@conference {13953,
	title = {Automatic matched filter recovery via the audio camera},
	booktitle = {Acoustics Speech and Signal Processing (ICASSP), 2010 IEEE International Conference on},
	year = {2010},
	month = {2010/03//},
	pages = {2826 - 2829},
	abstract = {The sound reaching the acoustic sensor in a realistic environment contains not only the part arriving directly from the sound source but also a number of environmental reflections. The effect of those on the sound is equivalent to a convolution with the room impulse response and can be undone via deconvolution - a technique known as matched filter processing. However, the filter is usually pre-computed in advance using known room geometry and source/receiver positions, and any deviations from those cause the performance to degrade significantly. In this work, an algorithm is proposed to compute the matched filter automatically using an audio camera - a microphone array based system that provides real-time audio images (essentially plots of steered response power in various directions) of environment. Acoustic sources, as well as their significant reflections, are revealed as peaks in the audio image. The reflections are associated with sound source(s) using an acoustic similarity metric, and an approximate matched filter is computed to align the reflections in time with the direct arrival. Preliminary experimental evaluation of the method is performed. It is shown that in case of two sources the reflections are identified correctly, the time delays recovered agree well with those computed from geometric constraints, and that the output SNR improves when the reflections are added coherently to the signal obtained by beamforming directly at the source.},
	keywords = {acoustic, array;real-time, arrays;transient, audio, camera;automatic, constraints;microphone, filter, filters;microphone, images;room, impulse, Matched, positions;acoustic, processing;cameras;matched, radiators;array, recovery;beamforming;geometric, response;, response;sound, sensor;acoustic, signal, source;source/receiver, sources;audio},
	doi = {10.1109/ICASSP.2010.5496187},
	author = {O{\textquoteright}Donovan,A.E. and Duraiswami, Ramani and Zotkin,Dmitry N}
}
@conference {14026,
	title = {The manifolds of spatial hearing},
	booktitle = {Acoustics, Speech, and Signal Processing, 2005. Proceedings. (ICASSP {\textquoteright}05). IEEE International Conference on},
	volume = {3},
	year = {2005},
	month = {2005/03//},
	pages = {iii/285 - iii/288 Vol. 3 - iii/285 - iii/288 Vol. 3},
	abstract = {We present exploratory studies on learning the non-linear manifold structure, in head related impulse responses (HRIRs). We use the recently popular locally linear embedding technique. The lower dimensional manifold encodes the perceptual information in the HRIRs, namely the direction of the sound source. Based on this, we propose a new method for HRIR interpolation. We also propose that the distance between two HRIRs of an individual be taken as the geodesic distance on the learned manifold.},
	keywords = {acoustics;, dimensional, direction, distance;, EMBEDDING, encoding;, geodesic, Head, hearing, hearing;, HRIR, impulse, information, interpolation;, learned, linear, localization;, locally, low, manifold, manifold;, manifolds;, nonlinear, perceptual, related, response;, responses;, sound, source, spatial, structure;, technique;, transient},
	doi = {10.1109/ICASSP.2005.1415702},
	author = {Duraiswami, Ramani and Raykar,V.C.}
}
@conference {14025,
	title = {A robust and self-reconfigurable design of spherical microphone array for multi-resolution beamforming},
	booktitle = {Acoustics, Speech, and Signal Processing, 2005. Proceedings. (ICASSP {\textquoteright}05). IEEE International Conference on},
	volume = {4},
	year = {2005},
	month = {2005/03//},
	pages = {iv/1137 - iv/1140 Vol. 4 - iv/1137 - iv/1140 Vol. 4},
	abstract = {We describe a robust and self-reconfigurable design of a spherical microphone array for beamforming. Our approach achieves a multi-resolution spherical beamformer with performance that is either optimal in the approximation of desired beampattern or is optimal in the directivity achieved, both robustly. Our implementation converges to the optimal performance quickly while exactly satisfying the specified frequency response and robustness constraint in each iteration step without accumulated round-off errors. The advantage of this design lies in its robustness and self-reconfiguration in microphone array reorganization, such as microphone failure, which is highly desirable in online maintenance and anti-terrorism. Design examples and simulation results are presented.},
	keywords = {3D, anti-terrorism;, array, array;, arrays;, audio, beam, beamforming;, beampattern, directivity, Frequency, microphone, multiresolution, omnidirectional, optimisation;, optimization;, processing;, reorganization, response;, robustness;, sampling;, self-reconfigurable, signal, soundfield, spherical, steering;},
	doi = {10.1109/ICASSP.2005.1416214},
	author = {Zhiyun Li and Duraiswami, Ramani}
}
@article {15374,
	title = {A parallel-in-time method for the transient simulation of SOI devices with drain current overshoots},
	journal = {Computer-Aided Design of Integrated Circuits and Systems, IEEE Transactions on},
	volume = {13},
	year = {1994},
	month = {1994/08//},
	pages = {1035 - 1044},
	abstract = {This paper presents a new parallel-in-time algorithm for the two dimensional transient simulation of SOI devices. With this approach, simulation in both space and time domains is performed in parallel As a result, the CPU time is reduced significantly from the conventional serial-in-time method. This new approach fully exploits the inherent parallelism of the finite difference formulation of the basic semiconductor device equations and the massively parallel architecture of SIMD computers. The space domain computations are inherently parallel due to the nature of our technique of solving the finite-difference equations. Time domain parallelism is achieved by shifting the potentials from previous time points to subsequent points one-step forward along the time axis with each Gummel iteration. This algorithm employs a fixed-point iteration technique, therefore a direct solution of matrix equations is avoided. The algorithm is especially suitable for the transient simulation of SOI devices that exhibit transient drain current overshoot. Numerical experiments show that the new parallel-in-time method is up to eight times faster than the conventional serial-in-time method in SOI transient simulations. The program is coded in CM Fortran for the Connection Machine},
	keywords = {2D, algorithms;semiconductor, analysis;transient, architecture;parallel-in-time, boundaries;silicon;time-domain, computations;time, computers;SOI, computing;finite, current, device, devices;drain, devices;parallel, difference, dimensional, domain, effect, engineering, equations;fixed-point, equations;space, field, Fortran;CPU, gate, iteration, iteration;SIMD, Machine;Gummel, method;semiconductor, methods;insulated, methods;metal-insulator-semiconductor, models;semiconductor-insulator, overshoots;finite-difference, Parallel, parallelism;transient, response;, simulation;CM, simulation;digital, simulation;electronic, simulation;two, technique;massively, time;Connection, transistors;iterative},
	isbn = {0278-0070},
	doi = {10.1109/43.298039},
	author = {Tai,G.-C. and Korman,C.E. and Mayergoyz, Issak D}
}
