@conference {18224,
	title = {Image acquisition forensics: Forensic analysis to identify imaging source},
	booktitle = {Acoustics, Speech and Signal Processing, 2008. ICASSP 2008. IEEE International Conference on},
	year = {2008},
	month = {2008/04/31/4},
	pages = {1657 - 1660},
	abstract = {With widespread availability of digital images and easy-to-use image editing softwares, the origin and integrity of digital images has become a serious concern. This paper introduces the problem of image acquisition forensics and proposes a fusion of a set of signal processing features to identify the source of digital images. Our results show that the devices{\textquoteright} color interpolation coefficients and noise statistics can jointly serve as good forensic features to help accurately trace the origin of the input image to its production process and to differentiate between images produced by cameras, cell phone cameras, scanners, and computer graphics. Further, the proposed features can also be extended to determining the brand and model of the device. Thus, the techniques introduced in this work provide a unified framework for image acquisition forensics.},
	keywords = {ACQUISITION, acquisition;image, analysis;, analysis;image, analysis;interpolation;statistical, cameras;color, cell, coefficients;computer, colour, editing, forensics;image, graphics;digital, identification;noise, images;forensic, Interpolation, phone, processing;data, softwares;imaging, source, statistics;scanners;signal},
	doi = {10.1109/ICASSP.2008.4517945},
	author = {McKay,C. and Swaminathan,A. and Gou,Hongmei and M. Wu}
}
@article {13996,
	title = {Fast Evaluation of the Room Transfer Function Using Multipole Expansion},
	journal = {Audio, Speech, and Language Processing, IEEE Transactions on},
	volume = {15},
	year = {2007},
	month = {2007/02//},
	pages = {565 - 576},
	abstract = {Reverberation in rooms is often simulated with the image method due to Allen and Berkley (1979). This method has an asymptotic complexity that is cubic in terms of the simulated reverberation length. When employed in the frequency domain, it is relatively computationally expensive if there are many receivers in the room or if the source or receiver positions are changing with time. The computational complexity of the image method is due to the repeated summation of the fields generated by a large number of image sources. In this paper, a fast method to perform such summations is presented. The method is based on multipole expansion of the monopole source potential. For offline computation of the room transfer function for N image sources and M receiver points, use of the Allen-Berkley algorithm requires O(NM) operations, whereas use of the proposed method requires only O(N+M) operations, resulting in significantly faster computation of reverberant sound fields. The proposed method also has a considerable speed advantage in situations where the room transfer function must be rapidly updated online in response to source/receiver location changes. Simulation results are presented, and algorithm accuracy, speed, and implementation details are discussed. For problems that require frequency-domain computations, the algorithm is found to generate sound fields identical to the ones obtained with the frequency-domain version of the Allen-Berkley algorithm at a fraction of computational cost},
	keywords = {acoustics;computational, algorithm;frequency-domain, Alien-Berkley, analysis;reverberation;, complexity;frequency-domain, computations;image, expansion;receivers;reverberant, field;architectural, fields;reverberation;room, function;acoustic, method;image, potential;multipole, sound, source, sources;monopole, transfer},
	isbn = {1558-7916},
	doi = {10.1109/TASL.2006.876753},
	author = {Duraiswami, Ramani and Zotkin,Dmitry N and Gumerov, Nail A.}
}
@conference {14024,
	title = {Approximate expressions for the mean and the covariance of the maximum likelihood estimator for acoustic source localization},
	booktitle = {Acoustics, Speech, and Signal Processing, 2005. Proceedings. (ICASSP {\textquoteright}05). IEEE International Conference on},
	volume = {3},
	year = {2005},
	month = {2005/03//},
	pages = {iii/73 - iii/76 Vol. 3 - iii/73 - iii/76 Vol. 3},
	abstract = {Acoustic source localization using multiple microphones can be formulated as a maximum likelihood estimation problem. The estimator is implicitly defined as the minimum of a certain objective function. As a result, we cannot get explicit expressions for the mean and the covariance of the estimator. We derive approximate expressions for the mean vector and covariance matrix of the estimator using Taylor{\textquoteright}s series expansion of the implicitly defined estimator. The validity of our expressions is verified by Monte-Carlo simulations. We also study the performance of the estimator for different microphone array configurations.},
	keywords = {(mathematics);, acoustic, approximate, approximation, array, array;, covariance, estimation;, expansion;, expressions;, function;, likelihood, localization;, matrices;, matrix;, maximum, mean, microphone, objective, processing;, series, signal, source, Taylor, theory;, vector;, vectors;},
	doi = {10.1109/ICASSP.2005.1415649},
	author = {Raykar,V.C. and Duraiswami, Ramani}
}
@conference {12279,
	title = {Differentiated traffic engineering for QoS provisioning},
	booktitle = {INFOCOM 2005. 24th Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings IEEE},
	volume = {4},
	year = {2005},
	month = {2005/03//},
	pages = {2349 - 2359 vol. 4 - 2349 - 2359 vol. 4},
	abstract = {We introduce a new approach for QoS provisioning in packet networks based on the notion of differentiated traffic engineering (DTE). We consider a single AS network capable of source based multi-path routing. We do not require sophisticated queuing or per-class scheduling at individual routers; instead, if a link is used to forward QoS sensitive packets, we maintain its utilization below a threshold. As a consequence, DTE eliminates the need for per-flow (IntServ) or per-class (DiffServ) packet processing tasks such as traffic classification, queueing, shaping, policing and scheduling in the core and hence poses a lower burden on the network management unit. Conversely, DTE utilizes network bandwidth much more efficiently than simple over-provisioning. In this paper, we propose a complete architecture and an algorithmic structure for DTE. We show that our scheme can be formulated as a non-convex optimization problem, and we present an optimal solution framework based on simulated annealing. We present a simulation-based performance evaluation of DTE, and compare our scheme to existing (gradient projection) methods.},
	keywords = {based, Computer, differentiated, DiffServ, DTE;, engineering;, evaluation;, link;, links;, management;, multipath, network, networks;, nonconvex, of, optimisation;, OPTIMIZATION, packet, performance, problem;, provisioning;, QoS, QUALITY, routing;, service;, simulation-based, source, Telecommunication, traffic, traffic;},
	doi = {10.1109/INFCOM.2005.1498521},
	author = {Tabatabaee,V. and Bhattacharjee, Bobby and La,R.J. and Shayman,M.A.}
}
@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.}
}
@article {14023,
	title = {Speaker Localization Using Excitation Source Information in Speech},
	journal = {Speech and Audio Processing, IEEE Transactions on},
	volume = {13},
	year = {2005},
	month = {2005/09//},
	pages = {751 - 761},
	abstract = {This paper presents the results of simulation and real room studies for localization of a moving speaker using information about the excitation source of speech production. The first step in localization is the estimation of time-delay from speech collected by a pair of microphones. Methods for time-delay estimation generally use spectral features that correspond mostly to the shape of vocal tract during speech production. Spectral features are affected by degradations due to noise and reverberation. This paper proposes a method for localizing a speaker using features that arise from the excitation source during speech production. Experiments were conducted by simulating different noise and reverberation conditions to compare the performance of the time-delay estimation and source localization using the proposed method with the results obtained using the spectrum-based generalized cross correlation (GCC) methods. The results show that the proposed method shows lower number of discrepancies in the estimated time-delays. The bias, variance and the root mean square error (RMSE) of the proposed method is consistently equal or less than the GCC methods. The location of a moving speaker estimated using the time-delays obtained by the proposed method are closer to the actual values, than those obtained by the GCC method.},
	keywords = {correlation, correlation;, cross, Delay, error, error;, estimation;, excitation, generalized, information;, localization;, mean, methods;, processing;, production;, root, source, speaker, speech, square, TIME},
	isbn = {1063-6676},
	doi = {10.1109/TSA.2005.851907},
	author = {Raykar,V.C. and Yegnanarayana,B. and Prasanna,S. R.M and Duraiswami, Ramani}
}
@conference {18200,
	title = {Distortion management of real-time MPEG-4 video over downlink multicode CDMA networks},
	booktitle = {Communications, 2004 IEEE International Conference on},
	volume = {5},
	year = {2004},
	month = {2004/06//},
	pages = {3071 - 3075 Vol.5 - 3071 - 3075 Vol.5},
	abstract = {In this paper, a protocol is designed to manage source rate/channel coding rate adaptation, code allocation, and power control to transmit real-time MPEG-4 FGS video over downlink multicode CDMA networks. We develop a fast adaptive scheme of distortion management to reduce the overall distortion received by all users subject, to the limited number of codes and maximal transmitted power. Compared with a modified greedy method in literature, our proposed algorithm can reduce the overall system{\textquoteright}s distortion by at least 45\%.},
	keywords = {access;, adaptation;, allocation;, CDMA, channel, code, coding, coding;, combined, communication;, compression;, control;, data, distortion, division, downlink, links;, management;, MPEG-4, multicode, multiple, networks;, power, radio, rate, real-time, resource, source, source-channel, video, video;, visual},
	doi = {10.1109/ICC.2004.1313096},
	author = {Su,Guan-Ming and Han,Zhu and Kwasinski,A. and M. Wu and Liu,K. J.R and Farvardin,N.}
}
