JAVA synchronized collaborative multimedia toolkit: A collaborative communication tool

In this project a collaboration multimedia toolkit, JSCMT (Java Synchronized Collaborative Multimedia Toolkit) was developed which is intended to connect a group of people located in different geographical locations who are working on the same project

A Novel Combined System of Direction Estimation and Sound Zooming of Multiple Speakers

This article presents a new system for estimation the direction of multiple speakers and zooming the sound of one of them at a time. The proposed system is a combination of two levels; namely, sound source direction estimation, and acoustic zooming. The sound source direction estimation uses so-called the energetic analysis method for estimation the direction of multiple speakers, whereas the acoustic zooming is based on modifying the parameters of the directional audio coding (DirAC) in order to zoom the sound of a selected speaker among the others. Both listening tests and objective assessments are performed to evaluate this system using different time-frequency transforms

Localization and Rendering of Sound Sources in Acoustic Fields

Disertační práce se zabývá lokalizací zdrojů zvuku a akustickým zoomem. Hlavním cílem této práce je navrhnout systém s akustickým zoomem, který přiblíží zvuk jednoho mluvčího mezi skupinou mluvčích, a to i když mluví současně. Tento systém je kompatibilní s technikou prostorového zvuku. Hlavní přínosy disertační práce jsou následující: 1. Návrh metody pro odhad více směrů přicházejícího zvuku. 2. Návrh metody pro akustické zoomování pomocí DirAC. 3. Návrh kombinovaného systému pomocí předchozích kroků, který může být použit v telekonferencích.This doctoral thesis deals with sound source localization and acoustic zooming. The primary goal of this dissertation is to design an acoustic zooming system, which can zoom the sound of one speaker among multiple speakers even when they speak simultaneously. The system is compatible with surround sound techniques. In particular, the main contributions of the doctoral thesis are as follows: 1. Design of a method for multiple sound directions estimations. 2. Proposing a method for acoustic zooming using DirAC. 3. Design a combined system using the previous mentioned steps, which can be used in teleconferencing.

Structured Sparsity Models for Multiparty Speech Recovery from Reverberant Recordings

We tackle the multi-party speech recovery problem through modeling the acoustic of the reverberant chambers. Our approach exploits structured sparsity models to perform room modeling and speech recovery. We propose a scheme for characterizing the room acoustic from the unknown competing speech sources relying on localization of the early images of the speakers by sparse approximation of the spatial spectra of the virtual sources in a free-space model. The images are then clustered exploiting the low-rank structure of the spectro-temporal components belonging to each source. This enables us to identify the early support of the room impulse response function and its unique map to the room geometry. To further tackle the ambiguity of the reflection ratios, we propose a novel formulation of the reverberation model and estimate the absorption coefficients through a convex optimization exploiting joint sparsity model formulated upon spatio-spectral sparsity of concurrent speech representation. The acoustic parameters are then incorporated for separating individual speech signals through either structured sparse recovery or inverse filtering the acoustic channels. The experiments conducted on real data recordings demonstrate the effectiveness of the proposed approach for multi-party speech recovery and recognition.Comment: 31 page

Digital Signal Processing

Contains introduction and reports on seventeen research projects.U.S. Navy - Office of Naval Research (Contract N00014-81-K-0742)U.S. Navy - Office of Naval Research (Contract N00014-77-C-0266)National Science Foundation (Grant ECS80-07102)Bell Laboratories FellowshipAmoco Foundation FellowshipSchlumberger-Doll Research Center FellowshipSanders Associates, Inc.Toshiba Company FellowshipM.I.T. Vinton Hayes FellowshipHertz Foundation Fellowshi

A Microphone Array System for Multimedia Applications with Near-Field Signal Targets

A microphone array beamforming system is proposed for multimedia communication applications using four sets of small planar arrays mounted on a computer monitor. A new virtual array approach is employed such that the original signals received by the array elements are weighted and delayed to synthesize a large, nonuniformly spaced, harmonically nested virtual array covering the frequency band [50, 7000] Hz of the wideband telephony. Subband multirate processing and near-field beamforming techniques are then used jointly by the nested virtual array to improve the performances in reverberant environments. A new beamforming algorithm is also proposed using a broadband near-field spherically isotropic noise model for array optimization. The near-field noise model assumes a large number of broadband random noises uniformly distributed over a sphere with a finite radius in contrast to the conventional far-field isotropic noise model which has an infinite radius. The radius of the noise model, thus, adds a design parameter in addition to its power for tradeoffs between performance and robustness. It is shown that the near-field beamformers designed by the new algorithm can achieve more than 8-dB reverberation suppression while maintaining sufficient robustness against background noises and signal location errors. Computer simulations and real room experiments also show that the proposed array beamforming system reduces beampattern variations for broadband signals, obtains strong noise and reverberation suppression, and improves the sound quality for near-field targets