Verfügbarkeit: Conference proceedings

Conference proceedings: May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 2

Gespeichert in:

Bibliographische Detailangaben
Körperschaft:	ICASSP Atlanta, Ga (VerfasserIn)
Format:	Tagungsbericht Buch
Sprache:	Undetermined
Veröffentlicht:	Piscataway, NJ IEEE Service Center 1996
Schlagworte:	Konferenzschrift
Online-Zugang:	Inhaltsverzeichnis
Beschreibung:	LVII, S. 581 - 1271, 7 S. Ill., graph. Darst.

Internformat

MARC


LEADER	00000nam a2200000 cc4500
001	BV010834428
003	DE-604
005	00000000000000.0
007	t
008	960704s1996 ad\|\| \|\|\|\| 10\|\|\| und d
035			\|a (OCoLC)634118403
035			\|a (DE-599)BVBBV010834428
040			\|a DE-604 \|b ger \|e rakddb
041			\|a und
049			\|a DE-91 \|a DE-83
111	2		\|a ICASSP \|n 21 \|d 1996 \|c Atlanta, Ga. \|j Verfasser \|0 (DE-588)5200316-4 \|4 aut
245	1	0	\|a Conference proceedings \|b May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA \|n 2 \|c The 1996 International Conference on Acoustics, Speech, and Signal Processing
264		1	\|a Piscataway, NJ \|b IEEE Service Center \|c 1996
300			\|a LVII, S. 581 - 1271, 7 S. \|b Ill., graph. Darst.
336			\|b txt \|2 rdacontent
337			\|b n \|2 rdamedia
338			\|b nc \|2 rdacarrier
655		7	\|0 (DE-588)1071861417 \|a Konferenzschrift \|2 gnd-content
773	0	8	\|w (DE-604)BV010834406 \|g 2
856	4	2	\|m Digitalisierung TU Muenchen \|q application/pdf \|u http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=007241636&sequence=000002&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA \|3 Inhaltsverzeichnis
999			\|a oai:aleph.bib-bvb.de:BVB01-007241636

Datensatz im Suchindex

_version_	1804125321035776000
adam_text	Volume II Volume II SP16 Speech Recognition: Training Sequences Discriminative Training of Stochastic Markov Graphs for Speech Recognition .........................................................11-581 F. Wolferstetter, G. Ruske - Munich University of Technology, Germany Incremental ML Estimation of HMM Parameters for Efficient Training ...............................................................11-585 Y. Gotoh, H. Silverman - Brown University, USA Improved HMM Training and Scoring Strategies .............................................................................................11-589 L. Arslan, J. Hansen - Duke University, USA Maximum Mutual Information Codebook Mapping for Discrete Hidden Markov Models ..........................................11-593 V. Fontaine, C. Ris, H. Leich - Faculte Polytechnique de Mons, Belgium Minimum Classification Error Training for a Small Amount of Data Enhanced by Vector-Field-Smoothed Bayesian Learning 11-597 J. Takahashi, S. Sagayama - NTT Human Interface Laboratories, Japan Maximum Likelihood Successive Splitting ......................................................................................................11-601 H. Singer - ATR Interpreting Telecommunications Research, Japan M. Ostendorf - Boston University, USA Lattice-Based Discriminative Training for Large Vocabulary Speech Recognition ......................................................11-605 V Valtchev, J. Odeli, P. Woodland, S. Young - Cambridge University, UK Training Algorithms fora Predictive Classifier ................................................................................................ II-609 P. Rao, R. Raveendran - Tata Institute of Fundamental Research, India Discriminative Training of Gaussian Mixture Models for Large Vocabulary Speech Recognition Systems ........................11-613 L· Bahl, M. Padmanabhan, D. Nahamoo, P. Gopalakrishnan - IBM T.J. Watson Research Center, USA Stochastic Observation Hidden Markov Models ................................................................................................11-617 С Mitchell - AT&T Bell Laboratories, USA M. Harper, L. Jamieson - Purdue University, USA SP17 Speech Enhancement Recursive Speech Enhancement Using the EM Algorithm with Initial Conditions Trained by HMMs ........................... П-621 K. Lee - Changwon National University, Korea B. Lee - Seoul National University, Korea I. Song, J. Yoo - KAIST, Korea Robust Noise Reduction for Speech and Audio Signals .......................................................................................11-625 S. Godsill, P. Rayner - Cambridge University, UK Speech Enhancement Based on a Priori Signal-to-Noise Ratio Estimation ...............................................................11-629 P. Scalari - CNET, France J. Vteira-Filho - UNESP, Brazil Noise Model Adaptation in Model Based Speech Enhancement ...........................................................................11-633 B. McKinley - Signal Processing Consultants, USA G. Whipple - U.S. Department of Defense, USA Analysis of Two Structures for Combined Acoustic Echo Cancellation and Noise Reduction .......................................11-637 Y. Guelou, A. Benamar, P. Scalari - CNET, France Selective All-Pole Modeling of Degraded Speech Using M-Band Decomposition ......................................................11-641 C. Yoo - Massachusetts Institute of Technology, USA Text-Directed Speech Enhancement Using Phoneme Classification and Feature Map Constrained Vector Quantization ......11-645 B. Pellom, J. Hansen - Duke University, USA Multi-Channel Signal Separation .................................................................................................................. П-649 D. Chan, P. Rayner, S. Godsill - University of Cambridge, UK Localization by Harmonic Structure and Its Application to Harmonic Sound Stream Segregation .................................11-653 T. Nakatani - NTT Basic Research Laboratories, Japan M. Goto - Waseda University, Japan H. Okuno - NTT Basic Research Laboratories, Japan xix Volume II SP18 Speaker Recognition II A Kind of Fuzzy-Neural Networks for Text-Independent Speaker Identification ...................................................... П-657 Z Yuan, B. Xu, C. Yu - Nanjing University, China A High Performance Text Independent Speaker Recognition System Based on Vowel Spotting and Neural Nets ...............11-661 N. Fakotakis, J. Sirigos - University of Pairas, Greece Signal Modeling for Speaker Identification ......................................................................................................11-665 L. Liu, J. He, G. Palm - University of Ulm, Germany A General Framework of Feature Extraction: Application to Speaker Recognition ...................................................11-669 C. Liu ■ Telecomm Labs, MOTC, Taiwan An HMM Approach to Text-Prompted Speaker Verification ................................................................................. H-673 С Che, Q. Lin - Rutgers University, USA Text-Independent Speaker Identification .........................................................................................................11-677 M. Birnbaum, К. Brown, S. Bardenhagen - Lockheed-Sanders, USA Normalized Discriminant Analysis with Application to a Hybrid Speaker-Verification System ....................................11-681 Q. Li, S. Parthasarathy, A. Rosenberg ■ AT&T Bell Labs, USA D. Tufts - University of Rhode Island, USA Language Independent Gender Identification ................................................................................................... П-685 E. Parris, M. Carey - Ensigma, UK Fine Structure Features for Speaker Identification .............................................................................................11-689 C. Jankowski, T. Quatieri, D. Reynolds - MIT Lincoln Laboratory, USA Speaker Verification with Low Storage Requirements ....................................................................................... П-693 J. Schalkwyk, N. Jain, E. Barnard - Oregon Graduale Institute, USA SP19 Speech Recognition: Speaker and Acoustic Adaptation A Markov Random Field Approach to Bayesian Speaker Adaptation .....................................................................11-697 B. Shahshahani - Corona Corporation, USA Speaker Clustering and Transformation for Speaker Adaptation in Large-Vocabulary Speech Recognition Systems .........11-701 M. Padmanabhan, L Bahl, D. Nahamoo, M. Picheney - IBM T.J. Watson Research Center, USA A Study of On-Line Quasi-Bayes Adaptation For CDHMM-Based Speech Recognition .............................................11-705 Q. Huo - ATR-ITL, Japan С Lee - AT&T Bell Laboratories, USA Speaker Time-Drifting Adaptation Using Trajectory Mixture Hidden Markov Models ................................................11-709 J. Su - Hong Kong City University, Hong Kong H. U, J. Haton - CRIN-CNRS/INRIA - Lorraine, France K. Ng - Hong Kong City University, Hong Kong An Experimental Study of Acoustic Adaptation Algorithms ................................................................................. П-713 A. Sankar, L Neumeyer, M. Weintraub - SRI International, USA Speaker Adaptation with Autonomous Model Complexity Control By MDL Principle ................................................ H-717 K. Shinoda, T. Watanabe - NEC Corporation, Japan An Approach to Speaker Adaptation Based on Analytic Functions ........................................................................11-721 J. McDonough, G. Zavaliagkos, H. Gish - BBN Systems and Technologies, USA Maximum a Posteriori Adaptation for Large Scale HMM Recognizers .................................................................. П-725 G. Zavaliagkos, R. Schwartz, J. McDonough - BBN Systems & Technologies, USA Acoustic Adaptation Using Non-linear Transformations of HMM Parameters ......................................................... П-729 V. Abrash, A. Sankar, H. Franco, M. Cohen - SRI International, USA A Vector Taylor Series Approach for Environment Independent Speech Recognition ................................................ П-733 P. Moreno, B. Raj, R. Stern - Carnegie Mellon University, USA xx Volume II SP20 Spectral Quantization Two-Stage Vector Quantization-Pyramid Lattice Vector Quantization and Application to Speech LSP Coding ..................11-737 J. Pan - Rockwell Telecommunications, USA Linked Split-Vector Quantizer of LPC Parameters .............................................................................................11-741 M. Kim, N. Ha, S. Kim - Samsung Advanced Institute of Technology, Korea Scalar Quantization Using Vector Measure with Application to Quantization of LSF Parameters .................................11-745 H. Ng, S. Leung - City University of Hong Kong, Hong Kong A Robust LSP Encoding Scheme for Noisy Channel ..........................................................................................11-749 J. Du, S. Kim - Polytechnic University, USA An Efficient Coding of LSP Parameters Using Multiple Type Frame Segmentation ...................................................11-753 К Lee, K. Kim, H. Lee - KA1ST, Korea Switched Prediction and Quantization of LSP Frequencies .................................................................................11-757 H. Zarrinkoub, P. Mermelstein - INRS - Telecommunications, Canada Classified Nonlinear Predictive Vector Quantization of Speech Spectral Parameters ................................................11-761 J. Loo - McGill University, Canada W. Chan - Illinois Institute of Technology, USA P. Kabal - McGill University, Canada Exploiting Interframe Correlations in Spectral Quantization: A Study of Different Memory VQ Schemes .....................11-765 T. Eriksson, J. Linden, J. Skoglund - Chalmers University of Technology, Sweden Low Rate Coding of the Spectral Envelope Using Channel Gains ........................................................................11-769 E. Singer, R. McAulay, R. Dunn, T. Quatieri - MIT Lincoln Laboratory, USA A Hybrid Input/Output Spectrum Adaptation Scheme for LD-CELP Coding .........................................................11-773 5. Chid, C. Chan ■ City University of Hong Kong, Hong Kong SP21A Language Identification Automatic Dialect Identification of Extemporaneous, Conversational, Latin American Spanish Speech ........................... П-777 M. Zissman, T. Gleason - MIT Lincoln Laboratory, USA D. Rekart - Texas Instruments, USA B. Losiewicz - Colorado College, USA LVCSR-Based Language Identification .........................................................................................................11-781 T. Schultz, I- Rogina, A. Waibel - University of Karlsruhe, Germany Automatic Language Identification Using Large Vocabulary Continuous Speech Recognition ....................................11-785 5. Mendoza, L Gillick, Y. Ito, S. Lowe, M. Newman - Dragon Systems, USA Experiments with Conversational Telephone Speech for Language Identification ...................................................... II-789 Y. Yan, E. Barnard - Oregon Graduate Institute, USA Statistical Language Identification Based on Untranscribed Training .....................................................................11-793 M. Lund, K Ma, H. Gish - BBN Systems & Technologies, USA A Model for Efficient Formant Estimation ...................................................................................................... П-797 L Welling, H. Ney - Aachen University of Technology, Germany Improved Vocal Tract Model for the Analysis of Nasal Speech Sounds ..................................................................11-801 M. Liu, A. Lacroix ■ University of Frankfurt, Germany A Scalar Homotopy Method for Parallel and Robust Tracking of Line Spectral Pairs ................................................11-805 U. Pillai, V. Stoniek - Carnegie Mellon University, USA Detection of Syntactic Boundaries by Partial Analysis-By-Synthesis of Fundamental Frequency Contours .....................11-809 K. H irose, A. Sakurai - University of Tokyo, Japan A Robust Method for the Estimation of Formant Frequency Modulation in Speech Signals .......................................11-813 P. Rao - Indian Institute of Technology, India Volume II SP22 Speech Recognition: Special Topics Visual Speech Recognition Using Active Shape Models And Hidden Markov Models ................................................ П-817 J. Luetlin, N. Thacker, S. Beet - University of Sheffield, UK Integrating Audio and Visual Information to Provide Highly Robust Speech Recognition ..........................................11-821 M. Tomlinson, M. Russell, N. Brooke ■ SRU, UK An Estimation of Speaker Sampling in Voice Across Japan Database .....................................................................11-825 /. Kudo, T. Nakama, T. Watanabe - Texas instruments Tsukuba R&D Center Ltd., Japan Automatic Articulator)· Annotation of Multi-Sensor Speech Database .....................................................................11-829 N. Parlangeau, R. Andre-Obrecht, A. Marchai - IRIT, France Adaptive Bimodal Sensor Fusion for Automatic Speechreading ...........................................................................11-833 U. Meier, W. Hurst - University of Karlsruhe, Germany P. Duchnowski - MIT Cambridge, USA The Bucket Box Intersection (ВВІ) Algorithm for Fast Approximative Evaluation of Diagonal Mixture Gaussiane ............11-837 J. F ritsch, I. Rogina - University of Karlsruhe, Germany Mulü-Channel HMM ................................................................................................................................. П-841 D. Хи, С. Fancourt, С. Wang - University of Florida, USA On-Line Garbage Modeling for Word and Utterance Verification in Natural Numbers Recognition ..............................11-845 C. de la Torre - Telefonica, Spain L Hernandez-Gomez, F. Caminero-Gal, C. Martin-del Alamo - ETSI Telecom, Spain Creation of Two Children s Speech Databases ...................................................................................................11-849 J. Miller, S. Lee, R. Uclianski, A. Heidbreder, B. Richman - Central Institute for the Deaf, USA J. Tadlock - Southwestern Bell Technology Resources, USA The Contribution of Consonants Versus Vowels in Word Recognition of Fluent Speech .............................................11-853 R. Cole, Y Yan, B. Mak, M. Fanty, T. Bailey - Oregon Graduate Institute, USA SP23 Speech Recognition: Emerging Topics A Novel Word Pre-Selection Method Based on Phonetic Set Indexing ..................................................................... П-857 R. Sarukkai, D. Ballard - University of Rochester, USA DP-Based Wordgraph Pruning ..................................................................................................................... П-861 T. Kuhn, P. Fetter, A. Kaltenmeier, P. Regel-Brietzmann - Research Center Daimler-Benz, Germany A New Hybrid System Based on MMI-Neural Networks for the RM Speech Recognition Task ....................................11-865 G. Rigoll, C. Neukirchen, J. Rottland - Gerhard-Mercator-University Duisburg, Germany Automatic Modeling of User Specific Words fora Speaker Independent Recognition System .......................................11-869 J. Duchateau, D. Van Compernolle - K. U. Leuven, ESAT, Belgium A Dependence Tree Model of Phone Correlation ................................................................................................11-873 O. Ronen, M. Ostendorf- Boston University, USA On the Use of Syllable Phonotactics for Word Hypothesization ...........................................................................11-877 R. Demori, M. Galler - McGill University, USA Creating Speaker-Specific Phonetic Templates with a Speaker-Independent Phonetic Recognizer: Implications for Voice Dialing П-881 N. Jain, R. Cole, E. Barnard - Oregon Graduate Institute, USA Deleted Interpolation and Density Sharing for Continuous Hidden Markov Models ...................................................11-885 X. Huang, M. Hwang, L. Jiang, M. Mahajan - Microsoft Corporation, USA ATime Continuous Model for Speech Recognition .............................................................................................11-889 5. Euler - Bosch Telecom, Germany Fast Output Probability Computation Using Scalar Quantization and Independent Dimension Multi-Mixture ...............11-893 M. Yantada, H. Yamamoto, T. Kosáka, Y. Komori, Y. Ohora - Canon Inc., Japan xxü Volume II ΑΕΙ Microphone Arrays Beamforming and Hearing Aids Microphone-Array Speech Recognition via Incremental MAPTraining ..................................................................11-897 /. Adcock, Y. Gotoh, D. Mashao, H. Silverman - Brown University, USA A Localization Error-Based Method for Microphone-Array Design ........................................................................11-901 M. Brandstein, J. Adcock, H. Silverman - Brown University, USA Nearfield Broadband Frequency Invariant Beamforming 11-905 R. Kennedy, T. Abliayapala, D. Ward, R. Williamson - Australian National University, Australia A Source Subspace Tracking Array of Microphones for Double Talk Situations ......................................................11-909 S. Affes, Y. Grenier - ENST, France The True Duration of the Impulse Response Used to Estimate Reverberation Time ...................................................11-913 L Faiget, R. Ruiz, C. Legros - Univ. de Toulouse - Le Mirati, France Sound Capture from Spatial Volumes: Matched-Filter Processing of Microphone Arrays Having Randomly-Distributed Sensors 11-917 E. Jan, J. Flanagan - Rutgers University, USA Acoustic Source Location in Noisy and Reverberant Environment Using CSP Analysis .............................................11-921 M. Omologo, P. Svaizer - I.R.S.T., Italy A Robust Adaptive Beamformer for Microphone Arrays with a Blocking Matrix Using Constrained Adaptive Filters .........11-925 O. Hoshuyama, A. Sugiyama - NEC Corporation, Japan Voice-Activated AGC for Teleconferencing ...................................................................................................... II-929 P. Chu - PictureTel Corporation, USA A Two-Microphone Adaptive Broadband Array for Hearing Aids ........................................................................... II-933 E. McKinney, V. Debrunner - University of Oklahoma, USA Adaptive Spectral Contrast Enhancement Based on Masking Effect for the Hearing Impaired ....................................11-937 Z. Ribic, J. Yang, M. Lattei - Viennatone GmbH, Austria AE2 Echo Cancelation and Active Noise Control Stereo Echo Cancellation Algorithm Using Imaginary Input-Output Relationships ...................................................11-941 S. Shimauchi, S. Makino - NTT Human Interface Labs, Japan SSB Subband Echo Canceller Using Low-Order Projection Algorithm ..................................................................11-945 S. Makino, J. Noebauer, Y. Haneda, A. Nakagawa - NTT Human Interface Laboratories, Japan A Fast Two Channel Projection Algorithm for Stereophonic Acoustic Echo Cancellation .............................................11-949 F. Amand, J. Benesty, A. Gilloire, Y. Grenier ■ CNET, France Optimization of a Noise Reduction Preprocessing inan Acoustic Echo and Noise Controller ....................................... П-953 B. Ayad, G. Faucon, R. Le Bouquin Jeannes - LTSI - Universite de Rennes I, France Dynamically Regularized Fast RLS with Application to Echo Cancellation ............................................................11-957 S. Gay - AT&T Bell Laboratories, USA Experimental Results of Subband Acoustic Echo Cancelers Under Spherically Invariant Random Processes ..................11-961 P. DeLeon, D. Etter - University of Colorado at Boulder, USA The Effect of Structured Uncertainty in Multi-Channel Feedforward Control System ................................................ II-965 A. Omoto - Kyushu Institute of Design, UK S. Elliott - /SVR University of Southampton, UK Automated Placement of Transducers for Active Noise Control: Performance Measures .............................................11-969 J. Olkin, L. Heck - SRI international, USA K. Naghshineh - Western Michigan University, USA Optimisation of Controlled Acoustic Shadows ...................................................................................................11-973 S. Wright, B. Vuksanovic - University of Huddersfield, UK xxm Volume II АЕЗ Auditory Modeling and Music AM-FM Separation Using Auditory-Motivated Filters .......................................................................................11-977 T. Quatieri - MIT Lincoln Laboratory, USA T Hanna - NSMRL, USA G. O Leary - MIT Lincoln Laboratory, USA A Gammachirp Function as an Optimal Auditory Filter with Mellin Transform ...................................................11-981 T. trino - NTT Basic Research Labs, Japan A Binaural Auditory Model for Sound Quality Measurements and Spatial Hearing Studies .......................................11-985 M. Karjalainen - Helsinki University of Technology, Finland A Study of Auditory Resolution by Using Bark-FAMlet Clicks ..............................................................................11-989 U. Laine - Helsinki University of Technology, Finland M. Huotilainen - University of Helsinki, Finland Real-time Discrimination of Speech/Music on Broadcast Radio ...........................................................................11-993 J. Sounders - Sanders, USA Analysis and Resynthesis of Musical Instrument Sounds Using Energy Separation ...................................................11-997 R. Sussman, M. Kahrs - Rutgers University, USA Automatic Audio Morphing ........................................................................................................................11-1001 M. Slaney, M. Covell, B. Lassiter ■ Interval Research Corporation, USA Residual Modeling in Music Analysis-Synthesis .............................................................................................11-1005 M. Goodwin - University of California at Berkeley, USA Computationally Efficient Algorithm for Time Scale Modification (GLS - TSM) ...................................................... 1Ы009 S. Yim, B. Pawate - Texas Instruments, Japan Multichannel Dynamic Range Compression for Music Signals ...........................................................................11-1013 J. Schmidt, J. Rutledge - Northwestern University, USA AE4 Audio Coding A Bi-Dimensional Coding Scheme Applied to Audio Bitrate Reduction ..................................................................11-1017 L Mainard, M. Lever ■ CCETT, France Audio Coding with a Dynamic Wavelet Packet Decomposition Based on Frequency-Varying Modulated Lapped Transforms ... П-1021 M. Purat, P. Noll - Technical University of Berlin, Germany A Test of MPEG Using Time-Inverted Spoken Audio ....................................................................................... 1Ы025 T. McLaughlin, J. Cookson, L Rasmussen - Library of Congress, USA Extension and Complexity Reduction of Twin VQ Audio Coder ...........................................................................11-1029 T. Moriya, N. Iwakami, K. Ikeda, S. Miki - NTT Human Interface Laboratories, Japan Minimizing the Effects of Subband Quantisation of the Time Domain Alaising Cancellation Filterbank ........................11-1033 C. Jakob, A. Bradley - Royal Melbourne Institute of Technology Speech Analysis and Coding Using a Multi-Resolution Sinusoidal Transform ......................................................... H-1037 D. Anderson - Georgia Institute of Technology, USA Audio Coding Using the Wavelet Packet Transform and a Combined Scalar-Vector Quantization ..............................11-1041 S. Bolond, M. Deriche - Queensland University of Technology, Australia Low Bit Rate High Quality Audio Coding with Combined Harmonic and Wavelet Representations ..............................11-1045 K. Hamdy, M. Ali, A. Tewfik - University of Minnesota, USA A High Performance Software Implementation of MPEG Audio Encoder ............................................................11-1049 M. Kumar, M. Zubair - IBM, Research Division, USA Audio Compression at Low Bit Rates Using a Signal Adaptive Switched Filterbank ................................................11-1053 D. Sinha, J. Johnston ■ AT&T Bell Laboratories, USA xxiv Volume II SPECI Signal Processing for Wireless Communication Systems Signal Processing Techniques for Efficient Use of Transmit Diversity in Wireless Communications ..............................11-1057 G. Wornell, M. Trott - Massachusetts Institute of Technology, USA Adaptive Supression of Narrowband Digital Interferers from Spread Spectrum Signals ..........................................11-1061 H. Poor, X. Wang - Princeton University, USA Signal Processing for Interference Suppression in Direct-Sequence CDMA Systems ................................................11-1065 U. Madhow - University of Illinois, USA Digital Video in a Fading Interference Wireless Environment ..............................................................................11-1069 L. Yun, D. Messerschmitt ■ University of California Berkeley, USA Singular Value Analysis of Space-Time Equalization in the GSM Mobile System ...................................................... II-1073 A. Van der Veen - Delft University of Technology, The Netherlands A. Paulraj - Stanford University, USA Adaptive Equalization for Underwater Wireless Communications ........................................................................11-1077 J. Preisig, D. Brady - Northeastern University, USA Unconditional Maximum Likelihood Approach for Blind Estimation of Digital Signals .............................................11-1081 B. Haider, B. Ng, A. Paulraj, T. Kailath - Stanford University, USA Geometric Properties of the Blind Digital Co-Channel Communications Problem ...................................................11-1085 L. Hansen, G. Хи - University of Texas at Austin, USA Low Complexity Optimal Multiple Access Joint Detection for Linearly Dependent User Sets ....................................11-1089 R. Learned, H. Krim ■ Massachusetts Institute of Technology, USA Equalization of Fading Channels with Random Coefficients ..............................................................................11-1093 M. Tsatsanis - Stevens Institute of Technology, USA G. Giannakis - University of Virginia, USA G. Zhou - Georgia Institute of Technology, USA SPEC2 Education Real-Time DSP for Sophomores ..................................................................................................................11-1097 K. Chiang, B. Evans, W. Huang, F Kovac, E. Lee, D. Messerschmitt, H. Reekie, S. Sastry - University of California at Berkeley, USA Multi-Media and Worldwide Web Resources for Teaching DSP ........................................................................11-1101 J. McClellan, R. Schäfer, J. Schodorf - Georgia Institute of Technology, USA M. Yoder ■ Rose-Hulman institute of Technology, USA Developing Internet-Based VHDL Course Material ..........................................................................................11-1105 J. Calhoun - Mississippi State University, USA Distance Teaming Experiments in Undergraduate DSP Education ........................................................................11-1109 D. Etter - University of Colorado, USA G. Orsak - George Mason University, USA D. Johnson - Rice University, USA A System Characterization/Identification Laboratory Tool for Internet ............................................................... П-ШЗ S. Chatfteld, D. Cochran, M. Sadaka, D. Sinno - Arizona State University, USA The Telecomputing Laboratory: A Multipurpose Facility Used in DSP Education at University of Oklahoma ...............11-1117 V. DeBrunner, L Debrunner, S. Radhakrishnan, К. Khan - University of Oklahoma, USA An Integrated Environment for Modeling, Simulation, Digital Signal Processing, and Control ....................................11-1121 С Crane, R. Kozick - Bucknell University, USA SPECTRA -А Hands-On DSP Learning Experience ..........................................................................................11-1125 E Taylor, J. Mellott, M. Lewis - University of Florida, USA A Laboratory Course for Designing and Testing Spoken Dialogue Systems ............................................................11-1129 D. Cotton, /?. Cole, D. Novick, S. Sutton - Oregon Graduate Institute, USA A MATLAB Software Tool for the Introduction of Speech Coding Fundamentals in a DSP Course .............................. П-1133 T. Painter, A. Spanias - Arizona State University, USA Volume II SPEC3 US DoD Selection of 2400 bps Standard Philosophy and Goals of the DoD 2400bps Vocoder Selection Process ..................................................................11-1137 T. Tremain, M. Kohier- United States Department of Defense, USA T. Champion - United States Air Force, USA Developing a Test Program for the DoD 2400bps Vocoder Selection Process ............................................................11-1141 M. Bielefeld ■ MITRE, USA L Supplee - United States Department of Defense, USA Vocoder Intelligibility and Quality Test Methods .............................................................................................11-1145 J. Tardelli, E. Kreamer - ARCON Corporation, USA Speaker Recognizability Testing for Voice Coders .............................................................................................11-1149 A. Schmidt-Nielsen, D. Brock ■ Naval Research Laboratory, USA Communicability Testing for Voice Coders II- 1153 E. Kreamer, J. Tardelli - ARCON Corporation, USA Host Laboratory Functions for the DoD 2400bps Vocoder Selection Process .........................................................11-1157 P. Gatewood, P. LaFollette - ARCON Corporation, USA Criteria for the DoD 2400bps Vocoder Selection .............................................................................................11-1161 ¥. Kohier - United States Department of Defense, USA P. LaFollete - ARCON Corporation, USA M. Bielefeld - MITRE, USA SPEC4 Sensor Arrays Datasets The ARPA/NAVY Mountaintop Test Program: Adaptive Signal Processing for Airborne Early Warning Radar ............ П-1165 G. Ш, D. Marshall - MIT Lincoln Laboratory, USA Evaluation Of Partially Adaptive STAP Algorithms on the Mountain Top Data Set ................................................11-1169 Y. Seliktar, D. Williams, J. McClellan - Georgia Institute of Technology, USA Reduced Rank Space-Time Adaptive Radar Processing .................................................................................... П-1173 S. Goldstein - University of Southern California, USA P. Zulch - USAF Rome Laboratory, USA I. Reed - University of Southern California, USA Beamspace Techniques for Hot Clutter Cancellation .......................................................................................11-1177 S. Kogon, D. Williams, E. Holder - Georgia Institute of Technology, USA Linear Constraints in Hot Clutter Cancellation ................................................................................................11-1181 L Griffiths - University of Colorado, USA Robust Matched-field Beamforming with Benchmark Shallow-water Acoustic Array Data .......................................11-1185 J. Królik - Duke University, USA A Coherent Approach to Broadband Matched-field Processing: Application in the Hudson Canyon ...........................11-1189 M. Porter, Z. Michalopoulou ■ New Jersey Institute of Technology, USA Matched-Field Localization with Many Uncertain Environmental Parameters: Experimental Data Results ..................11-1193 B. Harrison - Naval Undersea Warfare Center, USA R. Vaccaro, D. Tufts - University of Rhode Island, USA A Posteriori Probability Source Localization in an Uncertain Shallow Water Environment (Hudson Canyon) ...............11-1197 L Noite - Duke University, USA J. Shorey - Digital Systems Resources, Inc., USA SPEC5 Digital Video: Content Processing A New 3D Segmentation Method for Region-based Video Coding ........................................................................11-1201 H. Kang - Samsung Advanced Institute of Technology, Korea A Very Low Bit Rate Video Coding System Using Adaptive Region-Classified Vector Quantization ..............................11-1205 Y. Chen, L Chen, M. Juan - National Taiwan University XXVI Volume II A Hierarchical Video Coder with Cache Motion Information ..............................................................................11-1209 K. Truong, С Richardson - ASPI, USA Evaluation of a Mosaic-Based Approach to Video Compression ...........................................................................11-1213 B. Tannenbaum, R. Suryadevara, S. Hsu - David Sarnoff Research Center, USA JACOB: Just A Content-Based Query System for Video Databases .....................................................................11-1216 M. La Cascia, E. Ardizzone - Universila di Palermo, Italy Nonlinear Editing by Generative Video .........................................................................................................11-1220 R. Jasinschi, J. Moura ■ Carnegie Mellon University, USA Tracking Areas of Interest for Content-based Functionalities in Segmentation-based Video Coding ..............................11-1224 F. Marques - Universität Politecnica de Catalunya, Spain B. Marcotegui, F. Meyer - Centre de Morphologie Mathématique, France Key Frame Selection by Motion Analysis ...................................................................................................... 11- 1228 W. Wolf- Princeton University, USA SPEC6 Methods and Tools for Rapid Prototyping of DSP Systems Executable Requirements: Opportunities and Impediments ..............................................................................11-1232 G. Shaw, A. Anderson - MIT Lincoln Laboratory, USA Rapid Prototyping of DSP Chip-Sets via Functional Reuse ................................................................................. II-1236 M. Ben Romdhane, V. Madisetti - Georgia Institute of Technology, USA Heuristic Techniques for Synthesis of Hard Real-Time DSP Application Specific Systems ..........................................11-1240 M. Potkonjak ■ NEC USA, USA W. Wolf- Princeton University, USA A CAD Tool for the Optimization of Video Signal Processor Architectures ............................................................11-1244 H. Kropp, M. Schwiegershausen, P. Pirsch - University of Hannover, Germany Hardware/Software Co-Design for DSP Applications Via the HMS Framework ...................................................... H-1248 M. Sheliga, E. Hsing-Mean Sha - University of Notre Dame, USA An Architectural Trade Capability Using the Ptolemy Kernel ........................................................................... II-1252 E. Pauer, J. Prime - Sanders, USA Rapid Prototyping of DSP Systems via Standard Interfaces ..............................................................................11-1256 S. Famonadeh - Georgia Institute of Technology, USA Autocoding: An Enabling Technology for Rapid Prototyping ..............................................................................11-1260 C. Robbins - Management Communications and Control, USA Middleware for Realtime Multicomputing Tool Development ..............................................................................11-1264 fi. Isenstein, M. Krueger, A. Pool - Mercury Computer Systems, USA Interface Synthesis in Heterogeneous System-Level DSP Design Tools ..................................................................11-1268 J. Pino, M. Williamson, E. Lee - University of California at Berkeley, USA
any_adam_object	1
author_corporate	ICASSP Atlanta, Ga
author_corporate_role	aut
author_facet	ICASSP Atlanta, Ga
author_sort	ICASSP Atlanta, Ga
building	Verbundindex
bvnumber	BV010834428
ctrlnum	(OCoLC)634118403 (DE-599)BVBBV010834428
format	Conference Proceeding Book
fullrecord	<?xml version="1.0" encoding="UTF-8"?><collection xmlns="http://www.loc.gov/MARC21/slim"><record><leader>01214nam a2200277 cc4500</leader><controlfield tag="001">BV010834428</controlfield><controlfield tag="003">DE-604</controlfield><controlfield tag="005">00000000000000.0</controlfield><controlfield tag="007">t</controlfield><controlfield tag="008">960704s1996 ad\|\| \|\|\|\| 10\|\|\| und d</controlfield><datafield tag="035" ind1=" " ind2=" "><subfield code="a">(OCoLC)634118403</subfield></datafield><datafield tag="035" ind1=" " ind2=" "><subfield code="a">(DE-599)BVBBV010834428</subfield></datafield><datafield tag="040" ind1=" " ind2=" "><subfield code="a">DE-604</subfield><subfield code="b">ger</subfield><subfield code="e">rakddb</subfield></datafield><datafield tag="041" ind1=" " ind2=" "><subfield code="a">und</subfield></datafield><datafield tag="049" ind1=" " ind2=" "><subfield code="a">DE-91</subfield><subfield code="a">DE-83</subfield></datafield><datafield tag="111" ind1="2" ind2=" "><subfield code="a">ICASSP</subfield><subfield code="n">21</subfield><subfield code="d">1996</subfield><subfield code="c">Atlanta, Ga.</subfield><subfield code="j">Verfasser</subfield><subfield code="0">(DE-588)5200316-4</subfield><subfield code="4">aut</subfield></datafield><datafield tag="245" ind1="1" ind2="0"><subfield code="a">Conference proceedings</subfield><subfield code="b">May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA</subfield><subfield code="n">2</subfield><subfield code="c">The 1996 International Conference on Acoustics, Speech, and Signal Processing</subfield></datafield><datafield tag="264" ind1=" " ind2="1"><subfield code="a">Piscataway, NJ</subfield><subfield code="b">IEEE Service Center</subfield><subfield code="c">1996</subfield></datafield><datafield tag="300" ind1=" " ind2=" "><subfield code="a">LVII, S. 581 - 1271, 7 S.</subfield><subfield code="b">Ill., graph. Darst.</subfield></datafield><datafield tag="336" ind1=" " ind2=" "><subfield code="b">txt</subfield><subfield code="2">rdacontent</subfield></datafield><datafield tag="337" ind1=" " ind2=" "><subfield code="b">n</subfield><subfield code="2">rdamedia</subfield></datafield><datafield tag="338" ind1=" " ind2=" "><subfield code="b">nc</subfield><subfield code="2">rdacarrier</subfield></datafield><datafield tag="655" ind1=" " ind2="7"><subfield code="0">(DE-588)1071861417</subfield><subfield code="a">Konferenzschrift</subfield><subfield code="2">gnd-content</subfield></datafield><datafield tag="773" ind1="0" ind2="8"><subfield code="w">(DE-604)BV010834406</subfield><subfield code="g">2</subfield></datafield><datafield tag="856" ind1="4" ind2="2"><subfield code="m">Digitalisierung TU Muenchen</subfield><subfield code="q">application/pdf</subfield><subfield code="u">http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=007241636&sequence=000002&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA</subfield><subfield code="3">Inhaltsverzeichnis</subfield></datafield><datafield tag="999" ind1=" " ind2=" "><subfield code="a">oai:aleph.bib-bvb.de:BVB01-007241636</subfield></datafield></record></collection>
genre	(DE-588)1071861417 Konferenzschrift gnd-content
genre_facet	Konferenzschrift
id	DE-604.BV010834428
illustrated	Illustrated
indexdate	2024-07-09T17:59:42Z
institution	BVB
institution_GND	(DE-588)5200316-4
language	Undetermined
oai_aleph_id	oai:aleph.bib-bvb.de:BVB01-007241636
oclc_num	634118403
open_access_boolean
owner	DE-91 DE-BY-TUM DE-83
owner_facet	DE-91 DE-BY-TUM DE-83
physical	LVII, S. 581 - 1271, 7 S. Ill., graph. Darst.
publishDate	1996
publishDateSearch	1996
publishDateSort	1996
publisher	IEEE Service Center
record_format	marc
spelling	ICASSP 21 1996 Atlanta, Ga. Verfasser (DE-588)5200316-4 aut Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 2 The 1996 International Conference on Acoustics, Speech, and Signal Processing Piscataway, NJ IEEE Service Center 1996 LVII, S. 581 - 1271, 7 S. Ill., graph. Darst. txt rdacontent n rdamedia nc rdacarrier (DE-588)1071861417 Konferenzschrift gnd-content (DE-604)BV010834406 2 Digitalisierung TU Muenchen application/pdf http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=007241636&sequence=000002&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA Inhaltsverzeichnis
spellingShingle	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
subject_GND	(DE-588)1071861417
title	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
title_auth	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
title_exact_search	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
title_full	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 2 The 1996 International Conference on Acoustics, Speech, and Signal Processing
title_fullStr	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 2 The 1996 International Conference on Acoustics, Speech, and Signal Processing
title_full_unstemmed	Conference proceedings May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA 2 The 1996 International Conference on Acoustics, Speech, and Signal Processing
title_short	Conference proceedings
title_sort	conference proceedings may 7 10 1996 marriott marquis hotel atlanta georgia usa
title_sub	May 7 - 10, 1996, Marriott Marquis Hotel, Atlanta, Georgia, USA
topic_facet	Konferenzschrift
url	http://bvbr.bib-bvb.de:8991/F?func=service&doc_library=BVB01&local_base=BVB01&doc_number=007241636&sequence=000002&line_number=0001&func_code=DB_RECORDS&service_type=MEDIA
volume_link	(DE-604)BV010834406
work_keys_str_mv	AT icasspatlantaga conferenceproceedingsmay7101996marriottmarquishotelatlantageorgiausa2

Verfügbarkeit

Es ist kein Print-Exemplar vorhanden.

Fernleihe Bestellen Achtung: Nicht im THWS-Bestand! Inhaltsverzeichnis

MARC

Datensatz im Suchindex

Es ist kein Print-Exemplar vorhanden.

Ähnliche Einträge