Marc A. Zissman
Massachusetts Institute of Technology
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Featured researches published by Marc A. Zissman.
darpa information survivability conference and exposition | 2000
Richard P. Lippmann; David J. Fried; Isaac Graf; Joshua W. Haines; Kristopher R. Kendall; David McClung; Dan Weber; Seth E. Webster; Dan Wyschogrod; Robert K. Cunningham; Marc A. Zissman
An intrusion detection evaluation test bed was developed which generated normal traffic similar to that on a government site containing 100s of users on 1000s of hosts. More than 300 instances of 38 different automated attacks were launched against victim UNIX hosts in seven weeks of training data and two weeks of test data. Six research groups participated in a blind evaluation and results were analyzed for probe, denial-of-service (DoS) remote-to-local (R2L), and user to root (U2R) attacks. The best systems detected old attacks included in the training data, at moderate detection rates ranging from 63% to 93% at a false alarm rate of 10 false alarms per day. Detection rates were much worse for new and novel R2L and DoS attacks included only in the test data. The best systems failed to detect roughly half these new attacks which included damaging access to root-level privileges by remote users. These results suggest that further research should focus on developing techniques to find new attacks instead of extending existing rule-based approaches.
Speech Communication | 2001
Marc A. Zissman; Kay M. Berkling
Automatic language identification of speech is the process by which the language of a digitized speech utterance is recognized by a computer. In this paper, we will describe the set of available cues for language identification of speech and discuss the different approaches to building working systems. This overview includes a range of historical approaches, contemporary systems that have been evaluated on standard databases, and promising future approaches. Comparative results are also reported.
international conference on acoustics, speech, and signal processing | 1994
Marc A. Zissman; Elliot Singer
The paper compares the performance of four approaches to automatic language identification (LID) of telephone speech messages: Gaussian mixture model classification (GMM), language-independent phoneme recognition followed by language-dependent language modeling (PRLM), parallel PRLM (PRLM-P), and language-dependent parallel phoneme recognition (PPR). These approaches span a wide range of training requirements and levels of recognition complexity. All approaches were tested on the development test subset of the OGI multi-language telephone speech corpus. Generally, system performance was directly related to system complexity, with PRLM-P and PPR performing best. On 45 second test utterances, average two language, closed-set, forced-choice classification performance reached 94.5% correct. The best 10 language, closed-set, forced-choice performance was 79.2% correct.<<ETX>>
international conference on acoustics, speech, and signal processing | 1995
Douglas A. Reynolds; Marc A. Zissman; Thomas F. Quatieri; Gerald C. O'Leary; Beth A. Carlson
The two largest factors affecting automatic speaker identification performance are the size of the population and the degradations introduced by noisy communication channels (e.g., telephone transmission). To examine experimentally these two factors, this paper presents text-independent speaker identification results for varying speaker population sizes up to 630 speakers for both clean, wideband speech and telephone speech. A system based on Gaussian mixture speaker models is used for speaker identification and experiments are conducted on the TIMIT and NTIMIT databases. This is believed to be the first speaker identification experiments on the complete 630 speaker TIMIT and NTIMIT databases and the largest text-independent speaker identification task reported to date. Identification accuracies of 99.5% and 60.7% are achieved on the TIMIT and NTIMIT databases, respectively. This paper also presents experiments which examine and attempt to quantify the performance loss associated with various telephone degradations by systematically degrading the TIMIT speech in a manner consistent with measured NTIMIT degradations and measuring the performance loss at each step. It is found that the standard degradations of filtering and additive noise do not account for all of the performance gap between the TIMIT and NTIMIT data. Measurements of nonlinear microphone distortions are also described which may explain the additional performance loss.
ieee aerospace conference | 2002
Lee M. Rossey; Robert K. Cunningham; David J. Fried; Jesse C. Rabek; Richard P. Lippmann; Joshua W. Haines; Marc A. Zissman
The Lincoln adaptable real-time information assurance testbed, LARIAT, is an extension of the testbed created for DARPA 1998 and 1999 intrusion detection (ID) evaluations. LARIAT supports real-time, automated and quantitative evaluations of ID systems and other information assurance (IA) technologies. Components of LARIAT generate realistic background user traffic and real network attacks, verify attack success or failure, score ID system performance, and provide a graphical user interface for control and monitoring. Emphasis was placed on making LARIAT easy to adapt, configure and run without requiring a detailed understanding of the underlying complexity. LARIAT is currently being exercised at four sites and is undergoing continued development and refinement.
international conference on acoustics, speech, and signal processing | 1993
Marc A. Zissman
Ergodic, continuous-observation, hidden Markov models (HMMs) were used to perform automatic language classification and detection of speech messages. State observation probability densities were modeled as tied Gaussian mixtures. The algorithm was evaluated on four multilanguage speech databases: a three language subset of the Spoken Language Library, a three language subset of a five-language Rome Laboratory database, the 20-language CCITT database, and the ten-language OGI (Oregon Graduate Institute) telephone speech database. In general, the performance of a single state HMM (i.e., a static Gaussian mixture classifier) was comparable with that of the multistate HMMs, indicating that the sequential modeling capabilities of HMMs were not exploited.<<ETX>>
international conference on acoustics, speech, and signal processing | 1995
Marc A. Zissman
A language identification technique using multiple single-language phoneme recognizers followed by n-gram language models yielded top performance at the March 1994 NIST language identification evaluation. Since the NIST evaluation, work has been aimed at further improving performance by using the acoustic likelihoods emitted from gender-dependent phoneme recognizers to weight the phonotactic likelihoods output from gender-dependent language models. We have investigated the effect of restricting processing to the most highly discriminating n-grams, and we have also added explicit duration modeling at the phonotactic level. On the OGI Multi-language Telephone Speech Corpus, accuracy on an 11-language, closed-set, language identification task has risen to 89% on 45-s utterances and 79% on 10-s utterances. Two-language classification accuracy is 98% and 95% for the 45-s and 10-s utterances, respectively. Finally, we have started to apply these same techniques to the problem of dialect identifications.
international conference on acoustics, speech, and signal processing | 1987
Marc A. Zissman; G. C. O'Leary; D. H. Johnson
A Block Diagram Compiler (BOC) has been designed and implemented for converting graphic block diagram descriptions of signal processing tasks into source code to be executed on a Multiple Instruction Stream - Multiple Data Stream (MIMD) array computer. The compiler takes as input a block diagram of a real-time DSP application, entered on a graphics CAE workstation, and translates it into efficient real-time assembly language code for the target multiprocessor array. The current implementation produces code for a rectangular grid of Texas Instruments TMS32010 signal processors built at Lincoln Laboratory, but the concept could be extended to other processors or other geometries in the same way that a good assembly language programmer would write it. This report begins by examining the current implementation of the BOC including relevant aspects of the target hardware. Next, we describe the task-assignment module, which uses a simulated annealing algorithm to assign the processing tasks of the DSP application to individual processors in the array. Finally, our experiences with the current version of the BOC software and hardware are reported.
Wiley Encyclopedia of Electrical and Electronics Engineering | 2007
Pedro A. Torres-Carrasquillo; Marc A. Zissman
The sections in this article are 1 Language-Identification Cues 2 Language Identification Systems 3 Evaluations 4 Conclusions 5 Acknowledgment Keywords: phone recognition; spectral similiarity; speech-to-text systems; speech recognition
network computing and applications | 2006
Seth E. Webster; Richard P. Lippmann; Marc A. Zissman
Passive network mapping has often been proposed as an approach to maintain up-to-date information on networks between active scans. This paper presents a comparison of active and passive mapping on an operational network. On this network, active and passive tools found largely disjoint sets of services and the passive system took weeks to discover the last 15% of active services. Active and passive mapping tools provided different, not complimentary information. Deploying passive mapping on an enterprise network does not reduce the need for timely active scans due to non-overlapping coverage and potentially long discovery times