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Last Updated: 20070405, meiP1 - Spatial Audio Perception and Processing - 1
Saturday, May 5, 09:00 — 12:00
Chair: Thomas Sporer, Fraunhofer Institut IDMT - Ilmenau, Germany
P1-1 Using Transient Suppression in Blind Multichannel Up-Mix Algorithms—Andreas Walther, Christian Uhle, Sacha Disch, Fraunhofer Institute IIS - Erlangen, Germany
In the field of blind up-mixing, many algorithms exist for generating multichannel sound from mono or stereo sources. One of the important blind up-mix scenarios is the ambience-based up-mix. This approach aims at extracting the ambient parts of a given signal and their reproduction by taking the best possible advantage of a multichannel loudspeaker setup. Depending on the number of channels and the signal characteristics of the input signal, the quality of the extracted ambience can vary. In this paper transient suppression is suggested as a method for improving an extracted ambience signal. Two methods for suppressing transient components are proposed and contrasted to existing techniques. The ability of those methods to improve the perceived quality of the ambience signal and overall up-mix is evaluated in a subjective listening test.
Convention Paper 6990 (Purchase now)
P1-2 A Novel Approach to Up-Mix Stereo to Surround Based on MPEG Surround Technology—Heiko Purnhagen, Coding Technologies - Stockholm, Sweden; Andreas Ehret, Coding Technologies - Nuremberg, Germany; Jonas Rödén, Coding Technologies - Stockholm, Sweden; Alexander Gröschel, Coding Technologies - Nuremberg, Germany
With the increasing number of installed surround sound systems in consumer homes and cars, the demand for surround sound content is rising. However, the vast majority of content is still only available in stereo. Furthermore, in many cases it is difficult to create a surround mix for content previously released in mono or stereo due to the high effort necessary or simply because the original multitrack recordings are unavailable. Consequently the need for tools that allow an automated or semi-automated stereo to surround up-mix is growing. In this paper a novel approach is described that is based on technology that is part of the MPEG Surround standard. The basic algorithm and some proposed extensions are outlined and potential use-cases described. Finally, the subjective quality of the presented approach is compared to existing solutions.
Convention Paper 6991 (Purchase now)
P1-3 Coding of "2+2+2" Surround Sound Content Using the MPEG Surround Standard—Andreas Ehret, Alexander Gröschel, Coding Technologies - Nuremberg, Germany; Heiko Purnhagen, Jonas Roedén, Coding Technologies - Stockholm, Sweden
An increasing number of recordings is available in the so-called 2+2+2 surround sound format, where in addition to two front and two rear loudspeakers, a third pair of loudspeakers is placed at an elevated position above the front speakers. In 2006, the MPEG Surround standard was finalized as an efficient stereo backward-compatible surround sound coding format. The present paper studies the applicability of MPEG Surround for efficient coding of "2+2+2" content. Several alternative approaches are outlined and evaluated by means of subjective listening tests.
Convention Paper 6992 (Purchase now)
P1-4 Quality Taxonomies for Auditory Virtual Environments—Andreas Silzle, Ruhr-Universität Bochum - Bochum, Germany
The aim of the here developed new taxonomies is to describe the components involved in the quality process of Auditory Virtual Environments (AVE) and to quantify the relations between them for different applications. The taxonomy should allow an overview and identify the relations that are most important in the software development process and the design of listening experiments. For the first time the multivariate relations between the quality elements in the physical domain and the quality features in the perceptual domain of such a quality taxonomy are evaluated for three different AVE applications. This evaluation and quantification is done by means of an expert survey (DELPHI method) to objectify the results. Principal component analysis reveals that five dimensions are necessary to describe about 95 percent of the variance in the data. This indicates that the selected seven quality features are clearly distinguishable for the experts, but not orthogonal to each other. Most of the quality features are introduced in meaningful terms in the audio engineering field and therefore usable without training for the participating experts. The results of the expert survey are compared to listening test results, which use the same quality features. The bottom line is that the expert survey is not only a much faster method to get a good overview about a specific application as compared to the listening test, but it also reveals more information about it.
Convention Paper 6993 (Purchase now)
P1-5 Individual Localization Behavior for Perception of Virtual Sound Sources—Cornelius Bradter, University of Applied Science - Berlin, Germany; Klaus Hobohm, Film and Television Academy - Potsdam, Germany
Test results normally indicate very large variability in perception of lateral virtual sound sources with a 5-channel loudspeaker setup. Three recent studies indicate that up to a third of stimuli from a side loudspeaker pair were perceived surprisingly accurately as virtual sound sources. In other cases sound sources were perceived as coming from only one loudspeaker or from its vicinity. Therefore, we specified prototypical localization behaviors. We examined effects on localization by reproduction rooms, exact position of test persons in relation to loudspeakers, test persons’ head movements, and trading between time delay and level differences.
Convention Paper 6994 (Purchase now)
P1-6 Comparison of Different Sound Capture and Reproduction Techniques in a Virtual Acoustic Window—Timo Haapsaari, Werner De Bruijn, Aki Härmä, Philips Research Laboratories - Eindhoven, The Netherlands
In this paper we describe a two-way audio communication system using arrays of microphones and loudspeakers to create a virtual acoustic window. We compare three different methods for capturing the sound: wave field sampling using a line array of microphones, an adaptive beamformer, and close-talk microphones. For sound reproduction, we employ wave field synthesis. In the paper we review the acoustic and perceptual requirements for a real-time virtual acoustic window system and report results of a set of listening experiments performed with the system.
Convention Paper 6995 (Purchase now)