TELUS Centre for Performance and Learning at the Royal Conservatory of Music in Toronto, Canada. Since its opening in September 2009, Koerner Hall's beautiful design, flexible performance characteristics and superb acoustics have been praised by critics and performers alike. The hall achieved the highest possible acoustic rating—N1—rendering it ideal for the finest acoustical performances of classical music, jazz, and world music. The incorporation of variable acoustics makes it equally well suited to amplified music, lectures, and film presentations. This Engineering Brief will review the process and design of the sound reinforcement system. It features an innovative and almost invisible 'voice-stick' to maximize intelligibility, rather than sound reinforcement. The system must provide coverage for the audience as to performers on and around the stage in an extremely intimate venue. Testing the design with a computer and mock-up will also be discussed.
Engineering Brief 122 (Download now)
EB4-7 Consistently Stable Loudspeaker Measurements Using a Tetrahedral Enclosure—Geoff Hill, Hill Acoustics Limited - Leigh on Sea, Essex, United Kingdom
A major problem for the loudspeaker and transducer industries throughout the world is an inability to rely upon measurements
routinely exchanged between suppliers and customers. A system is proposed that offers a unique and stable test environment giving an opportunity to standardize and compare results between measurement sites. It works by having an enclosure shape that eliminates standing-waves and having acoustic foam to eliminate any remaining high frequencies. It then rigidly defines the measurement geometry together with interchangeable sub baffles, ensuring rapid and accurate change over and repeatable measurements. So that with several in use in the design, production and customer chain
results will be comparable unit to unit throughout the world to an
unprecedented degree.
Engineering Brief 123 (Download now)
EB5 - E-Brief Posters—Part 2
Saturday, October 19, 5:00 pm — 6:30 pm (1EFoyer)
EB5-1 Digital Model of the Passive James/Baxandall Tonestack—Christopher Bennett, University of Miami - Coral Gables, FL, USA; Oygo Sound LLC - Miami, FL, USA; Jonathon Toft-Nielsen, Intelligent Hearing Systems - Miami, FL, USA; Connor McCullough, University of Miami - Coral Gables, FL, USA
E. J. James described a two-knob tone control in 1949 with easily selectable boost/cut depths as well as cutoff frequencies. This design was later popularized by P. J. Baxandall to provide negative feedback in active circuits, and was subsequently popular in many high-end amplifiers. Here, the authors analyzed the circuit in the s-domain, preserving parametric control of bass and treble potentiometer values. Poles and zeros were found using Ferrari’s solution to a quartic equation, followed by bilinear transformation to the z-domain, and finally lumping into second-order sections to produce a computationally efficient and faithful emulation of this classic tonestack.
Engineering Brief 124 (Download now)
EB5-2 Automatic Analog Preamp Gain Control Using Digital Command—Nicolas Sturmel, Digigram S.A. - Montbonnot Saint Martin, France; Fusheng Yu, ENSEEIHT–INP - Toulouse, France
Automatic Gain Control (AGC) is a common tool for field recording, but it usually requires specific hardware such as voltage controlled amplifiers. In this paper, we address the problem of designing an AGC when none of this hardware is present, using an ubiquitous digitally controlled high end analog preamp. To do this, we have to overcome two problems: fixed gain steps and variable delay of the gain command. In order to propose an efficient solution, we will first study the effects of each of those two problems. Finally, a very simple digitally controlled automatic gain, but of high quality will be proposed, using only 10MIPS of processing power from our high end USB sound card.
Engineering Brief 125 (Download now)
EB5-3 Testing Watermark Robustness against Application of Audio Restoration Algorithms—Bozena Kostek, Gdansk University of Technology - Gdansk, Poland; Audio Acoustics Lab.; Janusz Cichowski, Gdansk University - Gdansk, Poland; Andrzej Czyzewski, Gdansk University of Technology - Gdansk, Poland
The purpose of this study was to test to what extent watermarks embedded in distorted audio signals are immune to audio restoration algorithm performing. Several restoration routines such as noise reduction, spectrum expansion, clipping or clicks reduction were applied in the online website system. The online service was extended with some copyright protection mechanisms proposed by the authors. They contain low-level music features embedded as watermarks using the non-blind approach. After applying restoration algorithms, the watermark is extracted from the audio track. It was shown in experiments, that a watermark “attacked” by the restoration procedures may still be detected. However in some cases it is possible to retrieve only a binary information about the watermark presence in the audio carrier.
Engineering Brief 126 (Download now)
EB5-4 PsychoMasker: An iOS Application for the Visualization of PsychoAcoustic Principles—Andrew Ayers, University of Miami - Coral Gables, FL, USA; Robert Rehrig, University of Miami - Coral Gables, FL, USA; Christopher Bennett, University of Miami - Coral Gables, FL, USA; Oygo Sound LLC - Miami, FL, USA; Colby N. Leider, University of Miami - Coral Gables, FL, USA
The concept of masking in psychoacoustics has invaded the daily lives of almost every audio listener since the initial release of the MPEG-1 standard. With the ubiquity of the MP3, the consumption of
perceptually coded audio is impossible to avoid. While many people understand the concept of perceptual coding, it can be difficult to visualize what is actually happening to the information in the audio files. PsychoMasker is an App that provides real-time visualization of the psychoacoustic principles used in MPEG encoding to anyone with an iPad. The PsychoMasker App shows the user how the encoding process
aff ects any song in the user's iTunes library step-by-step.
Engineering Brief 127 (Download now)
EB5-5 Using MIDI Control Surfaces with MATLAB Programs and Simulink—Charlie DeVane, MathWorks - Natick, MA, USA
MATLAB and Simulink are widely used in the design of software and hardware for audio products. MATLAB programs and Simulink models can simulate signal processing algorithms, control logic, and other aspects of the system design in real time, yielding substantial improvements in designer productivity and product quality. When exploring a new algorithm or product concept, designers often need to simultaneously tune multiple system parameters while simulating. This can become cumbersome using GUIs, but MIDI control surfaces provide a natural, intuitive interface, further enhancing the designer's work flow. In some work flows, such as rapid prototyping, MIDI control surfaces can eliminate the need to create a GUI. Using numerous examples, including a simple reverberator, this brief shows how to use MIDI control surfaces to interactively control running MATLAB programs and Simulink models.
Engineering Brief 128 (Download now)
EB5-6 MIDI to CV Conversion Using a Livid BrainV2 and I2C Protocol—Mark Gill, University of Miami - Coral Gables, FL, USA
Bob Moog developed Control voltage (CV) in the 1960s. His introduction of CV called for an oscillator’s pitch to vary at the rate of 1 volt per octave. This scheme is widely used today in analog circuits, and can be mimicked digitally. The CV output is determined by converting various MIDI messages including USB MIDI, and physical controls including analog potentiometer inputs and momentary on, momentary off buttons. The Livid BrainV2 handles all the MIDI inputs and directs them to a digital to analog converter to create the CV signal controlled by I2C communication. This paper documents the hardware used to create the converter, the mathematical onsiderations for conversion, and techniques used to overcome the limited ability of 8-bit MIDI messages to be portrayed as an analog signal. Sound clips are available at my website markolgill.com
Engineering Brief 129 (Download now)
