AES London 2010
Paper Session P17
P17 - Multichannel and Spatial Audio: Part 1
Monday, May 24, 14:00 — 17:30 (Room C3)
Chair: Wieslaw Woszczyk, McGill University - Montreal, Quebec, Canada
P17-1 Individualization of Dynamic Binaural Synthesis by Real Time Manipulation of ITD—Alexander Lindau, Jorgos Estrella, Stefan Weinzierl, Technical University of Berlin - Berlin, Germany
The dynamic binaural synthesis of acoustic environments is usually constrained to the use non-individual impulse response datasets, measured with dummy heads or head and torso simulators. Thus, fundamental cues for localization such as interaural level differences (ILD) and interaural time differences (ITD) are necessarily corrupted to a certain degree. For ILDs, this is a minor problem as listeners may swiftly adapt to spectral coloration at least as long as an external reference is not provided. In contrast, ITD errors can be expected to lead to a constant degradation of localization. Hence, a method for the individual customization of dynamic binaural reproduction by means of real time manipulation of the ITD is proposed. As a prerequisite, subjectively artifact free techniques for the decomposition of binaural impulse responses into ILD and ITD cues are discussed. Finally, based on listening test results, an anthropometry-based prediction model for individual ITD correction factors is presented. The proposed approach entails further improvements of auditory quality of real time binaural synthesis.
Convention Paper 8088 (Purchase now)
P17-2 Perceptual Evaluation of Physical Predictors of the Mixing Time in Binaural Room Impulse Responses—Alexander Lindau, Linda Kosanke, Stefan Weinzierl, Technical University of Berlin - Berlin, Germany
The mixing time of room impulse responses denotes the moment when the diffuse reverberation tail begins. A diffuse sound field can physically be defined by (1) equi-distribution of acoustical energy and (2) a uniform acoustical energy flux over the complete solid angle. Accordingly, the perceptual mixing time is the moment when the diffuse tail cannot be distinguished from that of any other position in the room. This provides an opportunity for reducing the length of binaural impulse responses that are dynamically exchanged in virtual acoustic environments (VAEs). Numerous model parameters and empirical features for the prediction of perceptual mixing time in rooms have been proposed. This paper aims at a perceptual evaluation of all potential estimators. Therefore, binaural impulse response data sets were collected with an adjustable head and torso simulator for a representative sample of rectangular-shaped rooms. Prediction performance was evaluated by linear regression using results of a listening test where mixing times could be adaptively altered in real time to determine a just audible transition time into a homogeneous diffuse tail. Regression formulae for the perceptual mixing time are presented, conveniently predicting perceptive mixing times to be used in the context of VAEs.
Convention Paper 8089 (Purchase now)
P17-3 HRTF Measurements with a Continuously Moving Loudspeaker and Swept Sines—Ville Pulkki, Mikko-Ville Laitinen, Ville Pekka Sivonen, Aalto University School of Science and Technology - Aalto Finland
An apparatus is described, which is designed to measure head-related transfer functions (HRTFs) for audio applications. A broadband, two-driver loudspeaker is rotated around the subject with continuous movement, and responses are measured with a swept-sine technique. Potential error sources are discussed and quantified, and it is shown that the responses are almost identical to responses measured with a static, small single-driver loudspeaker. It is also shown that the method can be used to measure a large number of HRTFs in a relatively short time period.
Convention Paper 8090 (Purchase now)
P17-4 In Situ Microphone Array Calibration for Parameter Estimation in Directional Audio Coding—Oliver Thiergart, Giovanni Del Galdo, Maja Taseska, Jose Angel Pineda Pardo, Fabian Kuech, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
Directional audio coding (DirAC) provides an efficient representation of spatial sound using a downmix audio signal and parametric information, namely direction-of-arrival (DOA) and diffuseness of sound. Input to the DirAC analysis are B-format signals, usually obtained via microphone arrays. The DirAC parameter estimation is impaired when phase mismatch between the array sensors occurs. We present an approach for the in situ microphone array calibration solely based on the DirAC parameters. The algorithm aims at providing consistent parameter estimates rather than matching the sensors explicitly. It does neither require to remove the sensors from the array, nor depend on a priori knowledge such as the array size. We further propose a suitable excitation signal to assure robust calibration in reverberant environments.
Convention Paper 8093 (Purchase now)
P17-5 Sound Field Recording by Measuring Gradients—Mihailo Kolundzija, Christof Faller, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland; Martin Vetterli, Ecole Polytechnique Fédérale de Lausanne - Lausanne, Switzerland, University of California at Berkeley, Berkeley, CA, USA
Gradient-based microphone arrays, the horizontal sound field's plane wave decomposition, and the corresponding circular harmonics decomposition are reviewed. Further, a general relation between directivity patterns of the horizontal sound field gradients and the circular harmonics of any order is derived. A number of example differential microphone arrays are analyzed, including arrays capable of approximating the sound pressure gradients necessary for obtaining the circular harmonics up to order three.
Convention Paper 8092 (Purchase now)
P17-6 Evaluation of a Binaural Reproduction System Using Multiple Stereo-Dipoles—Yesenia Lacouture Parodi, Per Rubak, Aalborg University - Aalborg, Denmark
The sweet spot size of different loudspeaker configurations was investigated in a previous study carried out by the authors. Closely spaced loudspeakers showed a wider control area than the standard stereo setup. The sweet spot with respect to head rotations showed to be especially large when the loudspeakers are placed at elevated positions. In this paper we describe a system that uses the characteristics of the loudspeakers placed above the listener. The proposed system is comprised of three pairs of closely spaced loudspeakers: one pair placed in front, one placed behind, and one placed above the listener. The system is based on the idea of dividing the sound reproduction into regions to reduce front-back confusions and enhance the virtual experience without the aid of a head tracker. A set of subjective experiments with the intention of evaluating and comparing the performance of the proposed system are also discussed.
Convention Paper 8091 (Purchase now)
P17-7 Conditioning of the Problem of a Source Array Design with Inverse Approach—Jeong-Guon Ih, KAIST - Daejeon, Korea; Wan-Ho Cho, KAIST - Daejeon, Korea, Chuo University, Bunkyoku, Tokyo, Japan
An inverse approach based on the acoustical holography concept can be effectively applied to the acoustic field rendered for achieving a target sound field given as a relative response distribution of sound pressure. To implement this method, the source configuration should be determined a priori, and a meaningful inverse solution of an ill-conditioned transfer matrix should be obtained. To choose efficient source positions that are almost mutually independent, the redundancy detection algorithm like the effective independence method was employed to decide the proper positions for a given number of sources. In this way, an efficient and stable filter set for a source array in controlling the sound field can be obtained. An interior domain with irregularly shaped boundaries was adopted as the target field to control for testing the suggested inverse method.
Convention Paper 8094 (Purchase now)