AES Amsterdam 2008
Home Visitors Exhibitors Press Students Authors
Technical Program

Sessions by Industry

Detailed Calendar

Convention Planner

Paper Sessions



Master Classes

Live Sound Seminars

Exhibitor Seminars

Special Events

Student Program


Technical Tours

Technical Council

Standards Committee

Heyser Lecture

AES Amsterdam 2008
Poster Session P28

P28 - Software, Instrumentation, and Measurement

Tuesday, May 20, 13:30 — 15:00
P28-1 An Anatomy of Graph-Based User Interfaces for Media ProcessingChristopher Schultz, Universität Bremen - Bremen, Germany, now at mediaclipping, Bremen, Germany; Jörn Loviscach, Hochschule Bremen - Bremen, Germany; Shailendra Mathur, Softimage Corp., Avid Technology, Inc. - Montreal, Quebec, Canada; Jay LeBoeuf, Digidesign - Daly City, CA, USA, now at Imagine Research, Inc., San Francisco, CA, USA
Graph-based user interfaces are employed in a variety of software such as audio synthesizers, video compositing tools, and database application builders. All of these uses afford the graphical metaphor of a graph: “Nodes” such as sound generators or filters are tied together by “links,” which may represent signal flow or conceptual relations. Focusing on media production tools, we have examined a large range of current software products to find out which de-facto standards have evolved in the field of graph-based interfaces and which features can be considered unique. We categorize a multitude of interface concepts employed in actual graph-based interfaces and describe differences in their implementation. The findings provide guidelines for developers of media production software.
Convention Paper 7495 (Purchase now)

P28-2 A Framework for Automatic Mixing Using Timbral Similarity Measures and Genetic OptimizationBennett Kolasinski, New York University - New York, NY, USA
A novel method is introduced for automatic mix recreation using timbral classification techniques and an optimization algorithm. This approach uses the Euclidean distance between modified Spectral Histograms to calculate the distance between a mix and a target sound and uses a genetic optimization algorithm to figure out the best coefficients for that mix. The implementation has been shown to successfully recreate multitrack mixes accurately and may pave the way toward the automatic mixing of novel multitrack sessions based on a desired target sound.
Convention Paper 7496 (Purchase now)

P28-3 Delta-Sigma DAC Topologies for Improved Jitter PerformanceIvar Løkken, Anders Vinje, Trond Sæther, Norwegian University of Science and Technology - Trondheim, Norway
Specifications for audio digital-to-analog-converters (DACs) place requirements on the analog circuit design that contradict physical design conditions in a modern, digital-oriented system on a chip process. Because of low supply voltages, use of current-steering DACs has become the dominant choice for high resolution applications. Fed by a delta-sigma modulator that requantizes the digital signal to a manageable number of bits, the current-steering DAC is a continuous time type converter without any discrete time filtering. This makes it very susceptible to sampling clock jitter. In this paper jitter distortion is addressed at a topology level, investigating design choices for the delta-sigma requantizer and the possible use of semidigital multi-bit current-steering filter DACs to reduce problems with jitter susceptibility.
Convention Paper 7497 (Purchase now)

P28-4 New Measurement Methods for Anechoic Chamber CharacterizationJuan Gómez-Alfageme, José Luis Sánchez-Bote, Elena Blanco-Martín, Universidad Politécnica de Madrid - Madrid, Spain
As a continuation of the work presented at the 122nd AES Convention (Paper 7153), this paper tries to study in depth the anechoic chambers qualification. The purpose of this paper is to find parameters that allow the characterization of this type of enclosure. The proposal tries to obtain data of the anechoic chambers absorption by means of the transfer functions between pairs of microphones or by means of the impulse response between pairs of microphones. The results of the transfer functions between pairs of microphones can be easily checked by the agreement of the inverse squared law, allowing determination of the chamber cut-off frequency. Making a band filtering confirmed the anechoic chamber’s qualifications.
Convention Paper 7498 (Purchase now)

P28-5 Acoustic Feedback Reduction Based on LMS and Normalized LMS Algorithms in WOLA Filters Bank Based Digital Hearing AidsRaúl Vicen-Bueno, Universidad de Alcalá - Alcalá de Henares, Madrid, Spain; Almudena Martínez-Leira, Dimetronic Signals - San Fernando de Henares, Madrid, Spain; Manuel Rosa-Zurera, Lucas Cuadra-Rodríguez, Universidad de Alcalá - Alcalá de Henares, Madrid, Spain
Acoustic feedback phenomenon can disturb a digital hearing aid performance at high gains, causing instability in the haring aid and degradation in the speech. In order to restore a stable situation, an acoustic feedback reduction (AFR) subsystem using adaptive algorithms such as the least-mean square (LMS) algorithm is needed. This algorithm has a reduced computational cost, but it is very unstable. In order to avoid this situation, another feedback reduction system based on a modified version of the LMS algorithm is used. Such algorithm is: the Normalized LMS (NLMS). These two algorithms are tested in two digital hearing aid categories: the In-The-Ear and the In-The-Canal. These categories are selected because they have great feedback effects, so robust AFR subsystems are needed. The added stable gain (ASG) over the limit gain when an AFR subsystem is working in the digital hearing aid is obtained for each category. The ASG is determined as a trade-off between two measurements: the segmented signal-to-noise ratio (objective measurement) and the speech quality (subjective measurement). The results show how the digital hearing aids working with a feedback reduction adaptive filter adapted with the NLMS algorithm is able to achieve up to 18 dB of increase over the limit gain.
Convention Paper 7499 (Purchase now)

P28-6 Nonlinear Distortions in CapacitorsMenno van der Veen, ir.bureau Vanderveen bv - Zwolle, The Netherlands; Hans van Maanen, Temporal Coherence
Many people have claimed that capacitors have a notable influence on the audible quality of systems. We have identified one of the major causes of nonlinear distortions in capacitors. Charging the capacitor will result in an attractive force acting on the conducting plates. As no material is infinitely stiff, this force will reduce the thickness of the dielectricum and thus increase the capacitance. This process occurs in both phases of an AC signal in the same way and is thus nonlinear. In this paper the consequences of this process are discussed. It should be noted that other passive components like resistors and inductors can also show similar nonlinear behavior.
Convention Paper 7500 (Purchase now)

Last Updated: 20080612, tendeloo