3:30, AUDIO-P3.1
OUT-OF-HEAD SOUND LOCALIZATION USING ADAPTIVE INVERSE FILTER
T. HORIUCHI, H. HOKARI, S. SHIMADA
It is well known that the transfer functions for out-of-head sound localization differ with the listener. Our goal is to find a method that easily realizes excellent sound localization without measuring the individual transfer functions. This paper proposes an out-of-head sound localization system that uses adaptive inverse filtering without any measuring preprocesses. This system estimates the inverse Ear Canal Transfer Function (ECTF), and can adaptively obtain the transfer function to fit the listener in real-time. This paper also proposes an adaptive inverse filtering method for out-of-head sound localization that differs from the filtered-x method. In addition, the relationship between convergence time and initial value is assessed. It is clarified that the proposed method is more effective, in terms of convergence, if the initial value is the average of many listeners' impulse responses.

3:30, AUDIO-P3.2
COMMON-ACOUSTIC-POLES/ZEROS APPROXIMATION OF HEAD-RELATED TRANSFER FUNCTIONS
C. LIU, S. HSIEH
Common-acoustic-poles/zeros (CAPZ) approximation is a more efficient way to model head-related transfer functions (HRTF's). In CAPZ approximation, a group of HRTF's can share a set of poles but use their own zeros. Previous CAPZ works, such as the Prony, Shanks and iterative prefiltering methods, were all based on the linearized least-square criterion. A new state-space approach, jointly balanced model truncation, is proposed by using singular value decomposition of a joint Hankel matrix. The proposed approach can choose the suitable order of IIR filters for HRTF's approximation according to the distribution of singular values but previous works can't. The proposed method is also modified to permit different orders for pole and zero. Computer simulations of these approaches are included for comparison.

3:30, AUDIO-P3.3
INTERPOLATED 3-D DIGITAL WAVEGUIDE MESH WITH FREQUENCY WARPING
L. SAVIOJA, V. VALIMAKI
An interpolated 3-D digital waveguide mesh algorithm is elaborated. We introduce an optimized technique that improves a formerly proposed interpolated 3-D mesh and renders the 3-D mesh more homogeneous in different directions. Frequency-warping techniques are used to shift the frequencies of the output signal of the mesh in order to cancel the effect of dispersion error. The extensions improve the accuracy of 3-D digital waveguide mesh simulations enough so that in the future it can be used for acoustical simulations needed in the design of listening rooms, for example.

3:30, AUDIO-P3.4
PREDICTION AND MEASUREMENT OF THE ACOUSTIC CROSSTALK CANCELLATION ROBUSTNESS
F. ORDUŅA-BUSTAMANTE, J. LOPEZ, A. GONZALEZ
The condition number of the matrix of electro-acoustic head-related transfer functions (HRTF) in a two-channel sound reproduction system has been used as a measure of robustness of the Atal-Schroeder crosstalk canceler. A comparative study has been made using results produced by computer simulations and HRTFs measured in an anechoic chamber by means of a dummy head. It has been found that acoustic scattering by the head has a very important and beneficial influence on robustness, specially for large loudspeaker separations. For narrow loudspeaker separations of less than about 40 degrees it is found that crosstalk cancellation exhibits a large variation of alternating very low and very high robustness. Also, simulations and measurements have been made of the natural channel separation under the same conditions. Scattering by the head is seen to provide a good level of natural channel separation at high frequencies and large loudspeaker angles. At low frequencies or small loudspeaker angles natural channel separation is poor.

3:30, AUDIO-P3.5
ARTIFACT-FREE ASYNCHRONOUS GEOMETRY-BASED AUDIO RENDERING
N. TSINGOS
Most audio rendering systems include a geometry-based simulation engine which computes and updates sound propagation paths and an auralization engine which renders audible the resulting sound field. In the case of dynamic environments the attributes of the sound paths are variable, which can cause severe artifacts in the auralized sound signal. To avoid this problem, geometrical calculations have to be conducted synchronously with the signal processing tasks and the necessary high update rates can be difficult to achieve in complex environments. In this paper, we describe a technique which allows for conducting geometrical calculations completely asynchronously from the signal processing. In particular, we introduce a prediction mechanism of path attributes which ensures that artifact free signal processing can be achieved even when geometrical information is updated at low rates. This technique can be applied to any geometry-based simulations, regardless of the technique used for finding the sound propagation paths.

3:30, AUDIO-P3.6
MULTIRATE ADAPTIVE FILTERING FOR IMMERSIVE AUDIO
J. LIM, C. KYRIAKAKIS
This paper describes a method for implementing immersive audio rendering filters for single or multiple listeners and loud- speakers. In particular, the paper is focused on the case of single or two listeners with different loudspeaker arrays to determine the weighting vectors for the necessary FIR and IIR filters using the LMS (least-mean-squares) adaptive inverse algorithm. It describes transform-domain LMS adaptive inverse algorithm that is designed for crosstalk cancellation necessary in loudspeaker-based immersive audio rendering. Specifically, each weighting vector of the inverse filter is generated based on psychoacoustic critical band filters and uses the LMS adaptive inverse algorithm to improve performance in the sensitive frequency bands. We also investigate the sensitivity of the listening position under different number of listeners and loud- speakers with various loudspeaker geometries. Performance is measured based on the ipsilateral signal to contralateral signal (crosstalk) ratio that results from the different filter types with and without psychoacoustic critical band filtering.

3:30, AUDIO-P3.7
SMART HEADPHONES: ENHANCING AUDITORY AWARENESS THROUGH ROBUST SPEECH DETECTION AND SOURCE LOCALIZATION
S. BASU, B. CLARKSON, A. PENTLAND
We describe a method for enhancing auditory awareness by selectively passing speech sounds in the environment to the user. We develop a robust far-field speech detection algorithm for noisy environments and a source localization algorithm for flexible arrays. We then combine these methods to give a user control over the spatial regions from which speech will be passed through. Using this technique, we have implemented a ``smart headphones'' system in which a user can be listening to music over headphones and hear speech from specified directions mixed in. We show our preliminary results on the algorithms and describe initial user feedback about the system.

3:30, AUDIO-P3.8
LOUDSPEAKER FAULT DETECTION USING TIME-FREQUENCY REPRESENTATIONS
M. DAVY, H. COTTEREAU, C. DONCARLI
This paper addresses the problem of loudspeaker test. Two applications are investigated: loudspeaker manufacturing fault detection and maintenance. In order to comply with practical requirements, we propose a new, brief, nonstationary test signal. Recorded loudspeaker responses are processed using an improved time-frequency decision algorithm. The fault detection procedure is tested with real data. Results show its accuracy and practical interest.

3:30, AUDIO-P3.9
MODELING THE EFFECT OF A NEARBY BOUNDARY ON THE HRTF
N. GUMEROV, R. DURAISWAMI
Understanding and simplified modeling of the Head Related Transfer Function (HRTF) holds the key to many applications in spatial audio. We develop an exact solution to the problem of scattering of sound from a sphere in the vicinity of an infinite plane. Using this solution we study the influence of a nearby scattering surface, on a spherical model for the HRTF. Our results show how the HRTF is modified by a nearby surface such as the wall/ground plane, and when simpler models may be used to account for the effect of boundaries on the HRTF.