• 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
P25 - Multichannel Sound
Paper Session P25
Tuesday, May 20, 11:30 — 16:00
Chair: Günther Theile, Institut für Rundfunktechnik - Munich, Germany
P25-1 Bitstream Format for Spatio-Temporal Wave Field Coder—Francisco Pinto, Martin Vetterli, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland
We present a non-parametric method for compressing multichannel audio data for reproduction through Wave Field Synthesis. The method consists of applying a two-dimensional filterbank to the input multichannel signal, in both time and channel dimensions, and coding the two-dimensional spectra using a spatio-temporal frequency masking model. The coded spectral data is organized into a bitstream together with side information containing scale factors and Huffman codebook information. We demonstrate how this coding method can be applied to any smooth distribution of loudspeakers in space, while obtaining a stable bit rate that is 15% lower compared to coding each channel independently.
Convention Paper 7472 (Purchase now)
P25-2 The Design of Ambisonic Decoders for the ITU 5.1 Layout with Even Performance Characteristics—David Moore, Jonathan Wakefield, University of Huddersfield - Huddersfield, West Yorkshire, UK
All previously published Ambisonic decoders for irregular loudspeaker layouts have localization performance that varies significantly by angle around the listener. This contrasts with decoders designed for evenly spaced arrangements of loudspeakers where performance characteristics are isotropic. Furthermore, even localization performance around the listener is desirable for a number of application areas of 5.1 surround sound. New decoder design criteria are presented that aim to reduce this variation in localization performance. These criteria are added to a multi-objective fitness function, based on auditory localization theory, which guides a heuristic search algorithm to derive decoder parameter sets for the ITU5.1 layout. The derived decoders exhibit a significant improvement in localization performance variation by angle around the 360-degree sound stage.
Convention Paper 7473 (Purchase now)
P25-3 Methods for Sharing Stereo and Multichannel Recordings among Planetariums—Leslie Gaston, Peter Dougall, Erick D. Thompson, University of Colorado at Denver - Denver, CO, USA
There is a demand for research on the transferability of surround sound audio from one planetarium to another, so that (1) audiences have similar experiences and (2) audio engineers can easily create this experience. This paper will consider: acoustics, production, delivery, equipment, and seating arrangements. Our recent survey of over 100 planetariums worldwide in the fall of 2007 will provide a look at current practices. The University of Colorado Denver and Gates Planetarium have collaborated in order to explore the potential of current audio technology, and to discover what similarities and differences exist between planetariums in order to achieve this goal of transferability.
Convention Paper 7474 (Purchase now)
P25-4 Optimal Hierarchical Bandwidth Limitation of Surround Sound—Yu Jiao, Slawomir Zielinski, Francis Rumsey, University of Surrey - Guildford, Surrey, UK
In order to save the transmission bandwidth of surround sound, a technique named Hierarchical Bandwidth Limitation (HBL) was proposed by the authors. In HBL, a psychoacoustically hierarchical transform is used as the preprocessing algorithm prior to bandwidth limitation. In our former experiments we found that the Karhunen-Lòeve transform (KLT) is a suitable hierarchical transform for HBL. Besides the hierarchical transform, the choice of an appropriate strategy for bandwidth allocation is also essential from the point of view of the resultant audio quality. In order to find the optimal bandwidth allocation strategy that achieves the best audio quality, the authors attempted to build up the mathematical relationship between audio quality and the bandwidth allocation strategy using a MUSHRA listening test. The experiment design and results of this listening test are reported in this paper.
Convention Paper 7475 (Purchase now)
P25-5 Frequency-Dependent Signal-Correlation in Surround- and Stereo-Microphone Systems and the Blumlein-Pfanzagl-Triple (BPT)—Edwin Pfanzagl-Cardone, Salzburg Festival - Salzburg, Austria; Robert Höldrich, Institute of Electronic Music and Acoustics - Graz, Austria
With the aim to recreate the original concert-hall sound field as faithfully as possible in the control- or living-room, recordings were made simultaneously with an artificial head and several surround microphone techniques (among them the new BPT method). The surround recordings were rerecorded using the same dummy-head as in the concert hall. The results of subjective listening tests (loudspeaker as well as binaural) were assessed using ANOVA and correlation analysis. Acoustical analysis of the dummy-head recordings was performed by measuring the Frequency-Dependent Inter Aural Cross-Correlation Coefficient (FIACC): the low-correlation AB-PC microphone system was capable of reproducing the original sound field better than any of the other systems under test (DECCA, KFM, OCT). A microphone systems Critical Frequency, below which correlation raises toward 1, is defined.
Convention Paper 7476 (Purchase now)
P25-6 Holographic Design of Source Array for Achieving a Desired Sound Field—Wan-Ho Cho, Jeong-Guon Ih, Korea Advance Institute of Science and Technology (KAIST) - Daejeon, Korea; Marinus M. Boone, Delft University of Technology - Delft, The Netherlands
For realizing a desired complicated sound field, an acoustic source array should be designed appropriately to obtain the acoustic source parameters. To this end, we suggest a method utilizing the acoustical holography technique based on the inverse boundary element method. Acoustical analogy between the problems of source reconstruction and source design was the initial motivation of the study. In the design of the source array, the pressure distribution at specific field points is the constraint of the problem and the signal distribution at the source surface points is the object function of the problem. The whole procedure of the application consists of three stages. First, a condition of the desired sound field should be set as the constraint. Second, the geometry and boundary condition of the source array system and the target field, i.e., points in the sound field of concern, are modeled by the boundary elements. Actual characteristics of source and space can be considered to generate the accurate condition of the target field. Finally, the source parameters are inversely calculated by the backward projection. As an example, a source array to fulfill the plane wave propagating zone and another quiet zone near the propagation zone was designed and tested by simulation and measurement.
Convention Paper 7477 (Purchase now)
P25-7 New Dimensions for Ambisonics—Michael Chapman - Culoz, France
Both two-dimensional (pantophonic) and three-dimensional (periphonic) representations of soundfields are common place in ambisonics. Reproducing either on rigs essentially designed for the other is common place. What though if one synthesizes a four (or more) dimensional soundfield and reproduces this on a standard rig? As there appears to be no source on hyperspherical harmonics applicable to ambisonics, the mathematical basis is first set out. The manipulation of hyperambisonic soundfields (rotation, mirroring, dominance) is then discussed. During that discussion various “proofs” are advanced as to the finite range of transformations that can be applied to ambisonic soundfields, of whatever dimension.
Convention Paper 7478 (Purchase now)
P25-8 Improving Spherical Microphone Arrays—Nicolas Epain, Jérome Daniel, France Télécom R&D - Lannion, France
Spherical microphone arrays are useful for numerous applications, such as spatial audio capture and beamforming. However, these sensor arrays are known to have a limited frequency range, due to poor directivity at low frequencies and spatial aliasing at high frequencies. In this paper we study two methods aiming at enhancing the frequency range of spherical microphone arrays without using more sensors. First, the benefit of locating the sensors at the end of cavities within the sphere is assessed through measurements and simulations. Second, we study the influence of using large membrane microphones. Finally, results show that the frequency range could be increased in both cases studied.
Convention Paper 7479 (Purchase now)
P25-9 Migration of 5.0 Multichannel Microphone Array Design to Higher Order MMAD (6.0, 7.0, and 8.0) with or without the Inter-Format Compatibility Criteria—Michael Williams, Sounds of Scotland - Paris, France
The severe limitations of the 5.0 Multichannel Reproduction Standard in reproducing good quality audio-visual or stand-alone audio surround sound reproduction has increased the pressure on recording and reproduction system designers to increase the number of channels in an attempt to give an even more satisfactory envelopment experience. This paper extends the MMAD process to show how higher order channel array designs (6.0, 7.0, and 8.0) can be developed from the existing data on 4.0 or 5.0 Multichannel Front Sound Stage Coverage Array Designs with almost perfectly seamless and linear surround sound reproduction. Designing for inter-format compatibility can also be accommodated from the existing multi-format array design data described in a previous paper on Multichannel Arrays Generating Inter-format Compatibility (MAGIC arrays).
Convention Paper 7480 (Purchase now)
Last Updated: 20080612, tendeloo