@conference {18467,
	title = {Regularized HRTF fitting using spherical harmonics},
	booktitle = {IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2009. WASPAA {\textquoteright}09},
	year = {2009},
	month = {2009/10//},
	pages = {257 - 260},
	publisher = {IEEE},
	organization = {IEEE},
	abstract = {By the Helmholtz reciprocity principle, the head-related transfer function (HRTF) is equivalent to an acoustic field created by a transmitter placed at the ear location. Therefore, it can be represented as a spherical harmonics spectrum - a weighted sum of spherical harmonics. Such representations are useful in theoretical and computational analysis. Many different (often severely undersampled) grids are used for HRTF measurement, making the spectral reconstruction difficult. In this paper, two methods of obtaining the spectrum are presented and analyzed both on synthetic (ground-truth data available) and real HRTF measurements.},
	keywords = {Acoustic applications, acoustic field, Acoustic fields, acoustic intensity measurement, Acoustic measurements, acoustic signal processing, Acoustic testing, acoustic waves, array signal processing, audio acoustics, circular arrays, computational analysis, Ear, ear location, head-related transfer function, Helmholtz reciprocity principle, HRTF, HRTF fitting, Loudspeakers, Microphones, Position measurement, signal reconstruction, spatial audio, spectral reconstruction, spherical harmonics, Transfer functions},
	isbn = {978-1-4244-3678-1},
	doi = {10.1109/ASPAA.2009.5346521},
	author = {Zotkin,Dmitry N and Duraiswami, Ramani and Gumerov, Nail A.}
}
@article {14009,
	title = {Fast head-related transfer function measurement via reciprocity},
	journal = {The Journal of the Acoustical Society of America},
	volume = {120},
	year = {2006},
	month = {2006///},
	pages = {2202 - 2215},
	abstract = {An efficient method for head-related transfer function (HRTF) measurement is presented. By applying the acoustical principle of reciprocity, one can swap the speaker and the microphone positions in the traditional (direct) HRTF measurement setup, that is, insert a microspeaker into the subject{\textquoteright}s ear and position several microphones around the subject, enabling simultaneous HRTF acquisition at all microphone positions. The setup used for reciprocal HRTF measurement is described, and the obtained HRTFs are compared with the analytical solution for a sound-hard sphere and with KEMAR manikin HRTF obtained by the direct method. The reciprocally measured sphere HRTF agrees well with the analytical solution. The reciprocally measured and the directly measured KEMAR HRTFs are not exactly identical but agree well in spectrum shape and feature positions. To evaluate if the observed differences are significant, an auditory localization model based on work byJ. C. Middlebrooks [J. Acoust. Soc. Am. 92, 2607{\textendash}2624 (1992)
] was used to predict where a virtual sound source synthesized with the reciprocally measured HRTF would be localized if the directly measured HRTF were used for the localization. It was found that the predicted localization direction generally lies close to the measurement direction, indicating that the HRTFs obtained via the two methods are in good agreement.
},
	keywords = {hearing, Microphones, Transfer functions},
	doi = {10.1121/1.2207578},
	url = {http://link.aip.org/link/?JAS/120/2202/1},
	author = {Zotkin,Dmitry N and Duraiswami, Ramani and Grassi,Elena and Gumerov, Nail A.}
}
@article {18462,
	title = {Processing of reverberant speech for time-delay estimation},
	journal = {IEEE Transactions on Speech and Audio Processing},
	volume = {13},
	year = {2005},
	month = {2005/11//},
	pages = {1110 - 1118},
	abstract = {In this paper, we present a method of extracting the time-delay between speech signals collected at two microphone locations. Time-delay estimation from microphone outputs is the first step for many sound localization algorithms, and also for enhancement of speech. For time-delay estimation, speech signals are normally processed using short-time spectral information (either magnitude or phase or both). The spectral features are affected by degradations in speech caused by noise and reverberation. Features corresponding to the excitation source of the speech production mechanism are robust to such degradations. We show that these source features can be extracted reliably from the speech signal. The time-delay estimate can be obtained using the features extracted even from short segments (50-100 ms) of speech from a pair of microphones. The proposed method for time-delay estimation is found to perform better than the generalized cross-correlation (GCC) approach. A method for enhancement of speech is also proposed using the knowledge of the time-delay and the information of the excitation source.},
	keywords = {Acoustic noise, acoustic signal processing, array signal processing, data mining, Degradation, delay estimation, Feature extraction, Hilbert envelope, localization algorithm, microphone arrays, microphone location, Microphones, Phase estimation, reverberation, short-time spectral information, Signal processing, source features, source information excitation, speech enhancement, Speech processing, speech production mechanism, speech signal, time-delay, time-delay estimation},
	isbn = {1063-6676},
	doi = {10.1109/TSA.2005.853005},
	author = {Yegnanarayana,B. and Prasanna,S. R.M and Duraiswami, Ramani and Zotkin,Dmitry N}
}
@conference {18463,
	title = {An audio-video front-end for multimedia applications},
	booktitle = {2000 IEEE International Conference on Systems, Man, and Cybernetics},
	volume = {2},
	year = {2000},
	month = {2000///},
	pages = {786-791 vol.2 - 786-791 vol.2},
	publisher = {IEEE},
	organization = {IEEE},
	abstract = {Applications such as video gaming, virtual reality, multimodal user interfaces and videoconferencing, require systems that can locate and track persons in a room through a combination of visual and audio cues, enhance the sound that they produce, and perform identification. We describe the development of a particular multimodal sensor fusion system that is portable, runs in real time and achieves these objectives. The system employs novel algorithms for acoustical source location, video-based person tracking and overall system control, which are also described},
	keywords = {Acoustic noise, acoustical source location, Application software, audio cues, audio-video front-end, CAMERAS, Computer vision, Microphones, multimedia applications, multimedia systems, multimodal sensor fusion system, multimodal user interfaces, Position measurement, REAL TIME, Real time systems, real-time systems, sensor fusion, sound, Speech recognition, User interfaces, video cameras, video gaming, video-based person tracking, Videoconference, videoconferencing, Virtual reality, visual cues, Working environment noise},
	isbn = {0-7803-6583-6},
	doi = {10.1109/ICSMC.2000.885945},
	author = {Zotkin,Dmitry N and Duraiswami, Ramani and Davis, Larry S. and Haritaoglu,I.}
}
@conference {18464,
	title = {Smart videoconferencing},
	booktitle = {2000 IEEE International Conference on Multimedia and Expo, 2000. ICME 2000},
	volume = {3},
	year = {2000},
	month = {2000///},
	pages = {1597-1600 vol.3 - 1597-1600 vol.3},
	publisher = {IEEE},
	organization = {IEEE},
	abstract = {The combination of acoustical and video processing to achieve a smart audio and video feed from a set of N microphones and M cameras is a task that might conventionally be accomplished by camera persons and control room staff. However, in the context of videoconferencing, this process needs to be performed by control software. We discuss the use of a multi-camera multi-microphone set-up for unattended videoconferencing, and present details of a prototype implementation being developed},
	keywords = {acoustical processing, Automatic control, CAMERAS, computerised control, control software, Control systems, Intelligent sensors, Layout, Microphones, multi-camera multi-microphone set-up, multimedia systems, Protocols, prototype implementation, Prototypes, sensor fusion, smart videoconferencing, Switches, Teleconferencing, unattended videoconferencing, video processing},
	isbn = {0-7803-6536-4},
	doi = {10.1109/ICME.2000.871075},
	author = {Zotkin,Dmitry N and Duraiswami, Ramani and Philomin,V. and Davis, Larry S.}
}
